kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-05 04:11:08 +00:00
Code Issues Packages Projects Releases Wiki Activity
Files
d5b0570bb2936938fd149fcb2e4c540e9af1c406
asterisk/ChangeLog
Leif Madsen d5b0570bb2 Importing files for 1.4.26-rc6 release. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.4.26-rc6@206223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-13 18:34:50 +00:00

25254 lines
1.1 MiB
Raw Blame History

2009-07-13 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.26-rc6
2009-07-13 15:12 +0000 [r206126] Russell Bryant <russell@digium.com>
* main/pbx.c: Print CID match in "show dialplan". (closes issue
#14702) Reported by: klaus3000 Patches:
patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000
(license 65)
2009-07-10 17:39 +0000 [r205877] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Properly ACK 487 responses to canceled
INVITEs. From the review board request: The fix from review 298
has exposed a new bug in chan_sip. When we hang up an outgoing
call, we first will dump all the outstanding packets on the
sip_pvt using __sip_pretend_ack. Then, if we can, we send a
CANCEL. The problem with this is that since destroyed all the
outstanding packets on the dialog, we cannot match the incoming
487 response to our INVITE. Because we cannot match the response,
we do not send an ACK. To correct this, instead of using
__sip_pretend_ack, I have changed the code to loop through the
list of packets and call __sip_semi_ack on each one instead. This
causes us to stop retransmitting the requests, but we still have
them around in case we get responses for them. When discussing
this earlier today with Josh Colp, we both agreed that in the
majority of cases, this would be enough of a fix. However, we
also agreed that we should have a safety net in place in case we
never receive a response to our initial INVITE. To handle this, I
have modified __sip_autodestruct to behave similar to the way it
does in Asterisk 1.4. If there are outstanding packets on the
sip_pvt, the needdestroy flag is not set, and the last request on
the dialog was either a CANCEL or BYE, then we set the
needdestroy flag and reschedule destruction for 10 seconds in the
future. If, though, the needdestroy flag is set, then we use
__sip_pretend_ack to kill the remaining outstanding packets so
that the monitor thread can destroy the sip_pvt. I ran two
separate tests. First, I placed a call from my Aastra phone to my
Polycom phone. I hung up the Aastra before the Polycom answered.
I verified through sip debug output that Asterisk properly ACKed
the 487 received from the Polycom. For my second test, I set up a
SIPp UAS scenario so that it would send a 200 OK in response to a
CANCEL but would not send a 487 for the original INVITE. I
verified that after about 40 seconds, Asterisk properly cleans up
the outgoing sip_pvt for the call. Review:
https://reviewboard.asterisk.org/r/308
2009-07-10 16:23 +0000 [r205804] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: SIP registration auth loop caused by stale
nonce If an endpoint sends two registration requests in a very
short period of time with the same nonce, both receive 401
responses from Asterisk, each with a different nonce (the second
401 containing the current nonce and the first one being stale).
If the endpoint responds to the first 401, it does not match the
current nonce so Asterisk sends a third 401 with a newly
generated nonce (which updates the current nonce)... Now if the
endpoint responds to the second 401, it does not match the
current nonce either and Asterisk sends a fourth 401 with a newly
generated nonce... This loop goes on and on. There appears to be
a simple fix for this. If the nonce from the request does not
match our nonce, but is a good response to a previous nonce,
instead of sending a 401 with a newly generated nonce, use the
current one instead. This breaks the loop as the nonce is not
updated until a response is received. Additional logic has been
added to make sure no nonce can be responded to twice though.
(closes issue #15102) Reported by: Jamuel Patches:
patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff
uploaded by dvossel (license 671) Tested by: Jamuel Review:
https://reviewboard.asterisk.org/r/289/
2009-07-10 15:51 +0000 [r205775] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Ensure that outbound NOTIFY requests are
properly routed through stateful proxies. With this change, we
make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will
have the proper Route headers in them. (closes issue #14725)
Reported by: ibc
2009-07-09 23:37 +0000 [r205728] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: No audio on calls from Asterisk to various
ISDN devices until DTMF sent by caller. Add missing clearing of
the dialing flag when the ISDN call is CONNECTED. (i.e. When
libpri generates the event PRI_EVENT_ANSWER.) (closes issue
#15420) Reported by: scottbmilne Patches:
bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
Tested by: scottbmilne, alecdavis (closes issue #15416) Reported
by: avinoash (closes issue #15389) Reported by: alecdavis This
patch should also fix the following issue: (issue #15205)
Reported by: vinsik
2009-07-09 16:18 +0000 [r205409-205599] David Vossel <dvossel@digium.com>
* include/asterisk/time.h: Changing ast_samp2tv to not use floating
point.
* main/rtp.c, channels/chan_iax2.c, include/asterisk/frame.h: Fixes
8khz assumptions Many calculations assume 8khz is the codec rate.
This is not always the case. This patch only addresses chan_iax.c
and res_rtp_asterisk.c, but I am sure there are other areas that
make this assumption as well. Review:
https://reviewboard.asterisk.org/r/306/
* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
include/asterisk/pbx.h: moving ast_devstate_to_extenstate to
pbx.c from devicestate.c ast_devstate_to_extenstate belongs in
pbx.c. This change fixes a compile time error with chan_vpb as
well.
2009-07-08 19:26 +0000 [r205349] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Prevent phantom calls to queue members. If a
caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would
incorrectly indicate that the caller was still in the queue. With
these changes, the problem does not occur. (closes issue #14631)
Reported by: latinsud Patches: queue_announce_ghost_call2.diff
uploaded by latinsud (license 745) (with small modification from
me)
2009-07-08 18:19 +0000 [r205288] Jason Parker <jparker@digium.com>
* config.guess, config.sub: Update config.guess and config.sub from
the savannah.gnu.org git repo.
2009-07-08 16:53 +0000 [r205215] David Vossel <dvossel@digium.com>
* include/asterisk/time.h: ast_samp2tv needs floating point for
16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate
is 16000. The .5 is currently stripped off because we don't
calculate using floating points. This causes madness with 16khz
audio. (issue ABE-1899) Review:
https://reviewboard.asterisk.org/r/305/
2009-07-08 16:26 +0000 [r205188] Tilghman Lesher <tlesher@digium.com>
* main/say.c: Add redirection warnings for the invalid language
codes previously removed.
2009-07-08 15:54 +0000 [r205149] Russell Bryant <russell@digium.com>
* res/res_crypto.c: Make OpenSSL usage thread-safe. OpenSSL is not
thread-safe by default. However, making it thread safe is very
easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more
information, start with the "Is OpenSSL thread-safe?" question on
the FAQ page of openssl.org.
2009-07-02 21:59 +0000 [r204834] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Removed confusing warning message "Got
Busy in Connected State" If an incoming mISDN call is answered
with the Answer application and a subsequent Dial gets a busy
endpoint then it is valid for that already connected channel to
get the busy indication. Asterisk will play the busy tones until
the dialplan plays something else or hangs up the call. (closes
issue #11974) Reported by: fvdb
2009-07-02 18:15 +0000 [r204755] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
include/asterisk/pbx.h: moving device state functions from pbx.h
to devicestate.h to sync with other branches
2009-07-02 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.26-rc5
2009-07-02 15:05 +0000 [r204681] David Vossel <dvossel@digium.com>
* include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c,
include/asterisk/pbx.h: Improved mapping of extension states from
combined device states. This fixes a few issues with incorrect
extension states and adds a cli command, core show
device2extenstate, to display all possible state mappings.
(closes issue #15413) Reported by: legart Patches:
exten_helper.diff uploaded by dvossel (license 671) Tested by:
dvossel, legart, amilcar Review:
https://reviewboard.asterisk.org/r/301/
2009-06-30 20:23 +0000 [r204556] Tilghman Lesher <tlesher@digium.com>
* main/say.c, UPGRADE.txt: More incorrect language codes, plus
ensuring that regionalizations use the specified language, and
not English for grammar. (closes issue #15022) Reported by:
greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by
tilghman (license 14)
2009-06-30 18:47 +0000 [r204474] Jason Parker <jparker@digium.com>
* main/say.c: Fix ast_say_counted_noun to correctly handle Polish.
Fix a comment typo in passing.
2009-06-30 18:23 +0000 [r204469] Tilghman Lesher <tlesher@digium.com>
* main/say.c, UPGRADE.txt: "tw" is the language specification for
Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported
by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded
by tilghman (license 14) 20090617__issue15346__trunk.diff.txt
uploaded by tilghman (license 14)
20090617__issue15346__1.6.0.diff.txt uploaded by tilghman
(license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by
tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt
uploaded by tilghman (license 14) Tested by: volivier
2009-06-29 22:45 +0000 [r204243-204300] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add error message so that it is clear why a
SIP peer was not processed when a DNS lookup fails on a host or
outboundproxy. (closes issue #13432) Reported by: p_lindheimer
Patches: outboundproxy.patch uploaded by p (license 558)
* channels/chan_sip.c: Fix build oops.
* channels/chan_sip.c: Fix a problem where chan_sip would ignore
"old" but valid responses. chan_sip has had a problem for quite a
long time that would manifest when Asterisk would send multiple
SIP responses on the same dialog before receiving a response. The
problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two
requests out, and a response arrived for the first request sent,
then Asterisk would ignore the response. The result was that
Asterisk would continue retransmitting the requests and ignoring
the responses until the maximum number of retransmissions had
been reached. The fix here is to rearrange the code a bit so that
instead of simply comparing the sequence number of the response
to our latest outgoing sequence number, we walk our list of
outstanding packets and determine if there is a match. If there
is, we continue. If not, then we ignore the response. In doing
this, I found a few completely useless variables that I have now
removed. (closes issue #11231) Reported by: flefoll Review:
https://reviewboard.asterisk.org/r/298
2009-06-29 19:36 +0000 [r204170] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c, funcs/func_strings.c: Revision 189537 was
supposed to make 1.4 more correct. Instead, it broke func_odbc.
Reverting. (closes issue #15317, issue #14614)
2009-06-29 17:04 +0000 [r204067] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: segfault after SPINLOCK schedule delete
Using the SPINLOCK schedule delete macro can result in the
iax_pvt lock being given up. This makes it possible for the
iax_pvt to dissappear when we thought we held the mutex the
entire time. To resolve this, the iax_pvt's ref count is
incremented. (closes issue #15377) Reported by: aragon Patches:
iax_spin_issue_1.4.diff uploaded by dvossel (license 671) Tested
by: aragon, dvossel
2009-06-29 15:04 +0000 [r204012] Mark Michelson <mmichelson@digium.com>
* apps/app_mixmonitor.c: Place unlock of mutex in an else block so
that it does not get unlocked twice. (closes issue #15400)
Reported by: aragon
2009-06-27 00:55 +0000 [r203908] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: The ISDN CPE side should not exclusively
pick B channels normally. Before this patch, Asterisk
unconditionally picked B channels exclusively on the CPE side and
normally allowed alternative B channels on the network side. Now
Asterisk does the opposite. Reasons for the CPE side to normally
not pick B channels exclusively: * For CPE point-to-multipoint
mode (i.e. phone side), the CPE side does not have enough
information to exclusively pick B channels. (There may be other
devices on the line.) * Q.931 gives preference to the network
side picking B channels. * Some telcos require the CPE side to
not pick B channels exclusively. (closes issue #14383) Reported
by: mbrancaleoni
2009-06-26 22:09 +0000 [r203848] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Make sure to recreate the dahdi pseudo
channel after dahdi restart (closes issue #14477) Reported by:
timking
2009-06-26 21:16 +0000 [r203785] Russell Bryant <russell@digium.com>
* main/file.c: Don't fast forward past the end of a message. This
is nice change for users of the voicemail application. If someone
gets a little carried away with fast forwarding through a
message, they can easily get to the end and accidentally exit the
voicemail application by hitting the fast forward key during the
following prompt. This adds some safety by not allowing a fast
forward past the end of a message. (closes issue #14554) Reported
by: lacoursj Patches: 21761.patch uploaded by lacoursj (license
707) Tested by: lacoursj
2009-06-26 20:03 +0000 [r203719] David Brooks <dbrooks@digium.com>
* apps/app_voicemail.c: Fixing voicemail's error in checking max
silence vs min message length Max silence was represented in
milliseconds, yet vmminsecs (minmessage) was represented as
seconds. Also, the inequality was reversed. The warning, if
triggered, was "Max silence should be less than minmessage or you
may get empty messages", which should have been logged if max
silence was greater than minmessage, but the check was for less
than. Also, conforming if statement to coding guidelines. closes
issue #15331) Reported by: markd Review:
https://reviewboard.asterisk.org/r/293/
2009-06-25 21:13 +0000 [r203380] Terry Wilson <twilson@digium.com>
* main/cli.c: I didn't see that Mark already fixed the underlying
issue! Yay for removing useless code.
2009-06-25 21:02 +0000 [r203375] Russell Bryant <russell@digium.com>
* res/res_features.c: Fix a case where CDR answer time could be
before the start time involving parking. (closes issue #13794)
Reported by: davidw Patches: 13794.patch uploaded by murf
(license 17) 13794.patch.160 uploaded by murf (license 17) Tested
by: murf, dbrooks
2009-06-25 20:09 +0000 [r203311] Terry Wilson <twilson@digium.com>
* main/cli.c: Don't try to free NULL
2009-06-25 18:52 +0000 [r203230] Mark Michelson <mmichelson@digium.com>
* main/astmm.c: Prevent false positives when freeing a NULL pointer
with MALLOC_DEBUG enabled.
2009-06-25 16:02 +0000 [r203115] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Resolve a crash related to a T.38 reinvite
race condition. This change resolves a crash observed locally
during some T.38 testing. A call was set up using a call file,
and when the T.38 reinvite came in, the channel state was still
AST_STATE_DOWN. The reason is explained by a comment in the code
that previously lived in the handling of AST_STATE_RINGING. This
change modifies the logic to handle the same race condition for
any channel state that is not UP. (closes ABE-1895)
2009-06-24 21:01 +0000 [r203036] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c: Improved chan_dahdi.conf pritimer error
checking. Valid format is: pritimer=timer_name,timer_value *
Fixed segfault if the ',' is missing. * Completely check the
range returned by pri_timer2idx() to prevent possible access
outside array bounds.
2009-06-24 18:28 +0000 [r202966] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Use the handy UNLINK macro instead of
hand-coding the same thing in-line.
2009-06-24 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.26-rc4
2009-06-23 16:28 +0000 [r202671] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: MWI NOTIFY contains a wrong URI if Asterisk
listens to non-standard port and transport (closes issue #14659)
Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt
uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff
uploaded by dvossel (license 671) Tested by: dvossel, klaus3000
Review: https://reviewboard.asterisk.org/r/288/
2009-06-23 15:22 +0000 [r202572-202601] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix more memory leaks that may result if rtp
is not successfully allocated.
* channels/chan_sip.c: Fix potential memory leak in chan_sip when
video rtp is not allocated properly.
2009-06-22 20:08 +0000 [r202414-202496] Russell Bryant <russell@digium.com>
* main/channel.c: Report CallerID change during a masquerade.
Reported by: markster
* channels/chan_sip.c: Make Polycom subscription type override
check more explicit.
2009-06-22 14:44 +0000 [r202336-202342] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Remove an extra debug line left from
previous commit.
* channels/chan_sip.c: Fix a situation in which Asterisk would not
stop retransmitting 487s. If a CANCEL were received by Asterisk,
we would send a 487 in response to the original INVITE and a 200
OK for the CANCEL. If there were a network hiccup which caused
the 200 OK and the 487 to be lost, then the UA communicating with
Asterisk may try to retransmit its CANCEL. Asterisk's response to
this used to be to try sending another 487 to the canceled INVITE
and another 200 OK to the CANCEL. The problem here is that the
originally-sent 487 was sent "reliably" meaning that it will be
retransmitted until it is received properly. So when we receive
the second CANCEL it is likely that the first batch of 487s we
sent is still going strong and reaches the UA. The result was
that the second set of 487s would be retransmitted constantly
until the maximum number of retries had been reached. The fix for
this is that if we receive a second CANCEL for an INVITE, then we
cancel the retransmission of the first set of 487s and start a
second set. This causes the dialog to be terminated reasonably.
(closes issue #14584) Reported by: klaus3000 Patches:
14584_v2.patch uploaded by mmichelson (license 60) Tested by:
klaus3000
* channels/chan_sip.c: Fix a possible infinite loop in SDP parsing
during glare situation. There was a while loop in
get_ip_and_port_from_sdp which was controlled by a call to
get_sdp_iterate. The loop would exit either if what we were
searching for was found or if the return was NULL. The problem is
that get_sdp_iterate never returns NULL. This means that if what
we were searching for was not present, the loop would run
infinitely. This modification of the loop fixes the problem.
(closes issue #15213) Reported by: schmidts (closes issue #15349)
Reported by: samy (closes issue #14464) Reported by: pj (closes
issue #15345) Reported by: aragon Patches: sip_inf_loop.patch
uploaded by mmichelson (license 60) Tested by: aragon
2009-06-20 17:51 +0000 [r202153] Sean Bright <sean@malleable.com>
* channels/chan_sip.c: Since we don't have sip_pvt_lock() in 1.4,
we need to use ast_mutex_* directly. (closes issue #15366)
Reported by: loloski
2009-06-19 21:21 +0000 [r202022] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Added deadlock protection to
try_suggested_sip_codec in chan_sip.c. Review:
https://reviewboard.asterisk.org/r/287/
2009-06-19 20:22 +0000 [r201993] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: timestamp was being converted to host order
as a short rather than a long (closes issue #15361) Reported by:
ffloimair Patches: ts_issue.diff uploaded by dvossel (license
671)
2009-06-19 00:40 +0000 [r201828] Tilghman Lesher <tlesher@digium.com>
* res/res_features.c: If the "h" extension fails, give it another
chance in main/pbx.c. If the "h" extension fails, give it another
chance in main/pbx.c, when it returns from the bridge code. Fixes
an issue where the "h" extension may occasionally not fire, when
a Dial is executed from a Macro. Debugged in #asterisk with user
tompaw.
2009-06-18 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.26-rc3
2009-06-18 15:24 +0000 [r201600] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Fix memory corruption and leakage related
reloads of non files mode MoH classes. For Music on Hold classes
that are not files mode, meaning that we are executing an
application that will feed us audio data, we use a thread to
monitor the external application and read audio from it. This
thread also makes use of the MoH class object. In the MoH class
destructor, we used pthread_cancel() to ask the thread to exit.
Unfortunately, the code did not wait to ensure that the thread
actually went away. What needed to be done is a pthread_join() to
ensure that the thread fully cleans up before we proceed. By
adding this one line, we resolve two significant problems: 1)
Since the thread was never joined, it never fully goes away. So,
on every reload of non-files mode MoH, an unused thread was
sticking around. 2) There was a race condition here where the
application monitoring thread could still try to access the MoH
class, even though the thread executing the MoH reload has
already destroyed it. (issue #15109) Reported by: jvandal (issue
#15123) Reported by: axisinternet (issue #15195) Reported by:
amorsen (issue AST-208)
2009-06-17 19:59 +0000 [r201450] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Change the datastore traversal in
ast_do_masquerade to use a safe list traversal. It is possible
for datastore fixup functions to remove the datastore from the
list and free it. In particular, the queue_transfer_fixup in
app_queue does this. While I don't yet know of this causing any
crashes, it certainly could. Found while discussing a separate
issue with Brian Degenhardt.
2009-06-17 19:28 +0000 [r201423] David Vossel <dvossel@digium.com>
* apps/app_mixmonitor.c: StopMixMonitor race condition (not giving
up file immediately) StopMixMonitor only indicates to the
MixMonitor thread to stop writing to the file. It does not
guarantee that the recording's file handle is available to the
dialplan immediately after execution. This results in a race
condition. To resolve this, the filestream pointer is placed in a
datastore on the channel. When StopMixMonitor is called, the
datastore is retrieved from the channel and the filestream is
closed immediately before returning to the dialplan.
Documentation indicating the use of StopMixMonitor to free files
has been updated as well. (closes issue #15259) Reported by:
travisghansen Tested by: dvossel Review:
https://reviewboard.asterisk.org/r/283/
2009-06-17 18:45 +0000 [r201380] David Brooks <dbrooks@digium.com>
* channels/chan_sip.c: Checks for NULL sip_pvt pointer in
chan_sip.c->acf_channel_read() Zombie channels could be passed,
and chan_sip.c wasn't checking for it. Could crash Asterisk. Now
checking for NULL pointer. (closes issue #15330) Reported by:
okrief Tested by: dbrooks
2009-06-17 12:03 +0000 [r200991-201261] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/linkedlists.h: Correct AST_LIST_APPEND_LIST
behavior when list to be appended is empty. When the list to be
appended is empty, and the list to be appended to is *not*,
AST_LIST_APPEND_LIST would actually cause the target list to
become broken, and no longer have a pointer to its last entry.
This patch fixes the problem. (reported by Stanislaw Pitucha on
the asterisk-dev mailing list)
* apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c,
build_tools/cflags-devmode.xml, main/autoservice.c, main/frame.c,
apps/app_meetme.c, main/slinfactory.c,
include/asterisk/linkedlists.h, main/file.c,
include/asterisk/channel.h, include/asterisk/frame.h: Improve
support for media paths that can generate multiple frames at
once. There are various media paths in Asterisk (codec
translators and UDPTL, primarily) that can generate more than one
frame to be generated when the application calling them expects
only a single frame. This patch addresses a number of those
cases, at least the primary ones to solve the known problems. In
addition it removes the broken TRACE_FRAMES support, fixes a
number of bugs in various frame-related API functions, and cleans
up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
2009-06-16 13:25 +0000 [r200875] Eliel C. Sardanons <eliels@gmail.com>
* res/res_smdi.c: Show the interface name on error, if it is not
found. If the smdiport specified is not found, show the interface
name instead of '(null)'.
2009-06-15 15:21 +0000 [r200513] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add INFO to our allowed methods so that
endpoints know they may send it to us. AST-223
2009-06-12 19:06 +0000 [r200360] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Suppress a warning message and give a better
return code when generating inband ringing after a call is
answered. (closes issue #15158) Reported by: madkins Patches:
15158.patch uploaded by mmichelson (license 60) Tested by:
madkins
2009-06-11 22:20 +0000 [r200185] Sean Bright <sean.bright@gmail.com>
* Makefile: Backport fix for parallel build warnings from trunk
r199781.
2009-06-11 12:12 +0000 [r200037] Leif Madsen <lmadsen@digium.com>
* build_tools/make_version_h: Fix path for .flavor and .version.
(issue #14737) Reported by: davidw Patches: flavor.patch uploaded
by davidw (license 780) Tested by: davidw
2009-06-10 16:08 +0000 [r199856] Sean Bright <sean.bright@gmail.com>
* include/asterisk/utils.h: __WORDSIZE is not available on all
platforms, so use sizeof(void *) instead.
2009-06-09 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.26-rc2
2009-06-08 19:28 +0000 [r199626-199628] Sean Bright <sean.bright@gmail.com>
* include/asterisk/utils.h: Fix a typo in the stack size
calculation just introduced.
* include/asterisk/utils.h: Increase the size of our thread stack
on 64 bit processors. We were setting the stack size for each
thread to 240KB regardless of architecture, which meant that in
some scenarios we actually had less available stack space on 64
bit processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's
__WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of
#asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
issue #14932) Reported by: jpiszcz Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright
2009-06-05 21:19 +0000 [r199297] David Vossel <dvossel@digium.com>
* main/pbx.c: Fixes issue with hints giving unexpected results.
Hints with two or more devices that include ONHOLD gave
unexpected results. (closes issue #15057) Reported by:
p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
(license 671) pbx.c.1.4.patch uploaded by p (license 558)
devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
p_lindheimer, dvossel Review:
https://reviewboard.asterisk.org/r/254/
2009-06-04 19:00 +0000 [r199138] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Additional updates to AST-2009-001
2009-06-04 14:14 +0000 [r198957-199022] Sean Bright <sean.bright@gmail.com>
* main/asterisk.c, main/loader.c, include/asterisk.h: Safely handle
AMI connections/reload requests that occur during startup. During
asterisk startup, a lock on the list of modules is obtained by
the primary thread while each module is initialized. Issue 13778
pointed out a problem with this approach, however. Because the
AMI is loaded before other modules, it is possible for a module
reload to be issued by a connected client (via Action: Command),
causing a deadlock. The resolution for 13778 was to move
initialization of the manager to happen after the other modules
had already been lodaded. While this fixed this particular issue,
it caused a problem for users (like FreePBX) who call AMI scripts
via an #exec in a configuration file (See issue 15189). The
solution I have come up with is to defer any reload requests that
come in until after the server is fully booted. When a call comes
in to ast_module_reload (from wherever) before we are fully
booted, the request is added to a queue of pending requests. Once
we are done booting up, we then execute these deferred requests
in turn. Note that I have tried to make this a bit more
intelligent in that it will not queue up more than 1 request for
the same module to be reloaded, and if a general reload request
comes in ('module reload') the queue is flushed and we only issue
a single deferred reload for the entire system. As for how this
will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in
a deadlock. After 13778, you simply couldn't connect to the
manager during startup (which causes problems with
#exec-that-calls-AMI configuration files). I believe this is a
good general purpose solution that won't negatively impact
existing installations. (closes issue #15189) (closes issue
#13778) Reported by: p_lindheimer Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright
(license 71) Tested by: p_lindheimer, seanbright Review:
https://reviewboard.asterisk.org/r/272/
* pbx/pbx_spool.c: Fix a possible crash in pbx_spool. We were
trying to reference members of a struct that had previously been
freed. This patch makes sure that we free the struct after it has
been removed from the spooler queue. (closes issue #15072)
Reported by: garlew Patches: spool.diff uploaded by garlew
(license 376)
2009-06-03 15:49 +0000 [r198891] David Vossel <dvossel@digium.com>
* main/channel.c, res/res_features.c, include/asterisk/channel.h:
Generic call forward api, ast_call_forward() The function
ast_call_forward() forwards a call to an extension specified in
an ast_channel's call_forward string. After an ast_channel is
called, if the channel's call_forward string is set this function
can be used to forward the call to a new channel and terminate
the original one. I have included this api call in both
channel.c's ast_request_and_dial() and res_feature.c's
feature_request_and_dial(). App_dial and app_queue already
contain call forward logic specific for their application and
options. (closes issue #13630) Reported by: festr Review:
https://reviewboard.asterisk.org/r/271/
2009-06-01 20:07 +0000 [r198665] Tilghman Lesher <tlesher@digium.com>
* res/res_musiconhold.c: If using the old deprecated format, a
reload would cause the class to disappear. (closes issue #14759)
Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt
uploaded by tilghman (license 14) Tested by: lmadsen
2009-05-30 19:36 +0000 [r198370] Sean Bright <sean.bright@gmail.com>
* res/res_jabber.c: Properly terminate AMI JabberSend response
messages. The response message (either Error or Success) needs an
extra trailing \r\n after the fields to inform the client that
the message is complete. (closes issue #14876) Reported by: srt
Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
(license 71) asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73)
2009-05-30 03:42 +0000 [r198311] Russell Bryant <russell@digium.com>
* res/res_smdi.c: Fix a crash that occurred when MWI SMDI messages
expired. (closes issue #14561) Reported by: cmoss28
2009-05-30 02:46 +0000 [r198251] Sean Bright <sean.bright@gmail.com>
* apps/app_dial.c: Treat an empty FORWARD_CONTEXT the same way we
treat a missing one. (closes issue #15056) Reported by:
p_lindheimer Patches: 05292009_bug15056.diff uploaded by
seanbright (license 71) Tested by: p_lindheimer
2009-05-29 18:53 +0000 [r198068] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, main/channel.c, res/res_features.c,
include/asterisk/cdr.h: Use AST_CDR_NOANSWER instead of
AST_CDR_NULL as the default CDR disposition. This change also
involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is
used on originated channels to distinguish: them from dialed
channels. (closes issue #12946) Reported by: meral Patches:
null-cdr2.diff uploaded by mnicholson (license 96) Tested by:
mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested
by: sum
2009-05-29 18:14 +0000 [r197998] Sean Bright <sean.bright@gmail.com>
* Makefile: Fix 'make config' target for Slackware. There was a
missing semi-colon after the echo statement in the Makefile that
was causing problems for some users. Fix suggested by reporter.
(closes issue #15225) Reported by: pdavis
2009-05-28 23:57 +0000 [r197895] Leif Madsen <lmadsen@digium.com>
* apps/app_mixmonitor.c: Update MixMonitor documentation. Updated
the MixMonitor documentation for the 'b' option so that it is
more obvious that you must not optimize awat the Local channel
when using this option. (issue #14829)
2009-05-28 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.26-rc1
2009-05-28 15:51 +0000 [r197620] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: 'iax show peer blah' now outputs whether or
not peer 'blah' is in trunk mode or not.
2009-05-28 15:27 +0000 [r197588] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow
for media to arrive from an alternate source when responding to a
reinvite with 491. When we receive a SIP reinvite, it is possible
that we may not be able to process the reinvite immediately since
we have also sent a reinvite out ourselves. The problem is that
whoever sent us the reinvite may have also sent a reinvite out to
another party, and that reinvite may have succeeded. As a result,
even though we are not going to accept the reinvite we just
received, it is important for us to not have problems if we
suddenly start receiving RTP from a new source. The fix for this
is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP
layer so that it will know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
2009-05-28 15:21 +0000 [r197562] Eliel C. Sardanons <eliels@gmail.com>
* channels/chan_sip.c: Use the address we already know when
reloading a peer with nat=yes. If we already have an address for
a peer, and we are reloading the sip configuration, try to use
that address to contact the peer, instead of getting it from the
Contact. (closes issue #15194) Reported by: ibc Patches:
sip.patch uploaded by eliel (license 64) Tested by: manwe
2009-05-28 14:49 +0000 [r197537] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c, include/asterisk/audiohook.h,
main/audiohook.c: Add flags to chanspy audiohook so that audio
stays in sync. There are two flags being added to the chanspy
audiohook here. One is the pre-existing
AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
the read and write slinfactories on the audiohook do not skew
beyond a certain tolerance. In addition, there is a new audiohook
flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
we do not allow for a slinfactory to build up a substantial
amount of audio before flushing it. For this particular issue,
this means that the person spying on the call will hear the
conversations in real time with very little delay in the audio.
(closes issue #13745) Reported by: geoffs Patches: 13745.patch
uploaded by mmichelson (license 60) Tested by: snblitz
2009-05-28 13:44 +0000 [r197466] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where the flag indicating the
presence of rport would get overwritten by the nat setting. The
presence of rport is now stored as a separate flag. Once the
dialog is setup and authenticated (or it passes through
unauthenticated) the proper nat flag is set. (closes issue
#13823) Reported by: dimas
2009-05-27 20:12 +0000 [r197264] Sean Bright <sean.bright@gmail.com>
* Makefile: Use bash explicitly when calling
build_tools/mkpkgconfig from the Makefile. Since we use bashisms
in build_tools/mkpkgconfig, we should call on bash explicitly
when running from the Makefile, otherwise we get errors during a
'make install.' (closes issue #15209) Reported by: seandarcy
2009-05-27 20:07 +0000 [r197259] Olle Johansson <oej@edvina.net>
* doc/asterisk-conf.txt: Typo fix
2009-05-27 19:09 +0000 [r197194] Tilghman Lesher <tlesher@digium.com>
* funcs/func_cut.c: Use a different determinator on whether to
print the delimiter, since leading fields may be blank. (closes
issue #15208) Reported by: ramonpeek Patch by me, though inspired
in part by a patch from ramonpeek
2009-05-27 16:49 +0000 [r197124] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, include/asterisk/channel.h: Fix broken attended
transfers The bridge was terminating immediately after the
attended transfer was completed. The problem was because upon
reentering ast_channel_bridge nexteventts was checked to see if
it was set and if so could possibly return AST_BRIDGE_COMPLETE.
(closes issue #15183) Reported by: andrebarbosa Tested by:
andrebarbosa, tootai, loloski
2009-05-27 13:54 +0000 [r197024] Sean Bright <sean.bright@gmail.com>
* apps/app_queue.c: Fix handling of the 'state_interface' option of
the 'queue add member' CLI command. This change relates to
r184980, which was a backport of the state interface changes to
app_queue from trunk. trunk and all of the 1.6.x branches are not
affected. 'queue add member' allows for specifying an interface
to use for device state when adding a queue member via CLI, but
the validation code was not properly updated to reflect this
optional argument. (closes issue #15198) Reported by: loloski
Patches: 05272009_app_queue.diff uploaded by seanbright (license
71) Tested by: loloski
2009-05-26 18:14 +0000 [r196826] Russell Bryant <russell@digium.com>
* res/res_convert.c: Resolve a file handle leak. The frames here
should have always been freed. However, out of luck, there was
never any memory leaked. However, after file streams became
reference counted, this code would leak the file stream for the
file being read. (closes issue #15181) Reported by: jkroon
2009-05-26 13:06 +0000 [r196657] Joshua Colp <jcolp@digium.com>
* contrib/scripts/safe_asterisk: Remove some bash specific stuff
from safe_asterisk. (closes issue #10812) Reported by: paravoid
Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license
46)
2009-05-22 13:54 +0000 [r196116] Joshua Colp <jcolp@digium.com>
* channels/chan_misdn.c: Fix a bug where using immediate with mISDN
caused a cause code of 16 to get sent back instead of 1 if the
's' extension did not exist. (closes issue #12286) Reported by:
lmamane
2009-05-21 19:04 +0000 [r195991] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Sign problem calculating timestamp for iax
frame leads to no audio on the receiving peer. There are rare
cases in which a frame's delivery timestamp is slightly less than
the iax2_pvt's offset. This causes the pvt's timestamp to be a
small negative number, but since the timestamp value is unsigned
it looks like a huge positive number. This patch checks for this
negative case and sets the ms to zero. A similar check is already
done right below this one in the 'else' statement. (closes issue
#15032) Reported by: guillecabeza Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
380) Tested by: guillecabeza (closes issue #14216) Reported by:
Andrey Sofronov
2009-05-21 15:25 +0000 [r195881] Matthew Nicholson <mnicholson@digium.com>
* main/cdr.c, res/res_features.c, include/asterisk/cdr.h: This
commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and
AST_CDR_FLAG_LOCKED from being updated in certain cases. This is
accomplished by adding two functions to update the answer time
and disposition of calls that checks for the proper lock flags.
These functions are used in the ast_bridge_call() function so
that ForkCDR(A) calls are respected. This patch also modifies the
way ast_bridge_call() chooses the cdr record to base the
bridged_cdr on. Previously the first unlocked cdr record would be
chosen, now instead the first cdr record is chosen and forked cdr
records are moved to the bridge_cdr. This allows the original cdr
record and any forked cdr records to be properly updated with
answer and end times. (closes issue #13797) Reported by: sh0t
Tested by: sh0t (closes issue #14744) Reported by: deepesh
2009-05-21 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.25
2009-05-13 Leif Madsen <lmadsen@digium.com>
* Release Asterisk 1.4.25-rc1
2009-05-13 13:38 +0000 [r194208] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Fix RFC2833 issues with DTMF getting duplicated and
with duration wrapping over. (closes issue #14815) Reported by:
geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
#14460) Reported by: moliveras Tested by: moliveras
2009-05-13 00:52 +0000 [r194137] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Fix logic for how to proceed with a single digit
extension. (closes issue #15091) Reported by: andrew Patches:
20090512__issue15091.diff.txt uploaded by tilghman (license 14)
Tested by: andrew
2009-05-12 22:15 +0000 [r194028] Matthew Nicholson <mnicholson@digium.com>
* apps/app_queue.c: This change modifies app_queue to properly
generate CDR records in failure situations. This involves setting
a proper cdr disposition coresponding to the given failure
condition and ensuring the proper information is stored in the
cdr record. (closes issue #13691) Reported by: dferrer Tested by:
mnicholson (closes issue #13637) Reported by: atis Tested by:
atis
2009-05-12 20:39 +0000 [r193955] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Avoid initializing routines if the
authentication fails. Fixes a crash (RR) issue. (closes issue
#14508) Reported by: tiziano Patches:
20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
377)
2009-05-12 18:18 +0000 [r193880] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Set the invitestate to INV_CANCELLED only if
we are actually sending a SIP CANCEL. The problem was that the
hangup code was setting the invitestate too early. The result of
this was that we would always send a CANCEL request, even if it
was not an appropriate time to do so (e.g. we have not yet
received a provisional response for our INVITE). Note that this
same fix had been applied to trunk and the 1.6.X branches
starting with revision 155467. This is why you will see this
revision being blocked from those places. AST-216
2009-05-11 22:48 +0000 [r193755] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Move 300 bytes around on the stack, to make
more room for an extension buffer. This allows more concurrent
extensions to be copied for a single voicemail, without creating
a possibility of upsetting existing users, where a dialplan could
run out of stack space where it had run fine before.
Alternatively, we could have allocated off the heap, but that is
a larger change and would have increased the chance for
instability introduced by this change. This is really solved
starting in 1.6.0.11, as the use of an ast_str buffer allows an
unlimited number of extensions (up to available memory). We
additionally create a new warning message when the buffer length
is exceeded, permitting administrators to see an issue after the
fact, whereas previously the list was silently truncated. (closes
issue #14739) Reported by: p_lindheimer Patches:
20090417__bug14739.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
2009-05-11 19:09 +0000 [r193613] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Sent wrong message to clear a call we
started if the other end has not responed yet. In the state
MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
yet), it is not allowed to clear the call with RELEASE_COMPLETE.
It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
2009-05-11 17:35 +0000 [r193544] Leif Madsen <lmadsen@digium.com>
* funcs/func_channel.c: Document CHANNEL(transfercapability) in CLI
documentation. (issue #15073) Reported by: pkempgen Patches:
20090511__issue15073.diff.txt uploaded by tilghman (license 14)
2009-05-08 21:01 +0000 [r193391] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c: Set the proper disposition on originated calls.
(closes issue #14167) Reported by: jpt Patches:
call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
Tested by: dlotina, rmartinez, mnicholson
2009-05-08 14:51 +0000 [r193262] David Vossel <dvossel@digium.com>
* channels/misdn_config.c: "misdn show config" segfaults asterisk,
if no MSN lists (closes issue #14976) Reported by: alecdavis
Patches: misdn_config.diff.txt uploaded by alecdavis (license
585) Tested by: alecdavis, FabienToune
2009-05-08 14:03 +0000 [r193193] Kevin P. Fleming <kpfleming@digium.com>
* configs/logger.conf.sample, main/logger.c: Make absolute paths
for logger channels work properly (Note: This is not a new
feature, it was previously undocumented and broken.) The Asterisk
logger has a feature to support absolute pathnames for logger
channels, but the code implementing the feature was broken. This
has been fixed, and the absolute path feature is now documented
in the sample logger.conf.
2009-05-07 23:41 +0000 [r193119] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Fix Background within a Macro for FreePBX. If the
single digit DTMF is an extension in the specified context, then
go there and signal no DTMF. Otherwise, we should exit with that
DTMF. If we're in Macro, we'll exit and seek that DTMF as the
beginning of an extension in the Macro's calling context. If
we're not in Macro, then we'll simply seek that extension in the
calling context. Previously, someone complained about the
behavior as it related to the interior of a Gosub routine, and
the fix (#14011) inadvertently broke FreePBX (#14940). This
change should fix both of these situations, but with the possible
incompatibility that if a single digit extension does not exist
(but a longer extension COULD have matched), it would have
previously gone immediately to the "i" extension, but will now
need to wait for a timeout. (closes issue #14940) Reported by:
p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
tilghman (license 14) Tested by: p_lindheimer
2009-05-07 22:17 +0000 [r193050] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Give a more helpful message when an
incoming call's dialed extension does not match. Added the dialed
extension and context to the chan_misdn messages warning that the
dialed number cannot be matched in the dialplan.
2009-05-07 16:29 +0000 [r192932] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Eliminate repetition of fullcontact during
reconstruction. If the fullcontact field appears in both the
sippeers and the sipregs table, then during reconstruction of the
field, it will otherwise be doubled. (closes issue #14754)
Reported by: Alexei Gradinari Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
2009-05-06 22:15 +0000 [r192858] Jeff Peeler <jpeeler@digium.com>
* res/res_features.c: Make ParkedCall application stop execution of
the dialplan after hang up Just changed park_exec to always
return non-zero. I really wasn't entirely sure at first if this
was a bug. Decided it was since it would be surprising when not
using ParkedCall in the dialplan to hang up and have dialplan
execution continue. (closes issue #14555) Reported by:
francesco_r
2009-05-06 13:30 +0000 [r192633] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Update some old logic to stop both begin and
end DTMF frames from reaching the core if rfc2833 is not enabled.
(closes issue #15036) Reported by: dimas Patches: v1-15036.patch
uploaded by dimas (license 88)
2009-05-05 19:56 +0000 [r192524] Sean Bright <sean.bright@gmail.com>
* static-http/astman.js: Fix Javascript error when using astman.js
in Internet Explorer. Internet Explorer (tested with 7.0) does
not like trailing commas on constructs like object initializers,
so get rid of them to avoid some errors. (closes issue #15026)
Reported by: rajnishgiri Patches: bug15026.patch uploaded by
seanbright (license 71) Tested by: seanbright
2009-05-05 18:22 +0000 [r192429-192454] Joshua Colp <jcolp@digium.com>
* res/res_features.c: Fix an incorrect assumption that certain
values on the channel will always exist when they may not. The
CDR code involved with bridges wrongly assumed that the currently
executing application and data values will always exist. It is
possible for this to be false when call forwarding is involved.
(closes issue #14984) Reported by: gincantalupo
* apps/app_followme.c: Fix a bug where the followme application
would continue trying numbers after the caller hung up. (closes
issue #13624) Reported by: sgenyuk
2009-05-04 22:37 +0000 [r192213] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: global mohinterpret setting is ignored
mohinterpret and mohsuggest global variables were not copied over
during build_users and build_peers. (closes issue #14728)
Reported by: dimas Patches: v1-14728.patch uploaded by dimas
(license 88) Tested by: dimas, dvossel
2009-05-02 18:48 +0000 [r191628-191778] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix a bug which resulted from the Hebrew
voicemail commit. This fixes a case where a certain message could
get played twice. (closes issue #13155) Reported by:
greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded
by greenfieldtech (license 369) Tested by: greenfieldtech
* apps/app_chanspy.c: Kevin has informed me that thi sort of thing
is not necessary.
* apps/app_chanspy.c: Move static buffers to outside for loops in
app_chanspy. Similar to seanbright's commit 191422, this moves
some static buffers to be defined outside of for loops since it
is undefined if memory will be re-used or if the stack will grow
with each iteration of the loop.
2009-05-01 20:00 +0000 [r191559] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: SIP Response 410 maps to cause code 22 (or
23), not 1. (closes issue #14993) Reported by: BigJimmy Patches:
causepatch uploaded by BigJimmy (license 371)
2009-05-01 17:40 +0000 [r191488] Jeff Peeler <jpeeler@digium.com>
* main/channel.c: Fix DTMF not being sent to other side after a
partial feature match This fixes a regression from commit 176701.
The issue was that ast_generic_bridge never exited after the
feature digit timeout had elapsed, which prevented the queued
DTMF from being sent to the other side. This issue was reported
to me directly.
2009-05-01 15:42 +0000 [r191422] Sean Bright <sean.bright@gmail.com>
* apps/app_queue.c: Move the defintion of the a couple arrays out
of loops. According to Kevin, it is unspecified as to whether a
variable defined inside a block is allocated once by the compiler
or for each pass through the block (loops being the only
interesting case), so just define these before we get into our
loop to be sure.
2009-04-29 23:10 +0000 [r191220] Tilghman Lesher <tlesher@digium.com>
* channels/h323/ast_h323.cxx, channels/chan_h323.c: Allow H.323 to
compile with FDLEAK checking enabled.
2009-04-29 18:07 +0000 [r191096] David Brooks <dbrooks@digium.com>
* pbx/pbx_config.c: Patch to fix tab-completion crash on "remove
extension" This patch simply removes some old code back before
Asterisk used editline. This fixes the crash that occurred when
tab-completing "remove extension". (closes issue #14689) Reported
by: isaacgal
2009-04-29 15:23 +0000 [r191041] Sean Bright <sean.bright@gmail.com>
* apps/app_queue.c: Fix a crash in app_queue with very long member
lists. A user reported via #asterisk that with very long lists of
members, a crash occurs in ast_strdupa, so just use a single
buffer and ast_copy_string instead of stack allocating copys of
each interface name.
2009-04-27 19:29 +0000 [r190721] Kevin P. Fleming <kpfleming@digium.com>
* configure, include/asterisk/autoconfig.h.in: Fix 'inconsistent
line endings' when autoconf 2.63 is used Attempt to make
configure script regeneration 'safe' using autoconf 2.63, which
embeds a bare CR into the script, thus making Subversion complain
about inconsistent line endings This commit changes the MIME type
of the configure script to be 'binary' thus making Subversion no
longer inspect line endings, and as a bonus 'svn diff' will no
longer try to generate diff output for it, which is not generally
useful anyway.
2009-04-27 19:03 +0000 [r190661-190662] Russell Bryant <russell@digium.com>
* res/res_smdi.c: Fix a typo from 190661.
* res/res_smdi.c: Resolve a crash in res_smdi when used with
chan_dahdi. When chan_dahdi goes to get an SMDI message, it
provides no search criteria. It just grabs the next message that
arrives. This code was written with the SMDI dialplan functions
in mind, since that is now the preferred method of using SMDI.
However, this broke support of it being used from chan_dahdi.
(closes AST-212)
2009-04-23 21:07 +0000 [r190356] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove a bogus ast_channel_unlock().
2009-04-23 19:13 +0000 [r190286] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c: Fix a bug in chan_local glare hangup
detection. If both sides of a Local channel were hung up at
around the same time it was possible for one thread to destroy
the local private structure and have the other thread immediately
try to remove the already freed structure from the local channel
list.
2009-04-23 10:07 +0000 [r190187] Olle Johansson <oej@edvina.net>
* include/asterisk/lock.h: unistd.h is required for usleep() on
Darwin. It will not hurt to include it always on other platforms
either.
2009-04-22 21:35 +0000 [r190092] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Detect availability of
pthread_rwlock_timedwrlock() before using it. (closes issue
#14930) Reported by: tilghman Patches:
20090420__bug14930.diff.txt uploaded by tilghman (license 14)
Tested by: mvanbaak, tilghman
2009-04-22 19:20 +0000 [r189991] Jeff Peeler <jpeeler@digium.com>
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
channels/h323/chan_h323.h: Make chan_h323 respect packetization
settings Previously, packetization settings were ignored and now
they are not. A new config option 'autoframing' has been added to
mirror the way chan_sip handles it. Turning on the autoframing
option (available both as a global option or per peer) overrides
the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the
following codecs: ulaw, alaw, gsm, and g729. (closes issue
#12415) Reported by: pj Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak
(license 7), modified by me
2009-04-22 14:29 +0000 [r189849] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/get_ilbc_source.sh: replace sed with tr to remove
\r from downloaded file On some systems, sed does not recognize
\r in the pattern the way it was used here. Use tr instead
because this works the same across systems. (closes issue #14936)
Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded
by mvanbaak (license 7) Tested by: leobrown, mvanbaak
2009-04-21 15:52 +0000 [r189601-189664] Doug Bailey <dbailey@digium.com>
* utils/muted.c: Remove daemon call on systems that do not support
forking.
* main/config.c, configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, configure.ac: Add check in configure
script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This
allows config.c to compile when linked against uclibc that does
not support these parameters
2009-04-20 22:02 +0000 [r189537] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c, funcs/func_strings.c: Add a workaround for
func_odbc/ARRAY() for problems that occur with certain special
characters. In certain cases, due to the way Set() works in 1.4,
values may not get set properly. This is a workaround for 1.4
only that corrects for these issues, without making func_odbc
more difficult to use properly. (closes issue #14614) Reported
by: wdoekes Patches: 20090309__bug14614__2.diff.txt uploaded by
tilghman (license 14)
double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff
uploaded by wdoekes (license 717) Tested by: wdoekes, tilghman
2009-04-20 21:10 +0000 [r189463-189465] Terry Wilson <twilson@digium.com>
* apps/app_dial.c: Update CDR appropriately when
AST_CAUSE_NO_ANSWER is set
* apps/app_dial.c: Don't treat a NOANSWER like a CHANUNAVAIL
2009-04-20 20:58 +0000 [r189462] Sean Bright <sean.bright@gmail.com>
* pbx/ael/ael.tab.c, pbx/ael/ael.y: Properly handle @s within hints
in AEL. AEL was not handling the case of a device hint containing
an @ symbol, which caused parking hints (e.g.
hint(park:exten@context)) to error out the parser. This patch
makes AEL treat the @ the same way it treats colon and ampersand
now, meaning the characters are included in verbatim. (closes
issue #14941) Reported by: bpgoldsb Patches: bug14941.patch
uploaded by seanbright (license 71) Tested by: bpgoldsb
2009-04-20 19:10 +0000 [r189391] Doug Bailey <dbailey@digium.com>
* main/manager.c, main/db1-ast/recno/rec_open.c,
channels/chan_iax2.c: Clean up problem with manager
implementation of mmap where it was not testing against
MAP_FAILED response. Got rid of shadowed variable used in
processign the mmap results. Change test of mmap results to
compare against MAP_FAILED
2009-04-20 14:04 +0000 [r189277] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Move the check for chan->fdno == -1 to after the
zombie/hangup check. Many users were finding that their hung up
channels were staying up and causing 100% CPU usage. (issue
#14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
uploaded by mmichelson (license 60) Tested by: falves11, bamby
2009-04-18 01:27 +0000 [r189203] David Vossel <dvossel@digium.com>
* channels/chan_agent.c: Fixed autologoff in agents.conf not
working when agent logs in via AgentLogin app An agent logs in by
calling an extension that calls the AgentLogin app. In
agents.conf ackcall=always is set, so when they get a call they
have the choice to either acknowledge it or ignore it.
autologoff=10 is set as well, so if the agent ignores the call
over 10sec one may assume that the agent should be logged out
(and in this case hungup on as well), but this was not happening.
(closes issue #14091) Reported by: evandro Patches:
autologoff.diff uploaded by dvossel (license 671) Review:
http://reviewboard.digium.com/r/225/
2009-04-17 21:27 +0000 [r189134] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c: Modifed/added some debug messages.
JIRA ABE-1835
2009-04-17 15:43 +0000 [r189009] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c: Make Busy() application set the CDR disposition to
BUSY. (closes issue #14306) Reported by: cristiandimache
2009-04-17 14:41 +0000 [r188937-188946] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix a bug where a value used to create the
channel name was bogus. This commit fixes the scenario where an
incoming call is authenticated using a peer entry. Previously the
channel name was created using either the username setting from
the sip.conf entry or the IP address that the call came from. Now
the channel name will be created using the peer name itself. This
commit will not change the way the channel name is generated for
users or friends. (closes issue #14256) Reported by: Nick_Lewis
Patches: chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
* channels/chan_dahdi.c: Fix a situation where the DAHDI channel
private structure lock was not unlocked when it should have been.
(issue AST-210)
2009-04-16 21:41 +0000 [r188835] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Only update realtime, if global option
rtupdate != false (closes issue #14885) Reported by: deepesh
Patches: 20090413__bug14885.diff.txt uploaded by tilghman
(license 14) Tested by: deepesh
2009-04-16 21:37 +0000 [r188833] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Only disable mISDN DSP if Asterisk DSP is
enabled. Leave jitter setting alone. JIRA ABE-1835
2009-04-16 21:02 +0000 [r188773] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Umask should not be exported into global
namespace. (closes issue #14912) Reported by: jcapp
2009-04-15 22:08 +0000 [r188646] David Vossel <dvossel@digium.com>
* channels/chan_dahdi.c: National prefix inserted even when caller
ID not available When the caller ID is restricted, the expected
behavior is for the caller id to be blank. In chan_dahdi, the
national prefix is placed onto the callers number even if its
restricted (empty) causing the caller id to be the national
prefix rather than blank. (closes issue #13207) Reported by:
shawkris Patches: national_prefix.diff uploaded by dvossel
(license 671) Review: http://reviewboard.digium.com/r/220/
2009-04-15 20:04 +0000 [r188582] Mark Michelson <mmichelson@digium.com>
* main/file.c: Update ast_readvideo_callback to match
ast_readaudio_callback. This fixes potential refcount errors that
may occur on ast_filestreams. AST-208
2009-04-14 15:02 +0000 [r188287] David Vossel <dvossel@digium.com>
* main/audiohook.c: audio_audiohook_write_list() does not correctly
update sample size after ast_translate.
audio_audiohook_write_list() does not take into account that the
sample size may change after translation depending on if the
original frame is is 8khz or 16khz. While no 16kz codecs are
supported in 1.4 at the moment, this will save headaches in the
future if they ever are. the sample size is now updated after
translating to reflect this possibility. Thanks to jcolp and
mmichelson for helping me work this out. (issue AST-197)
2009-04-13 23:04 +0000 [r188149] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: If fileconfig limit exceeds our maximum, then set
the limit to the maximum. (Closes issue #14888) Reported by:
falves11
2009-04-10 22:16 +0000 [r187962] Jeff Peeler <jpeeler@digium.com>
* channels/Makefile: Fix module embedding for chan_h323. Include
libchanh323.a in the modules.link file so that all the symbols
can be resolved at link time. (closes issue #11966) Reported by:
dome
2009-04-10 19:26 +0000 [r187865] Russell Bryant <russell@digium.com>
* channels/chan_dahdi.c: Support "signaling" in addition to
"signalling". The sample configuration file has references to
both spellings.
2009-04-10 17:28 +0000 [r187763] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/realtime_pgsql.sql,
contrib/scripts/sip-friends.sql: Add lastms column to the
contributed table designs
2009-04-09 18:51 +0000 [r187484] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Handle a SIP race condition (reinvite before
an ACK) properly. RFC 5047 explains the proper course of action
to take if a reINVITE is received before the ACK from a previous
invite transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of the
sip_pvt representing this dialog. (closes issue #13849) Reported
by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson
(license 60) Tested by: mmichelson, klaus3000
2009-04-09 18:39 +0000 [r187209-187482] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Oops, typo
* main/manager.c, include/asterisk/lock.h: Race condition between
ast_cli_command() and 'module unload' could cause a deadlock. Add
lock timeouts to avoid this potential deadlock. (closes issue
#14705) Reported by: jamessan Patches:
20090320__bug14705.diff.txt uploaded by tilghman (license 14)
Tested by: jamessan
* channels/chan_sip.c, apps/app_sendtext.c: Permit zero-length text
messages in SIP. (Related to an issue posted to the -users list,
subject "AEL2, BASE64_DECODE and hexadecimal")
* main/astfd.c (added): Oops, missed this file in the last commit.
* main/asterisk.c, agi/Makefile, build_tools/cflags.xml,
utils/Makefile, include/asterisk.h, main/Makefile, main/file.c:
Add debugging mode for diagnosing file descriptor leaks. (Related
to issue #14625)
* main/manager.c: Backport resolution for file descriptor leak in
1.6.0 to 1.4. This fixes short reads in http manager sessions,
such as those done by the ast-gui branch. (Fixes AST-198)
2009-04-08 19:16 +0000 [r186832-187135] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Fix a crash due to too few arguments to
RetryDial. (closes issue #14852) Reported by: junky Patches:
retry_fix.diff uploaded by junky (license 177)
* res/res_musiconhold.c: Fix a small logical error when loading moh
classes. We were unconditionally incrementing the number of
mohclasses registered. However, we should actually only increment
if the call to moh_register was successful. While this probably
has never caused problems, I noticed it and decided to fix it
anyway.
* main/channel.c: Make a couple of changes with regards to a new
message printed in ast_read(). "ast_read() called with no
recorded file descriptor" is a new message added after a bug was
discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this
error message to be displayed. This commit does two things to
help to make this message appear less. First, the message has
been downgraded to a debug level message if dev mode is not
enabled. The message means a lot more to developers than it does
to end users, and so developers should take an effort to be sure
to call ast_read only when a channel is ready to be read from.
However, since this doesn't actually cause an error in operation
and is not something a user can easily fix, we should not spam
their console with these messages. Second, the message has been
moved to after the check for any pending masquerades. ast_read()
being called with no recorded file descriptor should not
interfere with a masquerade taking place. This could be seen as a
simple way of resolving issue #14723. However, I still want to
try to clear out the existing ways of triggering this message,
since I feel that would be a better resolution for the issue.
* formats/format_wav.c, formats/format_wav_gsm.c: Fix a few typos
of the word "frequency." (closes issue #14842) Reported by:
jvandal Patches: frequency-typo.diff uploaded by jvandal (license
413)
* main/channel.c: Set the AST_FEATURE_WARNING_ACTIVE flag when a
p2p bridge returns AST_BRIDGE_RETRY. Without this flag set,
warning sounds will not be properly played to either party of the
bridge. (closes issue #14845) Reported by: adomjan
2009-04-07 22:16 +0000 [r186775] Tilghman Lesher <tlesher@digium.com>
* apps/app_macro.c: Fix Macro documentation to match current (and
intended) behavior. (See -dev mailing list)
2009-04-07 20:43 +0000 [r186719] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Ensure that \r\n is printed after the ActionID in
an OriginateResponse. (closes issue #14847) Reported by: kobaz
2009-04-06 13:54 +0000 [r186565] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Revert commit 186445 because it causes the
build to fail when IMAP_STORAGE is used.
2009-04-03 20:19 +0000 [r186458] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: Fix a bug where DAHDI/Zaptel channels
would not properly switch formats when requested Don't offer
AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could
provide a slight performance benefit, the translation core in
Asterisk has some flaws when a channel driver offers multiple raw
formats. this fix is much simpler than fixing the translation
core to solve that issue (although that will be done later).
2009-04-03 19:56 +0000 [r186415-186445] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Found a conflict in the last commit, due to
multiple targets
* apps/app_voicemail.c, configs/voicemail.conf.sample: Distinguish
in a sent email between simple sends and forwards. (closes issue
#11678) Reported by: jamessan Patches:
20090330__bug11678.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, lmadsen
2009-04-03 15:48 +0000 [r186320] Joshua Colp <jcolp@digium.com>
* include/asterisk/crypto.h: Fix a problem with the crypto variable
definitions not actually being defined properly. (closes issue
#14804) Reported by: jvandal
2009-04-03 01:57 +0000 [r186229] Russell Bryant <russell@digium.com>
* cdr/cdr_radius.c: Fix a memory leak in cdr_radius. I came across
this while doing some testing of my ast_channel_ao2 branch. After
running a test overnight that generated over 5 million calls,
Asterisk had taken up about 1 GB of my system memory. So, I
re-ran the test with MALLOC_DEBUG turned on. However, it showed
no leaks in Asterisk during the test, even though Asterisk was
still consuming it somehow. Instead, I turned to valgrind, which
when run with --leak-check=full, told me exactly where the leak
came from, which was from allocations inside the radiusclient-ng
library. This explains why MALLOC_DEBUG did not report it. After
a bit of analysis, I found that we were leaking a little bit of
memory every time a CDR record was passed to cdr_radius. I don't
actually have a radius server set up to receive CDR records.
However, I always have my development systems compile and install
all modules. In addition to making sure there are not build
errors across modules, always loading modules helps find bugs
like this, too, so it is strongly recommend for all developers.
2009-04-02 21:55 +0000 [r186174] Mark Michelson <mmichelson@digium.com>
* configs/features.conf.sample: Fix instructions in one-step
parking comment to make more sense. Changed a capital K to a
lowercase k.
2009-04-02 17:21 +0000 [r186081] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: ensure that the buffer passed to
DAHDI_SET_BUFINFO is fully initialized
2009-04-02 17:09 +0000 [r186057-186059] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions
186056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009)
| 2 lines Fix for AST-2009-003 ........
* channels/chan_sip.c: Avoid multiple warning messages in SIP, due
to this column not existing
2009-04-02 13:43 +0000 [r185952] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: the DAHDI_GETCONF, DAHDI_SETCONF and
DAHDI_GET_PARAMS ioctls were recently corrected to show that they
do, in fact, read data from userspace as part of their work. due
to this fix, valgrind now reports a number of cases where
chan_dahdi passed an uninitialized (or partially) buffer to these
ioctls, which could lead to unexpected behavior. this patch
corrects chan_dahdi to ensure that buffers passed to these ioctls
are always fully initialized.
2009-04-01 19:02 +0000 [r185845] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Fixes issue with dropped calles due to
re-Invite glare and re-Invites never executing after a 491
Acknowledgement for 491 responses were never being processed
because it didn't match our pending invite's seqno. Since the ACK
was never processed, the 491 frame would continue to be
retransmitted until eventually the call was dropped due to max
retries. Now during a pending invite, if we receive another
invite, we send an 491 and hold on to that glare invite's seqno
in the "glareinvite" variable for that sip_pvt struct. When ACK's
are received, we first check to see if it is in response to our
pending invite, if not we check to see if it is in response to a
glare invite. In this case, it is in response to the glare invite
and must be dealt with or the call is dropped. I've changed the
wait time for resending the re-Invite after receving a 491
response to comply with RFC 3261. Before this patch the scheduled
re-Invite would only change a flag indicating that the re-Invite
should be sent out, now it actually sends it out as well. (closes
issue #12013) Reported by: alx Review:
http://reviewboard.digium.com/r/213/
2009-04-01 13:47 +0000 [r185771] Russell Bryant <russell@digium.com>
* main/channel.c: Fix a case where DTMF could bypass audiohooks.
This change fixes a situation where an audiohook that wants DTMF
would not actually get it. This is in the code path where we end
DTMF digit length emulation while handling a NULL frame.
2009-03-31 22:00 +0000 [r185468-185599] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix crash that would occur if an empty member
was specified in queues.conf. (closes issue #14796) Reported by:
pida
* channels/chan_sip.c: Use AST_SCHED_DEL_SPINLOCK instead of
manually using the logic.
* apps/app_voicemail.c: Fix Russian voicemail intro to say the word
"messages" properly. (closes issue #14736) Reported by: chappell
Patches: voicemail_no_messages.diff uploaded by chappell (license
8)
2009-03-31 16:37 +0000 [r185362] David Brooks <dbrooks@digium.com>
* channels/chan_gtalk.c: Fix incorrect parsing in chan_gtalk when
xmpp contains extra whitespaces To drill into the xmpp to find
the capabilities between channels, chan_gtalk calls iks_child()
and iks_next(). iks_child() and iks_next() are functions in the
iksemel xml parsing library that traverse xml nodes. The bug here
is that both iks_child() and iks_next() will return the next
iks_struct node *regardless* of type. chan_gtalk expects the next
node to be of type IKS_TAG, which in most cases, it is, but in
this case (a call being made from the Empathy IM client), there
exists iks_struct nodes which are not IKS_TAG data (they are
extraneous whitespaces), and chan_gtalk doesn't handle that case,
so capabilities don't match, and a call cannot be made.
iks_first_tag() and iks_next_tag(), on the other hand, will not
return the very next iks_struct, but will check to see if the
next iks_struct is of type IKS_TAG. If it isn't, it will be
skipped, and the next struct of type IKS_TAG it finds will be
returned. This assures that chan_gtalk will find the iks_struct
it is looking for. This fix simply changes all calls to
iks_child() and iks_next() to become calls to iks_first_tag() and
iks_next_tag(), which resolves the capability matching. The
following is a payload listing from Empathy, which, due to the
extraneous whitespace, will not be parsed correctly by iksemel:
<iq from='dbrooksjab@235-22-24-10/Telepathy'
to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'>
<session xmlns='http://www.google.com/session'
initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate'
id='1837267342'> <description
xmlns='http://www.google.com/session/phone'> <payload-type
clockrate='16000' name='speex' id='96'/> <payload-type
clockrate='8000' name='PCMA' id='8'/> <payload-type
clockrate='8000' name='PCMU' id='0'/> <payload-type
clockrate='90000' name='MPA' id='97'/> <payload-type
clockrate='16000' name='SIREN' id='98'/> <payload-type
clockrate='8000' name='telephone-event' id='99'/> </description>
</session> </iq>
2009-03-31 15:34 +0000 [r185298] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix some state_interface stuff that was in
trunk but not in the backport to 1.4. Issue #14359 was fixed
between the time that I posted the review of the backport of the
state interface change for 1.4. This merges the changes from that
issue back into 1.4. (closes issue #14359) Reported by:
francesco_r
2009-03-31 14:06 +0000 [r185196] Joshua Colp <jcolp@digium.com>
* main/audiohook.c: Fix crash when moving audiohooks between
channels. Handle the scenario where we are called to move
audiohooks between channels and the source channel does not
actually have any on it. (closes issue #14734) Reported by:
corruptor
2009-03-30 20:40 +0000 [r185120-185121] Richard Mudgett <rmudgett@digium.com>
* channels/misdn_config.c, configs/misdn.conf.sample: Update the
channel allocation method documentation.
* channels/misdn/isdn_lib.c: Make chan_misdn BRI TE side normally
defer channel selection to the NT side. Channel allocation
collisions are not handled by chan_misdn very well. This patch
simply avoids the problem for BRI only. For PRI, allocation
collisions are still possible but less likely since there are
simply more channels available and each end could use a different
allocation strategy. misdn.conf options available:
te_choose_channel - Use to force the TE side to allocate
channels. method - Specify the channel allocation strategy.
(closes issue #13488) Reported by: Christian_Pinedo Patches:
isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes,
festr
2009-03-30 16:17 +0000 [r184980-185031] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix queue weight behavior so that calls in
low-weight queues are not inappropriately blocked. (This is
copied and pasted from the review request I made for this patch)
Asterisk has some odd behavior when queue weights are used. The
current logic used when potentially calling a queue member is: If
the member we are going to call is part of another queue and
_that other queue has any callers in it_ and has a higher weight
than the queue we are calling from, then don't try to contact
that member. The issue here is what I have marked with
underscores. If the higher-weighted queue has any callers in it
at all, then the queue member will be unreachable from the
lower-weighted queue. This has the potential to be really really
bad if using a queue strategy, such as leastrecent or
fewestcalls, with the potential to call the same member
repeatedly. The fix proposed by garychen on issue 13220 is very
simple and, as far as I can see, works well for this situation.
With this set of changes, the logic used becomes: If the member
we are going to call is part of another queue, the other queue
has a higher weight than the queue we are calling from, and the
higher weight queue has at least as many callers as available
members, then do not try to contact the queue member. If the
higher weighted queue has fewer callers than available members,
then there is no reason to deny the call to this member since the
other queue can afford to spare a member. Since the fix involved
writing a generic function for determining the number of
available members in the queue, I also modified the is_our_turn
function to make use of the new num_available_members function to
determine if it is our turn to try calling a member. There is one
small behavior change. Before writing this patch, if you had
autofill disabled, then if you were the head caller in a queue,
you would automatically be told that it was your turn to try
calling a member. This did not take into account whether there
were actually any queue members available to take the call. Now
we actually make sure there is at least one member available to
take the call if autofill is disabled. (closes issue #13220)
Reported by: garychen Review:
http://reviewboard.digium.com/r/202/
* configs/queues.conf.sample, apps/app_queue.c: Backport state
interface changes to app_queue from trunk. After several issues
raised on the Asterisk bugtracker against the 1.4 branch were
determined to be fixable with the state interface change
available in the 1.6.X series, it finally came time to just suck
it up and backport the change. For a detailed explanation of what
this change entails, the original trunk commit for this feature
may be found here:
http://svn.digium.com/view/asterisk?view=revision&revision=97203
In addition, the details for the use of this change to fix the
problems stated in issue #12970 may be found in the review
request I made for this change. It is linked below. (closes issue
#12970) Reported by: edugs15 Review:
http://reviewboard.digium.com/r/116
2009-03-30 14:35 +0000 [r184947] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Improve our handling of T38 in the initial
INVITE from a device. We now answer with matching media streams
to what is requested. If an INVITE is received with both a T38
and RTP media stream this means we answer with both. For any
outgoing calls created as a result of this inbound one no T38 is
requested in the initial INVITE. Instead if we start receiving
udptl packets we trigger a reinvite on the outbound side. (closes
issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief,
file, afu Review: http://reviewboard.digium.com/r/208/
2009-03-29 05:51 +0000 [r184842] Russell Bryant <russell@digium.com>
* apps/app_followme.c: Ensure targs variable is fully initialized.
(closes issue #14758) Reported by: tim_ringenbach
2009-03-27 13:06 +0000 [r184565] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix an issue where nat=yes would not always
take effect for the RTP session on outgoing calls. If calls were
placed using an IP address or hostname the global nat setting was
copied over but was not set on the RTP session itself. This
caused the RTP stack to not perform symmetric RTP actions.
(closes issue #14546) Reported by: acunningham
2009-03-26 22:17 +0000 [r184447] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: use new, improved 8kHz prompts
2009-03-26 21:07 +0000 [r184388] David Vossel <dvossel@digium.com>
* apps/app_test.c: pri loop TestClient/TestServer fails: server
SEND DTMF 8 app_test was failing when sending the last DTMF
digit, 8, because of the 100ms pause issued after DTMF is sent.
During this pause the other side would hang up causing the test
to look like it failed. Now the other side waits a second before
hanging up. (closes issue #12442) Reported by: tzafrir
2009-03-25 14:12 +0000 [r184188] Eliel C. Sardanons <eliels@gmail.com>
* main/asterisk.c: Avoid destroying the CLI line when moving the
cursor backward and trying to autocomplete. When moving the
cursor backward and pressing TAB to autocomplete, a NULL is put
in the line and we are loosing what we have already wrote after
the actual cursor position. (closes issue #14373) Reported by:
eliel Patches: asterisk.c.patch uploaded by eliel (license 64)
Tested by: lmadsen
2009-03-24 22:34 +0000 [r184078] Mark Michelson <mmichelson@digium.com>
* apps/app_senddtmf.c: Change NULL pointer check to be
ast_strlen_zero. The 'digit' variable is guaranteed to be
non-NULL, so the if statement could never evaluate true. Changing
to ast_strlen_zero makes the logic correct. This was found while
reviewing ast_channel_ao2 code review.
2009-03-24 15:25 +0000 [r183913] Tilghman Lesher <tlesher@digium.com>
* configs/voicemail.conf.sample: Additionally note that the
operator option needs an 'o' extension. (Related to issue #14731)
2009-03-23 17:59 +0000 [r183700] Mark Michelson <mmichelson@digium.com>
* res/res_monitor.c: Fix a memory leak in res_monitor.c The only
way that this leak would occur is if Monitor were started using
the Manager interface and no File: header were given. Discovered
while reviewing the ast_channel_ao2 review request.
2009-03-20 16:53 +0000 [r183559] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a crash in IAX2 registration handling
found during load testing with dvossel.
2009-03-19 23:37 +0000 [r183481] Terry Wilson <twilson@digium.com>
* apps/app_dial.c: Add missing datastore inherit (exists in all
other branches)
2009-03-19 19:40 +0000 [r183386] David Vossel <dvossel@digium.com>
* include/asterisk/features.h, apps/app_dial.c, res/res_features.c:
Cleaning up a few things in detect disconnect patch Initialized
ast_call_feature in detect_disconnect to avoid accessing
uninitialized memory. Cleaned up /param tags in features.h. No
longer send dynamic features in ast_feature_detect. issue #11583
2009-03-19 19:21 +0000 [r183319-183342] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Reordering, to change prior to unlocking
* channels/chan_dahdi.c: Delay signalling progress until a PRI
channel really signals progress. (closes issue #13034) Reported
by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by
tilghman (license 14) patch_trunk_183progress_klaus3000.txt
uploaded by klaus3000 (license 65) Tested by: klaus3000
2009-03-19 18:28 +0000 [r183291] Jason Parker <jparker@digium.com>
* main/asterisk.exports: Export some more required symbols.
2009-03-19 17:52 +0000 [r183145-183241] Russell Bryant <russell@digium.com>
* main/loader.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: Remove the use of RTLD_NOLOAD, as it is not
behaving like expected.
* main/asterisk.exports: Allow the AES API to work.
* main/asterisk.exports: Add missing semicolon in exports script.
2009-03-19 16:15 +0000 [r183126] David Vossel <dvossel@digium.com>
* include/asterisk/features.h, apps/app_dial.c, res/res_features.c,
res/res_features.exports: Allow disconnect feature before a call
is bridged feature.conf has a disconnect option. By default this
option is set to '*', but it could be anything. If a user wishes
to disconnect a call before the other side answers, only '*' will
work, regardless if the disconnect option is set to something
else. This is because features are unavailable until bridging
takes place. The default disconnect option, '*', was hardcoded in
app_dial, which doesn't make any sense from a user perspective
since they may expect it to be something different. This patch
allows features to be detected from outside of the bridge, but
not operated on. In this case, the disconnect feature can be
detected before briding and handled outside of features.c.
(closes issue #11583) Reported by: sobomax Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671) Tested
by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/
2009-03-19 16:13 +0000 [r183123] Russell Bryant <russell@digium.com>
* main/asterisk.exports: Allow the CallerID API to work again.
2009-03-19 16:04 +0000 [r183115] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix an issue where cancelled outgoing SIP
calls would erroneously report the device as "in use." A user was
having an issue where if an outgoing SIP call was canceled, the
SIP device would remain in use if we had not received any
response to the initial INVITE we sent out. The SIP device would
remain in use until the autocongestion timer was exhausted. I
tracked down the cause of this to be the section of code I am
removing here. I asked several people what the purpose of this
code was meant to be, but no one could give me any sort of answer
as to why this was here. The person who was having this issue has
been using this patch for several months and it has stopped the
problems they have had. AST-196
2009-03-18 20:02 +0000 [r182963-182965] Jeff Peeler <jpeeler@digium.com>
* configure, autoconf/ast_check_openh323.m4: fix typo which broke
configure
* channels/h323/compat_h323.cxx, channels/h323/ast_h323.cxx,
configure, autoconf/ast_check_openh323.m4,
channels/h323/compat_h323.h, channels/chan_h323.c,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Allow H.323
Plus library to be used in addition to the OpenH323 library
Chan_h323 can now be compiled against both the previously
supported versions of OpenH323 as well as the current H.323 Plus
(version 1.20.2). The configure script has been modified to look
in the default install location of h323 to hopefully help avoid
using the environment variables OPENH323DIR and PWLIBDIR. Also,
the CLI command "h323 show version" has been added which
indicates which version of h323 is in use. (closes issue 0011261)
Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch
uploaded by jthurman (license 614)
2009-03-18 11:31 +0000 [r182882] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/callerid.h, channels/chan_dahdi.c,
main/callerid.c: fix another symbol namespace issue (reported by
Andrew on asterisk-dev)
2009-03-18 02:09 +0000 [r182810] Russell Bryant <russell@digium.com>
* main/poll.c, main/io.c, main/channel.c, main/manager.c,
channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c,
include/asterisk/poll-compat.h, channels/chan_alsa.c,
main/asterisk.c, apps/app_nbscat.c, main/Makefile,
include/asterisk/autoconfig.h.in, configure.ac, main/utils.c,
include/asterisk/io.h, include/asterisk/channel.h: Fix cases
where the internal poll() was not being used when it needed to
be. We have seen a number of problems caused by poll() not
working properly on Mac OSX. If you search around, you'll find a
number of references to using select() instead of poll() to work
around these issues. In Asterisk, we've had poll.c which
implements poll() using select() internally. However, we were
still getting reports of problems. vadim investigated a bit and
realized that at least on his system, even though we were
compiling in poll.o, the system poll() was still being used. So,
the primary purpose of this patch is to ensure that we're using
the internal poll() when we want it to be used. The changes are:
1) Remove logic for when internal poll should be used from the
Makefile. Instead, put it in the configure script. The logic in
the configure script is the same as it was in the Makefile.
Ideally, we would have a functionality test for the problem, but
that's not actually possible, since we would have to be able to
run an application on the _target_ system to test poll()
behavior. 2) Always include poll.o in the build, but it will be
empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll()
throughout the source tree to ast_poll(). I feel that it is good
practice to give the API call a new name when we are changing its
behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems
where AST_POLL_COMPAT is defined, ast_poll() is redefined to
ast_internal_poll(). 4) Change poll() in main/poll.c to be
ast_internal_poll(). It's worth noting that any code that still
uses poll() directly will work fine (if they worked fine before).
So, for example, out of tree modules that are using poll() will
not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used. (closes issue
#13404) Reported by: agalbraith Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
2009-03-18 01:55 +0000 [r182802-182808] Kevin P. Fleming <kpfleming@digium.com>
* main/astobj2.c, main/asterisk.exports (added),
res/res_odbc.exports (added), res/res_speech.exports (added),
res/res_config_odbc.c, res/res_features.exports (added),
build_tools/strip_nonapi (removed), res/res_adsi.exports (added),
res/res_indications.c, default.exports (added), makeopts.in,
res/res_jabber.exports (added), res/res_monitor.exports (added),
res/res_config_pgsql.c, res/res_snmp.c, main/Makefile,
res/res_smdi.exports (added), include/asterisk/astobj2.h,
res/res_crypto.c, res/res_agi.exports (added), Makefile.rules,
res/res_musiconhold.c: Improve the build system to *properly*
remove unnecessary symbols from the runtime global namespace.
Along the way, change the prefixes on some internal-only API
calls to use a common prefix. With these changes, for a module to
export symbols into the global namespace, it must have *both* the
AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
the linker to leave the symbols exposed in the module's .so file
(see res_odbc.exports for an example).
* main/astobj2.c, main/asterisk.exports (removed),
res/res_odbc.exports (removed), main/channel.c,
res/res_config_odbc.c, res/res_features.exports (removed),
default.exports (removed), include/asterisk/frame.h,
res/res_jabber.exports (removed), res/res_config_pgsql.c,
main/Makefile, res/res_smdi.exports (removed),
include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c,
res/res_agi.exports (removed), res/res_speech.exports (removed),
include/asterisk/linkedlists.h, main/file.c,
build_tools/strip_nonapi (added), res/res_adsi.exports (removed),
res/res_indications.c, makeopts.in, apps/app_mixmonitor.c,
apps/app_chanspy.c, res/res_monitor.exports (removed),
main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c,
apps/app_meetme.c, res/res_snmp.c, Makefile.rules,
res/res_musiconhold.c: revert commit that included extranous
changes
* /: remove accidentally merged properties
* main/astobj2.c, main/asterisk.exports (added),
res/res_odbc.exports (added), main/channel.c,
res/res_config_odbc.c, res/res_features.exports (added),
default.exports (added), include/asterisk/frame.h,
res/res_jabber.exports (added), res/res_config_pgsql.c,
main/Makefile, res/res_smdi.exports (added),
include/asterisk/astobj2.h, main/slinfactory.c, res/res_crypto.c,
res/res_agi.exports (added), res/res_speech.exports (added),
include/asterisk/linkedlists.h, main/file.c,
build_tools/strip_nonapi (removed), res/res_adsi.exports (added),
res/res_indications.c, makeopts.in, apps/app_mixmonitor.c,
apps/app_chanspy.c, res/res_monitor.exports (added),
main/autoservice.c, build_tools/cflags-devmode.xml, main/frame.c,
/, apps/app_meetme.c, res/res_snmp.c, Makefile.rules,
res/res_musiconhold.c: Improve the build system to *properly*
remove unnecessary symbols from the runtime global namespace.
Along the way, change the prefixes on some internal-only API
calls to use a common prefix. With these changes, for a module to
export symbols into the global namespace, it must have *both* the
AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows
the linker to leave the symbols exposed in the module's .so file
(see res_odbc.exports for an example).
2009-03-17 20:13 +0000 [r182652] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c, apps/app_flash.c: Allow dahdichanname to
work as advertised. (closes issue #14056) Reported by: dsedivec
Patches: load_from_zapata_conf.patch uploaded by dsedivec
(license 638)
2009-03-17 05:50 +0000 [r182449] Tilghman Lesher <tlesher@digium.com>
* main/db.c: Fix race in astdb The underlying db1 implementation
does not fully isolate the pages retrieved from astdb, so the
lock protecting accesses needs to be extended until the copy from
the shared memory structure is done. (closes issue #14682)
Reported by: makoto
2009-03-16 Leif Madsen <lmadsen@digium.com>
* Released 1.4.24
2009-03-06 Leif Madsen <lmadsen@digium.com>
* Released 1.4.24-rc1
2009-03-06 18:23 +0000 [r180567] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Make compilation succeed in dev-mode when
IMAP storage is enabled.
2009-03-06 17:19 +0000 [r180532] David Vossel <dvossel@digium.com>
* main/enum.c: Fix handling of backreferences for ENUM lookups
enum.c did not handle regex backtraces correctly. The '\1' in the
regex is a backreference that requires a pattern match to be
inserted. The way the code used to work is that it would find the
backreference and insert the entire input string minus the '+'.
This is incorrect. The regexec() function takes in a variable
called pmatch which is an array of structs containing the start
and end indexes for each backreference substring. The original
code actually passed the pmatch array pointer into regexec but
never did anything with it. Now when a backtrace is found, the
backtrace number is looked up in the pmatch array and the correct
substring is inserted. (closes issue #14576) Reported by:
chris-mac Review: http://reviewboard.digium.com/r/187/
2009-03-05 23:26 +0000 [r180380-180464] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: [IMAP] Fix message retrieval issues when
identical mailbox names were defined in separate contexts. There
was a fix put in a while back so that an X-Asterisk-VM-Context
message header was added to stored IMAP voicemails. This would
allow for us to differentiate if the same mailbox name was used
in multiple contexts. The problem still left was that not all
places where messages were retrieved actually attempted to use
this header for information when retrieving messages. This commit
fixes that so that MWI and message retrieval from VoiceMailMain
work as expected. (closes issue #13853) Reported by: vicks1
Patches: 13853_v2.patch uploaded by mmichelson (license 60)
Tested by: lmadsen
* apps/app_voicemail.c, configs/voicemail.conf.sample: Fix broken
mailbox parsing when searchcontexts option is enabled. When using
the searchcontexts option in voicemail.conf, the code made the
assumption that all mailbox names defined were unique across all
contexts. However, the code did nothing to actually enforce this
assumption, nor did it do anything to alert a user that he may
have created an ambiguity in his voicemail.conf file by defining
the same mailbox name in multiple contexts. With this change, we
now will issue a nice long warning if searchcontexts is on and we
encounter the same mailbox name in multiple contexts and ignore
any duplicates after the first box. Whether searchcontexts is
enabled or not, if we come across a duplicate mailbox in the same
context, then we will issue a warning and ignore the duplicated
mailbox. I have also added a small note to voicemail.conf.sample
in the explanation for searchcontexts explaining that you cannot
define the same mailbox in multiple contexts if you have enabled
the option. (closes issue #14599) Reported by: lmadsen Patches:
14599.patch uploaded by mmichelson (license 60) (with slight
modification) Tested by: lmadsen
2009-03-05 18:22 +0000 [r180372] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp.c, main/frame.c, include/asterisk/frame.h: Fix problems
when RTP packet frame size is changed During some code analysis,
I found that calling ast_rtp_codec_setpref() on an ast_rtp
session does not work as expected; it does not adjust the
smoother that may on the RTP session, in fact it summarily drops
it, even if it has data in it, even if the current format's
framing size has not changed. This is not good. This patch
changes this behavior, so that if the packetization size for the
current format changes, any existing smoother is safely updated
to use the new size, and if no smoother was present, one is
created. A new API call for smoothers,
ast_smoother_reconfigure(), was required to implement these
changes. Review: http://reviewboard.digium.com/r/184/
2009-03-04 19:22 +0000 [r180194] Joshua Colp <jcolp@digium.com>
* main/callerid.c: Look for the number in a callerid string
starting from the end. This way a value using <> can exist in the
name portion. (issue #AST-194)
2009-03-03 23:01 +0000 [r180010] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c: Make sure we still support zapchan in
users.conf, in addition to dahdichan.
2009-03-03 22:48 +0000 [r180006] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample, apps/app_queue.c: Clarify some
documentation of queues.conf.sample It had always been possible
to explicitly specify a "blank" value for a sound file in
queues.conf and have no sound played back. The problem with this
is that it would result in some ugly CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in
queues.conf.sample (closes issue #14227) Reported by: caspy
2009-03-03 18:27 +0000 [r179840] Joshua Colp <jcolp@digium.com>
* res/res_features.c: Do not assume that the bridge_cdr is still
attached to the channel when the 'h' exten is finished executing.
It is possible for a masquerade operation to occur when the 'h'
exten is operating. This operation moves the CDR records around
causing the bridge_cdr to no longer exist on the channel where it
is expected to. We can not safely modify it afterwards because of
this, so don't even try. (closes issue #14564) Reported by: meric
2009-03-03 18:11 +0000 [r179807] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
utils/expr2.testinput, main/ast_expr2.h, main/ast_expr2.y,
main/ast_expr2f.c: These changes allow AEL to better check ${}
constructs within $[...], that are concatenated with text. I
modified and added rules in ast_expr2.fl to better handle the
concatenations. I added some default routines to ast_expr2.y so
the standalone would compile. It also looks like I haven't run
this thru bison since 2.1, so it's good to get this updated. The
Makefile has comments added now for check_expr2 and check_expr to
explain what they are for, and how to run them. The testexpr2s
stuff has been removed, in favor of check_expr2. expr2.testinput
has been updated to include the two expressions that inspired
these changes (from mcnobody on #asterisk this morning) The
regression has been run and all looks well.
2009-03-03 16:45 +0000 [r179741] Russell Bryant <russell@digium.com>
* main/channel.c: Ensure chan->fdno always gets reset to -1 after
handling a channel fd event. Since setting fdno to -1 had to be
moved, a couple of other code paths that do process an fd event
return early and do not pass through the code path where it was
moved to. So, set it to -1 in a few other places, too.
2009-03-03 14:38 +0000 [r179671] Joshua Colp <jcolp@digium.com>
* main/channel.c: Move where fdno is set to the default value to
*after* the read callback of the channel driver is called. We
have to do this as the underlying channel driver may need the
fdno value to determine what to read.
2009-03-03 13:53 +0000 [r179608] Russell Bryant <russell@digium.com>
* main/channel.c: Make it easier to detect an improper call to
ast_read(). When you call ast_waitfor() on a channel, the index
into the channel fds array that holds the file descriptor that
poll() determines has input available is stored in fdno. This
patch clears out this value after a call to ast_read() and also
reports errors if ast_read() is called without an fdno set. From
a discussion on the asterisk-dev list.
2009-03-02 23:54 +0000 [r179536] Jeff Peeler <jpeeler@digium.com>
* main/channel.c: Fix bridging regression from commit 176701 This
fixes a bad regression where the bridge would exit after an
attended transfer was made. The problem was due to nexteventts
getting set after the masquerade which caused the bridge to
return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
tim_ringenbach
2009-03-02 23:34 +0000 [r179532] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Move ast_waitfor() down to avoid the results
of the API call becoming stale. This call to ast_waitfor() was
being done way too soon in this section of code. Specifically,
there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice. By putting
the channel in autoservice, the previous results of ast_waitfor()
become meaningless, as the autoservice thread will do it's own
ast_waitfor() and ast_read() on the channel. So, when we came
back out of autoservice and eventually hit the block of code that
calls ast_read() on the channel, there may not actually be any
input on the channel available. Even though the previous call to
ast_waitfor() in app_meetme said there was input, the autoservice
thread has since serviced the channel for some period of time.
This bug manifested itself while dvossel was doing some testing
of MeetMe in Asterisk trunk. He was using the timerfd timing
module. When the code hit ast_read() erroneously, it determined
that it must have been called because of input on the timer fd,
as chan->fdno was set to AST_TIMING_FD, since that was the cause
of the last legitimate call to ast_read() done by autoservice. In
this test, an IAX2 channel was calling into the MeetMe
conference. It was _much_ more likely to be seen with an IAX2
channel because of the way audio is handled. Every audio frame
that comes in results in a call to ast_queue_frame(), which then
uses ast_timer_enable_continuous() to notify the channel thread
that a frame is waiting to be handled. So, the chances of
ast_waitfor() indicating that a channel needs servicing due to a
timer event on an IAX2 event is very high. Finally, it is
interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When
ast_read() is called and erroneously thinks that there is a timer
event to handle, it calls the ast_timer_ack() function. The
pthread and dahdi timing modules handle the ack() function being
called when there is no event by simply ignoring it. In the case
of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused
Asterisk to lock up very quickly. Thanks to dvossel and
mmichelson for the fun debugging session. :-)
2009-03-02 23:09 +0000 [r179468] Tilghman Lesher <tlesher@digium.com>
* main/app.c: When ending a recording with silence detection,
remember to reduce the duration. The end of the recording is
correspondingly trimmed, but the duration was not trimmed by the
number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration. (closes issue #14406)
Reported by: sasargen Patches: 20090226__bug14406.diff.txt
uploaded by tilghman (license 14) Tested by: sasargen
2009-03-02 22:58 +0000 [r179461] Russell Bryant <russell@digium.com>
* main/channel.c: Ensure that only one thread is calling
ast_settimeout() on a channel at a time. For example, with an
IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so
twice at the same time is a bad thing. (Found in a debugging
session with dvossel and mmichelson)
2009-03-02 20:14 +0000 [r179395] Jason Parker <jparker@digium.com>
* main/editline/configure, main/editline/np/unvis.c,
main/editline/sys.h, main/editline/configure.in: Remove several
silly warnings in editline. One about a broken preprocessor
directive, and another about strlcpy/strlcat. (closes issue
#14264) Reported by: dimas
2009-02-27 19:03 +0000 [r179056] Jason Parker <jparker@digium.com>
* doc/channelvariables.txt: Update documentation for DIALEDTIME and
ANSWEREDTIME variables. (closes issue #14566) Reported by:
klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
klaus3000 (license 65)
2009-02-26 21:27 +0000 [r178956] Steve Murphy <murf@digium.com>
* configs/features.conf.sample, res/res_features.c: This change
moves the default feature digit timeout to 1000 ms from the
previous default of 500. As per bug 14515, a dev discussion
arrived at a "mediated concensus" of a default feature digit
timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for
distracted phone users in phone booths; kpfleming put his foot
down at 1.0 sec. Users who found the previous default max delay
of 250 msec perfect, are welcome to override the new default.
Notice that I said that 250 msec was the default; wait a minute,
you might say, the config file said it was 500 msec!; well,
because of the bug fix for 14515, we found that 500 msec was
actually enforcing a max of 250. The bug fix would restore 500
msec, but we felt even that was a bit tight for most users...
2000 msec was pushed earlier by mmichelson, so that reduces to
1000 msec after the bug fix. Enjoy!
2009-02-26 17:24 +0000 [r178838] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: IAX2 prune realtime fix Now prune_users()
and prune_peers() are called instead of reload_config() to prune
all users/peers that are realtime. These functions remove all
users/peers with the rtfriend and delme flags set.
iax2_prune_realtime() also lacked the code to properly delete a
single friend. For example. if iax2 prune realtime <friend> was
called, only the peer instance would be removed. The user would
still remain. (closes issue #14479) Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/
2009-02-26 17:09 +0000 [r178640-178804] Steve Murphy <murf@digium.com>
* res/res_features.c: This patch prevents the feature detection
timeout from being cut in half. Because the ast_channel_bridge()
call will return 0 and pass a frame pointer for both DTMF_BEGIN
and DTMF_END, the feature_timer field in hte config struct is
getting decremented twice, which effectively cuts the
digittimeout in half. I added conditions to the if statement to
only let DTMF_END frames to flow thru, which solved the problem.
Also, when the frame pointer is null, let control flow thru--
this usually happens on timeouts. I added a comment to the code
to explain what's going on and why. Many thanks to sodom for
reporting this problem. Personnally, it always seemed like
something was wrong with the featuredigittimeout, but I never
could quite decide what... and was too busy to investigate. This
bug forced the issue, and now we know. Sodom had other issues in
14515, but I couldn't reproduce them. If he still has problems,
and wants to get them solved, he is welcome to reopen 14515.
(closes issue #14515) Reported by: sodom Patches: 14515.patch
uploaded by murf (license 17) Tested by: murf, sodom
* main/ast_expr2.fl, main/ast_expr2f.c: This patch completes the
fixes nec. to make 1.4 asterisk dialplan expressions ($[...])
8-bit transparent While I was updating ast_expr2.fl, I missed one
rule that would allow 8-bit chars to be caught in tokens; and in
so doing, it absorbs the ${ sequence and messes up the checking
of raw exprs by AEL. Trunk already has these changes. (closes
issue #14543) Reported by: klaus3000 Patches: patch.14543
uploaded by murf (license 17) Tested by: murf
2009-02-25 12:43 +0000 [r178508] Russell Bryant <russell@digium.com>
* main/asterisk.c: Update the copyright year for the main page of
the doxygen documentation.
2009-02-24 23:25 +0000 [r178445] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Add section about the #exec
command in configuration files. (closes issue #14540) Reported
by: jtodd Patch by: jtodd, with additional notes by tilghman
(license 14)
2009-02-24 20:36 +0000 [r178373] Russell Bryant <russell@digium.com>
* main/rtp.c: Only set dtmfcount on BEGIN, and ensure it gets reset
to 0 properly. (issue #14460) Reported by: moliveras Tested by:
russell
2009-02-24 17:02 +0000 [r178266] Terry Wilson <twilson@digium.com>
* apps/app_dahdiras.c, res/res_musiconhold.c: Change include order
to make compile on Centos 5 with DAHDI If BIT_TYPES_DEFINED gets
defined before linux/types.h is included, the __s32 type doesn't
get defined
2009-02-24 15:16 +0000 [r178205] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Skip check for extension when subscribing
for MWI. Since the remote side is not actually subscribing to a
specific extension when subscribing for MWI just skip the check
to see if the extension exists. They can't use it to specify the
mailbox either since we require configuration of that in sip.conf
(closes issue #14531) Reported by: festr
2009-02-23 23:09 +0000 [r178141] Russell Bryant <russell@digium.com>
* main/rtp.c: Fix infinite DTMF when a BEGIN is received without an
END. This commit is related to rev 175124 of 1.4 where a previous
attempt was made to fix this problem. The problem with the
previous patch was that the inserted code needed to go _before_
setting the lastrxts to the current timestamp. Because those were
the same, the dtmfcount variable was never decremented, and so
the END was never sent. In passing, I removed the dtmfsamples
variable which was completed unused. I also removed a redundant
setting of the lastrxts variable. (closes issue #14460) Reported
by: moliveras
2009-02-20 22:59 +0000 [r177701-177786] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Don't print the CR-NL combination when we aren't
outputting to the manager. An embedded CR-NL in a CLI command
screws up several AMI parsers that don't expect to see that
combination in the middle of output. (Closes issue #14305)
Reported by: martins Patch by: tilghman
* include/asterisk/threadstorage.h: This exception does not appear
to still be true for Solaris 10, and OpenSolaris definitely needs
it to be removed. Fixed for snuff-home on -dev channel.
2009-02-20 20:17 +0000 [r177696] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
undefined audio codecs in chan_iax2 During iax2 call negotiation,
supported codecs are passed in an Information Element containing
a 2 byte field where each bit correlates to a specific codec. In
1.4 only audio codec bits 0-12 are defined, leaving bits 13-15
undefined. By default all bits are enabled unless specified
otherwise. Since its a 2 byte field and 13-15 are not defined,
these bits are never turned off. In trunk, bits 13-15 are
defined, which means 1.4 is advertising support for codecs it
does not have when talking to trunk. I fixed this by adding
#define for undefined audio codec bits. These bits are then
removed from iax2's full bandwidth capabilities. (closes issue
#14283) Reported by: jcovert
2009-02-19 22:51 +0000 [r177540] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This patch
fixes a problem with 8-bit input to the ast_expr2 scanner. The
real culprit was the --full argument to flex in the Makefile!
This causes a 7-bit scanner to be generated. I reviewed the rules
and found one rule where I needed to specifically include 8-bit
chars for a token. I tested against the text supplied by ibercom,
and all looks very well. This has been there a surprisingly long
time! (closes issue #14498) Reported by: ibercom Patches:
14498.patch uploaded by murf (license 17) Tested by: murf
2009-02-19 22:26 +0000 [r177536] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Fix up potential crashes, by reducing the
sharing between interactive and non-interactive threads. (closes
issue #14253) Reported by: Skavin Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin
2009-02-19 18:58 +0000 [r177450] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Force a MWI notification after subscribe
request. Reported by the Resiprocate dev team. Thanks!
2009-02-19 16:37 +0000 [r177383] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: If we are able to create a speech
structure unset the ERROR variable in case it was previously set.
(issue #LUMENVOX-13)
2009-02-18 22:43 +0000 [r177225] Steve Murphy <murf@digium.com>
* pbx/ael/ael.tab.c, pbx/ael/ael.y: This patch fixes a regression
of sorts that was introduced in rev 24425. It basically fixes
AST-190/ABE-1782. What was wrong: the user has 6000 extensions in
one context; and then 6000 contexts, one per extension. The
parser could only handle about 4893 of the 6000 extens in the
single context. This was due to the regression I mentioned. To
get rid of shift/reduce conflicts, Luigi set up right-recursive
lists for globals, context elements, switch lists, and
statements. Right recursive lists got rid of the warnings, but
instead, they use up a tremendous amount of stack space when the
lists are long. I saw this a few years back, and resolved not to
fix it until someone complained. That day has arrived! After the
changes were made, I ran the regression test suite, and there
were no problems. I took the test case the user provided, and
added 100,000 extensions to the single context, that already had
6,000 extens in it. (I'll see your 6, and raise you 100!) It
takes a few minutes to read it all in, check it and generate code
for it, but no problems. So, I think I can say that
fundamentally, there are no longer any limits on the number of
items you can place in contexts, statement blocks, switches, or
globals, beyond your virt mem constraints.
2009-02-18 20:06 +0000 [r177160] Jeff Peeler <jpeeler@digium.com>
* channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx,
autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h,
channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
channels/h323/ast_ptlib.h (added), configure,
channels/h323/compat_h323.h, configure.ac,
channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Modify h323
to build against PTLib as well as the older PWLib Several changes
in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option
change, install location change, and a boolean type change which
is handled by ast_ptlib.h. Also, the sed check has been modified
to properly work with autoconf >= 2.62. (closes issue #14224)
Reported by: bergolth Patches: asterisk-autoconf-sed.patch
uploaded by bergolth (license 661) asterisk-pwlib-v3.patch
uploaded by bergolth (license 661) Tested by: jpeeler
2009-02-18 18:30 +0000 [r177096] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/config.h: Document the return value of the
update method (as requested on -dev list)
2009-02-18 17:41 +0000 [r176945-177039] Doug Bailey <dbailey@digium.com>
* main/utils.c: Merged revisions 177035 manually from
https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 |
dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
Fixed error where a check for an zero length, terminated string
was needed. ........
* main/utils.c: Need to take into account the \0 terminator of the
old string to determine the amount available.
2009-02-18 00:34 +0000 [r176810] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
with G723. This commit brings in the changes that were living out
on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch.
codec_dahdi.c now always uses signed linear as the simple codec
so that a soft g729 codec will not end up being preferred to the
hardware codec. There are also changes to allow codec_dahdi.c to
feed packets to the hardware in the native sample size of the
codec. This solves problems with choppy audio when using G723.
2009-02-17 21:54 +0000 [r176701] Jeff Peeler <jpeeler@digium.com>
* main/channel.c, res/res_features.c, include/asterisk/channel.h:
Modify bridging to properly evaluate DTMF after first warning is
played The main problem is currently if the Dial flag L is used
with a warning sound, DTMF is not evaluated after the first
warning sound. To fix this, a flag has been added in
ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played
back (due to a feature evaluation or waiting for digits).
ast_channel_bridge was modified to store the nexteventts in the
ast_bridge_config structure as that information was lost every
time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations. (closes issue #14315) Reported by:
tim_ringenbach Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
2009-02-17 21:21 +0000 [r176426-176661] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c: Backport change to 1.4: Prior to
masquerade, move the group definitions to the channel performing
the masq, so that the group count lingers past the bridge.
(closes issue #14275) Reported by: kowalma Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
* channels/chan_sip.c: After a 'sip reload', qualifies for realtime
peers weren't immediately restarted, instead waiting until the
next registration. We're now caching the qualify across a
reload/restart and starting the qualify immediately upon loading
the peer. (closes issue #14196) Reported by: pdf Patches:
20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
Tested by: pdf
2009-02-16 23:30 +0000 [r176354] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE not
being relayed correctly during bridging This should have been
committed with rev176247, but I missed it. srcupdate frames no
longer break out of the native bridge, but are not being sent to
the other call leg either. This fixs that. issue #13749
2009-02-16 21:41 +0000 [r176254] Kevin P. Fleming <kpfleming@digium.com>
* main/utils.c: correct a logic error in the last stringfields
commit... don't mark additional space as allocated if the string
was built using already-allocated space
2009-02-16 21:39 +0000 [r176249-176252] Mark Michelson <mmichelson@digium.com>
* apps/app_meetme.c: Remove unused variable and make dev-mode
compilation happy
* apps/app_meetme.c: Open the DAHDI pseudo device and set it to be
nonblocking atomically Apparently on FreeBSD, attempting to set
the O_NONBLOCKING flag separately from opening the file was
causing an "inappropriate ioctl for device" error. While I cannot
fathom why this would be happening, I certainly am not opposed to
making the code a bit more compact/efficient if it also fixes a
bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
uploaded by ys (license 281) Tested by: ys
2009-02-16 21:28 +0000 [r176247] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE
breaking out of native bridge In iax2, when a
AST_CONTROL_SRCUPDATE is received it breaks from the native
bridge, but since there is no code path to handle srcupdate it
just goes to be beginning of the loop. This was causing packet
storms of srcupdate frames between servers. Now srcupdate frames
do not break the native bridge for processing. (closes issue
#13749) Reported by: adiemus
2009-02-16 21:10 +0000 [r176216] Kevin P. Fleming <kpfleming@digium.com>
* main/utils.c: fix a flaw in the ast_string_field_build() family
of API calls; these functions made no attempt to reuse the space
already allocated to a field, so every time the field was written
it would allocate new space, leading to what appeared to be a
memory leak.
2009-02-16 15:33 +0000 [r176029] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't have the Via header stored as a
stringfield as it can change often during the lifetime of a
dialog. This issue crept up with subscriptions on the AA50. When
an outgoing NOTIFY is sent a new branch value is created and the
Via header is changed to reflect it. Since this was a stringfield
a new spot in the pool was used for the value while the old was
left untouched/unused. If the current pool was full a new pool
was created. This would cause memory usage to increase steadily.
(issue #AA50-2332)
2009-02-15 23:37 +0000 [r175921] Michiel van Baak <michiel@vanbaak.info>
* main/pbx.c, channels/chan_sip.c, main/devicestate.c,
include/asterisk/manager.h: fix mis-spelling of the word
registered. Reported by De_Mon on #asterisk-dev.
2009-02-15 20:33 +0000 [r175777-175825] Olle Johansson <oej@edvina.net>
* formats/format_ilbc.c: format_ilbc does not depend on codec
libraries and can therefore always be made. My mistake. Ursäkta!
* formats/format_ilbc.c: Disable format_ilbc.so by default, like
codec_ilbc.so
* channels/chan_sip.c: Make sure that the debug line is not printed
on debug level 0
2009-02-13 21:53 +0000 [r175698] Jason Parker <jparker@digium.com>
* include/asterisk/dahdi_compat.h: Zaptel is not DAHDI. Rather,
Zaptel is actually Zaptel. (in case you're confused, DAHDI is
still DAHDI)
2009-02-13 19:47 +0000 [r175407-175590] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix a potential crash situation when using
IMAP voicemail If calling into VoiceMailMain when using IMAP
storage, it was possible to crash Asterisk by hanging up the
phone when prompted for a voicemail mailbox. This patch fixes the
issue. While it may appear that this patch is superficial, it
allows code execution to continue to the failure case just below
the IMAP_STORAGE code block where this patch has been applied
(closes issue #14473) Reported by: dwpaul Patches:
voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
689)
* main/file.c: Fix a place where filestreams were not refcounted
properly This section was already present in trunk and other
branches, but did not exist in 1.4. (closes issue #14395)
Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut
(license 60) Tested by: ZX81
2009-02-12 21:19 +0000 [r175311] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c: Fix crashes when receiving certain T.38 packets.
Also, increase the maximum size of T.38 packets and warn users
when they try to set the limits above those maximums. (closes
issue #13050) Reported by: schern Patches:
20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
Tested by: schern
2009-02-12 20:34 +0000 [r175187-175294] Jeff Peeler <jpeeler@digium.com>
* res/res_features.c: Fix ParkedCall event information for From
field in the case of a blind transfer If the parker information
can not be obtained from the peer, try and see if the
BLINDTRANSFER channel variable has been set. Previously, a blind
transfer to the ParkAndAnnounce app would return nothing for the
From. Closes AST-189
* res/res_features.c: Fix crash in event of failed attempt to
transfer to parking The peer may not necessarily exist, such as
in the case of a transfer to ParkAndAnnounce. In this case don't
try to play a sound to it.
2009-02-12 16:51 +0000 [r175124] Russell Bryant <russell@digium.com>
* main/rtp.c: Don't send DTMF for infinite time if we do not
receive an END event. I thought that this was going to end up
being a pretty gnarly fix, but it turns out that there was
actually already a configuration option in rtp.conf, dtmftimeout,
that was intended to handle this situation. However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option
got lost. So, this commit brings it back to life. The default
timeout is 3 seconds. However, it is worth noting that having
this be configurable at all is not really the recommended
behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the
time period of extending the tone is necessary to avoid that a
tone "gets stuck". Regardless of the algorithm used, the tone
SHOULD NOT be extended by more than three packet interarrival
times. A slight extension of tone durations and shortening of
pauses is generally harmless. Three seconds will pretty much
_always_ be far more than three packet interarrival times.
However, that behavior is not required, so I'm going to leave it
with our legacy behavior for now. Code from
svn/asterisk/team/russell/issue_14460 (closes issue #14460)
Reported by: moliveras
2009-02-12 10:16 +0000 [r175029] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Set the initiator attribute to lowercase
in our replies when receiving calls. This attribute contains a
JID that identifies the initiator of the GoogleTalk voice
session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters. (closes issue
#13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
by jcovert (license 551) Tested by: jcovert
2009-02-12 00:19 +0000 [r174997] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Revert RTP changes for continuation of DTMF. Proxy
commit by russell via SMS.
2009-02-12 00:01 +0000 [r174985-174986] Russell Bryant <russell@digium.com>
* main/rtp.c: Clear out the current event after forcing the end of
a digit
* main/rtp.c: Fixify infinite DTMF in the case that no RFC2833 END
event is ever received
2009-02-11 20:54 +0000 [r174885] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, apps/app_macro.c: Restore a behavior that was
recently changed, when we fixed issue #13962 and issue #13363
(related to issue #6176). When a hangup occurs during a Macro
execution in earlier 1.4, the h extension would execute within
the Macro context, whereas it was always supposed to execute only
within the main context (where Macro was called). So this fix
checks for an "h" extension in the deepest macro context where a
hangup occurred; if it exists, that "h" extension executes,
otherwise the main context "h" is executed. (closes issue #14122)
Reported by: wetwired Patches: 20090210__bug14122.diff.txt
uploaded by Corydon76 (license 14) Tested by: andrew
2009-02-10 18:50 +0000 [r174644] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Go off hold when we get an empty reinvite
telling us to. (closes issue #14448) Reported by: frawd Patches:
hold_invite_nosdp.patch uploaded by frawd (license 610)
2009-02-10 17:52 +0000 [r174583] Matthew Nicholson <mnicholson@digium.com>
* main/jitterbuf.c: Improve behavior of jitterbuffer when
maxjitterbuffer is set. This change improves the way the
jitterbuffer handles maxjitterbuffer and dramatically reduces the
number of frames dropped when maxjitterbuffer is exceeded. In the
previous jitterbuffer, when maxjitterbuffer was exceeded, all new
frames were dropped until the jitterbuffer is empty. This change
modifies the code to only drop frames until maxjitterbuffer is no
longer exceeded. Also, previously when maxjitterbuffer was
exceeded, dropped frames were not tracked causing stats for
dropped frames to be incorrect, this change also addresses that
problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
by mnicholson (license 96) Tested by: mnicholson Review:
http://reviewboard.digium.com/r/144/
2009-02-10 02:27 +0000 [r174369] Steve Murphy <murf@digium.com>
* apps/app_rpt.c: This patch solves some compiler complaints in
both 32 and 64-bit environments.
2009-02-09 17:11 +0000 [r174282] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Don't do an SRV lookup if a port is
specified RFC 3263 says to do A record lookups on a hostname if a
port has been specified, so that's what we're going to do. See
section 4.2. (closes issue #14419) Reported by: klaus3000
Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by
klaus3000 (license 65)
2009-02-09 14:48 +0000 [r174218] Joshua Colp <jcolp@digium.com>
* res/res_musiconhold.c: Don't overwrite our pointer to the music
class when music on hold stops. We will use this if it starts
again to see if we can resume the music where it left off.
(closes issue #14407) Reported by: mostyn
2009-02-07 16:15 +0000 [r174148] Russell Bryant <russell@digium.com>
* res/snmp/agent.c: Fix a race condition that could cause a crash.
2009-02-06 23:36 +0000 [r174082] Dwayne M. Hubbard <dhubbard@digium.com>
* channels/chan_sip.c: check ast_strlen_zero() before calling
ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp()
The reporter didn't actually upload a properly-formed patch,
instead a modified chan_sip.c file was uploaded. I created a
patch to determine the changes, then modified the suggested
changes to create a proper fix. The summary above is a complete
description of the changes. (closes issue #13547) Reported by:
tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license
258) Tested by: tecnoxarxa
2009-02-06 17:15 +0000 [r173967-173968] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Remove a debug message I put in by accident.
* channels/chan_sip.c: Some clients do not put the call-id for
replaces at the beginning, so support it being anywhere in the
string. (closes issue #14350) Reported by: fhackenberger
2009-02-06 16:20 +0000 [r173917] Matthew Nicholson <mnicholson@digium.com>
* channels/chan_sip.c: Limit the addition of the Contact header in
SIP responses according to various SIP RFCs. (closes issue
#13602) Reported by: hjourdain Tested by: mnicholson
2009-02-06 15:43 +0000 [r173900] Tilghman Lesher <tlesher@digium.com>
* utils/muted.c: Backport OS X fix from trunk (AGAIN, closes issue
#14360)
2009-02-05 23:19 +0000 [r173770] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix logic regarding when to perform an SRV
lookup for outgoing REGISTER requests With this fix, we only will
perform an SRV lookup at the following times: * The first time we
register with a remote registrar * If we send a REGISTER but do
not receive a response * If the sendto() function returns an
error While I wrote the patch that fixes this issue, a huge
amount of credit is due to Brett Bryant, who wrote the initial
patch on which I based this one. (closes issue #12312) Reported
by: jrast Patches: 12312.patch uploaded by putnopvut (license 60)
Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/
2009-02-05 20:47 +0000 [r173696] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c: Add new configuration option to make shared
IMAP mailboxes function as expected. The new option is
"imapvmshareid" which is an ID to tag multiple mailboxes using
the same IMAP storage location to function as one mailbox. This
allows all messages to be retrieved for any user in the group.
The patch alters the 'X-Asterisk-VM-Extension' header that is
responsible for matching voicemails for a given user. (closes
issue #13673) Reported by: howardwilkinson
2009-02-05 20:29 +0000 [r173392-173692] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix situations where queue members could be
autopaused unexpectedly Specifically, this patch prevents us from
autopausing members when we receive a busy or congestion frame
from them. (closes issue #14376) Reported by: fiddur Patches:
14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
* apps/app_mixmonitor.c: Add some missing cleanup to app_mixmonitor
* apps/app_mixmonitor.c: Fix a problem where a channel pointer
becomes invalid due to masquerading or hanging up. app_mixmonitor
runs its own thread to monitor the channel's activity and write
the mixed audio to a file. Since this thread runs independently
of the channel, it is possible that the mixmonitor thread's
channel pointer will point to freed memory when the channel
either is masqueraded or hangs up (technically, both cases are
hangups, but we need to handle the cases slightly differently).
The solution for this is to employ a datastore, which has the
nice benefit of allowing us to hook into channel masquerades and
hangups and update our pointer as necessary. If this looks
familiar, this same technique is employed in app_chanspy.
app_chanspy is a bit more involved since it does a lot more
operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't
have, though. Since there is a thread race between the channel's
thread and the mixmonitor thread on a hangup, we em- ploy a
condition-and-boolean combination to ensure that the channel
thread finishes with our structure before the mixmonitor thread
attempts to free it. No crashes! (closes issue #14374) Reported
by: aragon Patches: 14374.patch uploaded by putnopvut (license
60) Tested by: aragon, putnopvut
* apps/app_chanspy.c: Revert my previous change because it was
stupid
* apps/app_chanspy.c: Add a missing unlock. Extremely unlikely to
ever matter, but it's needed.
2009-02-03 23:35 +0000 [r173248] David Vossel <dvossel@digium.com>
* channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing
off calls. Fixes issue with IAX2 transfers not taking place. As
it was, a call that was being transfered would never be handed
off correctly to the call ends because of how call numbers were
stored in a hash table. The hash table, "iax_peercallno_pvt",
storing all the current call numbers did not take into account
the complications associated with transferring a call, so a
separate hash table was required. This second hash table
"iax_transfercallno_pvt" handles calls being transfered, once the
call transfer is complete the call is removed from the transfer
hash table and added to the peer hash table resuming normal
operations. Addition functions were created to handle storing,
removing, and comparing items in the iax_transfercallno_pvt
table. (issue #13468) Review:
http://reviewboard.digium.com/r/140/
2009-02-03 21:57 +0000 [r173211] Jeff Peeler <jpeeler@digium.com>
* res/res_features.c: Parking attempts made to one end of a bridge
no longer will hang up due to a parking failure. Parking attempts
made using either one-touch, or doing either a blind or assisted
transfer to the parking extension now keep up the bridge instead
of hanging up the attempted parked party. Normal causes for the
parking attempt to fail includes the specific specified extension
(via PARKINGEXTEN) not being available or if all the parking
spaces are currently in use. To avoid having to reverse a
masquerade park_space_reserve was made to provide foresight if a
parking attempt will succeed and if so reserve the parking space.
(closes issue #13494) Reported by: mdu113 Reviewed by Russell:
http://reviewboard.digium.com/r/133/
2009-02-03 00:15 +0000 [r173070] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.conf.sample: Add warning to standard config,
that globals may be overridden by other dialplan configuration
files. (closes issue #14388) Reported by: macli
2009-02-02 23:48 +0000 [r173066] Terry Wilson <twilson@digium.com>
* res/res_features.c: Fix a feature inheritance bug I added after
code review
2009-02-02 20:28 +0000 [r172962] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample:
channels/chan_dahdi.c * Added doxygen comments to the major dahdi
structures. * Fixed PRI using an incorrect string value if the
extension delimiter is not present in the Dial() function. *
Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample * Updated some documentation.
These changes are already in trunk -r172400
2009-01-31 00:15 +0000 [r172517-172639] Terry Wilson <twilson@digium.com>
* configs/features.conf.sample, res/res_features.c: Rename new
parkedcallparking option to parkedcallreparking Since this option
actually already existed in 1.6.0+, use the same name so as not
to confuse people when they upgrade
* configs/features.conf.sample, apps/app_dial.c,
main/global_datastores.c, res/res_features.c,
doc/channelvariables.txt, include/asterisk/global_datastores.h,
CHANGES: Fix feature inheritance with builtin features When using
builtin features like parking and transfers, the AST_FEATURE_*
flags would not be set correctly for all instances when either
performing a builtin attended transfer, or parking a call and
getting the timeout callback. Also, there was no way on a
per-call basis to specify what features someone should have on
picking up a parked call (since that doesn't involve the Dial()
command). There was a global option for setting whether or not
all users who pickup a parked call should have
AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
variable which can be set either in the dialplan or with setvar
in channels that support it. This variable can be set to any
combination of 't', 'k', 'w', and 'h' (case insensitive matching
of the equivalent dial options), to set what features should be
activated on this channel. The patch moves the setting of the
features datastores into the bridging code instead of app_dial to
help facilitate this. 2) adds global options parkedcallparking,
parkedcallhangup, and parkedcallrecording to be similar to the
parkedcalltransfers option for globally setting features. 3) has
builtin_atxfer call builtin_parkcall if being transfered to the
parking extension since tracking everything through multiple
masquerades, etc. is difficult and error-prone 4) attempts to fix
all cases of return calls from parking and completed builtin
transfers not having the correct permissions (closes issue
#14274) Reported by: aragon Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy
(license 396) Tested by: aragon, otherwiseguy Review
http://reviewboard.digium.com/r/138/
2009-01-29 22:54 +0000 [r172438] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, apps/app_nbscat.c, autoconf/ast_func_fork.m4,
apps/app_festival.c, build_tools/menuselect-deps.in, configure,
apps/app_dahdiras.c, apps/app_mp3.c, res/res_agi.c,
apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c:
Lose the CAP_NET_ADMIN at every fork, instead of at startup.
Otherwise, if Asterisk runs as a non-root user and the
administrator does a 'restart now', Asterisk loses the ability to
set QOS on packets. (closes issue #14004) Reported by: nemo
Patches: 20090105__bug14004.diff.txt uploaded by Corydon76
(license 14) Tested by: Corydon76
2009-01-29 08:48 +0000 [r172169] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Make sure that we always add the hangupcause
headers. In some cases, the owner was disconnected before we
checked for the cause. This patch implements a temporary storage
in the pvt and use that instead. The code is based on ideas from
code from Adomjan in issue #13385 (Add support for Reason:
header) Thanks to Klaus Darillion for testing! (closes issue
#14294) related to issue #13385 Reported by: klaus3000 and
adomjan Patches: bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by
adomjan (license 487) Tested by: oej, klaus3000
2009-01-28 18:51 +0000 [r172030] Steve Murphy <murf@digium.com>
* apps/app_channelredirect.c, main/pbx.c, main/manager.c,
res/res_features.c, include/asterisk/channel.h: This patch fixes
h-exten running misbehavior in manager-redirected situations.
What it does: 1. A new Flag value is defined in
include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which
used as a messenge to the bridge hangup exten code not to run the
h-exten there (nor publish the bridge cdr there). It will done at
the pbx-loop level instead. 2. In the manager Redirect code, I
set this flag on the channel if the channel has a non-null pbx
pointer. I did the same for the second (chan2) channel, which
gets run if name2 is set... and the first succeeds. 3. I restored
the ending of the cdr for the pbx loop h-exten running code.
Don't know why it was removed in the first place. 4. The first
attempt at the fix for this bug was to place code directly in the
async_goto routine, which was called from a large number of
places, and could affect a large number of cases, so I tested
that fix against a fair number of transfer scenarios, both with
and without the patch. In the process, I saw that putting the fix
in async_goto seemed not to affect any of the blind or attended
scenarios, but still, I was was highly concerned that some other
scenarios I had not tested might be negatively impacted, so I
refined the patch to its current scope, and jmls tested both. In
the process, tho, I saw that blind xfers in one situation, when
the one-touch blind-xfer feature is used by the peer, we got
strange h-exten behavior. So, I inserted code to swap CDRs and to
set the HANGUP_DONT field, to get uniform behavior. 5. I added
code to the bridge to obey the HANGUP_DONT flag, skipping both
publishing the bridge CDR, and running the h-exten; they will be
done at the pbx-loop (higher) level instead. 6. I removed all the
debug logs from the patch before committing. 7. I moved the
AUTOLOOP set/reset in the h-exten code in res_features so it's
only done if the h-exten is going to be run. A very minor
performance improvement, but technically correct. (closes issue
#14241) Reported by: jmls Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by
murf (license 17) Tested by: murf, jmls
2009-01-28 17:25 +0000 [r171963] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Clarify log message (suggested by
manxpower on #asterisk-dev)
2009-01-28 13:07 +0000 [r171837] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Add a better explanation of the
difference between the device namespace and the dialplan for
newbies.
2009-01-27 21:55 +0000 [r171621-171689] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Fix devicestate problems for "always-on"
agent channels A revision to chan_agent attempted to "inherit"
the device state of the underlying channel in order to report the
device state of an agent channel more accurately. The problem
with the logic here is that it makes no sense to use this for
always-on agents. If the agent is logged in, then to the
underlying channel, the agent will always appear to be "in use,"
no matter if the agent is on a call or not. The reason is that to
the underlying channel, the channel is currently in use on a call
to the AgentLogin application. The most common cause that I found
for this issue to occur was for a SIP channel to be the
underlying channel type for an Agent channel. If the SIP phone
re-registers, then the registration will cause the device state
core to query the device state of the SIP channel. Since the SIP
channel is in use, the Agent channel would also inherit this
status. Once the agent channel was set to "in use" there was no
way that the device state could change on that channel unless the
agent logged out. The solution for this problem is a bit
different in 1.4 than it is in the other branches. In 1.4, there
will be a one-line fix to make sure that only callback agents
will inherit device state from their underlying channel type. For
the other branches of Asterisk, since callback support has been
removed, there is also no need for device state inheritance in
chan_agent, so I will simply be removing it from the code. In
addition, the 1.4 source is getting a new comment to help the
next person who edits chan_agent.c. I'm adding a comment that a
agent_pvt's loginchan field may be used to determine if the agent
is a callback agent or not. (closes issue #14173) Reported by:
nathan Patches: 14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez
* main/slinfactory.c: Prevent a crash from occurring when a jitter
buffer interpolated frame is removed from a slinfactory
slinfactory used the "samples" field of an ast_frame in order to
determine the amount of data contained within the frame. In
certain cases, such as jitter buffer interpolated frames, the
frame would have a non-zero value for "samples" but have NULL
"data" This caused a problem when a memcpy call in
ast_slinfactory_read would attempt to access invalid memory. The
solution in use here is to never feed frames into the slinfactory
if they have NULL "data" (closes issue #13116) Reported by:
aragon Patches: 13116.diff uploaded by putnopvut (license 60)
2009-01-27 14:33 +0000 [r171527] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Use the same branch tag in CANCEL as in
INVITE Originally putnopvut implemented some changes in revision
142079 that according to the bug report seemed to have worked
then, but somehow fails now. I guess code, as humans, get old and
forget stuff. Anyway, this bug caused CANCEL not to work with
picky systems. Thanks Fredrik for pointing out where the bug in
the SIP messaging was. (closes issue #14346) Reported by: oej
Patches: bug14346.diff uploaded by oej (license 306) Tested by:
oej
2009-01-26 21:31 +0000 [r171452] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Resolve some synchronization issues in
chan_iax2 scheduler handling. The important changes here are
related to the synchronization between threads adding items into
the scheduler and the scheduler handling thread. By adjusting the
lock and condition handling, we ensure that the scheduler thread
sleeps no longer and no less than it is supposed to. We also
ensure that it does not wake up more often than it has to. There
is no bug report associated with this. It is just something that
I found while putting scheduler thread handling into a reusable
form (review 129). Review: http://reviewboard.digium.com/r/131/
2009-01-26 12:51 +0000 [r171264] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't retransmit 401 on REGISTER requests
when alwaysauthreject=yes (closes issue #14284) Reported by:
klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt
uploaded by klaus3000 (license 65) Tested by: klaus3000
2009-01-25 23:44 +0000 [r171120-171187] Tilghman Lesher <tlesher@digium.com>
* channels/chan_oss.c: Correctly track the hookstate (closes issue
#13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt
uploaded by Corydon76 (license 14)
* res/res_agi.c: Err, yeah.
* res/res_agi.c: Add thread to kill zombies, when child processes
don't die immediately on SIGHUP. (closes issue #13968) Reported
by: eldadran Patches: 20090114__bug13968.diff.txt uploaded by
Corydon76 (license 14) Tested by: eldadran
2009-01-25 13:33 +0000 [r170979] Sean Bright <sean.bright@gmail.com>
* apps/app_page.c: Resolve a logic error that was causing Page() to
crash when more than one channel was specified. (closes issue
#14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt
uploaded by seanbright (license 71) Tested by: kc0bvu
2009-01-24 13:55 +0000 [r170836] Tilghman Lesher <tlesher@digium.com>
* configs/res_odbc.conf.sample: Remove superfluous implementation
note (closes issue #14319)
2009-01-23 20:55 +0000 [r170671-170719] Mark Michelson <mmichelson@digium.com>
* configs/res_odbc.conf.sample: Add notes to the idlecheck
explanation in res_odbc.conf.sample (closes issue #14319)
Reported by: klaus3000 Patches:
patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000
(license 65)
* contrib/i18n.testsuite.conf: Update contrib/i18n.testsuite.conf
to not use deprecated syntax * Convert Wait,1 to Wait(1) *
Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all
priorities beyond the first Also added test for Chinese numbers,
too. (closes issue #14320) Reported by: dant Patches:
i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license
670)
2009-01-23 20:16 +0000 [r170648] Joshua Colp <jcolp@digium.com>
* main/channel.c: When a channel is answered make sure any
indications currently playing stop. Usually the phone would do
this but if the channel was already answered then they are being
generated by Asterisk and we darn well need to stop them. (closes
issue #14249) Reported by: RadicAlish
2009-01-23 Tilghman Lesher <tlesher@digium.com>
* Asterisk 1.4.23.1 released.
* channels/chan_iax2.c: Regression fix for AST-2009-001 security
fix.
2009-01-21 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.4.23 released.
2009-01-20 18:49 -0500 [r169581] Terry Wilson <twilson@digium.com>
* One-touch parking was calling back the wrong channel on timeout
2009-01-20 13:40 -0500 [r169485] Terry Wilson <twilson@digium.com>
* Don't play audio to the channel if we've masqueraded (closes
issue #14066) Reported by: bluefox Tested by: otherwiseguy,
bluefox
2009-01-16 Russell Bryant <russell@digium.com>
* Asterisk 1.4.23-rc4 released.
2009-01-16 00:19 +0000 [r168745] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump,
in this case) fails a consistency check because it can't find a
matching extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern into
a regex and uses the regex code to determine the match. I tested
using the AEL code the user supplied, and now, the consistency
check passes. (closes issue #14141) Reported by: dimas
2009-01-15 18:43 +0000 [r168721] Olle Johansson <oej@edvina.net>
* configs/extconfig.conf.sample: Meetme actually has realtime but
wasn't documented
2009-01-15 18:22 +0000 [r168716] Terry Wilson <twilson@digium.com>
* res/res_features.c: Convert call to park_call_full to
masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
return value, we need to use masqueraded parking, otherwise we
will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash. (closes issue
#14215) Reported by: waverly360 Tested by: otherwiseguy (closes
issue #14228) Reported by: kobaz Tested by: otherwiseguy
2009-01-15 01:20 +0000 [r168633] Tilghman Lesher <tlesher@digium.com>
* /: Blocked revision 168632 from /branches/1.2: 1.2 regression on
security fix AST-2009-001 (Closes issue #14238)
2009-01-15 00:11 +0000 [r168628] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix some crashes from bad datastore handling in
app_queue.c * The queue_transfer_fixup function was searching for
and removing the datastore from the incorrect channel, so this
was fixed. * Most datastore operations regarding the
queue_transfer datastore were being done without the channel
locked, so proper channel locking was added, too. (closes issue
#14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
putnopvut (license 60) Tested by: ZX81, festr
2009-01-14 21:48 +0000 [r168622] Richard Mudgett <rmudgett@digium.com>
* channels/misdn/isdn_lib.c: * Fixed create_process() allocation of
process ID values. The allocated process IDs could overflow their
respective NT and TE fields. Affects outgoing calls.
2009-01-14 20:52 +0000 [r168614] Sean Bright <sean.bright@gmail.com>
* contrib/scripts/autosupport: Update autosupport script to supply
info for both Zaptel and DAHDI in 1.4 and be sure to run
dahdi_test in 1.6.x and trunk instead of zttest. (closes issue
#14132) Reported by: dsedivec Patches:
asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
asterisk-trunk-autosupport.patch uploaded by dsedivec (license
638)
2009-01-14 19:34 +0000 [r168608] Steve Murphy <murf@digium.com>
* apps/app_page.c: app_page was failing to compile in dev-mode on
my gcc-4.2.4 system. This change gets rid of the warning.
2009-01-14 19:02 +0000 [r168603] Tilghman Lesher <tlesher@digium.com>
* main/udptl.c: Don't read into a buffer without first checking if
a value is beyond the end. (closes issue #13600) Reported by:
atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
(license 14) Tested by: atis
2009-01-14 16:19 +0000 [r168598] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Fix a logic error I found while searching
through chan_agent.c I found that the allow_multiple_logins
function would never return 0 due to an incorrect comparison
being used when traversing the list of agents. While I was
modifying this function, I also did a little bit of coding
guidelines cleanup, too.
2009-01-14 01:27 +0000 [r168593] Terry Wilson <twilson@digium.com>
* apps/app_page.c: Don't overflow when paging more than 128
extensions The number of available slots for calls in app_page
was hardcoded to 128. Proper bounds checking was not in place to
enforce this limit, so if more than 128 extensions were passed to
the Page() app, Asterisk would crash. This patch instead
dynamically allocates memory for the ast_dial structures and
removes the (non-functional) arbitrary limit. This issue would
have special importance to anyone who is dynamically creating the
argument passed to the Page application and allowing more than
128 extensions to be added by an outside user via some external
interface. The patch posted by a_villacis was slightly modified
for some coding guidelines and other cleanups. Thanks,
a_villacis! (closes issue #14217) Reported by: a_villacis
Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
uploaded by a (license 660) Tested by: otherwiseguy
2009-01-13 19:13 +0000 [r168561] Russell Bryant <russell@digium.com>
* main/indications.c, main/channel.c, apps/app_read.c,
channels/chan_misdn.c, funcs/func_channel.c,
include/asterisk/indications.h, apps/app_disa.c, main/app.c,
res/snmp/agent.c, include/asterisk/channel.h,
res/res_indications.c: Revert unnecessary indications API change
from rev 122314
2009-01-13 18:34 +0000 [r168551] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Don't pass a value with a side effect to a
macro (closes issue #14176) Reported by: paraeco Patches:
chan_sip.c.diff uploaded by paraeco (license 658)
2009-01-13 17:48 +0000 [r168546] Tilghman Lesher <tlesher@digium.com>
* funcs/func_logic.c: If either conditional is NULL, don't try
copying it. (closes issue #14226) Reported by: caspy Patches:
20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler <jpeeler@digium.com>
* res/res_agi.c: (closes issue #13881) Reported by: hoowa Update
the app CDR field for AGI commands that are not executing an
application via "exec".
* channels/chan_agent.c: (closes issue #12269) Reported by: IgorG
Tested by: denisgalvao This gits rid of the notion of an
owning_app allowing the request and hangup to be initiated by
different threads. Originating from an active agent channel
requires this. The implementation primarily changes __login_exec
to wait on a condition variable rather than a lock. Review:
http://reviewboard.digium.com/r/35/
2009-01-12 14:58 +0000 [r168482] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: I am reverting the fix made in revision
168128 (and its upward merges) after being contacted by Olle
Johansson and being shown how this fix is incorrect. Thanks to
Olle for clearing this up for me.
2009-01-12 14:57 +0000 [r168480] Russell Bryant <russell@digium.com>
* configs/indications.conf.sample: s/ringdance/ringcadence/ for
Bulgaria
2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming <kpfleming@digium.com>
* README: small commit to test new server
* README: small commit to test new server
* sounds/Makefile: update to use new sound file packages that
include license files
2009-01-09 22:14 +0000 [r168198] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Make this compile for mvanbaak
2009-01-09 21:28 +0000 [r168191] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 *
Miscellaneous doxygen comments added.
2009-01-09 20:08 +0000 [r168128] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add check_via calls to more request handlers
INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
checking the topmost Via to determine where to send the response.
Adding check_via calls to those request handlers solves this.
(closes issue #13071) Reported by: baron Patches: check_via.patch
uploaded by baron (license 531) Tested by: baron
2009-01-08 22:08 +0000 [r167840] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c: Don't truncate database results at 255 chars.
(closes issue #14069) Reported by: evandro Patches:
20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: remove an unnecessary argument to
queue_request()
* channels/chan_sip.c: When a SIP request or response arrives for a
dialog with an associated Asterisk channel, and the lock on that
channel cannot be obtained because it is held by another thread,
instead of dropping the request/response, queue it for later
processing when the channel lock becomes available.
http://reviewboard.digium.com/r/117/
2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant <russell@digium.com>
* main/file.c: Fix the last couple of places where free() was
improperly used directly.
* main/file.c: Don't fclose() the file early, the filestream
destructor will handle it.
* main/file.c: Only try to close the file if one was actually
opened
* main/file.c: Don't use free() directly. This caused a crash since
ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
#asterisk-dev
* main/indications.c: Treat an empty string the same way as a NULL
country argument. In passing, simplify the handling of returning
a default tone zone.
2009-01-06 21:35 +0000 [r167299] Mark Michelson <mmichelson@digium.com>
* main/db.c: Use the correct variable when creating the format
string (closes issue #14177) Reported by: nic_bellamy Patches:
asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
(license 299)
2009-01-06 20:48 +0000 [r167260] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
Jan 2009) | 2 lines Security fix AST-2009-001. ........
2009-01-05 16:51 +0000 [r167179] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: A couple of changes to T.38 SDP attribute
handling There are some boolean attributes for T.38 such as
T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we should treat
these as a "true" value. The current code, however, was requiring
a 1 or 0 as the value of the attribute in order to parse it. This
is due to the fact that there are some T.38 endpoints and
gateways that also transmit this information incorrectly. This
patch follows the "be liberal in what you accept and strict in
what you send" philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending information as
it is supposed to be sent. It was also discovered that a
particular type of T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
it used T38MaxDatagram and T38FaxMaxRate respectively. We now
will properly accept these attributes as well. Note that there
are a lot of patches cited in the below commit message template.
This is because the person who submitted these patches is an
awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
issue #13976) Reported by: linulin Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
(license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
(license 648) Tested by: arcivanov
2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher <tlesher@digium.com>
* channels/chan_alsa.c: Repeat attempts to write when we receive
-EAGAIN from the driver, as detailed in the ALSA sample code (see
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
Reported by: Jerry Geis (via the -users list) Fixed by: me
(license 14)
* channels/chan_local.c: Also inherit the musiconhold class.
(Closes #14153) Reported by: Jerry Geis, via the users list.
Patch by: me (license 14)
2008-12-28 15:13 +0000 [r166772] Russell Bryant <russell@digium.com>
* channels/misdn_config.c: Use strncat() instead of an sprintf() in
which source and target buffers overlap
http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
2008-12-23 15:35 +0000 [r166592] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, channels/chan_iax2.c: Compile, even if both
DAHDI and Zaptel are not installed. (Closes issue #14120)
2008-12-23 15:16 +0000 [r166568] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Fix a crash resulting from a datastore with
inheritance but no duplicate callback The fix for this is to
simply set the newly created datastore's data pointer to NULL if
it is inherited but has no duplicate callback. (closes issue
#14113) Reported by: francesco_r Patches: 14113.patch uploaded by
putnopvut (license 60) Tested by: francesco_r
2008-12-23 04:05 +0000 [r166509] Tilghman Lesher <tlesher@digium.com>
* main/channel.c: Use the integer form of condition for integer
comparisons. (closes issue #14127) Reported by: andrew
2008-12-22 20:56 +0000 [r166380] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c: Fix a deadlock relating to channel locks
and autoservice It has been discovered that if a channel is
locked prior to a call to ast_autoservice_stop, then it is likely
that a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure that
the thread running autoservice is not currently trying to
manipulate the channel we are about to pull out of autoservice.
The autoservice thread, however, cannot advance beyond where it
currently is, though, because it is trying to acquire the lock of
the channel for which autoservice is attempting to be stopped.
The gist of all this is that a channel MUST NOT be locked when
attempting to stop autoservice on the channel. In this particular
case, the channel was locked by a call to ast_read. A call to
ast_exists_extension led to autoservice being started and stopped
due to the existence of dialplan switches. It may be that there
are future commits which handle the same symptoms but in a
different location, but based on my looks through the code, it is
very rare to see a construct such as this one. (closes issue
#14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
by putnopvut (license 60) Tested by: rtrauntvein Review:
http://reviewboard.digium.com/r/107/
2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant <russell@digium.com>
* main/utils.c: Fix up timeout handling in ast_carefulwrite().
* include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref
count handling of MoH classes using astobj2 to resolve crashes.
(closes issue #13566) Reported by: igorcarneiro Tested by:
russell Review: http://reviewboard.digium.com/r/106/
2008-12-19 23:34 +0000 [r166157] Mark Michelson <mmichelson@digium.com>
* main/channel.c, funcs/func_audiohookinherit.c (added),
channels/chan_sip.c, include/asterisk/audiohook.h,
main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for
Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches:
13538.patch uploaded by putnopvut (license 60) Tested by:
putnopvut
2008-12-19 22:30 +0000 [r166093] Steve Murphy <murf@digium.com>
* apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h,
apps/app_queue.c: This merges the masqpark branch into 1.4 These
changes eliminate the need for (and use of) the KEEPALIVE return
code in res_features.c; There are other places that use this
result code for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk. The reason these
changes are being made in 1.4, is that parking ends a channel's
life, in some situations, and the code in the bridge (and some
other places), was not checking the result code properly, and
dereferencing the channel pointer, which could lead to memory
corruption and crashes. Calling the masq_park function eliminates
this danger in higher levels. A series of previous commits have
replaced some parking calls with masq_park, but this patch puts
them ALL to rest, (except one, purposely left alone because a
masquerade is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes. While
bug 13820 inspired this work, this patch does not solve all the
problems mentioned there. I have tested this patch (again) to
make sure I have not introduced regressions. Crashes that
occurred when a parked party hung up while the parking party was
listening to the numbers of the parking stall being assigned, is
eliminated. These are the cases where parking code may be
activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to
parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip
xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via
manager. The interesting testing cases for parking are: I. A
calls B, A parks B a. B hangs up while A is getting the numbers
announced. b. B hangs up after A gets the announcement, but
before the parking time expires c. B waits, time expires, A is
redialed, A answers, B and A are connected, after which, B hangs
up. d. C picks up B while still in parking lot. II. A calls B, B
parks A a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but before the
parking time expires c. A waits, time expires, B is redialed, B
answers, A and B are connected, after which, A hangs up. d. C
picks up A while still in parking lot. Testing this throroughly
involves acting all the permutations of I and II, in situations
1,2,3, and 4. Since I added a few more changes (ALL references to
KEEPALIVE in the bridge code eliimated (I missed one earlier), I
retested most of the above cases, and no crashes. H-extension
weirdness. Current h-extension execution is not completely
correct for several of the cases. For the case where A calls B,
and A parks B, the 'h' exten is run on A's channel as soon as the
park is accomplished. This is expected behavior. But when A calls
B, and B parks A, this will be current behavior: After B parks A,
B is hung up by the system, and the 'h' (hangup) exten gets run,
but the channel mentioned will be a derivative of A's... Thus, if
A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on
channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info
will be those relating to Channel A. And, in the case where A is
reconnected to B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten will be run at all.
In the case where C picks up A from the parking lot, when either
A or C hang up, the h-exten will be run for the C channel. CDR's
are a separate issue, and not addressed here. As to WHY this
strange behavior occurs, the answer lies in the procedure
followed to accomplish handing over the channel to the parking
manager thread. This procedure is called masquerading. In the
process, a duplicate copy of the channel is created, and most of
the active data is given to the new copy. The original channel
gets its name changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original thread (preserving its
role as a call originator, if it had this role to begin with),
while the new channel is without this info and becomes a call
target (a "peer"). In this case, the parking lot manager thread
is handed the new (masqueraded) channel. It will not run an
h-exten on the channel if it hangs up while in the parking lot.
The h exten will be run on the original channel instead, in the
original thread, after the bridge completes. See bug 13820 for
our intentions as to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
2008-12-19 19:48 +0000 [r165991] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c,
apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c,
apps/app_dahdiscan.c, codecs/codec_dahdi.c,
res/res_musiconhold.c, channels/chan_iax2.c: (closes issue
#13480) Reported by: tzafrir Replace a bunch of if defined checks
for Zaptel/DAHDI through several new defines in dahdi_compat.h.
This removes a lot of code duplication. Example from bug: #ifdef
HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd =
open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd =
open(DAHDI_FILE_PSEUDO, O_RDRW);
2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c: Ensure that the chanspy datastore is fully
initialized. This patch resolved some random crash issues
observed by a user on a BSD system (closes issue #14111) Reported
by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281)
* main/utils.c: Make ast_carefulwrite() be more careful. This patch
handles some additional cases that could result in partial writes
to the file description. This was done to address complaints
about partial writes on AMI. (issue #13546) (more changes needed
to address potential problems in 1.6) Reported by: srt Tested by:
russell Review: http://reviewboard.digium.com/r/99/
2008-12-18 21:14 +0000 [r165767] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Add mutexes around accesses to the IMAP
library interface. This prevents certain crashes, especially when
shared mailboxes are used. (closes issue #13653) Reported by:
howardwilkinson Patches:
asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
howardwilkinson (license 590) Tested by: jpeeler
2008-12-18 18:52 +0000 [r165661] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Set the process group ID on the MOH
process so that all children will get killed (closes issue
#14099) Reported by: caspy Patches:
res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license
645)
2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Only care about a compatible codec for early bridging
if we are actually bridging to another channel. If we are not we
actually want to bring the audio back to us. (closes issue
#13545) Reported by: davidw
* apps/app_followme.c: Do not crash if we are not passed in a
followme id. (closes issue #14106) Reported by: ys Patches:
app_followme.c.2.diff uploaded by ys (license 281)
2008-12-17 Russell Bryant <russell@digium.com>
* Asterisk 1.4.23-rc3 released.
2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher@digium.com>
* apps/app_macro.c: Reverse the fix from issue #6176 and add proper
handling for that issue. (Closes issue #13962, closes issue
#13363) Fixed by myself (license 14)
2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson@digium.com>
* apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
apps/app_queue.c: Fix some memory leaks found while looking at
how realtime configs are handled. Also cleaned up some coding
guidelines violations in app_realtime.c, mostly related to
spacing
* channels/chan_sip.c: After looking through SIP registration code
most of the day, this is one of the few things I could find that
was just plain wrong. Even though it probably isn't possible for
it to happen, it seems weird to have code that checks if a
pointer is NULL and then immediately dereferences that pointer if
it was NULL.
2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell@digium.com>
* main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
report that a thread is exiting while holding a lock. If the last
lock attempt was a trylock, and it failed, it will still be in
the list of locks so that it can be reported. (closes issue
#13219) Reported by: pj
* apps/app_macro.c: Do not dereference the channel if
AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
while looking at the code for app_macro. This return code means
that another thread has assumed ownership of the channel and it
can no longer be touched. (I hate this return code with a
passion, by the way.)
* main/manager.c: Add "restart gracefully" to the AMI blacklist of
CLI commands. "module unload" was already identified as a command
that can not be used from the AMI. "restart gracefully"
effectively unloads all modules, and will run in to the same
problems. (closes issue #13894) Reported by: kernelsensei
* include/asterisk/threadstorage.h, main/threadstorage.c: Fix
memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
One issue was that the ast_mutex_* API was being used within the
context of the thread local data destructors. We would go off and
allocate more thread local data while the pthread lib was in the
middle of destroying it all. This led to a memory leak. Another
issue was an invalid argument being provided to the the
object_add API call. (closes issue #13678) Reported by: ys Tested
by: Russell
* channels/chan_sip.c: Fix a memory leak related to the use of the
"setvar" configuration option. The problem was that these
variables were being appended to the list of vars on the sip_pvt
every time a re-registration or re-subscription came in. Since
it's just a waste of memory to put them there unless the request
was an INVITE, then the fix is to check the request type before
copying the vars. (closes issue #14037) Reported by: marvinek
Tested by: russell
2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf@digium.com>
* main/pbx.c: I added a sentence to clarify why - and ' ' are
ignored in patterns as per bug 14076. Leif says he'll put some
stuff about it in the extensions.conf sample, etc.
2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Don't try to change working directory if a
directory was not configured. (closes issue #14089) Reported by:
caspy
2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson@digium.com>
* include/asterisk/pbx.h: Add the deadlock note to
ast_spawn_extension as well
* include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
autoservice and pbx doxygen regarding a potential deadlock
scenario so that it is avoided in the future
2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Do not try to unlock a non-existant channel
if the transfer fails. (closes issue #13800) Reported by: dwagner
Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
(license 608)
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/channel.h: Use autoconf logic to determine
whether the system has timersub or not. Do not blindly assume
Solaris does not. (closes issue #13838) Reported by: ano
* apps/app_dial.c: Can we try not to assign an unsigned int to -1?
(closes issue #14074) Reported by: wetwired
2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell@digium.com>
* main/channel.c, res/res_features.c: Handle a case where a call
can be bridged to a channel that is still ringing. The issue that
was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once
the parked call got taken out of parking, it heard silence until
the other side answered. Ideally, the caller that was parked
would get a ringing indication. This patch fixes this case so
that the caller receives ringback once it comes out of parking
until the other side answers. The fixes are: - Make sure we
remember that a channel was an outgoing channel when doing a
masquerade. This prevents an erroneous ast_answer() call on the
channel, which causes a bogus 200 OK to be sent in the case of
SIP. - Add some additional comments to explain related parts of
code. - Update the handling of the ast_channel visible_indication
field. Storing values that are not stateful is pointless. Control
frames that are events or commands should be ignored. - When a
bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still
ringing. - Rework ast_indicate_data() a bit for the sake of
readability. (closes issue #13747) Reported by: davidw Tested by:
russell Review: http://reviewboard.digium.com/r/90/
2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c: Change the default calldurationlimit from the
special value 0 to -1, so we can better detect an exceptional
case. This follows on to the changes made in revision 156386.
Related to issue #13851. (closes issue #13974) Reported by:
paradise Patches: 20081208__bug13974.diff.txt uploaded by
Corydon76 (license 14) Tested by: file, blitzrage, ZX81
2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell@digium.com>
* /: Set the reviewboard:url property on 1.4, as well
2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
immediately exiting Asterisk, but also add a pointer inside
editline to look back to asterisk.c, so others don't spend as
much time as I did looking (in the wrong place) for the
appropriate function. Reported by: ZX81, via the #asterisk-users
channel Fixed by: me (license 14)
2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
so that it is actually 32 bits on 64-bit machines, as well.
(inspired by issue #13879)
* main/channel.c, main/autoservice.c, include/asterisk/channel.h:
Resolve issues that could cause DTMF to be processed out of
order. These changes come from team/russell/issue_12658 1) Change
autoservice to put digits on the head of the channel's frame
readq instead of the tail. If there were frames on the readq that
autoservice had not yet read, the previous code would have
resulted in out of order processing. This required a new API call
to queue a frame to the head of the queue instead of the tail. 2)
Change up the processing of DTMF in ast_read(). Some of the
problems were the result of having two sources of pending DTMF
frames. There was the dtmfq and the more generic readq. Both were
used for pending DTMF in various scenarios. Simplifying things to
only use the frame readq avoids some of the problems. 3) Fix a
bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit
emulation, and a digit arrived before emulation was complete,
digits would get processed out of order. (closes issue #12658)
Reported by: dimas Tested by: russell, file Review:
http://reviewboard.digium.com/r/85/
2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
on certain shells, the terminal is messed up. By intercepting
those events with a signal handler in the remote console, we can
avoid those issues. (closes issue #13464) Reported by: tzafrir
Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
(license 14) Tested by: blitzrage
2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson@digium.com>
* pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
(closes issue #13819) Reported by: adomjan Patches:
pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
dundi_clearecache3.diff uploaded by mnicholson (license 96)
Tested by: adomjan
2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell@digium.com>
* funcs/func_strings.c, funcs/func_cut.c: Fix some observed
slowdowns in dialplan processing. The change is to remove
autoservice usage from dialplan functions that do not need it
because they do not perform operations that potentially block.
(closes issue #13940) Reported by: tbelder
* res/res_features.c: Fix an issue that made it so you could only
have a single caller executing a custom feature at a time. This
was especially problematic when custom features ran for any
appreciable amount of time. The fix turned out to be quite
simple. The dynamic features are now stored in a read/write list
instead of a list using a mutex. (closes issue #13478) Reported
by: neutrino88 Fix suggested by file
2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c: Don't wait forever, if there's a specified
recording timeout. (closes issue #13885) Reported by: bamby
Patches: res_agi.c.patch uploaded by bamby (license 430)
2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Revert this cast to long. Using time_t here
causes build failures on a FreeBSD 32-bit build.
* apps/app_queue.c: Fix a potential crash due to unsafe datastore
handling. This patch also contains a conversion from using long
to time_t for representing times for a queue, as well as some
whitespace fixes. (closes issue #14060) Reported by: nivek
Patches: datastore_fixup.patch.corrected uploaded by nivek
(license 636) with slight modification from me Tested by: nivek
2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler@digium.com>
* res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
check. Pointed out by mmichelson, thanks!
* res/res_musiconhold.c: (closes issue #13229) Reported by:
clegall_proformatique Ensure that moh_generate does not return
prematurely before local_ast_moh_stop is called. Also, the sleep
in mp3_spawn now only occurs for http locations since it seems to
have been added originally only for failing media streams.
2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix subscription based MWI up a bit. We only
want to put sip: at the beginning of the URI if it is not already
there and revert code to ignore destination check if subscribing
for MWI. (closes issue #12560) Reported by: vsauer Patches:
patch001.diff uploaded by ramonpeek (license 266)
* channels/chan_sip.c: When a SIP peer unregisters set the expiry
time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599) Reported by: hjourdain Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf@digium.com>
* pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
Reported by: wetwired Tested by: murf I checked, and I added a
mod to the trunk version of Asterisk that would make it 8-bit
transparent on 27 Nov 2007, but I made no such updates to 1.4. My
best guess is that 1.4 was released, and it was not appropriate
to commit an enhancement. But I'm going to add the same fixes to
1.4 now, for the following reasons: 1. wetwired is correct; 1.4
is **mostly** 8-bit transparent now. This is because the lexical
token forming rules use . in most 'word' state continuances. It's
just the beginning of a 'word' that is picky. 2. Accepting 8-bit
chars in some places and not others leads to bug reports like
this.
2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson@digium.com>
* include/asterisk/stringfields.h: Update to stringfield handling
so that side-effects on parameters are not evaluated multiple
times. An example where this caused a problem was in chan_sip.c,
with the line ast_string_field_set(p, fromdomain, ++fromdomain);
This patch was originally uploaded to issue #13783 by jamessan.
While the issue was closed for other reasons, this patch is valid
and fixes a separate problem, and is thus being committed.
* channels/chan_sip.c: Revert fix for issue 13570. It has caused
more problems than it helped to fix. (closes issue #13783)
Reported by: navkumar (closes issue #14025) Reported by: ffs
* doc/misdn.txt: Add missing documentation to misdn.txt (closes
issue #14052) Reported by: festr Patches: misdn.txt.patch
uploaded by festr (license 443)
2008-12-10 16:05 +0000 [r162653] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Increment the sequence number on the end packets for
RFC2833. After reading the RFC some more and doing some testing I
agree with this change. (closes issue #12983) Reported by: vt
Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
520)
2008-12-09 23:08 +0000 [r162463] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Oops, should be "tz", not "zonetag".
2008-12-09 22:17 +0000 [r162413] Russell Bryant <russell@digium.com>
* main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove
the test_for_thread_safety() function completely. The test is not
valid. Besides, if we actually suspected that recursive mutexes
were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on. (inspired by a discussion on the
asterisk-dev list)
2008-12-09 21:53 +0000 [r162348] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: We appear to have documented tz= in the
[general] section of voicemail.conf, without actually having
implemented it. Oops. (Reported by Olivier on the -users list)
2008-12-09 21:14 +0000 [r162341] Joshua Colp <jcolp@digium.com>
* apps/app_directed_pickup.c: Add 'down' as a valid state for
directed call pickup. This creeps up when we receive session
progress when dialing a device and not ringing. (closes issue
#14005) Reported by: ddl
2008-12-09 20:57 +0000 [r162286] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Fix an issue where callers on an incoming call
on an SLA trunk would not hear ringback. We need to make sure
that we don't start writing audio to the trunk channel until
we're actually ready to answer it. Otherwise, the channel driver
will treat it as inband progress, even though all they are
getting is silence. (closes issue #12471) Reported by: mthomasslo
2008-12-09 20:44 +0000 [r162273] Joshua Colp <jcolp@digium.com>
* apps/app_festival.c: Fix double declaration of 'x' on the PPC
platform. (closes issue #14038) Reported by: ffloimair
2008-12-09 20:28 +0000 [r162265] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: If we fail to start a thread for the pbx to run in,
we need to be sure to decrease the number of active calls on the
system. This fix may relate to ABE-1713, but it is not certain
yet.
2008-12-09 20:20 +0000 [r162264] Steve Murphy <murf@digium.com>
* pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with
seanbright on #asterisk-dev, I have added a default rule, and an
option to suppress the default rule from being generated in the
flex output, for the sake of those OS's where they didn't tweak
flex's ECHO macro, and the compiler doesn't like it. The
regressions are OK with this.
2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Make sure that the timestamp for DTMF is not the same
as the previous voice frame and do not send audio when
transmitting DTMF as this confuses some equipment. (closes issue
#13209) Reported by: ip-rob Patches: 13209.diff uploaded by file
(license 11) Tested by: ip-rob, bujones
* main/rtp.c: Take video into account when early bridging RTP.
(closes issue #13535) Reported by: davidw
2008-12-09 18:13 +0000 [r162136] Steve Murphy <murf@digium.com>
* pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc
and ast_copy_string and messed up the standalone stuff. Fixed.
2008-12-09 17:07 +0000 [r162071] Tilghman Lesher <tlesher@digium.com>
* channels/chan_phone.c: For some reason, after a distclean, gcc
started returning 'value computed is not used'. Fixing (for
--enable-dev-mode).
2008-12-09 16:46 +0000 [r162014] Russell Bryant <russell@digium.com>
* apps/app_disa.c: Allow DISA to handle extensions that start with
#. (closes issue #13330) Reported by: jcovert
2008-12-09 16:31 +0000 [r162013] Steve Murphy <murf@digium.com>
* pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h,
pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme
Patches: 14019.diff uploaded by murf (license 17) Tested by:
ckjohnsonme, murf This crash was the result of a few small errors
that would combine in 64-bit land to result in a crash. 32-bit
land might have seen these combine to mysteriously drop the args
to an application call, in certain circumstances. Also, in trying
to find this bug, I spotted a situation in the flex input, where,
in passing back a 'word' to the parser, it would allocate a
buffer larger than necessary. I changed the usage in such
situations, so that strdup was not used, but rather, an
ast_malloc, followed by ast_copy_string. I removed a field from
the pval struct, in u2, that was never getting used, and set in
one spot in the code. I believe it was an artifact of a previous
fix to make switch cases work invisibly with extens. And, for
goto's I removed a '!' from before a strcmp, that has been there
since the initial merging of AEL2, that might prevent the proper
target of a goto from being found. This was pretty harmless on
its own, as it would just louse up a consistency check for users.
Many thanks to ckjohnsonme for providing a simplified and
complete set of information about the bug, that helped
considerably in finding and fixing the problem. Now, to get
aelparse up and running again in trunk, and out of its "horribly
broken" state, so I can run the regression suite!
2008-12-09 14:52 +0000 [r161948] Russell Bryant <russell@digium.com>
* main/app.c: Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old
channel to the new one. However, it did nothing with the group
settings that were already on the new channel. This patch removes
all group settings that already existed on the new channel. I
have a more complicated version of this patch which addresses
only the most blatant problem with this, which is that a channel
can end up with multiple group settings in the same category.
However, I could not think of a use case for keeping any of the
group settings from the old channel, so I went this route for
now. (closes AST-152)
2008-12-08 17:52 +0000 [r161725] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make the usereqphone option work again.
(closes issue #13474) Reported by: mmaguire Patches:
20080912_bug13474.diff uploaded by mmaguire (license 571)
2008-12-05 21:02 +0000 [r161426] Sean Bright <sean.bright@gmail.com>
* main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
161421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
int). (closes issue #14006) Reported by: alphaque Patches:
astobj2.h-patch uploaded by alphaque (license 259) (Slightly
modified by seanbright) ........
2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard <dhubbard@digium.com>
* utils/smsq.c: kill a warning
2008-12-05 14:12 +0000 [r161287] Russell Bryant <russell@digium.com>
* main/pbx.c: Fix a NULL format string warning found by buildbot.
2008-12-04 18:30 +0000 [r161013] Jeff Peeler <jpeeler@digium.com>
* main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by:
jpeeler This mirrors a check that was present in ast_rtp_read to
also be in ast_rtp_raw_write to not schedule sending the receiver
report if the remote RTCP endpoint address isn't present in the
RTCP structure. Closes AST-142.
2008-12-04 16:44 +0000 [r160943] Mark Michelson <mmichelson@digium.com>
* main/callerid.c: Fix a callerid parsing issue. If someone
formatted callerid like the following: "name <number>" (including
the quotation marks), then the parts would be parsed as name:
"name number: number This is because the closing quotation mark
was not discovered since the number and everything after was
parsed out of the string earlier. Now, there is a check to see if
the closing quote occurs after the number, so that we can know if
we should strip off the opening quote on the name. Closes AST-158
2008-12-03 21:54 +0000 [r160770] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Some compilers warn on null format strings;
some don't (caught by buildbot)
2008-12-03 21:38 +0000 [r160764] Jason Parker <jparker@digium.com>
* channels/chan_agent.c: Only show this warning when we want to
show it. (closes issue #13982) Reported by: coolmig Patches:
chan_agent.c.patch uploaded by coolmig (license 621)
2008-12-03 20:41 +0000 [r160703] Steve Murphy <murf@digium.com>
* funcs/func_callerid.c: (closes issue #13597) Reported by:
john8675309 Patches: patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309 This patch causes the setcid func to
update the CDR clid after setting the channel field.
2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: During bridge code, the channel bridge may
return a retry code, if a transfer was initiated but not yet
completed. If the bridge is immediately retried, then we may send
a storm of TXREQ packets, even though the first set is sent
reliably (retransmitted). Fixes AST-137.
* pbx/pbx_spool.c: If an entry is added to the directory during a
scan when another entry expires, then that new entry will not be
processed promptly, but must wait for either a future entry to
start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch
expire, then the new entries must wait until another new entry is
added to be processed. This was a rather weird race condition,
really. Fixes AST-147.
* pbx/pbx_spool.c: Don't start scanning the directory until all
modules are loaded, because some required modules (channels,
apps, functions) may not yet be in memory yet. Fixes AST-149.
* channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev
list: "I guess that having only ip-phones in mind is not a good
approach. Since it is possible to have a sip proxy connected to
asterisk we could receive a 407 (unauthorized) or 483 (too many
hops) as response and dialog ending would not be a good
behavior." So modified.
2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson <twilson@digium.com>
* res/res_features.c: Channel is masqueraded, don't keep alive
* res/res_features.c: A situation like A calls B, A builtin_atxfers
B to C, C parks B would lead to a crash. Thanks to file for
telling me how to fix it! (closes issue #13854) Reported by: Adam
Lee Tested by: otherwiseguy
2008-12-02 17:42 +0000 [r160297] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: When the text does not match exactly (e.g.
RTP/SAVP), then the %n conversion fails, and the resulting
integer is garbage. Thus, we must initialize the integer and
check it afterwards for success. (closes issue #14000) Reported
by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by
folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded
by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
uploaded by folke (license 626)
2008-12-02 01:16 +0000 [r160266] Terry Wilson <twilson@digium.com>
* include/asterisk/astmm.h: make compile with dev mode and malloc
debug
2008-12-02 00:25 +0000 [r160207] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/stringfields.h, apps/app_voicemail.c,
main/pbx.c, main/frame.c: Ensure that Asterisk builds with
--enable-dev-mode, even on the latest gcc and glibc.
2008-12-01 Tilghman Lesher <tilghman@digium.com>
* Released 1.4.23-rc2
2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
iax2_setoption(), as well, since they both have the potential to
send control frames in the middle of call setup. We have to wait
until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will
consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel@vanbaak.info>
* main/manager.c: Get rid of the useless format string and argument
in the Bogus/ manager channelname. Noted by kpfleming and name
Bogus/manager suggested by eliel
2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell@digium.com>
* .cleancount: Force a "make clean" to avoid a bizarre build issue
...
2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel@vanbaak.info>
* main/manager.c: make manager compile on OpenBSD. The last (10th)
argument to ast_channel_alloc here should be a pointer and NULL
is not really a pointer.
2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming@digium.com>
* main/enum.c, utils/frame.c, configure, res/res_agi.c,
include/asterisk/module.h, main/logger.c, main/dns.c,
include/asterisk/threadstorage.h, include/asterisk/utils.h,
include/asterisk/devicestate.h, channels/chan_sip.c,
include/asterisk/dundi.h, main/jitterbuf.c,
channels/chan_agent.c, configure.ac, utils/astman.c,
include/asterisk/cli.h, include/asterisk/channel.h,
include/jitterbuf.h, include/asterisk/manager.h,
main/ast_expr2.c, Makefile, include/asterisk/logger.h,
include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
include/asterisk/linkedlists.h, include/asterisk/lock.h,
include/asterisk/strings.h, makeopts.in,
include/asterisk/stringfields.h, utils/check_expr.c,
channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
update dev-mode compiler flags to match the ones used by default
on Ubuntu Intrepid, so all developers will see the same warnings
and errors since this branch already had some printf format
attributes, enable checking for them and tag functions that
didn't have them format attributes in a consistent way
2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_oss.c, channels/busy.h (removed),
channels/ring_tone.h (added), channels/chan_alsa.c,
channels/ringtone.h (removed), channels/busy_tone.h (added),
channels/Makefile: rename these files so as to avoid conflicts
when users update their working copies and have unversioned files
already in place
* channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
(added), channels/Makefile: simplify (and slightly bug-fix) the
recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
and channels/ringtone.h to the repository instead of generating
them repeatedtly; most users do not change the settings to build
them, but the Makefile rules are still there if they wish to do
so ensure that 'make clean' removes dependency files for .i files
that are created in COMPILE_DOUBLE mode
2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf@digium.com>
* main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
fix. the change to cdr.c allows no-answer to percolate up into
CDR's, and feels like the right place to locate this fix; if BUSY
is done here, no-answer should be, too.
2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
(closes issue #13919) Reported by: barthpbx Patches:
chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
barthpbx
* /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
Nov 2008) | 7 lines Regression fix for last security fix. Set the
iseqno correctly. (closes issue #13918) Reported by: ffloimair
Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
(license 14) Tested by: ffloimair ........
2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell@digium.com>
* main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
go along with ao2_lock() and ao2_unlock()
2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson@digium.com>
* apps/app_festival.c: Add missing variable declaration in the PPC
code
2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher@digium.com>
* apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
configure.ac: System call ioperm is non-portable, so check for
its existence in autoconf. (Closes issue #13863)
2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
issue #13786) Reported by: tzafrir When compiling against Zaptel
dahdi_compat will now only define all the DAHDI defines if the
Zaptel define is present. Also, there is no such thing as
DAHDI_PRI.
2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf@digium.com>
* res/res_features.c: In reference to the fix made for 13871, I was
merging the fix into 1.6.0 and realized I missed the code in the
h-exten block, and didn't catch it because my test case had the
h-exten commented out. So, this corrects the code I missed, as a
preventative against another crash report. Tested with the
h-exten defined, all is well.
2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: The passed extension may not be the same in the list
as the current entry, because we strip spaces when copying the
extension into the structure. Therefore, use the copied item to
place the item into the list. (found by lmadsen on -dev, fixed by
me)
2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell@digium.com>
* main/astobj2.c, include/asterisk/astobj2.h: When compiling with
DEBUG_THREADS, report the real file/func/line for
ao2_lock/ao2_unlock
2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf@digium.com>
* res/res_features.c: (closes issue #13871) Reported by: mdu113
This one is totally my fault. The code doesn't even create a
bridge if the channel CDR has POST_DISABLED. I didn't check for
that at the end of the bridge. Fixed with a few small insertions.
Tested. Looks good. No cdr generated, no crash, no unnecc. data
objects created either.
2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This change had somehow gotten reverted due to
a completely unrelated commit. Thanks to Theo Belder on the
Asterisk-dev list for pointing this out.
* include/asterisk/file.h, main/frame.c, main/file.c,
include/asterisk/frame.h: There was an issue when attempting to
reference an embedded frame in a freed ast_filestream. This patch
makes use of the ao2 functions to make sure that we do not free
an ast_filestream structure until the embedded ast_frame has been
"freed" as well. (closes issue #13496) Reported by: fst-onge
Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
60) Tested by: putnopvut Closes AST-89
* channels/chan_sip.c: We don't handle 4XX responses to BYE well.
According to section 15 of RFC 3261, we should terminate a dialog
if we receive a 481 or 408 in response to our BYE. Since I am
aware of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog. (closes issue
#12994) Reported by: pabelanger Patches: 12994.patch uploaded by
putnopvut (license 60) Closes AST-129
* apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
cause on the calling channel in the case that ast_call() fails.
For incoming SIP channels, this was causing us to send a 603
instead of a 486 when the call-limit was reached on the
destination channel. (closes issue #13867) Reported by: still_nsk
Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
blitzrage
2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
........ Fixes JIRA ABE-1726 The dial extension could be empty if
you are using MISDN_KEYPAD to control ISDN provider features.
2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming@digium.com>
* main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
res/snmp, res/Makefile: the gcc optimizer frequently finds broken
code (use of uninitalized variables, unreachable code, etc.),
which is good. however, developers usually compile with the
optimizer turned off, because if they need to debug the resulting
code, optimized code makes that process very difficult. this
means that we get code changes committed that weren't adequately
checked over for these sorts of problems. with this build system
change, if (and only if) --enable-dev-mode was used and
DONT_OPTIMIZE is turned on, when a source file is compiled it
will actually be preprocessed (into a .i or .ii file), then
compiled once with optimization (with the result sent to
/dev/null) and again without optimization (but only if the first
compile succeeded, of course). while making these changes, i did
some cleanup work in Makefile.rules to move commonly-used
combinations of flag variables into their own variables, to make
the file easier to read and maintain
2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Add some missing invite state changes
necessary in the sip_write function. Not setting the invite state
correctly on the call was resulting in the Record application
leaving empty files. I also have updated the doxygen comment next
to the declaration of the INV_EARLY_MEDIA constant to reflect
that we also use this state when we *send* a 18X response to an
INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
(license 162) Tested by: putnopvut
2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
This fix is the result of a bug fix in ast_app_separate_args
r124395. If an argument does not exist it should always be set to
a null string rather than a null pointer.
2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, channels/chan_local.c, res/res_features.c,
include/asterisk/channel.h, apps/app_followme.c: Fix a crash in
the end_bridge_callback of app_dial and app_followme which would
occur at the end of an attended transfer. The error occurred
because we initially stored a pointer to an ast_channel which
then was hung up due to a masquerade. This commit adds a "fixup"
callback to the bridge_config structure to allow for
end_bridge_callback_data to be changed in the case that a new
channel pointer is needed for the end_bridge_callback.
2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, Makefile.rules: when an individual directory dist-clean
is run, run clean in that directory first, and when running
top-level dist-clean, do not run subdirectory clean operations
twice
* Makefile.moddir_rules: dist-clean should remove dependency
information files as well
* contrib/asterisk-ng-doxygen: major update to doxygen
configuration file: 1) update to doxygen 1.5.x style file, as
used in trunk 2) tell doxygen where are header files are, so
include-file processing can be done 3) make all macros that are
used to define variables/functions be expanded, so that doxygen
will properly document the resulting variable/function 4) make
all macros that are used to provide the contents of a variable
(structure) be expanded, so that doxygen will be able to document
the resulting fields 5) suppress compiler attributes
(__attribute__(xxx)) from being seen by doxygen, so it will
properly match up function definition and usage (for an example
of th effect of this, look at the doxygen docs for ast_log() from
before and afte this commit)
2008-11-14 15:18 +0000 [r156816] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: If the prompt to reenter a voicemail
password timed out, it resulted in the password not being saved,
even if the input matched what you gave when first prompted to
enter a new password. This is because the return value of
ast_readstring was checked, but not checked properly. This bug
was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!
2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher <tlesher@digium.com>
* apps/app_while.c: ast_waitfordigit() requires that the channel be
up, for no good logical reason. This prevents While/EndWhile from
working within the "h" extension. Reported by: jgalarneau (for
ABE C.2) Fixed by: me
* main/manager.c: Provide more space for all the data which can
appear in an originating channel name. (closes issue #13398)
Reported by: bamby Patches: manager.c.diff uploaded by bamby
(license 430)
2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming <kpfleming@digium.com>
* configure, autoconf/ast_gcc_attribute.m4: revert this change...
non-functional changes don't belong here
* configure, autoconf/ast_gcc_attribute.m4: correct minor syntax
error... no functional change
2008-11-12 21:18 +0000 [r156386] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c: When using call limits under 1 second, infinite
call lengths are allowed, instead. (closes issue #13851) Reported
by: ruddy
2008-11-12 19:36 +0000 [r156297] Steve Murphy <murf@digium.com>
* main/pbx.c: It turns out that the 0x0XX00 codes being returned
for N, X, and Z are off by one, as per conversation with jsmith
on #asterisk-dev; he was teaching a class and disconcerted that
this published rule was not being followed, with patterns _NXX,
_[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
have been. This change, tested on these 3 patterns now picks the
proper one. However, this change may surprise users who set up
dialplans based on previous behavior, which has been there for
what, 2 and half years or so now.
2008-11-12 19:26 +0000 [r156294] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: If the SLA thread is not started, then reload
causes a memory leak. (closes issue #13889) Reported by: eliel
Patches: app_meetme.c.patch uploaded by eliel (license 64)
2008-11-12 19:10 +0000 [r156289] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c: For whatever reason, gcc only warned me about
the possible use of an uninitialized variable when compiling
1.6.1.
2008-11-12 18:39 +0000 [r156229] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Revert revision 132506, since it
occasionally caused IAX2 HANGUP packets not to be sent, and
instead, schedule a task to destroy the iax2 pvt structure 10
seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed. (closes issue
#13645) Reported by: dzajro Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
2008-11-12 17:53 +0000 [r156178] Jeff Peeler <jpeeler@digium.com>
* apps/app_meetme.c: (closes issue #13173) Reported by: pep This
change adds an announce_thread responsible for playing
announcements to an existing conference. This allows all
announcing to be immediately stopped if necessary but more
importantly allows other threads that need to play something to
not block. There are multiple benefits to this, but the actual
bug is for solving the scenario for a channel to be unusable
after hang up for the entire duration of the parting
announcement. The parting announcement can be extremely long
depending on what the user recorded upon joining the conference.
Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/
2008-11-12 17:38 +0000 [r156167] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: When doing some tests, I was having a crash at
the end of every call if an attended transfer occurred during the
call. I traced the cause to the CDR on one of the channels being
NULL. murf suggested a check in the end bridge callback to be
sure the CDR is non-NULL before proceeding, so that's what I'm
adding.
2008-11-12 17:29 +0000 [r156164] Russell Bryant <russell@digium.com>
* main/asterisk.c: Move the sanity check that makes sure "always
fork" is not set along with the console option to be after the
code that reads options from asterisk.conf. This resolves a
situation where Asterisk can start taking up 100% when
misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting
me log in to his system to figure out what was causing the 100%
CPU problem.)
2008-11-10 21:07 +0000 [r155861] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Channel drivers assume that when their
indicate callback is invoked, that the channel on which the
callback was called is locked. This patch corrects an instance in
chan_agent where a channel's indicate callback is called directly
without first locking the channel. This was leading to some
observed locking issues in chan_local, but considering that all
channel drivers operate under the same expectations, the generic
fix in chan_agent is the right way to go. AST-126
2008-11-10 20:49 +0000 [r155803] Tilghman Lesher <tlesher@digium.com>
* doc/valgrind.txt: I got tired of saying this in every single
bugnote referring to this file.
2008-11-09 01:08 +0000 [r155553] Sean Bright <sean.bright@gmail.com>
* apps/app_dial.c, res/res_features.c, include/asterisk/channel.h,
apps/app_followme.c: Use static functions here instead of nested
ones. This requires a small change to the ast_bridge_config
struct as well. To understand the reason for this change, see the
following post:
http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
2008-11-07 22:27 +0000 [r155398] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Clarify error message. (closes issue #13809)
Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded
by Corydon76 (license 14) Tested by: denke
2008-11-06 19:45 +0000 [r155011] Mark Michelson <mmichelson@digium.com>
* configs/voicemail.conf.sample: The documentation listed the
ability to set 'maxmsg' per context. The truth is that you can
only set this in the general section or per mailbox. Thus I am
updating the sample config file to be more accurate. Thanks to
sasargen on IRC for bringing up this issue.
2008-11-05 16:44 +0000 [r154724] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: The logic of a strcasecmp call was
reversed (closes issue #13841) Reported by: clegall_proformatique
2008-11-05 16:06 +0000 [r154685] Steve Murphy <murf@digium.com>
* main/channel.c: This fix was prompted by communication from user,
who was seeing thousands of error logs... looks like EAGAIN. Made
such uninteresting.
2008-11-04 20:49 +0000 [r154365] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: On busy systems, it's possible for the
values checked within a single line of code to change, unless the
structure is locked to ensure a consistent state. (closes issue
#13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
uploaded by Corydon76 (license 14) Tested by: kowalma
2008-11-04 19:01 +0000 [r154266] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to
the wrong state when it receives the indication
AST_CONTROL_RINGING.
2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher <tlesher@digium.com>
* channels/chan_h323.c: Make the monitor thread non-detached, so it
can be joined (suggested by Russell on -dev list).
* apps/app_voicemail.c: Attempting to expunge a mailbox when the
mailstream is NULL will crash Asterisk. (Closes issue #13829)
Reported by: jaroth Patch by: me (modified jaroth's patch)
* main/rtp.c: Remove the potential for a division by zero error.
(Closes issue #13810)
2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c,
main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c,
res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in
this branch on the first pass
2008-11-02 19:51 +0000 [r153651] Russell Bryant <russell@digium.com>
* include/asterisk/features.h: features.h depends on linkedlists.h,
so include it
2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming <kpfleming@digium.com>
* utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c,
funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c,
channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c,
pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c,
pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c,
formats/format_wav.c, apps/app_festival.c,
main/db1-ast/hash/hash_page.c, main/translate.c,
res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c,
utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c,
pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h,
apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c,
utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c,
apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y,
apps/app_authenticate.c, formats/format_wav_gsm.c,
res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of
potential problems found by gcc 4.3.x, primarily bare strings
being passed to printf()-like functions and ignored results from
read()/write() and friends
2008-10-31 22:36 +0000 [r153270] Terry Wilson <twilson@digium.com>
* res/res_features.c, apps/app_followme.c: Add end_bridge_callback
for app_follome and add AUTOLOOP flag to res_features
2008-10-31 Tilghman Lesher <tlesher@digium.com>
* Asterisk 1.4.23-rc1 released.
2008-10-31 16:30 +0000 [r153114] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Turn off qualify on uncached realtime peers.
(Closes issue #13383)
2008-10-31 15:45 +0000 [r153095] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, res/res_features.c, include/asterisk/channel.h:
Recent CDR fixes moved execution of the 'h' exten into the
bridging code, so variables that were set after ast_bridge_call
was called would not show up in the 'h' exten. Added a callback
function to handle setting variables, etc. from w/in the bridging
code. Calls back into a nested function within the function
calling ast_bridge_call (closes issue #13793) Reported by:
greenfieldtech
2008-10-30 20:58 +0000 [r152992] Sean Bright <sean.bright@gmail.com>
* bootstrap.sh: The -I argument to aclocal needs a space before the
include directory name.
2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher <tlesher@digium.com>
* channels/chan_h323.c: Cannot join detached threads. See
http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400)
* channels/chan_local.c: Unlock before returning, when extension
doesn't exist. (closes issue #13807) Reported by: eliel Patches:
chan_local.c.patch uploaded by eliel (license 64)
2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming <kpfleming@digium.com>
* main/cdr.c: instead of comparing the string pointer to 0, let's
compare the value that was actually parsed out of the string
(found by sparse)
2008-10-29 05:23 +0000 [r152539] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
issue #13795) Reported by: andrew53 Patches:
chan_sip_sizeof.patch uploaded by andrew53 (license 519)
2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy <murf@digium.com>
* configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c:
A little documentation cross-ref between features and dial and
queue... I wasted some time (stupidly) trying to get the
one-touch parking stuff working, because it didn't occur to me
that I had to also have the corresponding options in the dial
command! Duh! (In all this time, I never set this up before!) So,
to keep some poor fool from suffering the same fate, I made the
features.conf.sample file mention the corresponding opts in
dial/queue; and the docs for dial/app specifically mention the
corresponding decls in the feature.conf file. I hope this doesn't
spoil some vast, eternal plan...
* apps/app_dial.c, res/res_features.c, funcs/func_channel.c,
include/asterisk/pbx.h, apps/app_queue.c: The magic trick to
avoid this crash is not to try to find the channel by name in the
list, which is slow and resource consuming, but rather to pay
attention to the result codes from the ast_bridge_call, to which
I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are
returned when a channel is parked. If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer. If you get
AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
don't touch the peer pointer. Updated the several places where
the results from a bridge were not being properly obeyed, and
fixed some code I had introduced so that the results of the
bridge were not overridden (in trunk). All the places that
previously tested for AST_PBX_NO_HANGUP_PEER now have to check
for both AST_PBX_NO_HANGUP_PEER and
AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
while A is getting the parking slot announcement, immediately
after being put on hold. 2. A calls B; B answers; A parks B; B
hangs up after A has been hung up, but before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting
the parking slot announcement, immediately after being put on
hold. 4. A calls B; B answers; B parks A; A hangs up after B has
been hung up, but before the park times out. No crash. I also ran
the scenarios above against valgrind, and accesses looked good.
2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Quoting in the wrong direction (Fixes
AST-107)
* apps/app_dial.c: Reset all DIAL variables back to blank, in case
Dial is called multiple times per call (which could otherwise
lead to inconsistent status reports). (closes issue #13216)
Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
by Corydon76 (license 14) Tested by: ruddy
2008-10-27 23:28 +0000 [r152286] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Buffer policy setting for half is not
needed.
2008-10-27 21:32 +0000 [r152215] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c: Inherit ALL elements of CallerID across a
local channel. (closes issue #13368) Reported by: Peter Schlaile
Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
(license 14)
2008-10-26 20:23 +0000 [r152059] Sean Bright <sean.bright@gmail.com>
* funcs/func_strings.c: Since passing \0 as the second argument to
strchr is valid (and will match the trailing \0 of a string) we
need to check that first, otherwise we end up with incorrect
results. Fix suggested by reporter. (closes issue #13787)
Reported by: meitinger
2008-10-25 10:59 +0000 [r151905] Russell Bryant <russell@digium.com>
* main/asterisk.c: Move AMI initialization to occur after loading
modules. This prevents a deadlock when someone tries to initiate
a module reload from the AMI just as Asterisk is starting.
(closes issue #13778) Reported by: hotsblanc Fix suggested by
hotsblanc
2008-10-23 16:04 +0000 [r151763] Terry Wilson <twilson@digium.com>
* configs/features.conf.sample, res/res_features.c, CHANGES:
Backport fix from 1.6.0 that allows you to set
parkedcalltransfers=no|caller|callee|both, but default to both
which would be the equivalent of the existing behaviour. The
problem was that if someone parked a call, the callee and caller
would both get assigned the builtin transfer feature, which would
not only be potentially giving someone the ability to transfer
themselves when they shouldn't have it, but would also dissallow
reinviting the media off of the call. (closes issue #12854)
Reported by: davidw Patches: parkingfix4.diff.txt uploaded by
otherwiseguy Tested by: davidw, otherwiseguy
2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming <kpfleming@digium.com>
* autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4,
configure.ac: rename this macro to properly reflect what it does
* autoconf/ast_check_pwlib.m4 (added), autoconf (added),
autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4
(added), configure, autoconf/ast_gcc_attribute.m4 (added),
bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added),
autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added),
autoconf/ast_c_compile_check.m4 (added),
autoconf/ast_c_define_check.m4 (added),
autoconf/ast_prog_egrep.m4 (added),
autoconf/ast_check_openh323.m4 (added),
autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4
(added), acinclude.m4 (removed): break up acinclude.m4 into
individual files, which will make it easier to maintain, easier
to add new macros (less patching) and will ease maintenance of
these macros across Asterisk branches
2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke <bweschke@btwtech.com>
* main/asterisk.c: As per kpfleming's comments to the prior commit,
I'm reverting some of the changes here. A comment was made in bug
#13726 "3. The same mistake as in (2) is done in a few other
places in the code that check for: #if defined(HAVE_ZAPTEL) ||
defined(HAVE_DAHDI) Harmless, but still incorrect." In the case
of main/asterisk.c, this is not incorrect because without
HAVE_ZAPTEL defined, we're missing the include for ioctl and the
namespace that defines DAHDI_TIMERCONFIG which is still required
when using Zaptel with the 1.4 branch.
* main/asterisk.c: Fix the 1.4 branch compile again broken with
r150557 when using with Zaptel and not DAHDI (closes issue
#13740) reported by: jmls patch by: bweschke
2008-10-18 01:42 +0000 [r150816] BJ Weschke <bweschke@btwtech.com>
* main/manager.c: Using the GetVar handler in AMI is potentially
dangerous (insta-crash [tm]) when you use a dialplan function
that requires a channel and then you don't provide one or provide
an invalid one in the Channel: parameter. We'll handle this
situation exactly the same way it was handled in pbx.c back on
r61766. We'll create a bogus channel for the function call and
destroy it when we're done. If we have trouble allocating the
bogus channel then we're not going to try executing the function
call at all and run the risk of crashing. (closes issue #13715)
reported by: makoto patch by: bweschke
2008-10-17 17:18 +0000 [r150637] Steve Murphy <murf@digium.com>
* res/res_features.c: Interesting crash. In this case, you exit the
bridge with peer completely GONE. I moved the channel find call
up to cover the whole peer CDR reset code segment. This appears
to solve the crash without changing the logic at all.
2008-10-17 15:31 +0000 [r150557] Jason Parker <jparker@digium.com>
* main/asterisk.c, main/channel.c, channels/chan_dahdi.c,
configure, configure.ac: Correctly allow chan_dahdi to compile
against older versions of Zaptel. Don't always define
HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor
cleanup to make things clear. (closes issue #13726) Reported by:
tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46)
2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Reverting changes from commits 150298 and 150301
since I was mistakenly under the assumption that dialplan
functions *always* required that a channel be present. I need to
go home earlier, I think :)
* main/manager.c: And don't forget to return on the error condition
* main/manager.c: Don't try to call a dialplan function's read
callback from the manager's GetVar handler if an invalid channel
has been specified. Several dialplan functions, including CHANNEL
and SIP_HEADER, do not check for NULL-ness of the channel being
passed in. (closes issue #13715) Reported by: makoto
2008-10-16 15:56 +0000 [r150124] Richard Mudgett <rmudgett@digium.com>
* channels/chan_misdn.c: Fix memory leak found by customer
2008-10-16 15:26 +0000 [r150056] Steve Murphy <murf@digium.com>
* cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and
RYM-150398 and AST-103 in internal Digium bug trackers. These
fixes address a really subtle memory corruption problem that
would happen in machines heavily loaded in production
environments. The corruption would always take the form of the
STMT object getting nulled out and one of the unixODBC calls
would crash trying to access statement->connection. It isn't
fully proven yet, but the server has now been running 2.5 days
without appreciable memory growth, or any gain of %cpu, and no
crashes. Whether this is the problem or not on that server, these
fixes are still warranted. As it turns out, **I** introduced
these errors unwittingly, when I corrected another crash earlier.
I had formed the build_query routine, and failed to remove
mutex_unlock calls in 3 places in the transplanted code. These
unlocks would only happen in error situations, but unlocking the
mutex early set the code up for a catastrophic failure, it
appears. It would happen only once every 100K-200K or more calls,
under heavy load... but that is enough. If another crash occurs,
with the same MO, I'll come back and remove my confession from
the log, and we'll keep searching, but the fact that we have
Asterisk dying from an asynchronous wiping of the STMT object,
only on some connection error, and that the server has lived for
2.5 days on this code without a crash, sure make it look like
this was the problem! Also, in several points, Statement handles
are set to NULL after SQLFreeHandle. This was mainly for
insurance, to guarantee a crash. As it turns out, the code does
not appear to be attempting to use these freed pointers. Asterisk
owes a debt of gratitude to Federico Alves and Frediano Ziglio
for their untiring efforts in finding this bug, among others.
2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke <bweschke@btwtech.com>
* CHANGES: Another documentation fix. (closes issue #13708)
* configs/agents.conf.sample: An update to the
documentation/example of agents.conf.sample with the correct
parameter for this feature as defined in chan_agent.c (closes
issue #13709)
2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: fix some problems when parsing SIP messages
that have the maximum number of headers or body lines that we
support
2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Change this warning to an error message.
Suggestion comes from Sean Bright. Thanks Sean!
* channels/chan_sip.c: Call register_peer_exten even in the case
that the peer's IP/port does not change. (closes issue #13309)
Reported by: dimas Patches: v2-13309.patch uploaded by dimas
(license 88)
* include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
period for sync-triggered audiohooks so that if packetization of
audio is close (but not equal) we don't end up flushing the
audiohooks over small inconsistencies in synchronization. Related
to issue #13005, and solves the issue for most people who were
experiencing the problem. However, a small number of people are
still experiencing the problem on long calls, so I am not closing
the issue yet
* apps/app_queue.c: Update the queue with the correct number of
calls and whether the call was completed within the service level
when a transfer takes place. This way, we do not "break" the
leastrecent and fewestcalls strategies by not logging a call
until after the transferred call has ended. (closes issue #13395)
Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
by Marquis (license 32)
* channels/chan_sip.c: Don't allow reserved characters to be used
in register lines in sip.conf. (closes issue #13570) Reported by:
putnopvut
2008-10-14 20:09 +0000 [r149061] Tilghman Lesher <tlesher@digium.com>
* apps/app_waitforsilence.c: Check correct values in the return of
ast_waitfor(); also, get rid of a possible memory leak. (closes
issue #13658) Reported by: explidous Patch by: me
2008-10-14 19:05 +0000 [r148990] Leif Madsen <lmadsen@digium.com>
* CHANGES: Add in some missing updates to the CHANGES file for
sip.conf (closes issue #13100) Reported and patch by:
gknispel_proformatique
2008-10-14 19:03 +0000 [r148916-148987] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Some compilers warn, some don't. Fixing.
* apps/app_voicemail.c: Ensure that mail headers are 7-bit clean,
even when UTF-8 characters are used in headers like 'Subject' and
'To'. Closes AST-107.
2008-10-14 17:33 +0000 [r148912] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: Deadlock prevention in chan_local. (closes
issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
by putnopvut (license 60) Tested by: tacvbo
2008-10-14 10:30 +0000 [r148611-148736] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: on Ubuntu (at least), recent versions of ld in binutils
delete all debugging symbols when -x is supplied; since the
reasons why -x is being passed are lost in the mists of time,
remove it so debugging will work properly
* main/translate.c: it would be nice if this message printing code
had actually been tested before it was committed...
2008-10-10 16:25 +0000 [r147997-148257] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: User not notified of temporary greeting, if
ODBC storage is in use. (closes issue #13659) Reported by:
moliveras Patches: 20081009__bug13659.diff.txt uploaded by
Corydon76 (license 14) Tested by: moliveras
* apps/app_voicemail.c: When blank, callerid name and number should
display "unknown caller" in voicemail emails. (Closes issue
#13643)
2008-10-09 18:56 +0000 [r147941] Jeff Peeler <jpeeler@digium.com>
* res/res_features.c: (closes issue #13139) Reported by: krisk84
Tested by: krisk84 This change prevents a call that is placed in
the parkinglot to be picked up before the PBX is finished. If
another extension dials the parking extension before the PBX
thread has completed at minimum warnings will occur about the PBX
not properly being terminated. At worst, a crash could occur.
2008-10-08 22:22 +0000 [r147681] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: when parsing a text configuration option,
ensure that the buffer on the stack is actually large enough to
hold the legal values of that option, and also ensure that
sscanf() knows to stop parsing if it would overrun the buffer
(without these changes, specifying "buffers=...,immediate" would
overflow the buffer on the stack, and could not have worked as
expected)
2008-10-08 14:51 +0000 [r147517] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: If we receive DTMF make sure that the
state of the speech structure goes back to being not ready.
(issue #LUMENVOX-8)
2008-10-07 23:14 +0000 [r147429-147430] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: revert this change until i can understand
why it results in locking order changes
* channels/chan_dahdi.c: don't start a PBX on incoming PRI call
channels until after we're done setting channel variables and
other things on the channel, otherwise the channel might go away
(if the dialplan hangs up quickly) before we are done, which
results in a spectacular crash
2008-10-07 16:48 +0000 [r147193] Sean Bright <sean.bright@gmail.com>
* apps/app_voicemail.c: Make 'imapsecret' an alias to
'imappassword' in voicemail.conf.
2008-10-06 20:52 +0000 [r146711-146799] Tilghman Lesher <tlesher@digium.com>
* funcs/func_callerid.c, apps/app_speech_utils.c,
funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c,
channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c:
Dialplan functions should not actually return 0, unless they have
modified the workspace. To signal an error (and no change to the
workspace), -1 should be returned instead. (closes issue #13340)
Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt
uploaded by Corydon76 (license 14)
* channels/chan_local.c: Check whether an extension exists in the
_call method, rather than the _alloc method, because we need to
evaluate the callerid (since that data affects whether an
extension exists). (closes issue #13343) Reported by: efutch
Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
(license 14) Tested by: efutch
2008-10-06 15:57 +0000 [r146643] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: ensure that the private structure for
pseudo channels is created without 'leaking' configuration data
from other configured channels (closes issue #13555) Reported by:
jeffg Patches: issue_13555.patch uploaded by kpfleming (license
421) Tested by: jeffg
2008-10-05 21:17 +0000 [r146448] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Fix silly formatting.
2008-10-03 22:51 +0000 [r146244] Sean Bright <sean.bright@gmail.com>
* apps/app_rpt.c: Change some preprocessor macros to struct
definitions so that we get app_rpt to build with DAHDI. (closes
issue #13576) Reported by: blitzrage
2008-10-03 20:44 +0000 [r146129] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/features.h, res/res_features.c, res/res_agi.c:
(closes issue #13425) Reported by: mdu113 Tested by: mdu113
Similar to r143204, masquerade the channel in the case of Park
being called from AGI.
2008-10-03 17:12 +0000 [r146026] Steve Murphy <murf@digium.com>
* res/res_features.c: (closes issue #13579) Reported by: dwagner
(closes issue #13584) Reported by: dwagner Tested by: murf,
putnopvut The thought occurred to me that the res= from the
extension spawn was ending up being returned from the bridge.
"Thou shalt not poison the return value". Made the change and it
appears to allow blind xfers to work as normal. If I'm wrong,
reopen the bugs. But it looks good to me! Many thanks to
putnopvut for helping me reproduce this!
2008-10-02 16:39 +0000 [r145751-145839] Tilghman Lesher <tlesher@digium.com>
* funcs/func_odbc.c: Backport support for some of the keyword
modifications used in 1.6 (while warning that some options aren't
really supported) and add some warning messages. Some credit to
oej, who was complaining in #asterisk-dev.
* res/res_odbc.c: Some sanity checks that may have led to prior
crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup
of incorrectly-used constants.
2008-10-01 17:18 +0000 [r145479] Leif Madsen <lmadsen@digium.com>
* contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql
script to create the setinterfacevar column. (closes issue
#13549) Reported by: fiddur
2008-10-01 Russell Bryant <russell@digium.com>
* Asterisk 1.4.22 released.
2008-09-09 Russell Bryant <russell@digium.com>
* Asterisk 1.4.22-rc5 released.
2008-09-09 15:40 +0000 [r142063] Russell Bryant <russell@digium.com>
* res/res_features.c: Ensure that the stored CDR reference is still
valid after the bridge before poking at it. Also, keep the
channel locked while messing with this CDR. (fixes crashes
reported in issue #13409)
2008-09-08 21:10 +0000 [r141809] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix pedantic mode of chan_sip to only check
the remote tag of an endpoint once a dialog has been confirmed.
Up until that point, it is possible and legal for the far-end to
send provisional responses with a different To: tag each time.
With this patch applied, these provisional messages will not
cause a matching problem. (closes issue #11536) Reported by: ibc
Patches: 11536v2.patch uploaded by putnopvut (license 60)
2008-09-08 21:02 +0000 [r141806] Russell Bryant <russell@digium.com>
* main/pbx.c: When doing an async goto, detect if the channel is
already in the middle of a masquerade. This can happen when
chan_local is trying to optimize itself out. If this happens,
fail the async goto instead of bursting into flames. (closes
issue #13435) Reported by: geoff2010
2008-09-08 Russell Bryant <russell@digium.com>
* Asterisk 1.4.22-rc4 released.
2008-09-08 20:15 +0000 [r141741] Jason Parker <jparker@digium.com>
* Makefile, redhat (removed): Remove RPM package targets from
Makefile (and all associated parts). This has never worked in
1.4, and we decided that it makes no sense to be done here. There
are many distros out there that already have "proper" spec files
that can be (re)used. Closes issue #13113 Closes issue #10950
Closes issue #10952
2008-09-08 16:26 +0000 [r141678] Russell Bryant <russell@digium.com>
* configure, configure.ac: Actually use Zaptel CFLAGS if using
Zaptel instead of DAHDI This fixes building against Zaptel when
using a custom path
2008-09-06 20:13 +0000 [r141565] Steve Murphy <murf@digium.com>
* channels/chan_sip.c: This fix comes from Joshua Colp The
Brilliant, who, given the trace, came up with a solution. This
will most likely will close 13235 and 13409. I'll wait till
Monday to verify, and then close these bugs.
2008-09-06 15:23 +0000 [r141503] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c: Reverting behavior change (AGI should not exit
non-zero on SUCCESS) (closes issue #13434) Reported by:
francesco_r
2008-09-05 21:10 +0000 [r141217-141366] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Agent's should not try to call a channel's
indicate callback if the channel has been hung up. It will likely
crash otherwise ABE-1159
* apps/app_voicemail.c: Since greetings are not stored in IMAP, we
should not be DISPOSE'ing of them the same way we do with other
messages. (closes issue #13414) Reported by: mthomasslo Patches:
13414v2.patch uploaded by putnopvut (license 60) Tested by:
mthomasslo
* channels/chan_sip.c: Commit 140417 had a logic flaw in it which
caused port 5060 to always be used when dialing a peer if no
explicit port was specified. This broke the behavior of
implicitly using the port from which the peer registered if no
port is specified. This commit fixes the logic flaw. (closes
issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded
by putnopvut (license 60) Tested by: mdu113
2008-09-05 14:15 +0000 [r141094-141156] Steve Murphy <murf@digium.com>
* main/channel.c: A small change to prevent double-posting of
CDR's; thanks to Daniel Ferrer for bringing it to our attention
* pbx/ael/ael-test/ref.ael-vtest25 (added),
pbx/ael/ael-test/ael-vtest25/extensions.ael (added),
pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c,
pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue
#13357) Reported by: pj Tested by: murf (closes issue #13416)
Reported by: yarns Tested by: murf If you find this message
overly verbose, relax, it's probably not meant for you. This
message is meant for probably only two people in the whole world:
me, or the poor schnook that has to maintain this code because
I'm either dead or unavailable at the moment. This fix solves two
reports, both having to do with embedding a function call in a
${} construct. It was tricky because the funccall syntax has
parenthesis () in it. And up till now, the 'word' token in the
flex stuff didn't allow that, because it would tend to steal the
LP and RP tokens. To be truthful, the "word" token was the
trickiest, most unstable thing in the whole lexer. I was lucky it
made this long without complaints. I had to choose every
character in the pattern with extreme care, and I knew that
someday I'd have to revisit it. Well, the day has come. So, my
brilliant idea (and I'm being modest), was to use the surrounding
${} construct to make a state machine and capture everything in
it, no matter what it contains. But, I have to now treat the word
token like I did with comments, in that I turn the whole thing
into a state-machine sort of spec, with new contexts
"curlystate", "wordstate", and "brackstate". Wait a minute,
"brackstate"? Yes, well, it didn't take very many regression
tests to point out if I do this for ${} constructs, I also have
to do it with the $[] constructs, too. I had to create a separate
pcbstack2 and pcbstack3 because these constructs can occur inside
macro argument lists, and when we have two state machines
operating on the same structures we'd get problems otherwise. I
guess I could have stopped at pcbstack2 and had the brackstate
stuff share it, but it doesn't hurt to be safe. So, the pcbpush
and pcbpop routines also now have versions for "2" and "3". I had
to add the {KEYWORD} construct to the initial pattern for "word",
because previously word would match stuff like "default7",
because it was a longer match than the keyword "default". But,
not any more, because the word pattern only matches only one or
two characters now, and it will always lose. So, I made it the
winner again by making an optional match on any of the keywords
before it's normal pattern. I added another regression test to
make sure we don't lose this in future edits, and had to fix just
one regression, where it no longer reports a 'cascaded' error,
which I guess is a plus. I've given some thought as to whether to
apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
decided to put it in 1.4 because one of the bug reports was
against 1.4; and it is unexpected that AEL cannot handle this
situation. It actually reduced the amount of useless "cascade"
error messages that appeared in the regressions (by one line,
ehhem). There is a possible side-effect in that it does now do
more careful checking of what's in those ${} constructs, as far
as matching parens, and brackets are concerned. Some users may
find a an insidious problem and correct it this way. This should
be exceedingly rare, I hope.
2008-09-04 17:00 +0000 [r141028] Jeff Peeler <jpeeler@digium.com>
* res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes
multiple parking problems: Crash when executing a park on an
extension dialed by AGI due to not returning the proper return
code. Crash when using a builtin feature that was a subset of a
enabled dynamic feature. Crash due to always hanging up the peer
despite the fact that the peer was supposed to be parked.
2008-09-03 Russell Bryant <russell@digium.com>
* Asterisk 1.4.22-rc3 released.
2008-09-03 14:29 +0000 [r140850] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix voicemail forwarding when using ODBC
storage. (closes issue #13387) Reported by: moliveras Patches:
13387.patch uploaded by putnopvut (license 60) Tested by:
putnopvut, moliveras
2008-09-03 13:24 +0000 [r140816] Russell Bryant <russell@digium.com>
* main/poll.c: Don't freak out if the poll emulation receives NULL
for the pollfds array (closes issue #13307) Reported by: jcovert
2008-09-02 23:47 +0000 [r140751] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: After adding the context checking to
app_voicemail for IMAP storage, I left out a crucial place to
copy the context to the vm_state structure. This is the
correction.
2008-09-02 23:36 +0000 [r140670-140747] Steve Murphy <murf@digium.com>
* main/cdr.c: I am turning the warnings generated in ast_cdr_free
and post_cdr into verbose level 2 messages. Really, they matter
little to end users. You either get the CDR's you wanted, or you
don't, and it is a bug.
* main/channel.c: After reconsidering, with respect to 13409,
ast_cdr_detach should be OK, better in fact, than ast_cdr_free,
which generates lots of useless warnings that will undoubtably
generate complaints.
* main/channel.c, main/pbx.c: (closes issue #13409) Reported by:
tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by
tomaso (license 564) I basically spent the day, verifying that
this patch solves the problem, and doesn't hurt in non-problem
cases. Why valgrind did not plainly reveal this leak absolutely
mystifies and stuns me. Many, many thanks to tomaso for finding
and providing the fix.
2008-09-02 18:14 +0000 [r140605] Sean Bright <sean.bright@gmail.com>
* channels/chan_iax2.c: Make sure to use the correct length of the
mohinterpret and mohsuggest buffers when copying configuration
values. (closes issue #13336) Reported by:
decryptus_proformatique Patches:
chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
by decryptus (license 555)
2008-08-29 17:34 +0000 [r140417-140488] Mark Michelson <mmichelson@digium.com>
* main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After
working on the ao2_containers branch, I noticed something a bit
strange. In all cases where we provide a callback function to
ao2_container_alloc, the callback function would only return 0 or
CMP_MATCH. After inspecting the ao2_callback() code carefully, I
found that if you're only looking for one specific item, then you
should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will
continue traversing the current bucket until the end searching
for more matches. In cases like chan_iax2 where in 1.4, all the
peers are shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be traversed
even if the peer is one of the first ones come across in the
bucket. All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc so
that calls to ao2_find could find a unique instance of whatever
object was being stored in the container.
* apps/app_voicemail.c: Add context checking when retrieving a
vm_state. This was causing a problem for people who had
identically named mailboxes in separate voicemail contexts. This
commit affects IMAP storage only. (closes issue #13194) Reported
by: moliveras Patches: 13194.patch uploaded by putnopvut (license
60) Tested by: putnopvut, moliveras
* channels/chan_sip.c: Fix SIP's parsing so that if a port is
specified in a string to Dial(), it is not ignored. (closes issue
#13355) Reported by: acunningham Patches: 13355v2.patch uploaded
by putnopvut (license 60) Tested by: acunningham
2008-08-27 19:49 +0000 [r140299] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked
when in pedantic mode. The problem was that the wrong tags would
be compared depending on the direction of the call. (closes issue
#13353) Reported by: flefoll Patches:
chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
(license 244)
2008-08-26 16:49 +0000 [r140115] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: add HAVE_PRI if define around
dahdi_close_pri_fd
2008-08-26 16:07 +0000 [r140060] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip.
This code used the return value of a completely unrelated
function to determine whether the scheduler should be run or not.
This would have caused the scheduler to not run in cases where it
should have. Also, leave a note about another scheduler issue
that needs to be addressed at some point.
2008-08-26 15:57 +0000 [r140056] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir
Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
by: tzafrir, jpeeler This patch fixes closing open file
descriptors in the case of an error.
2008-08-26 15:27 +0000 [r140051] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a race condition with the IAX scheduler
thread. A lock and condition are used here to allow newly
scheduled tasks to wake up the scheduler just in case the new
task needs to run sooner than the current wakeup time when the
thread is sleeping. However, there was a race condition such that
a newly scheduled task would not properly wake up the scheduler
or affect the wake up period. The order of execution would have
been: 1) Scheduler thread determines wake up time of N ms. 2)
Another thread schedules a task and signals the condition, with
an execution time of < N ms. 3) Scheduler thread locks and goes
to sleep for N ms. By moving the sleep time determination to
inside the critical section, this possibility is avoided.
2008-08-26 15:22 +0000 [r140050] Terry Wilson <twilson@digium.com>
* Makefile: sounds/Makefile installs sounds using the "new"
language directory structure, but languageprefix needs to be set
= yes for sounds in subdirectories (digits/1, etc.) to play as
the correct language. Fix the generation of asterisk.conf to
include languageprefix=yes
2008-08-26 14:09 +0000 [r140029] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: correct a file location in an error
message
2008-08-25 21:47 +0000 [r139927] Jeff Peeler <jpeeler@digium.com>
* main/manager.c: Fix a typo I made. Lesson learned, apply the
patch if one exists.
2008-08-25 21:31 +0000 [r139909] Sean Bright <sean.bright@gmail.com>
* build_tools/get_moduleinfo, build_tools/get_makeopts: Some
versions of awk (nawk, for example) don't like empty regular
expressions so be slightly more verbose. (closes issue #13374)
Reported by: dougm Patches: 13374.diff uploaded by seanbright
(license 71) Tested by: dougm
2008-08-25 20:46 +0000 [r139869] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely
2008-08-25 15:52 +0000 [r139769] Mark Michelson <mmichelson@digium.com>
* main/config.c: Fix the logic in config_text_file_save so that if
an UpdateConfig manager action is issued and the file specified
in DstFileName does not yet exist, an error is not returned.
(closes issue #13341) Reported by: vadim Patches: 13341.patch
uploaded by putnopvut (license 60) (with small modification from
seanbright)
2008-08-25 15:33 +0000 [r139764] Steve Murphy <murf@digium.com>
* main/pbx.c, res/res_features.c: This patch reverts the changes
made via 139347, and 139635, as users are seeing adverse
difference. I will un-close 13251. Back to the drawing board/
concept/ beginning/ whatever!
2008-08-22 22:24 +0000 [r139635] Steve Murphy <murf@digium.com>
* res/res_features.c: I found some problems with the code I
committed earlier, when I merged them into trunk, so I'm coming
back to clean up. And, in the process, I found an error in the
code I added to trunk and 1.6.x, that I'll fix using this patch
also.
2008-08-22 21:36 +0000 [r139621] Jeff Peeler <jpeeler@digium.com>
* main/manager.c: (closes issue #13359) Reported by: Laureano
Patches: originate_channel_check.patch uploaded by Laureano
(license 265)
2008-08-22 19:45 +0000 [r139456-139553] Mark Michelson <mmichelson@digium.com>
* include/asterisk/threadstorage.h: Fix compilation when
DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported
by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy
(license 35)
* main/frame.c: Remove show_frame_stats_deprecated since it is not
used anywhere and causes build errors if building under dev-mode
with TRACE_FRAMES selected in menuselect. (closes issue #13362)
Reported by: snuffy
* channels/chan_iax2.c: Fix the build. Thanks, mvanbaak!
* channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting
from incorrect locking order between iax2_pvt and ast_channel
structures. AST-13
2008-08-21 23:39 +0000 [r139387] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Fixes loop that could possibly never exit
in the event of a channel never being able to be opened or
specify after a restart. (closes issue #11017)
2008-08-21 23:03 +0000 [r139347] Steve Murphy <murf@digium.com>
* main/pbx.c, res/res_features.c: (closes issue #13251) Reported
by: sergee Tested by: murf THis is a bold move for a static
release fix, but I wouldn't have made it if I didn't feel
confident (at least a *bit* confident) that it wouldn't mess
everyone up. The reasoning goes something like this: 1. We simply
cannot do anything with CDR's at the current point (in pbx.c,
after the __ast_pbx_run loop). It's way too late to have any
affect on the CDRs. The CDR is already posted and gone, and the
remnants have been cleared. 2. I was very much afraid that moving
the running of the 'h' extension down into the bridge code (where
it would be now practical to do it), would result in a lot more
calls to the 'h' exten, so I implemented it as another exten
under another name, but found, to my pleasant surprise, that
there was a 1:1 correspondence to the running of the 'h' exten in
the pbx_run loop, and the new spot at the end of the bridge. So,
I ifdef'd out the current 'h' loop, and moved it into the bridge
code. The only difference I can see is the stuff about the
AST_PBX_KEEPALIVE, and hopefully, if this is still an important
decision point, I can replicate it if there are complaints. To be
perfectly honest, the KEEPALIVE situation is not totally clear to
me, and how it relates to a post-bridge situation is less clear.
I suspect the users will point out everything in total clarity if
this steps on anyone's toes! 3. I temporarily swap the bridge_cdr
into the channel before running the 'h' exten, which makes it
possible for users to edit the cdr before it goes out the door.
And, of course, with the endbeforehexten config var set, the
users can also get at the billsec/duration vals. After the h
exten finishes, the cdr is swapped back and processing continues
as normal. Please, all who deal with CDR's, please test this
version of Asterisk, and file bug reports as appropriate!
2008-08-21 10:11 +0000 [r139283] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4,
too.
2008-08-20 22:14 +0000 [r139213] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c: Fix a crash in the ChanSpy application. The
issue here is that if you call ChanSpy and specify a spy group,
and sit in the application long enough looping through the
channel list, you will eventually run out of stack space and the
application with exit with a seg fault. The backtrace was always
inside of a harmless snprintf() call, so it was tricky to track
down. However, it turned out that the call to snprintf() was just
the biggest stack consumer in this code path, so it would always
be the first one to hit the boundary. (closes issue #13338)
Reported by: ruddy
2008-08-20 19:52 +0000 [r139151] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Fix bug where the samples were not accurate
when in G723 mode, which would cause the timestamp field of the
RTP header to be invalid.
2008-08-20 19:35 +0000 [r139145] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Backport support
for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not
doing so just makes it difficult for people with channels that
are in alarm when Asterisk starts up to get them going once the
alarm is cleared (closes issue #12160) Reported by: tzafrir
Patches: asterisk-chanalarms_14.patch uploaded by tzafrir
(license 46) Tested by: tzafrir
2008-08-20 17:14 +0000 [r139074] Steve Murphy <murf@digium.com>
* main/cdr.c: (closes issue #13263) Reported by: brainy Tested by:
murf The specialized reset routine is tromping on the flags field
of the CDR. I made a change to not reset the DISABLED bit. This
should get rid of this problem.
2008-08-20 15:37 +0000 [r139015] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: sip_read should properly handle a NULL
return from sip_rtp_read. (closes issue #13257) Reported by:
travishein
2008-08-19 23:22 +0000 [r138949] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL
compatability define for Zaptel
2008-08-19 23:17 +0000 [r138942] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Reset agent_pvt variables back to the
values in agents.conf (from what the corresponding channel
variables were set to) when the agent logs out. (closes issue
#13098) Reported by: davidw Patches:
20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
bbryant (license 36) Tested by: davidw
2008-08-19 22:56 +0000 [r138938] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Add configuration option to
chan_dahdi.conf to allow buffering policy and number of buffers
to be configured per channel. Syntax: buffers=<num of
buffers>,<policy> Where the number of buffers is some
non-negative integer and the policy is either "full", "half", or
"immediate".
2008-08-19 18:50 +0000 [r138685-138886] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Add a lock and unlock prior to the
destruction of the chanspy_ds lock to ensure that no other
threads still have it locked. While this should not happen under
normal circumstances, it appears that if the spyer and spyee hang
up at nearly the same time, the following may occur. 1.
ast_channel_free is called on the spyee's channel. 2. The chanspy
datastore is removed from the spyee's channel in
ast_channel_free. 3. In the spyer's thread, the spyer attempts to
remove and destroy the datastore from the spyee channel, but the
datastore has already been removed in step 2, so the spyer
continues in the code. 4. The spyee's thread continues and calls
the datastore's destroy callback, chanspy_ds_destroy. This
involves locking the chanspy_ds. 5. Now the spyer attempts to
destroy the chanspy_ds lock. The problem is that in step 4, the
spyee has locked this lock, meaning that the spyer is attempting
to destroy a lock which is currently locked by another thread.
The backtrace provided in issue #12969 supports the idea that
this is possible (and has even occurred). This commit does not
close the issue, but should help in preventing one type of crash
associated with the use of app_chanspy.
* apps/app_queue.c: Change the inequalities used in app_queue with
regards to timeouts from being strict to non-strict for more
accuracy. (closes issue #13239) Reported by: atis Patches:
app_queue_timeouts_v2.patch uploaded by atis (license 242)
2008-08-18 16:57 +0000 [r138663] Kevin P. Fleming <kpfleming@digium.com>
* codecs/codec_dahdi.c: look for transcoder in proper place based
on build against Zaptel or DAHDI
2008-08-18 11:57 +0000 [r138569] Sean Bright <sean.bright@gmail.com>
* channels/chan_dahdi.c: You know what's awesome? Code that
compiles... ;)
2008-08-18 02:05 +0000 [r138516] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: fix compilation warnings
2008-08-16 01:12 +0000 [r138309-138360] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: fixes use count to properly decrement if
an active dahdi channel is destroyed allowing module to be
unloaded
* channels/chan_dahdi.c: add forgotten locks around ss_thread_count
in ss_thread for dahdi restart
2008-08-15 22:33 +0000 [r138258] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample: More fixes for
realtime peers. (closes issue #12921) Reported by: Nuitari
Patches: 20080804__bug12921.diff.txt uploaded by Corydon76
(license 14) 20080815__bug12921.diff.txt uploaded by Corydon76
(license 14) Tested by: Corydon76
2008-08-15 21:28 +0000 [r138119-138238] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: initialize condition variable
ss_thread_complete using ast_cond_init
* channels/chan_dahdi.c: declared static mutexes using
AST_MUTEX_DEFINE_STATIC macro
* channels/chan_dahdi.c: Fixes the dahdi restart functionality.
Dahdi restart allows one to restart all DAHDI channels, even if
they are currently in use. This is different from unloading and
then loading the module since unloading requires the use count to
be zero. Reloading the module is different in that the signalling
is not changed from what it was originally configured. Also, this
fixes not closing all the file descriptors for D-channels upon
module unload (which would prevent loading the module
afterwards). (closes issue #11017)
2008-08-15 15:07 +0000 [r138027] Russell Bryant <russell@digium.com>
* main/autoservice.c: Ensure that when a hangup occurs in
autoservice, that a hangup frame gets properly deferred to be
read from the channel owner when it gets taken out of
autoservice. (closes issue #12874) Reported by: dimas Patches:
v1-12874.patch uploaded by dimas (license 88)
2008-08-15 14:51 +0000 [r137847-138023] Tilghman Lesher <tlesher@digium.com>
* funcs/func_strings.c: Additional check for more string specifiers
than arguments. (closes issue #13299) Reported by: adomjan
Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
(license 14) func_strings.c-sprintf.patch uploaded by adomjan
(license 487) Tested by: adomjan
* channels/chan_dahdi.c: Oops, wrong direction
* channels/chan_dahdi.c: When creating the secondary subchannel
name, it is necessary to compare to the existing channel name
without the "Zap/" or "DAHDI/" prefix, since our test string is
also without that prefix. (closes issue #13027) Reported by:
dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
(license 525) (Slightly modified by me, to compensate for both
names)
2008-08-14 14:05 +0000 [r137731] Russell Bryant <russell@digium.com>
* configs/sip.conf.sample: Comments in this config file were
aligned only if your tab size was set to 8. So, convert tabs to
spaces so that things should be aligned regardless of what tab
size you use in your editor.
2008-08-14 02:03 +0000 [r137677-137679] Kevin P. Fleming <kpfleming@digium.com>
* Zaptel-to-DAHDI.txt: forgot one module name that changed
* include/asterisk/dahdi_compat.h, channels/chan_dahdi.c,
build_tools/menuselect-deps.in, configure, configure.ac,
codecs/codec_dahdi.c: add support for Zaptel versions that
contain the new transcoder interface
2008-08-13 21:35 +0000 [r137580] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Register DAHDISendKeypadFacility
application if dahdi_chan_mode is set to DAHDI + Zap. Mark
ZapSendKeypadFacility application as deprecated on usage.
2008-08-13 20:46 +0000 [r137527-137530] Kevin P. Fleming <kpfleming@digium.com>
* Zaptel-to-DAHDI.txt (added): add document describing what users
will need to be aware of when upgrading to this version and using
DAHDI
* apps/app_meetme.c: remove some more chan_zap references
* doc/asterisk-conf.txt, channels/chan_dahdi.c: document
dahdichanname option in doc/asterisk-conf.txt make chan_dahdi
read its configuration from zapata.conf if dahdichanname has been
set to 'no'
2008-08-13 14:33 +0000 [r137348-137405] Sean Bright <sean.bright@gmail.com>
* doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds
* cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends
and have it try to store CDR(userfield) if it is set. The new
behavior is to check for the userfield column on module load, and
if it exists, we will store CDR(userfield) when CDRs are written.
A similar patch already went into trunk and 1.6.0. (closes issue
#13290) Reported by: falves11
2008-08-11 13:33 +0000 [r137188] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_meetme.c: convert this module to be able to handle DAHDI
or Zaptel (reported on asterisk-users, don't know how this got
missed before)
2008-08-11 00:20 +0000 [r137138] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Deallocate database connection handle on
disconnect, as we allocate another one on connect. (closes issue
#13271) Reported by: dveiga
2008-08-09 17:11 +0000 [r136999] Russell Bryant <russell@digium.com>
* configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate
to 0 if not found instead of empty. pointed out by tzafrir on
#asterisk-dev
2008-08-09 15:25 +0000 [r136946] Tilghman Lesher <tlesher@digium.com>
* /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
revisions 136945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
| 2 lines Regression fixes for Solaris ........
2008-08-08 00:15 +0000 [r136726] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
pbx/ael/ael-test/ref.ael-vtest13,
pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c,
include/asterisk/ael_structs.h: (closes issue #13236) Reported
by: korihor Wow, this one was a challenge! I regrouped and ran a
new strategy for setting the ~~MACRO~~ value; I set it once per
extension, up near the top. It is only set if there is a switch
in the extension. So, I had to put in a chunk of code to detect a
switch in the pval tree. I moved the code to insert the set of
~~exten~~ up to the beginning of the gen_prios routine, instead
of down in the switch code. I learned that I have to push the
detection of the switches down into the code, so everywhere I
create a new exten in gen_prios, I make sure to pass onto it the
values of the mother_exten first, and the exten next. I had to
add a couple fields to the exten struct to accomplish this, in
the ael_structs.h file. The checked field makes it so we don't
repeat the switch search if it's been done. I also updated the
regressions.
2008-08-07 18:25 +0000 [r136560] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/menuselect-deps.in, configure, configure.ac: change
the required dependency for codec_dahdi to only be satisfied by
DAHDI and not Zaptel, as the new transcoder interface is only in
DAHDI
2008-08-07 18:14 +0000 [r136544] Shaun Ruffell <sruffell@digium.com>
* codecs/codec_dahdi.c: Updated codec_dahdi to use the new
transcoder interface in the first DAHDI release. Codec dahdi no
longer functions with the transcoder interface in zaptel at this
time (which the last zaptel release was 1.4.11). NOTE: Still
needs an update to the configure script to make sure that
codec_dahdi is only built if the new transcoder interface is
present in the drivers. (Issue: DAHDI-42)
2008-08-07 16:50 +0000 [r136488] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: Update persistent state on all exit conditions.
(closes issue #12916) Reported by: sgenyuk Patches:
app_queue.patch.txt uploaded by neutrino88 (license 297) Tested
by: sgenyuk, aragon
2008-08-07 16:30 +0000 [r136404-136484] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/doxyref.h: add a raw list of all libraries that
any part of Asterisk links directly to
* apps/app_voicemail.c: work around a bug in gcc-4.2.3 that
incorrectly ignores the casting away of 'const' for pointers when
the developer knows it is safe to do so
* Makefile: remove config.cache during distclean, in case the user
is using autoconf caching
2008-08-07 01:31 +0000 [r136304-136348] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Also, parse
useincomingcalleridonzaptransfer (and add appropriate deprecation
warnings).
* channels/chan_dahdi.c: For backwards compatibility with previous
1.4 versions which used "zapchan" in users.conf, ensure that we
still support it.
2008-08-06 21:18 +0000 [r136241] Richard Mudgett <rmudgett@digium.com>
* channels/misdn_config.c, channels/chan_misdn.c,
configs/misdn.conf.sample: * The allowed_bearers setting in
misdn.conf misspelled one of its options: digital_restricted. *
Fixed some other spelling errors and typos.
2008-08-06 20:42 +0000 [r136238] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: We only need to unregister the QueueStatus
manager command once on an unload
2008-08-06 20:14 +0000 [r136190] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.redhat.asterisk: -C option takes a filename,
not a directory path. (closes issue #13007) Reported by:
klaus3000
2008-08-06 18:58 +0000 [r136168] Russell Bryant <russell@digium.com>
* Makefile: Remove the use of --no-print-directory when compiling
subdirectories. This allows vim :make functionality to work
properly when errors have occurred in the build. Without printing
the directories, vim did not know how to find the file that the
error occurred in. If the extra bit of build noise annoys anyone,
just let me know, and I'll make this optional.
2008-08-06 15:58 +0000 [r136062] Mark Michelson <mmichelson@digium.com>
* main/rtp.c, channels/chan_skinny.c: Since adding the
AST_CONTROL_SRCUPDATE frame type, there are places where
ast_rtp_new_source may be called where the tech_pvt of a channel
may not yet have an rtp structure allocated. This caused a crash
in chan_skinny, which was fixed earlier, but now the same crash
has been reported against chan_h323 as well. It seems that the
best solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL. This
change to ast_rtp_new_source also allows the removal of what is
now a redundant pointer check from chan_skinny. (closes issue
#13247) Reported by: pj
2008-08-06 03:53 +0000 [r135899-135949] Tilghman Lesher <tlesher@digium.com>
* main/channel.c: Fix a longstanding bug in channel walking logic,
and fix the explanation to make sense. (Closes issue #13124)
* main/translate.c: Since powerof() can return an error condition,
it's foolhardy not to detect and deal with that condition.
(Related to issue #13240)
* include/asterisk/threadstorage.h, include/asterisk/utils.h: 1)
Bugfix for debugging code 2) Reduce compiler warnings for another
section of debugging code (Closes issue #13237)
2008-08-06 00:29 +0000 [r135841-135850] Mark Michelson <mmichelson@digium.com>
* /: Remove properties that should not be here
* apps/app_skel.c: Revert inadvertent changes to app_skel that
occurred when I was testing for a memory leak
* include/asterisk/abstract_jb.h, main/channel.c, /,
apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h:
Merging the issue11259 branch. The purpose of this branch was to
take into account "burps" which could cause jitterbuffers to
misbehave. One such example is if the L option to Dial() were
used to inject audio into a bridged conversation at regular
intervals. Since the audio here was not passed through the
jitterbuffer, it would cause a gap in the jitterbuffer's
timestamps which would cause a frames to be dropped for a brief
period. Now ast_generic_bridge will empty and reset the
jitterbuffer each time it is called. This causes injected audio
to be handled properly. ast_generic_bridge also will empty and
reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
frame since the change in audio source could negatively affect
the jitterbuffer. All of this was made possible by adding a new
public API call to the abstract_jb called ast_jb_empty_and_reset.
(closes issue #11259) Reported by: plack Tested by: putnopvut
2008-08-05 23:13 +0000 [r135799] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c,
include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit
Tested by: murf I discovered that also, in the previous bug fixes
and changes, the cdr.conf 'unanswered' option is not being
obeyed, so I fixed this. And, yes, there are two 'answer' times
involved in this scenario, and I would agree with you, that the
first answer time is the time that should appear in the CDR. (the
second 'answer' time is the time that the bridge was begun). I
made the necessary adjustments, recording the first answer time
into the peer cdr, and then using that to override the bridge
cdr's value. To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as DIALED (with a
new flag), and outputting them if they bear that flag, and you
are in the right mode. I also corrected one small mention of the
Zap device to equally consider the dahdi device. I heavily tested
10-sec-wait macros in dial, and without the macro call; I tested
hangups while the macro was running vs. letting the macro
complete and the bridge form. Looks OK. Removed all the
instrumentation and debug.
2008-08-05 21:34 +0000 [r135747] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the
meaning of the flag IAX_HASCALLERID was perverted. This change
reverts IAX2 to the original meaning, which was, that the
callerid set on the client should be overridden on the server,
even if that means the resulting callerid is blank. In other
words, if you set "callerid=" in the IAX config, then the
callerid should be overridden to blank, even if set on the
client. Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field
existing, but set to blank (override callerid to blank).
2008-08-05 13:25 +0000 [r135597] Sean Bright <sean.bright@gmail.com>
* main/cli.c: Use PATH_MAX for filenames
2008-08-04 20:15 +0000 [r135536] Russell Bryant <russell@digium.com>
* configs/chan_dahdi.conf.sample: fix a config sample typo
2008-08-04 17:07 +0000 [r135479-135482] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script
* apps/app_voicemail.c: Memory leak on unload (closes issue #13231)
Reported by: eliel Patches: app_voicemail.leak.patch uploaded by
eliel (license 64)
2008-08-04 16:26 +0000 [r135473] Russell Bryant <russell@digium.com>
* configs/chan_dahdi.conf.sample: Add a minor clarification to the
documentation of mohinterpret and mohsuggest
2008-08-01 11:43 +0000 [r135055-135058] Michiel van Baak <michiel@vanbaak.info>
* apps/app_ices.c: make app_ices compile on OpenBSD.
* channels/chan_skinny.c: fix some potential deadlocks in
chan_skinny (closes issue #13215) Reported by: qwell Patches:
2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak
2008-07-31 22:18 +0000 [r134983] Kevin P. Fleming <kpfleming@digium.com>
* main/http.c: accomodate users who seem to lack a sense of humor
:-)
2008-07-31 21:53 +0000 [r134976] Tilghman Lesher <tlesher@digium.com>
* sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in
callfiles and manager, to allow video calls to be set up from
callfiles and AMI. (closes issue #9531) Reported by: Geisj
Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76
(license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by
Corydon76 (license 14) Tested by: Corydon76
2008-07-31 19:37 +0000 [r134915] Russell Bryant <russell@digium.com>
* apps/app_ices.c: Get app_ices working again (closes issue #12981)
Reported by: dlogan Patches:
20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
(license 36) 20080709__app_ices_v2_update_14.diff uploaded by
bbryant (license 36) Tested by: bbryant
2008-07-31 19:23 +0000 [r134883] Steve Murphy <murf@digium.com>
* res/res_features.c: (closes issue #11849) Reported by: greyvoip
Tested by: murf OK, a few days of debugging, a bunch of
instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and
5 solid notebook pages of notes later, I have made the small
tweek necc. to get the start time right on the second CDR when: A
Calls B B answ. A hits Xfer button on sip phone, A dials C and
hits the OK button, A hangs up C answers ringing phone B and C
converse B and/or C hangs up But does not harm the scenario
where: A Calls B B answ. B hits xfer button on sip phone, B dials
C and hits the OK button, B hangs up C answers ringing phone A
and C converse A and/or C hangs up The difference in start times
on the second CDR is because of a Masquerade on the B channel
when the xfer number is sent. It ends up replacing the CDR on the
B channel with a duplicate, which ends up getting tossed out. We
keep a pointer to the first CDR, and update *that* after the
bridge closes. But, only if the CDR has changed. I hope this
change is specific enough not to muck up any current CDR-based
apps. In my defence, I assert that the previous information was
wrong, and this change fixes it, and possibly other similar
scenarios. I wonder if I should be doing the same thing for the
channel, as I did for the peer, but I can't think of a scenario
this might affect. I leave it, then, as an exersize for the
users, to find the scenario where the chan's CDR changes and
loses the proper start time.
2008-07-31 16:45 +0000 [r134814] Russell Bryant <russell@digium.com>
* channels/iax2-parser.c: In case we have some processing threads
that free more frames than they allocate, do not let the frame
cache grow forever. (closes issue #13160) Reported by: tavius
Tested by: tavius, russell
2008-07-31 15:56 +0000 [r134758] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Add more timeout checks into app_queue,
specifically targeting areas where an unknown and potentially
long time has just elapsed. Also added a check to try_calling()
to return early if the timeout has elapsed instead of potentially
setting a negative timeout for the call (thus making it have *no*
timeout at all). (closes issue #13186) Reported by:
miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
(license 60) Tested by: miquel_cabrespina
2008-07-30 22:39 +0000 [r134704] Tilghman Lesher <tlesher@digium.com>
* main/sched.c, include/asterisk/sched.h: Oops, wrong define
2008-07-30 22:02 +0000 [r134652] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes
issue #13051) Reported by: pj This patch substitutes commas in
the expr supplied to the if () statement, as in if ( expr ) ...
This solves both the bugs above, and makes the source symmetric
with switch statements, which were earlier reported to need this
sort of treatment. I tested this using the examples, both for the
compiler and at run time. Looks good.
2008-07-30 21:38 +0000 [r134649] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Qwell pointed out, via IRC, that the
previous fix only worked when explicitly set. When nothing is
set, and the option is implied, it breaks, because configure sets
the prefix to 'NONE'. Fixing.
2008-07-30 20:37 +0000 [r134540-134595] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max
stack size to fix a crash when compiled with LOW_MEMORY. (closes
issue #13154) Reported by: edantie
* funcs/func_curl.c: Fix a memory leak in func_curl. Every thread
that used this function leaked an allocation the size of a
pointer. (reported by jmls in #asterisk-dev)
2008-07-30 19:47 +0000 [r134480-134536] Tilghman Lesher <tlesher@digium.com>
* configure, configure.ac: Only override sysconfdir and mandir when
prefix=/usr (closes issue #13093) Reported by: pabelanger
* res/res_agi.c: launch_netscript sometimes returns -1, which fails
to set AGISTATUS. Map failure to -1, so that AGISTATUS is always
set. (closes issue #13199) Reported by: smw1218
2008-07-30 18:31 +0000 [r134475] Mark Michelson <mmichelson@digium.com>
* main/app.c: Fix a spot where a function could return without
bringing a channel out of autoservice.
2008-07-30 15:29 +0000 [r134254-134352] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: use the proper method for building version.h
* include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c,
channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c,
apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c:
build against the now-typedef-free dahdi/user.h
2008-07-29 15:54 +0000 [r134223] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Merging the imap_consistency branch. The
main aim of this branch was to make the IMAP code function in the
same manner as the ODBC code does, eliminating the need for so
many IMAP-specific code chunks. The focal point of all of this
work was to make the various macros (e.g. RETRIEVE, DISPOSE)
functionally equivalent. While doing the above work, I also fixed
a few bugs that I came across in my testing. Among these were 1.
Fixed message forwarding. This was completely broken when using
IMAP. 2. Fixed the inability to save new messages as old and vice
versa. 3. Fixed the "delete" options in voicemail.conf when using
IMAP storage. Even though a few bugs were fixed and the code is a
lot more consistent, the one thing that was *not* improved in
this branch was performance. The merge of this to trunk may not
come immediately due to the amount of work it will probably
involve. (closes issue #12764) Reported by: balsamcn
2008-07-28 21:50 +0000 [r134161] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Detect when sox fails to raise the volume,
because sox can't read the file. (closes issue #12939) Reported
by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
Corydon76 (license 14) Tested by: rickbradley
2008-07-26 15:31 +0000 [r133980] Russell Bryant <russell@digium.com>
* main/asterisk.c, include/asterisk/doxyref.h: Add the licensing
section to the docs in 1.4, as well, so that we can work on
having an accurate list for each version of Asterisk that is
supported
2008-07-25 18:00 +0000 [r133649-133709] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue
#13161)
* main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix
some errant device states by making the devicestate API more
strict in terms of the device argument (only without the unique
identifier appended). (closes issue #12771) Reported by: davidw
Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
(license 14) Tested by: davidw, jvandal, murf
2008-07-25 15:00 +0000 [r133578] Russell Bryant <russell@digium.com>
* /, LICENSE: Merged revisions 133577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
| 2 lines Fix the IAX2 URI for calling Digium ........
2008-07-25 14:40 +0000 [r133572] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: We need to make sure to null-terminate the
"name" portion of SIP URI parameters so that there are no bogus
comparisons. Thanks to bbryant for pointing this out.
2008-07-24 21:17 +0000 [r133361-133488] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes
issue #12707)
* /: Blocked revisions 133360 via svnmerge ........ r133360 |
tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines
This part was not correctly patched for AST-2008-010. ........
2008-07-23 21:49 +0000 [r133295] Jason Parker <jparker@digium.com>
* channels/chan_dahdi.c: inbandrelease is gone - it's now
inbanddisconnect
2008-07-23 21:05 +0000 [r133226-133237] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/stringfields.h, main/utils.c: revert an
optimization that broke ABI... thanks russell!
* apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
apps/app_dahdibarge.c, channels/chan_dahdi.c,
apps/app_dahdiras.c: make some more changes to the dahdi/zap
channel name support stuff to ensure allthe globals are 'const',
and clean up mmichelson's changes to app_chanspy to simplify the
code
2008-07-23 19:39 +0000 [r132974-133169] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
channels/chan_dahdi.c: As suggested by seanbright, the
PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at
compile time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the static
qualifier. Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
the suggestion.
* apps/app_chanspy.c: Zap/pseudo is ten characters, but
DAHDI/pseudo is twelve. The strncmp call in next_channel should
account for this.
* apps/app_chanspy.c: Update the "last" channel in next_channel in
app_chanspy so that the same pseudo channel isn't constantly
returned. related to issue #13124
* channels/chan_dahdi.c: Small cleanup. Move the declaration of the
DAHDI_SPANINFO variable to the block where it is used. This
allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
Tzafrir for pointing this out on #asterisk-dev
* channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI
is not defined.
2008-07-23 15:52 +0000 [r132872-132942] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: ensure that after a channel is created, if
it happened to be in 'channel alarm' state, when that alarm
clears we won't generate a spurious 'alarm cleared' message
(closes issue #12160) Reported by: tzafrir
* include/asterisk/stringfields.h, main/utils.c: minor optimization
for stringfields: when a field is being set to a larger value
than it currently contains and it happens to be the most recent
field allocated from the currentl pool, it is possible to 'grow'
it without having to waste the space it is currently using (or
potentially even allocate a new pool)
2008-07-23 11:37 +0000 [r132826] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c: another Fix because of r119585, this
commit has broken high frequented BRI Ports, there was a
possibility that a channel, that was marked as in_use would be
reused later, the corresponding port could got stuck then. So it
is recommended to upgrade for chan_misdn users.
2008-07-22 22:14 +0000 [r132790] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Allow Spiraled INVITEs to work correctly
within Asterisk. Prior to this change, a spiraled INVITE would
cause a 482 Loop Detected to be sent to the caller. With this
change, if a potential loop is detected, the Request-URI is
inspected to see if it has changed from what was originally
received. If pedantic mode is on, then this inspection is fully
RFC 3261 compliant. If pedantic mode is not on, then a string
comparison is used to test the equality of the two R-URIs. This
has been tested by using OpenSER to rewrite the R-URI and send
the INVITE back to Asterisk. (closes issue #7403) Reported by:
stephen_dredge
2008-07-22 22:11 +0000 [r132784-132787] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/options.h, main/asterisk.c,
apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c,
apps/app_dahdiras.c: fix up namespace pollution for
dahdi_chan_mode enum correct registration of AMI actions in
chan_dahdi; in zap-only mode, only register the Zap flavors of
the actions (and use Zap prefixes for headers and acks), but in
dahdi+zap mode, register both Zap and DAHDI flavors of actions
* Makefile.rules: add rules to create preprocessor output... useful
for debugging macros
2008-07-22 21:19 +0000 [r132713] Tilghman Lesher <tlesher@digium.com>
* configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
revisions 132711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
| 2 lines Fixes for AST-2008-010 and AST-2008-011 ........
2008-07-22 21:17 +0000 [r132704-132712] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: ensure that if any alarms exist at channel
creation time, they are handled identically to if they occurred
later, so that later alarm clearing will work properly and 'make
sense' (closes issue #12160) Reported by: tzafrir
* configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK
able to print a 'pretty' description of what it is doing
2008-07-22 20:10 +0000 [r132645] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, doc/sip-retransmit.txt (added): The most
common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails.
This is frequently misunderstood as a failure of Asterisk, not a
failure of the network to reach the other party. This document
tries to assist the Asterisk user in sorting out these issues by
explaining the logic and pointing at some possible causes.
Hopefully, we will get other questions now :-)
2008-07-22 19:57 +0000 [r132571-132642] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: correct wording in comment
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
include/asterisk/autoconfig.h.in, configure.ac: use renamed
libpri API call for controlling this feature (was improperly
named before)
* channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel
on BRI spans, and don't attempt to use channel 24 as a D-channel
on spans of unexpected sizes
2008-07-21 20:51 +0000 [r132506-132507] Brett Bryant <bbryant@digium.com>
* apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set
properly when it isn't supported on a channel (yet _another_
useful patch by eliel). (issue #13081) Reported by: eliel
Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested
by: eliel
* channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels
not being removed when a call was rejected (from the calling box,
not the box that denied the registration). Related to revisions
132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently
tested 1.4 with a backport from those revisions, so I didn't see
this problem (oops).
2008-07-19 16:45 +0000 [r132311] Kevin P. Fleming <kpfleming@digium.com>
* LICENSE: grant a license exception to allow distribution of
Asterisk binaries that use the UW IMAP Toolkit (which is licensed
under a non-GPL-compatible license)
2008-07-18 19:06 +0000 [r131970-132112] Tilghman Lesher <tlesher@digium.com>
* main/say.c: Fix for Taiwanese number syntax (closes issue #12319)
Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
uploaded by CharlesWang (license 444)
* main/config.c: Textual clarification (closes issue #13106)
Reported by: flefoll Patches:
config.c.br14.120173.patch-unknown-directive uploaded by flefoll
(license 244)
* include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within
the destroy, to allow a scheduled job to continue, if it's
waiting on the mutex which the destroy thread has.
* main/sched.c: Oops
* main/sched.c, include/asterisk/sched.h: Preserve ABI
compatibility with last change
* main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c:
Make the ast_assert call within ast_sched_del report something
useful.
2008-07-18 16:15 +0000 [r131921] Kevin P. Fleming <kpfleming@digium.com>
* main/dlfcn.c (removed), main/loader.c, main/Makefile,
include/asterisk/dlfcn-compat.h (removed): remove the dlfcn
compatibility stuff, because no platforms that Asterisk currently
runs on it use it, and it doesn't build anyway
2008-07-18 15:34 +0000 [r131915] Brett Bryant <bbryant@digium.com>
* res/res_features.c: Fix a bug in blind transfers where the
BLINDTRANSFER variable isn't always set to the other end of the
blind transfer. (closes issue #12586)
2008-07-17 20:35 +0000 [r131790] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965)
as it appears that it was in error. This should fix talk call
progress on analog lines. (closes issue #12178) Reported by:
michael-fig Patches: 20080717__bug12178.diff.txt uploaded by
Corydon76 (license 14)
2008-07-16 22:17 +0000 [r131491] Brett Bryant <bbryant@digium.com>
* channels/chan_iax2.c: Fix a bug in iax2 registration that allowed
peers to register with case-insensitive names (user_cmp_cb and
peer_cmp_cb are now both case-sensitive). (closes issue #13091)
2008-07-16 21:46 +0000 [r131480] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Apparently, in certain cases, a callno is
already destroyed when iax2_destroy is called.
2008-07-16 20:47 +0000 [r131421] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Always ensure that the channel's tech_pvt
reference is NULL after calling the destroy callback. (closes
issue #13060) Reported by: jpgrayson Patches:
chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license
492)
2008-07-16 20:23 +0000 [r131299-131369] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Move the init_queue call back to where it used
to be (changed Sept 12 last year). It was moved then to prevent a
memory leak. Since then, the same memory leak recurred and was
fixed in a better way. Now it has been found that the placement
of this init_queue call can cause problems if a realtime queue
has values changed to an empty string. The problem is that the
default value for that queue parameter would not be set. (closes
issue #13084) Reported by: elbriga
* apps/app_queue.c: Apparently, "thread safety" is important,
whatever that means. :P (Thanks Russell!)
* apps/app_queue.c: Make absolutely certain that the transfer
datastore is removed from the calling channel once the caller is
finished in the queue. This could have weird con- sequences when
dialing local queue members when multiple transfers occur on a
single call. Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member. (closes issue
#13047) Reported by: festr
2008-07-16 17:53 +0000 [r131242] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The
problem was that, esoteric as it is, because the hangerupper
context immediately preceded the std-priv-extent macro, that the
checking code accidentally would fall from traversing hangerupper
into the std-priv-exten macro, where it would hit the hangerupper
in the 'includes', and proceed into an infinite recursion. A
small fix to traverse into the statements of the context instead
of the context solves this issue. I also added some commented out
printfs for debug, which were pretty handy in the face of a dorky
gdb. This was a problem around since the package was first
written; but evidently pretty rare in turning up in the field.
2008-07-15 17:47 +0000 [r131012] Michiel van Baak <michiel@vanbaak.info>
* main/cdr.c: remove 4 lines of redundant code. (closes issue
#13080) Reported by: gknispel_proformatique Patches:
trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
2008-07-15 17:19 +0000 [r130889-130959] Tilghman Lesher <tlesher@digium.com>
* main/manager.c, channels/chan_sip.c: astman_send_error does not
need a newline appended -- the API takes care of that for us.
(closes issue #13068) Reported by: gknispel_proformatique
Patches: asterisk_1_4_astman_send.patch uploaded by gknispel
(license 261) asterisk_trunk_astman_send.patch uploaded by
gknispel (license 261)
* channels/chan_iax2.c: Override the callerid in all cases when the
callerid is set in the user, not just when a remote callerid is
set. Also, if not set in the user, allow the remote CallerID to
pass through. (closes issue #12875) Reported by: dimas Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
2008-07-14 17:50 +0000 [r130792] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in
app_dial to be sure there are no audiohooks present on the
channels involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
2008-07-14 17:10 +0000 [r130735] Michiel van Baak <michiel@vanbaak.info>
* main/dnsmgr.c: notify the user that dnsmgr refresh wont work when
dnsmgr is not enabled. Previously this command would
automagically appear and disappear. This was confusing. (closes
issue #12796) Reported by: chappell Patches:
dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
russell, chappell, mvanbaak
2008-07-14 10:38 +0000 [r130634] Russell Bryant <russell@digium.com>
* main/audiohook.c: Bump up the debug level for a message.
2008-07-13 22:48 +0000 [r130573] Michiel van Baak <michiel@vanbaak.info>
* main/manager.c: fix memory leak when originate from manager
cannot create a thread (closes issue #13069) Reported by:
gknispel_proformatique Patches:
asterisk_trunk_action_originate.patch uploaded by gknispel
(license 261) Tested by: gknispel_proformatique, mvanbaak
2008-07-13 17:56 +0000 [r130514] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Reverting 2 changesets, as it breaks
incoming IAX2 calls (Related to issue #12963) Reported by:
mvanbaak
2008-07-12 10:25 +0000 [r130373] Michiel van Baak <michiel@vanbaak.info>
* pbx/pbx_ael.c: in 1.4 the functions still have | as argument
seperator. This commit fixes the use of RAND in the ael random
function. (closes issue #13061) Reported by: danpwi
2008-07-11 22:23 +0000 [r130298-130317] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: forcibly remove the modules that are changing names
* include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c,
Makefile, main/channel.c, apps/app_dahdibarge.c,
channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c,
apps/app_dahdiras.c, main/file.c,
contrib/utils/zones2indications.c, include/asterisk/channel.h,
channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility
work, with lots more to come... this tree is not yet ready for
users to be easily upgrading or switching, but it needs to be :-)
2008-07-11 20:03 +0000 [r130173-130236] Mark Michelson <mmichelson@digium.com>
* main/audiohook.c: Remove redundant logic
* main/audiohook.c: Fix a typo in audiohook_read_frame_both. While
this change has not been proven to fix any specific issue, it is
incorrect and could cause unforeseen problems.
2008-07-11 18:51 +0000 [r130102-130169] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Ensure that a destination callno of 0 will
not match for frames that do not start a dialog (new, lagrq, and
ping). (closes issue #12963) Reported by: russellb Patches:
chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
* channels/chan_agent.c: Pass the devicestate from an underlying
channel up through the Agent channel. This should make the Agent
always report the correct device state, even when the underlying
channel is used for other purposes. (closes issue #12773)
Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded
by Corydon76 (license 14) Tested by: davidw
2008-07-11 16:08 +0000 [r130039-130042] Kevin P. Fleming <kpfleming@digium.com>
* doc/configuration.txt, configs/extensions.conf.sample,
configs/sla.conf.sample, configs/zapata.conf.sample (removed),
contrib/scripts/autosupport, README,
configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c,
include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt,
configs/extensions.ael.sample, configs/smdi.conf.sample: new
installations should be using DAHDI instead of Zaptel, so the
sample config file is now chan_dahdi.conf instead of zapata.conf
also, convert remaining references to zapata.conf in various
places
* configs/zapata.conf.sample, channels/chan_dahdi.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: add support for a
configuration parameter for 'inband audio during RELEASE', which
is currently mandatory in libpri-1.4.4 but will become
configurable in libpri-1.4.5 later today (related to issue
#13042)
2008-07-11 14:18 +0000 [r129970] Russell Bryant <russell@digium.com>
* include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro
...
2008-07-11 14:14 +0000 [r129907-129967] Kevin P. Fleming <kpfleming@digium.com>
* main/astmm.c: simplify calculation
* main/astmm.c: fix a flaw found while experimenting with structure
alignment and padding; low-fence checking would not work properly
on 64-bit platforms, because the compiler was putting 4 bytes of
padding between the fence field and the allocation memory block
added a very obvious runtime warning if this condition reoccurs,
so the developer who broke it can be chastised into fixing it :-)
* sounds/Makefile: don't attempt to set user/group ownership of
extracted sound files (reported on asterisk-users) (closes issue
#13059)
2008-07-10 21:57 +0000 [r129741-129803] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Correctly deal with duplicate NEW frames
(due to retransmission). Also, fixup the destination call number
matching to be more strict and reliable. (closes issue #12963)
Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch
uploaded by jpgrayson (license 492) Tested by: jpgrayson,
Corydon76
* res/res_config_odbc.c: Oops
2008-07-10 16:03 +0000 [r129567] Russell Bryant <russell@digium.com>
* sample.call: Note that pbx_spool.so is the module used for call
files (inspired by a question in #asterisk)
2008-07-10 13:57 +0000 [r129505] Sean Bright <sean.bright@gmail.com>
* main/editline: Update svn:ignore
2008-07-09 19:32 +0000 [r129436] Mark Michelson <mmichelson@digium.com>
* main/rtp.c: Fix a problem where inbound rfc2833 audio would be
sent to the core instead of being P2P bridged. When the core
regenerated the rfc2833 packet for the outbound leg, the SSRC
would be different than the RTP audio on the call leg causing
DTMF detection issues on the far end. (closes issue #12955)
Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
tsearle (license 373) Tested by: tonyredstone
2008-07-09 13:41 +0000 [r129343] Sean Bright <sean.bright@gmail.com>
* main/editline/makelist (removed), main/editline/makelist.in
(added), main/editline/configure, main/editline/Makefile.in,
main/editline/configure.in: Look for the system installed awk
instead of assuming it's at /usr/bin/awk. Pointed out by jmls via
#asterisk-dev.
2008-07-08 21:31 +0000 [r129158-129208] Mark Michelson <mmichelson@digium.com>
* doc/imapstorage.txt: Update documentation to have the correct
option name
* apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related
timeouts to IMAP voicemail in 1.4 since it should solve bugs
people are experiencing. Specifically, there are times where
communication with the IMAP server causes system calls to block
forever. If this should happen when querying the mailbox so that
chan_sip's do_monitor thread can send MWI to a phone, it means
that SIP calls cannot be processed any more. The timeout options
are outlined in doc/imapstorage.txt. Defaults for the timeouts
are sixty seconds. (closes issue #12987) Reported by: mthomasslo
2008-07-08 20:27 +0000 [r129047-129149] Tilghman Lesher <tlesher@digium.com>
* apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h:
Cause SIP to return a 480 instead of a 404 when a sip peer
exists, but is not registered. (closes issue #12885) Reported by:
ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76
(license 14) Tested by: ibc
* channels/chan_iax2.c: Timestamp decoding for video mini-frames is
bogus, because the timestamp only includes 15 bits, unlike voice
frames, which contain a 16-bit timestamp. (closes issue #13013)
Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch
uploaded by jpgrayson (license 492)
2008-07-08 09:52 +0000 [r128912-128950] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't hangup the call if we can't resolve
the Contact if there's a proxy route set for the call. ---- This
comment was added a while ago and today it hit me badly. /* OEJ:
Possible issue that may need a check: If we have a proxy route
between us and the device, should we care about resolving the
contact or should we just send it? */
* channels/chan_sip.c: Fix issues where repeated messages where
ignored, but retransmitted reliably instead of unreliably.
Reported by: johan Patches: 12746.txt uploaded by oej (license
306) Tested by: johan (issue #12746)
2008-07-08 00:01 +0000 [r128812-128856] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Check for non-NULL before stripping
characters. (closes issue #12954) Reported by: bfsworks Patches:
20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
Tested by: deti
* apps/app_voicemail.c: Stop using deprecated method, as requested
by Kevin.
2008-07-07 22:41 +0000 [r128795] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix handling of when a pvt disappears.
Properly return the pvt locked and don't hold the pvt lock while
destroying the ast_channel. (closes issue #13014) Reported by:
jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by
jpgrayson (license 492)
2008-07-07 20:47 +0000 [r128737] Sean Bright <sean.bright@gmail.com>
* channels/chan_iax2.c: Remove spurious trailing whitespace from
log messages and fix a spelling error in a log message. (closes
issue #13017) Reported by: jpgrayson Patches:
chan_iax2_space_after_newline.patch uploaded by jpgrayson
(license 492) chan_iax2_spelling.patch uploaded by jpgrayson
(license 492)
2008-07-07 17:02 +0000 [r128639] Mark Michelson <mmichelson@digium.com>
* channels/chan_iax2.c: By using the iaxdynamicthreadcount to
identify a thread, it was possible for thread identifiers to be
duplicated. By using a globally-unique monotonically- increasing
integer, this is now avoided. (closes issue #13009) Reported by:
jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by
jpgrayson (license 492)
2008-07-07 16:51 +0000 [r128637] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac: use tzafrir's patch to fix this problem
properly... i made the previous set of changes without thoroughly
testing them, doh! (closes issue #12911) Reported by: tzafrir
Patches: custum_dahdi_configure_2.diff uploaded by tzafrir
(license 46) Tested by: tzafrir
2008-07-04 16:11 +0000 [r127973-128029] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_config.c: Move the free down one
* main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the
'dialplan remove extension' logic, so that it a) works with
cidmatch, and b) completes contexts correctly when the extension
is ambiguous. (closes issue #12980) Reported by: licedey Patches:
20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
2008-07-03 22:20 +0000 [r127754-127895] Kevin P. Fleming <kpfleming@digium.com>
* apps/Makefile: remove this, it has been moved to the main
Makefile
* Makefile, main/editline/np/vis.c: a couple of small
Solaris-related fixes (closes issue #11885) Reported by: snuffy,
asgaroth
* configure, main/Makefile, configure.ac, acinclude.m4: ensure that
DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places
needed improve AST_EXT_LIB_CHECK to accept (and remember)
additional CFLAGS data like it does in trunk already (closes
issue #12911) Reported by: tzafrir
2008-07-03 00:16 +0000 [r127663] Steve Murphy <murf@digium.com>
* main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h:
The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported
by: murf Tested by: murf, deeperror (closes issue #12907)
Reported by: falves11 Tested by: murf, falves11 (closes issue
#11849) Reported by: greyvoip As to 11849, I think these changes
fix the core problems brought up in that bug, but perhaps not the
more global problems created by the limitations of CDR's
themselves not being oriented around transfers. Reopen if necc,
but bug reports are not the best medium for enhancement
discussions. We need to start a second-generation CDR
standardization effort to cover transfers. (closes issue #11093)
Reported by: rossbeer Tested by: greyvoip, murf
2008-07-02 20:47 +0000 [r127560] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Fix thread-safety of some of the
pbx_builtin_getvar_helper calls
2008-07-02 19:47 +0000 [r127501] Tilghman Lesher <tlesher@digium.com>
* main/acl.c: Merged revisions 127466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
Solaris fix (closes issue #12949) Reported by: snuffy Patches:
bug_12949.diff uploaded by snuffy (license 35) ........
2008-07-01 23:36 +0000 [r127244] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Add error message to failed open(2) calls
inside the copy() function of app_voicemail. This idea came as
part of my work in helping to resolve issue #12764.
2008-07-01 20:25 +0000 [r126999-127133] Tilghman Lesher <tlesher@digium.com>
* build_tools/cflags.xml, channels/chan_iax2.c: Disable the old,
slow search for matching callno in chan_iax2 (but allow it to be
reenabled for debugging)
* channels/chan_iax2.c: Oops
* channels/chan_iax2.c: Change around how we schedule pings and
lagrqs, and fix a reason why the jobs were not getting properly
cancelled. (closes issue #12903) Reported by: stevedavies
Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76
(license 14) Tested by: stevedavies
* channels/chan_iax2.c: Suppress annoying warning by finding the
remaining cases where the callno is not in the hash.
2008-07-01 14:59 +0000 [r126735-126902] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Use domain part of SIP uri in register=
configuration as fromdomain. Reported by: one47 Patches:
sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes
issue #12474)
* channels/chan_sip.c: Handle escaped URI's in call pickups. Patch
by oej and IgorG. Reported by: IgorG Patches:
bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by:
IgorG, oej (closes issue #12299)
* configs/sip.conf.sample: Clear up documentation on "domain="
setting in sip.conf Reported by: davidw (closes issue #12413)
* channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs
requests (to confirm that the dialog exist). Don't bother
checking the request URI. (closes issue #11264) Reported by: ibc
* channels/chan_sip.c: Fix bad XML for hold notification. Reported
by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432)
(closes issue #12942)
2008-06-30 23:11 +0000 [r126680] Jeff Peeler <jpeeler@digium.com>
* channels/chan_dahdi.c: Load the proper channel configuration file
based on which driver was detected.
2008-06-30 22:30 +0000 [r126674] Tilghman Lesher <tlesher@digium.com>
* configs/zapata.conf.sample: Add note about other names for
EuroISDN
2008-06-30 16:05 +0000 [r126573] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS
version of the recently added DEADLOCK_AVOIDANCE() macro. This
caused the lock to not actually be released, and as a result, not
avoid deadlocks at all. This resolves the issues reported in the
last while about Asterisk locking up all over the place (and most
commonly, in chan_iax2). (closes issue #12927) (closes issue
#12940) (closes issue #12925) (potentially closes others ...)
2008-06-30 12:50 +0000 [r126516] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Send all responses to an INVITE reliably, so
that we retransmit if we don't get an ACK and also fail if we
don't get the very same precious ACK. Based on patch by tsearle,
with my own additions. (closes issue #12951) Reported by: tsearle
Patches: busy_retransmit.patch uploaded by tsearle (license 373)
2008-06-29 18:05 +0000 [r126395] Kevin P. Fleming <kpfleming@digium.com>
* pbx/Makefile: ignore warnings for prototypes in GTK headers
2008-06-27 22:01 +0000 [r125740-126056] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: When we get a 408 Timeout, don't stop trying
to re-register. (closes issue #12863) Reported by: ricvil
* include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined
to HAVE_ZAPTEL in dahdi_compat.h, we must first check for
HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches:
tonezone_compat.diff uploaded by opticron (license 267)
* main/utils.c, include/asterisk/lock.h: In this debugging
function, copy to a buffer instead of using potentially unsafe
pointers.
* channels/chan_local.c: Add proper deadlock avoidance. (closes
issue #12914) Reported by: ozan Patches:
20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
Tested by: ozan
2008-06-26 23:03 +0000 [r125587] Jason Parker <jparker@digium.com>
* main/utils.c: Make sure to unlock the lock_info lock (huh?).
Possible deadlock?
2008-06-26 22:52 +0000 [r125476-125585] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Add the interface of a queue member to the
output of the "queue show" command so that it can easily be
associated with a queue member's name. This helps so that the
appropriate queue member can be removed or paused since the
interface is required, not the member's name. (closes issue
#12783) Reported by: davevg Patches: app_queue.diff uploaded by
davevg (license 209) with small mod from me
* apps/app_queue.c: Backport of attended transfer queue_log patch
from trunk. This patch allows for attended transfers to be logged
in the queue_log the same way that blind transfers have always
been. It was decided by popular opinion on the asterisk-dev
mailing list that this should be backported to 1.4. Thanks to
everyone who gave an opinion.
* apps/app_queue.c: Prior to this patch, the "queue show" command
used cached information for realtime queues instead of giving
up-to-date info. Now realtime is queried for the latest and
greatest in queue info. (closes issue #12858) Reported by: bcnit
Patches: queue_show.patch uploaded by putnopvut (license 60)
2008-06-26 16:32 +0000 [r125384] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add support for peer realm based auth (a few
missing lines, the rest is well documented but never worked)
2008-06-26 15:30 +0000 [r125327] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: ensure that (whenever possible) if we
generate a log message because an ioctl() call to DAHDI/Zaptel
failed, that we include the reason it failed by including the
stringified error number (issue AST-80)
2008-06-26 11:01 +0000 [r125218-125276] Tilghman Lesher <tlesher@digium.com>
* main/rtp.c: Check for rtcp structure before trying to delete
schedule. (closes issue #12872) Reported by: destiny6628 Patches:
20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
Tested by: destiny6628
* configs/agents.conf.sample: Document ackcall=always. (closes
issue #12852) Reported by: davidw
2008-06-25 22:21 +0000 [r125132] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_rpt.c, include/asterisk/dahdi_compat.h,
channels/chan_dahdi.c, configure,
include/asterisk/tonezone_compat.h (added), configure.ac: allow
tonezone to live in a different place than DAHDI/Zaptel, since
dahdi-tools and dahdi-linux are now separate packages and can be
installed in different places don't include tonezone.h in
dahdi_compat.h, because only a couple of modules need it get
app_rpt building again after the DAHDI changes (closes issue
#12911) Reported by: tzafrir
2008-06-25 00:46 +0000 [r124908-124965] Tilghman Lesher <tlesher@digium.com>
* channels/chan_dahdi.c: Pvt deadlock causes some channels to get
stuck in Reserved status. (closes issue #12621) Reported by:
fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
Corydon76 (license 14) Tested by: fabianoheringer
* apps/app_voicemail.c: Occasionally control characters find their
way into CallerID. These need to be stripped prior to placing
CallerID in the headers of an email. (closes issue #12759)
Reported by: RobH Patches: 20080602__bug12759__2.diff.txt
uploaded by Corydon76 (license 14) Tested by: RobH
* channels/chan_sip.c: Don't access the pvt structure if unable to
acquire the lock. (closes issue #12162) Reported by: norman
Patches: 12162-lockfail.diff uploaded by qwell (license 4)
2008-06-23 21:22 +0000 [r124743] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: emit a warning if the old IAX2 call
searching code finds a call when the new code did not... so that
we can get rid of the old code in 2-3 months
2008-06-22 02:54 +0000 [r124540] Steve Murphy <murf@digium.com>
* apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac
Sorry, my testing did not contain the simple case of forkCDR(v),
I am much embarrassed to admit. If I had, I would have more
solidly initialized the opts element for varset.
2008-06-20 23:12 +0000 [r124395-124450] Tilghman Lesher <tlesher@digium.com>
* apps/app_rpt.c: usleep with a value over 1,000,000 is
nonportable. Changing to use sleep() instead. (closes issue
#12814) Reported by: pputman Patches: app_rtp_sleep.patch
uploaded by pputman (license 81)
* main/app.c: If the last character in a string to be parsed is the
delimiter, then we should count that final empty string as an
additional argument.
2008-06-20 21:14 +0000 [r124372] Jeff Gehlbach <jeffg@opennms.org>
* doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in
digium-mib.txt and asterisk-mib.txt to placate smilint - bug
12905
2008-06-20 20:16 +0000 [r124182-124315] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c: When using a Local channel, started by a
call file, with a destination of an AGI script, the AGI script
does not always get notified of a hangup if the underlying
channel hangs up early. (closes issue #11833) Reported by: IgorG
Patches: local_hangup-v1.diff uploaded by IgorG (license 20)
* channels/chan_dahdi.c: It's possible for a hangup to be received,
even just after the initial cid spill. (closes issue #12453)
Reported by: Alex728 Patches: 20080604__bug12453.diff.txt
uploaded by Corydon76 (license 14)
2008-06-19 20:28 +0000 [r124112] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix IMAP forwarding so that messages are
sent to the proper mailbox. (closes issue #12897) Reported by:
jaroth Patches: destination_forward.patch uploaded by jaroth
(license 50)
2008-06-19 19:55 +0000 [r124066] Brett Bryant <bbryant@digium.com>
* main/utils.c: Merge revision 124064 from trunk. Add errors that
report any locks held by threads when they are being closed.
2008-06-19 16:58 +0000 [r123710-123930] Tilghman Lesher <tlesher@digium.com>
* main/channel.c: Change informative messages to use the _multiple
variant when multiple formats are possible. (Closes issue #12848)
Reported by klaus3000
* build_tools/strip_nonapi: Only process 40 arguments (20 files) at
once with xargs, because some older shells may force xargs to
separate on an odd boundary. (Closes issue #12883) Reported by
Nik Soggia
* configs/sip.conf.sample: Correct description of notifyringing
option. (Closes issue #12890) Reported by gminet
* main/asterisk.c: The RDTSC instruction was introduced on the
Pentium line of microprocessors, and is not compatible with
certain 586 clones, like Cyrix. Hence, asking for i386
compatibility was always incorrect. See
http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported
by tecnoxarxa
* main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add
support for saying numbers in Hebrew. (closes issue #11662)
Reported by: greenfieldtech Patches: say.c.patch-12042008
uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
uploaded by greenfieldtech (with signficant changes to the
spreadsheet by me)
* pbx/pbx_spool.c: Set the variables top-down, so that if a script
sets a variable more than once, the last one will take
precedence. (closes issue #12673) Reported by: phber Patches:
20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
2008-06-17 20:26 +0000 [r123485] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Make chan_sip build under dev mode with
compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this
2008-06-17 18:56 +0000 [r123391] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Fix 3 more places where not locking the
structure could cause the wrong lock to be unlocked. (Closes
issue #12795)
2008-06-17 18:09 +0000 [r123274-123333] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Cisco BTS sends SIP responses with a tab
between the Cseq number and SIP request method in the Cseq:
header. Asterisk did not handle this properly, but with this
patch, all is well. (closes issue #12834) Reported by: tobias_e
Patches: 12834.patch uploaded by putnopvut (license 60) Tested
by: tobias_e
* apps/app_queue.c: davidw pointed out that the holdtime
calculation used by app_queue does not use "boxcar" filtering as
the comments say. The term "boxcar" means that the number of
samples used to calculate stays constant, with new samples
replacing the oldest ones. The queue holdtime calculation uses
all holdtime samples collected since the queue was loaded, so the
comment has been changed to be accurate. (closes issue #12781)
Reported by: davidw
2008-06-17 15:48 +0000 [r123271] Russell Bryant <russell@digium.com>
* main/astobj2.c: Fix a memory leak in astobj2 that was pointed out
by seanbright. When a container got destroyed, the underlying
bucket list entry for each object that was in the container at
that time did not get free'd.
2008-06-16 19:50 +0000 [r123110-123113] Tilghman Lesher <tlesher@digium.com>
* channels/chan_mgcp.c, channels/chan_dahdi.c,
channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_iax2.c: Port "hasvoicemail" change from SIP to
other channel drivers
* channels/chan_sip.c: People expect that if "hasvoicemail" is set
in users.conf, even if "mailbox" isn't set, that SIP will detect
a mailbox. (closes issue #12855) Reported by: PLL Patches:
20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: PLL
2008-06-16 12:31 +0000 [r122869-122919] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only compare the first 15 characters so that
even if the charset is specified we still accept it as SDP.
(closes issue #12803) Reported by: lanzaandrea Patches:
chan_sip.c.diff uploaded by lanzaandrea (license 496)
* channels/chan_sip.c: Don't send a BYE on a dialog that is already
gone during a REFER. (closes issue #12865) Reported by: flefoll
Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by
flefoll (license 244)
2008-06-13 21:44 +0000 [r122713] Mark Michelson <mmichelson@digium.com>
* main/autoservice.c: Short circuit the loop in autoservice_run if
there are no channels to poll. If we continued, then the result
would be calling poll() with a NULL pollfd array. While this is
fine with POSIX's poll(2) system call, those who use Asterisk's
internal poll mechanism (Darwin systems) would have a failed
assertion occur when poll is called. (related to issue #10342)
2008-06-13 18:57 +0000 [r122663] Jeff Peeler <jpeeler@digium.com>
* include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed
dahdi compatability header from assuming either dahdi or zaptel
is installed (may not have either)
2008-06-13 17:45 +0000 [r122617] Terry Wilson <twilson@digium.com>
* apps/app_dial.c: Remove extra option from previous solution
attempt
2008-06-13 17:36 +0000 [r122613] Jeff Peeler <jpeeler@digium.com>
* configure, configure.ac: (closes issue #12846) Reported by:
Netview Tested by: jpeeler Use correct location to search for
tonezone.
2008-06-13 16:29 +0000 [r122589] Terry Wilson <twilson@digium.com>
* apps/app_dial.c, res/res_features.c: This should fix the behavior
of the 'T' dial feature being passed incorrectly to the
transferee when builtin_atxfers are used. Also, doing a
builtin_atxfer to parking was broken and is fixed here as well.
(closes issue #11898) Reported by: sergee Tested by: otherwiseguy
2008-06-12 19:08 +0000 [r122314] Jeff Peeler <jpeeler@digium.com>
* main/indications.c, include/asterisk/dahdi_compat.h (added),
main/loader.c, main/channel.c, channels/chan_dahdi.c (added),
configure, apps/app_zapscan.c (removed), apps/app_zapras.c
(removed), main/app.c, include/asterisk/options.h,
apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c,
channels/chan_zap.c (removed), apps/app_page.c,
include/asterisk/indications.h, apps/app_dahdiras.c (added),
configure.ac, apps/app_disa.c, include/asterisk/channel.h,
apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c
(removed), channels/chan_misdn.c, apps/app_flash.c,
build_tools/menuselect-deps.in, funcs/func_channel.c,
main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c,
codecs/codec_dahdi.c (added), res/res_indications.c,
pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c,
main/asterisk.c, apps/app_dahdibarge.c (added),
apps/app_meetme.c, include/asterisk/autoconfig.h.in,
apps/app_dahdiscan.c (added), acinclude.m4,
res/res_musiconhold.c, codecs/codec_zap.c (removed),
channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI
usage favored, but all Zap stuff should continue working. Release
announcement to follow.
2008-06-12 18:50 +0000 [r122311] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Properly play a holdtime message if the
announce-holdtime option is set to "once." (closes issue #12842)
Reported by: ramonpeek Patches: patch001.diff uploaded by
ramonpeek (license 266)
2008-06-12 18:22 +0000 [r122259] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix some race conditions that cause
ast_assert() to report that chan_iax2 tried to remove an entry
that wasn't in the scheduler
2008-06-12 15:46 +0000 [r122208] Jeff Peeler <jpeeler@digium.com>
* apps/app_parkandannounce.c, res/res_features.c: (closes issue
#12193) Reported by: davidw Patch by: Corydon76, modified by me
to work properly with ParkAndAnnounce app
2008-06-12 15:18 +0000 [r122130-122137] Tilghman Lesher <tlesher@digium.com>
* apps/app_meetme.c: Flipflop the sections for two options, since
the section for 'X' (exit context) may otherwise absorb
keypresses meant for 's' (admin/user menu). (closes issue #12836)
Reported by: blitzrage Patches: 20080611__bug12836.diff.txt
uploaded by Corydon76 (license 14) Tested by: blitzrage
* main/channel.c: Occasionally, the alertpipe loses its nonblocking
status, so detect and correct that situation before it causes a
deadlock. (Reported and tested by ctooley via #asterisk-dev)
2008-06-12 14:51 +0000 [r122127] Steve Murphy <murf@digium.com>
* main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the
code added to ast_cdr_busy, _failed, and _noanswer was redundant.
Didn't spot it until I was resolving conflicts in trunk. Ugh.
Redundant code removed. It wasn't harmful. Just dumb.
2008-06-12 Russell Bryant <russell@digium.com>
* Asterisk 1.4.21 released.
2008-06-06 Russell Bryant <russell@digium.com>
* Asterisk 1.4.21-rc2 released.
2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell@digium.com>
* UPGRADE-1.2.txt: fix filename
* UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
from Asterisk 1.2, for handy reference.
2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Ignore appended resource when comparing JIDs.
2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell@digium.com>
* doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
forwarding station isn't very useful for most people, so pull in
the changes that allow searching for SMDI messages based on other
components of the SMDI message. Also, update the SMDI
documentation.
2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Make sure that the string we set will survive
the unref of the queue member. Thanks to Russell, who pointed
this out.
2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher@digium.com>
* channels/chan_zap.c: If we fail to setup the PRI request channel,
don't continue, exit with an error. (closes issue #11989)
Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
uploaded by Corydon76 (license 14)
2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell@digium.com>
* pbx/pbx_config.c: Make the "dialplan remove include" CLI command
actually work. Also, tweak some formatting, and make the success
message a little bit more clear. (closes AST-52)
2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Tab completion when removing a member should
give the member's interface, not the name, since the interface is
what is expected for the command. (closes issue #12783) Reported
by: davevg
2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp@digium.com>
* main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
for when the dialplan is done reloading. (closes issue #12716)
Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
chappell (license 8)
2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_loopback.c: Due to incorrect use of the
AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
any translation on the extension number before searching for it
in the target context. (closes issue #12473) Reported by:
chappell Patches: pbx_loopback.c.diff uploaded by chappell
(license 8)
2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler@digium.com>
* main/config.c: (closes issue #11594) Reported by: yem Tested by:
yem This change decreases the buffer size allocated on the stack
substantially in config_text_file_load when LOW_MEMORY is turned
on. This change combined with the fix from revision 117462
(making mkintf not copy the zt_chan_conf structure) was enough to
prevent the crash.
2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix another place where peer->callno could
change at a very bad time, and also fix a place where a peer was
used after the reference was released. (inspired by rev 120001)
2008-06-03 Russell Bryant <russell@digium.com>
* Asterisk 1.4.21-rc1 released.
2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: When listing the manager users, managers in
users.conf are not shown, even though they are allowed to
connect. (closes issue #12594) Reported by: bkruse Patches:
12594-managerusers-2.diff uploaded by qwell (license 4) Tested
by: bkruse
* channels/chan_iax2.c: Save the callno when we're poking, because
our peer structure could change during deadlock avoidance (and
thus we unlock the wrong callno, causing a cascade failure).
(closes issue #12717) Reported by: gewfie Patches:
20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
Tested by: gewfie
2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
pbx/ael/ael-test/ref.ael-vtest13,
pbx/ael/ael-test/ref.ael-vtest17,
pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
Hadn't updated this for the changes I made to preserve ${EXTEN}
in switches, which affected several tests because it adds extra
priorities, and at least one needed to be updated because of the
removal of the empty extension warning message.
* pbx/pbx_ael.c: as per
http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the
WARNING that an extension is empty be reduced to a NOTICE or
less, as empty extensions are syntactically possible, and no big
deal. With which I agree, and have removed that WARNING message
entirely. I think it is not necessary to see this message. It
didn't state that a NoOp() was inserted automatically on your
behalf, and really, as users, who cares? Why freak out dialplan
writers with unnecessary warnings? The details of the
machinations a compiler goes thru to produce working assembly
code is of little interest to most programmers-- we will follow
the unix principal of doing our work silently.
2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Treat ECONNREFUSED as an error that will
stop further retransmissions. (issue #AST-58, patch from
Switchvox)
2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Revert a change made for issue #12479. This
change caused a regression such that a dial string such as
(IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore. (closes issue #12770) Reported by: dagmoller
* main/manager.c: Improve CLI command blacklist checking for the
command manager action. Previously, it did not handle case or
whitespace properly. This made it possible for blacklisted
commands to get executed anyway. (closes issue #12765)
2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
account with any configured XMPP client (in jabber.conf). The
actual connection is made when a call comes in Asterisk. Fix the
ast_aji_get_client function that was not able to retrieve an XMPP
client from its JID. (closes issue #12085) Reported by: junky
Tested by: phsultan
2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
sent because it comes up too soon, make sure to reschedule so it
gets sent later.
2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
enabled
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
counter for unhandled_bmsg Print, this prevents the logs to be
flooded to fast and save CPU in this error scenario. Added
'last_used' element to bc structure, when a bchannel changes from
used to free this exact time will be marked in last_used. When a
new channel is requested the find_free_chan function will check
if the new empty channel was used within the last second, if yes
it will search for the next channel, if no it will return this
channel. This simple mechanism has prooven to prevent race
conditions where the NT and TE tried to allocate the exact same
channel at the same time (RELEASE cause: 44).
2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Change a debug message to an actual debug
message
* apps/app_dial.c: Fix another typo in documentation
2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel@vanbaak.info>
* apps/app_dial.c: small typo fix 'retires' => 'retries'
2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: When joinempty=strict, it only failed on join
if there were busy members. If all members were logged out OR
paused, then it (incorrectly) let callers join the queue. (closes
issue #12451) Reported by: davidw
2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp@digium.com>
* main/autoservice.c: Fix a bug I found while testing for another
issue.
2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel@vanbaak.info>
* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.mandrake.asterisk,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.gentoo.asterisk,
contrib/init.d/rc.slackware.asterisk: dont use a bashism way to
check the $VERSION variable. The rc/init.d scripts, and
safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
(me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
(closes issue #12687) Reported by: loloski Patches:
20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
mvanbaak (license 7) Tested by: loloski, mvanbaak
2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
May 2008) | 7 lines - Instead of only enforcing destination call
number checking on an ACK, check all full frames except for PING
and LAGRQ, which may be sent by older versions too quickly to
contain the destination call number. (As suggested by Tim Panton
on the asterisk-dev list) - Merge changes from
team/russell/iax2-frame-race, which prevents PING and LAGRQ from
being sent before the destination call number is known. ........
* main/autoservice.c: Fix a race condition in channel autoservice.
There was still a small window of opportunity for a DTMF frame,
or some other deferred frame type, to come in and get dropped.
(closes issue #12656) (closes issue #12656) Reported by: dimas
Patches: v3-12656.patch uploaded by dimas (license 88) -- with
some modifications by me
* include/asterisk/audiohook.h: Oddly enough, all of the contents
of audiohook.h were in there twice. I have removed the second
copy.
2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher@digium.com>
* channels/chan_zap.c: Call waiting tone occurs too often, because
it's getting serviced by both subchannels. (closes issue #11354)
Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
by Corydon76 (license 14)
2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell@digium.com>
* apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
use ast_safe_sleep() instead of calling the wait application
(thanks to tilghman for pointing these out!)
* /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
May 2008) | 7 lines Merge changes from
team/russell/iax2-another-fix-to-the-fix As described in the
following post to the asterisk-dev mailing list, only enforce
destination call numbers when processing an ACK.
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
(closes issue #12631) ........
* apps/app_milliwatt.c: - Mark app_milliwatt dependent on
res_indications (thanks to jsmith) - fix a typo in a log message
(thanks to qwell)
* apps/app_milliwatt.c: Change milliwatt to use the proper tone by
default (1004 Hz) instead of 1000 Hz. An option is there to use
1000 Hz for anyone that might want it.
2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Define also when not DEBUG_THREADS
* channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
code that ensures that when we do deadlock avoidance, we don't
lose the information about how a lock was originally acquired.
2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf@digium.com>
* main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
issue #11721) (closes issue #12726) Reported by: arkadia Tested
by: murf These changes: 1. revert the changes made via bug 10668;
I should have known that such changes, even tho they made sense
at the time, seemed like an omission, etc, were actually integral
to the CDR system via forkCDR. It makes sense to me now that
forkCDR didn't natively end any CDR's, but rather depended on
natively closing them all at hangup time via traversing and
closing them all, whether locked or not. I still don't completely
understand the benefits of setvar and answer operating on locked
cdrs, but I've seen enough to revert those changes also, and stop
messing up users who depended on that behavior. bug 12726 found
reverting the changes fixed his changes, and after a long review
and working on forkCDR, I can see why. 2. Apply the suggested
enhancements proposed in 10668, but in a completely compatible
way. ForkCDR will behave exactly as before, but now has new
options that will allow some actions to be taken that will
slightly modify the outcome and side-effects of forkCDR. Based on
conversations I've had with various people, these small tweaks
will allow some users to get the behavior they need. For
instance, users executing forkCDR in an AGI script will find the
answer time set, and DISPOSITION set, a situation not covered
when the routines were first written. 3. A small problem in the
cdr serializer would output answer and end times even when they
were not set. This is now fixed.
2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant@digium.com>
* channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
Fixes a bug in chan_iax that uses send_command to poke a peer
while a channel is unlocked in some cases, and because it can
cause seemingly random failures could be related to some bugs in
the tracker...
2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
option to use the source IP address of RTP as the destination IP
address of UDPTL when a specific option is enabled. If the remote
side is properly configured (ports forwarded) then UDPTL will
flow. (closes issue #10417) Reported by: cstadlmann
* channels/chan_sip.c: Fix an issue where codec preferences were
not set on dialogs that were not authenticated via a user or peer
and allow framing to work without rtpmap in the SDP. (closes
issue #12501) Reported by: slimey
2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher@digium.com>
* main/cli.c: When showing an error message for a command, don't
shorten the command output, as it tends to confuse the user (it's
fine for suggesting other commands, however). Reported by:
seanbright (on #asterisk-dev) Fixed by: me
2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Russell noted to me that in the case that
separate threads use their own addressing system, the fix I made
for issue 12376 does not guarantee uniqueness to the datastores'
uids. Though I know of no system that works this way, I am going
to change this right now to prevent trying to track down some
future bug that may occur and cause untold hours of debugging
time to track down. The change involves using a global counter
which increases with each new chanspy_ds which is created. This
guarantees uniqueness.
2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: NULL character should terminate only commands
back to the core, not log messages to the console. (closes issue
#12731) Reported by: seanbright Patches:
20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
Tested by: seanbright
2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel@vanbaak.info>
* apps/app_voicemail.c: small update to the g() option of
app_voicemail to note that gain changes only work on zap channels
right now. issue #12578 shows it's not clear right now.
2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Add a unique id to the datastore allocated in
app_chanspy since it is possible that multiple spies may be
listening to the same channel. (closes issue #12376) Reported by:
DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
(license 60) Tested by: destiny6628 (closes issue #12243)
Reported by: atis
2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher@digium.com>
* configs/queues.conf.sample: Add a note that pbx_config.so is
needed for Local channels. (Closes issue #12671)
2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Realtime flag affects construction in
multiple ways, so consulting whether rtcachefriends was set was
done too soon (needed to be done inside build_peer, not just as a
flag to build_peer). Also, fullcontact needed to be
reconstructed, because realtime separates the embedded ';' into
multiple fields. (closes issue #12722) Reported by: barthpbx
Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
(license 14) Tested by: barthpbx (Much of the discussion happened
on #asterisk-dev for diagnosing this issue)
2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler@digium.com>
* channels/chan_zap.c: Fix a few things I missed to ensure
zt_chan_conf structure is not modified in mkintf
2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/utils.h: Add format type checking for recently
de-inlined function
* doc/cli.txt (added), doc/00README.1st: Add information on using
the Asterisk console, including tab command line completion.
(Closes issue #12681)
2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell@digium.com>
* include/asterisk/utils.h, main/utils.c: Don't declare a function
that takes variable arguments as inline, because it's not valid,
and on some compilers, will emit a warning.
http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
issue #12289) Reported by: francesco_r Patches by Tilghman, final
patch by me
2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Also remove preamble from asynchronous events
(reported by jsmith on #asterisk-dev)
* funcs/func_realtime.c: Take into account the length of delimiters
when calculating result string length. (closes issue #12696)
Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
uploaded by adomjan (license 487)
2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler@digium.com>
* channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
is not modified in mkintf.
2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Apply the autoframing setting to dialogs
that do not get matched against a user or peer.
2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher@digium.com>
* pbx/pbx_spool.c: Revert accidental commit of the last change
* main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
output also when -rx is not being used (Related to issue #12702)
2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell@digium.com>
* main/asterisk.c: Don't filter the magic character in the network
verboser. It gets filtered once it reaches the client. (related
to issue #12702, pointed out by tilghman)
* main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
marker in front of every message from ast_verbose() being sent to
remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
marker. (related to issue #12702)
* main/asterisk.c: Don't display the verbose marker for calls to
ast_verbose() that do not include a VERBOSE_PREFIX in front of
the message. (closes issue #12702) Reported by: johnlange Patched
by me
2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler@digium.com>
* channels/chan_zap.c: Pass a pointer for the conf parameter to the
function mkintf rather than the whole zt_chan_conf structure.
2008-05-20 Russell Bryant <russell@digium.com>
* Asterisk 1.4.20 released.
2008-05-14 Russell Bryant <russell@digium.com>
* Asterisk 1.4.20-rc3 released.
2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Accept text messages even with Content-Type:
text/plain;charset=Södermanländska
2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson@digium.com>
* main/channel.c, include/asterisk/lock.h: A change to the way
channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
After debugging a deadlock, it was noticed that when
DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
of channel locks is obscured by the fact that all channel locks
appear to happen in the function ast_channel_lock(). This code
change redefines ast_channel_lock to be a macro which maps to
__ast_channel_lock(), which then relays the proper file name,
line number, and function name information to the core lock
functions so that this information will be displayed in the case
that there is some sort of locking error or core show locks is
issued.
2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell@digium.com>
* channels/chan_local.c: Fix a deadlock involving channel
autoservice and chan_local that was debugged and fixed by
mmichelson and me. We observed a system that had a bunch of
threads stuck in ast_autoservice_stop(). The reason these threads
were waiting around is because this function waits to ensure that
the channel list in the autoservice thread gets rebuilt before
the stop() function returns. However, the autoservice thread was
also locked, so the autoservice channel list was never getting
rebuilt. The autoservice thread was stuck waiting for the channel
lock on a local channel. However, the local channel was locked by
a thread that was stuck in the autoservice stop function. It
turned out that the issue came down to the local_queue_frame()
function in chan_local. This function assumed that one of the
channels passed in as an argument was locked when called.
However, that was not always the case. There were multiple cases
in which this channel was not locked when the function was
called. We fixed up chan_local to indicate to this function
whether this channel was locked or not. The previous assumption
had caused local_queue_frame() to improperly return with the
channel locked, where it would then never get unlocked. (closes
issue #12584) (related to issue #12603)
* main/autoservice.c: Fix an issue that I noticed in autoservice
while mmichelson and I were debugging a different problem. I
noticed that it was theoretically possible for two threads to
attempt to start the autoservice thread at the same time. This
change makes the process of starting the autoservice thread,
thread-safe.
2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp@digium.com>
* channels/chan_alsa.c: Use the right flag to open the audio in
non-blocking. (closes issue #12616) Reported by:
nicklewisdigiumuser
2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: If the socket dies (read returns 0=EOF), return
immediately. (Closes issue #12637)
2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson@digium.com>
* main/utils.c: If a thread holds no locks, do not print any
information on the thread when issuing a core show locks command.
This will help to de-clutter output somewhat. Russell said it
would be fine to place this improvement in the 1.4 branch, so
that's why it's going here too.
2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac:
Improve res_ninit and res_ndestroy autoconf logic on the Darwin
platform.
2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Remove debug output.
* /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
May 2008) | 25 lines Fix a race condition that bbryant just found
while doing some IAX2 testing. He was running Asterisk trunk
running IAX2 calls through a few Asterisk boxes, however, the
audio was extremely choppy. We looked at a packet trace and saw a
storm of INVAL and VNAK frames being sent from one box to
another. It turned out that what had happened was that one box
tried to send a CONTROL frame before the 3 way handshake had
completed. So, that frame did not include the destination call
number, because it didn't have it yet. Part of our recent work
for security issues included an additional check to ensure that
frames that are supposed to include the destination call number
have the correct one. This caused the frame to be rejected with
an INVAL. The frame would get retransmitted for forever, rejected
every time ... This race condition exists in all versions that
got the security changes, in theory. However, it is really only
likely that this would cause a problem in Asterisk trunk. There
was a control frame being sent (SRCUPDATE) at the _very_
beginning of the call, which does not exist in 1.2 or 1.4.
However, I am fixing all versions that could potentially be
affected by the introduced race condition. These changes are what
bbryant and I came up with to fix the issue. Instead of simply
dropping control frames that get sent before the handshake is
complete, the code attempts to wait a little while, since in most
cases, the handshake will complete very quickly. If it doesn't
complete after yielding for a little while, then the frame gets
dropped. ........
* channels/chan_sip.c: Don't give up on attempting an outbound
registration if we receive a 408 Timeout. (closes issue #12323)
* contrib/scripts/postgres_cdr.sql (removed): remove
postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
well (closes issue #9676)
* contrib/init.d/rc.debian.asterisk: Don't exit the script if
Asterisk is not running. (closes issue #12611)
* main/pbx.c: Don't use a channel before checking for channel
allocation failure. (closes issue #12609) Reported by: edantie
* contrib/init.d/rc.debian.asterisk: Use the same method for
executing Asterisk as the rest of the script. (closes issue
#12611) Reported by: b_plessis
2008-05-07 Russell Bryant <russell@digium.com>
* Asterisk 1.4.20-rc2 released.
2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Track peer references when stored in the
sip_pvt struct as the peer related to a qualify ping or a
subscription. This fixes some realtime related crashes. (closes
issue #12588) (closes issue #12555)
2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker@digium.com>
* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
7 lines read requires an argument on some non-bash shells (closes
issue #12593) Reported by: bkruse Patches:
getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
........
* res/res_musiconhold.c: Switch to using ast_random() rather than
just rand(). This does not fix the bug reported, but I believe it
is correct. (from issue #12446) Patches: bug_12446.diff uploaded
by snuffy (license 35)
2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Don't print the terminating NUL. (Closes issue
#12589)
2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp@digium.com>
* configure, configure.ac: Add in missing argument.
2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, main/logger.c: Separate verbose output from CLI
output, by using a preamble. (closes issue #12402) Reported by:
Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
uploaded by Corydon76 (license 14)
20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
Corydon76 (license 14)
2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
configure.ac: Make sure that either the main speex library
contains preprocess functions or that speexdsp does. If both fail
then speex stuff can not be built.
2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Don't consider a caller "handled" until the
caller is bridged with a queue member. There was too much of an
opportunity for the member to hang up (either during a delay,
announcement, or overly long agi) between the time that he
answered the phone and the time when he actually was bridged with
the caller. The consequence of this was that if the member hung
up in that interval, then proper abandonment details would not be
noted in the queue log if the caller were to hang up at any point
after the member hangup. (closes issue #12561) Reported by:
ablackthorn
2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher@digium.com>
* Makefile: Reverse order, such that user configs override default
selections
* include/asterisk/res_odbc.h: Err, the documentation on the return
value of ast_odbc_backslash_is_escape is exactly backwards.
2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Avoid putting opaque="" in Digest
authentication. This patch came from switchvox. It fixes
authentication with Primus in Canada, and has been in use for a
very long time without causing problems with any other providers.
(closes issue AST-36)
2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.mandrake.asterisk,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.gentoo.asterisk,
contrib/init.d/rc.slackware.asterisk: When starting Asterisk, bug
out if Asterisk is already running. (closes issue #12525)
Reported by: explidous Patches: 20080428__bug12525.diff.txt
uploaded by Corydon76 (license 14) Tested by: mvanbaak
2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp@digium.com>
* configure, acinclude.m4: Expand the test function for GCC
attributes so that more complex attributes are properly
recognized.
* include/asterisk/compiler.h: For my next trick I will make these
work with what our autoconf header file gives us.
* configure, acinclude.m4: Treat warnings as errors when checking
if a GCC attribute exists. We have to do this as GCC will just
ignore the attribute and pop up a warning, it won't actually fail
to compile.
2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant@digium.com>
* channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
configure.ac, CHANGES: Add new "pri show version" command to show
the libpri version for support reasons.
2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson@digium.com>
* include/asterisk/sched.h: Clarify a comment that was, well, just
wrong. It turns out that ignoring the way that macros expand.
Instead, I have clarified in the comment why the macro will work
even if the scheduler id for the task to be deleted changes
during the execution of the macro.
2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/res_odbc.h: Change the comment of deprecated to
an actual compiler deprecation
* main/utils.c: '#' is another reserved character for URIs that
also needs to be escaped. (closes issue #10543) Reported by:
blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
Corydon76 (license 14)
2008-05-01 Russell Bryant <russell@digium.com>
* Asterisk 1.4.20-rc1 released.
2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell@digium.com>
* include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
Merge changes from team/russell/iax2_find_callno and
iax2_find_callno_1.4 These changes address a critical performance
issue introduced in the latest release. The fix for the latest
security issue included a change that made Asterisk randomly
choose call numbers to make them more difficult to guess by
attackers. However, due to some inefficient (this is by far, an
understatement) code, when Asterisk chose high call numbers,
chan_iax2 became unusable after just a small number of calls. On
a small embedded platform, it would not be able to handle a
single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
more than about 16 IAX2 channels. Ouch. These changes address
some performance issues of the find_callno() function that have
bothered me for a very long time. On every incoming media frame,
it iterated through every possible call number trying to find a
matching active call. This involved a mutex lock and unlock for
each call number checked. So, if the random call number chosen
was 20000, then every media frame would cause 20000 locks and
unlocks. Previously, this problem was not as obvious since
Asterisk always chose the lowest call number it could. A second
container for IAX2 pvt structs has been added. It is an astobj2
hash table. When we know the remote side's call number, the pvt
goes into the hash table with a hash value of the remote side's
call number. Then, lookups for incoming media frames are a very
fast hash lookup instead of an absolutely insane array traversal.
In a quick test, I was able to get more than 3600% more IAX2
channels on my machine with these changes.
2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
in Huntsville together with Mark M (putnopvut) (closes issue
#12363) Reported by: jvandal Tested by: putnopvut, oej
2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming@digium.com>
* channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
for indexing through the iaxs/iaxsl arrays so that the size of
the arrays can be adjusted in one place, and change the size of
the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
defined
* Makefile.rules: pay attention to *all* header files for
dependency tracking, not just the local ones (inspired by r578 of
asterisk-addons by tilghman)
2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
variables instead of the channel's macrocontext and macroexten
fields. This is needed because if macros are daisy-chained, the
incorrect context and extension are placed on the new channel. I
also added locking to the channel prior to accessing these
variables as noted in trunk's janitor project file. (closes issue
#12549) Reported by: darren1713 Patches:
app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
(with modifications from me) Tested by: putnopvut
2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker@digium.com>
* res/res_config_pgsql.c: Change warning message to debug, since
there are cases where 0 results is perfectly fine.
2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming@digium.com>
* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
2008) | 2 lines stop script from appending source code if run
multiple times ........
2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking
if a module is loaded. (Closes issue #12534)
2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright@gmail.com>
* configure, configure.ac: When we don't explicitly pass a path to
the --with-tds configure option, we may end up finding tds.h in
/usr/local/include instead of /usr/include. If this happens, the
grep that looks for the version (from tdsver.h) will fail and
we'll have some problems during the build.
2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/vmail.cgi: Clicking forward without selecting a
message leaves an errant .lock file. (closes issue #12528)
Reported by: pukepail Patches: patch.diff uploaded by pukepail
(license 431)
2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Use consistent logic for checking to see if
a call number has been chosen yet. Also, remove some redundant
logic I recently added in a fix.
2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Move the unlock of the spyee channel to
outside the start_spying() function so that the channel is not
unlocked twice when using whisper mode.
2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher@digium.com>
* configs/zapata.conf.sample, configs/iax.conf.sample,
configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
documentation files that actually exist. (closes issue #12516)
Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
by linuxmaniac (license 472)
2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Re-invite RTP during a masquerade so that,
for instance, an AMI redirect of two channels which are natively
bridged will preserve audio on both channels. This prevents a
problem with Asterisk not re-inviting due to one of the channels
having being a zombie. (closes issue #12513) Reported by:
mneuhauser Patches:
asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
mneuhauser (license 425)
* apps/app_queue.c: Output of channel variables when
eventwhencalled=vars was set was being truncated two characters.
This patch corrects the problem. (closes issue #12493) Reported
by: davidw
* channels/chan_local.c: Resolve a deadlock in chan_local by
releasing the channel lock temporarily. (closes issue #11712)
Reported by: callguy Patches: 11712.patch uploaded by putnopvut
(license 60) Tested by: acunningham
2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher@digium.com>
* channels/chan_local.c: Ensure that when we set the accountcode,
it actually shows up in the CDR. (Fix for AMI Originate) (Closes
issue #12007)
2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a silly mistake in a change I made
yesterday that caused chan_iax2 to blow up very quickly. (issue
#12515)
2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Only have one max-forwards header in
outbound REFERs. Discovered in the Asterisk SIP Masterclass in
Orlando. Thanks Joe!
2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell@digium.com>
* main/http.c: Improve some broken cookie parsing code. Previously,
manager login over HTTP would only work if the mansession_id
cookie was first. Now, the code builds a list of all of the
cookies in the Cookie header. This fixes a problem observed by
users of the Asterisk GUI. (closes AST-20)
* apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
the correct next channel to not always work. Also, remove setting
the amount of time to wait for a digit from 5 seconds back down
to 1/10 of a second. I believe this was so the beep didn't get
played over and over really fast, but a while back I put in
another fix for that issue. (closes issue #12498) Reported by:
jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
jsmith (license 15)
2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker@digium.com>
* res/res_musiconhold.c: Fix reload/unload for res_musiconhold
module. (closes issue #11575) Reported by: sunder Patches:
M11575_14_rev3.diff uploaded by junky (license 177)
bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell@digium.com>
* main/manager.c, include/asterisk/manager.h: Store the manager
session ID explicitly as 4 byte ID instead of a ulong. The
mansession_id cookie is coded to be limited to 8 characters of
hex, and this could break logins from 64-bit machines in some
cases. (inspired by AST-20)
* channels/chan_iax2.c: Fix find_callno_locked() to actually return
the callno locked in some more cases.
2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add 502 support for both directions, not
only one... (see r114571)
2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp@digium.com>
* main/pbx.c: Instead of stopping dialplan execution when SayNumber
attempts to say a large number that it can not print out a
message informing the user and continue on. (closes issue #12502)
Reported by: bcnit
2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Treat a 502 just like a 503, when it comes
to processing a response code
2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: When we receive a full frame that is
supposed to contain our call number, ensure that it has the
correct one. (closes issue #10078) (AST-2008-006)
* main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
thought I was going to be able to leave 1.4 alone, but that was
not the case. I ran into some problems with G.722 in 1.4, so I
have merged in all of the fixes in this area that I have made in
trunk/1.6.0, and things are happy again.
* res/res_musiconhold.c: Trivial change to read the number of
samples from a frame before calling ast_write()
* res/res_features.c: After a parked call times out, allow the call
back to the parker to time out. (closes issue #10890)
* channels/chan_iax2.c: If the dial string passed to the call
channel callback does not indicate an extension, then consider
the extension on the channel before falling back to the default.
(closes issue #12479) Reported by: darren1713 Patches:
exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
116)
* channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
team/russell/issue_9520 These changes make sure that the
reference count for sip_peer objects properly reflects the fact
that the peer is sitting in the scheduler for a scheduled
callback for qualifying peers or for expiring registrations.
Without this, it was possible for these callbacks to happen at
the same time that the peer was being destroyed. This was
especially likely to happen with realtime peers, and for people
making use of the realtime prune CLI command. (closes issue
#9520) Reported by: kryptolus Committed patch by me
2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only drop audio if we receive it without a
progress indication. We allow other frames through such as DTMF
because they may be needed to complete the call. (closes issue
#12440) Reported by: aragon
2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher@digium.com>
* apps/app_playback.c: Ensure that help text terminates with a
newline
* res/res_musiconhold.c: MOH usage information needs a terminating
newline, or else "asterisk -rx 'help moh reload'" will hang.
Reported via -dev list, fixed by me.
2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell@digium.com>
* main/manager.c: Don't destroy a manager session if poll() returns
an error of EAGAIN.
* Makefile: ensure directories are created before we try to install
stuff into them
* Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
bininstall
2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson@digium.com>
* channels/chan_zap.c, main/callerid.c: Clearing up error messages
so they make a bit more sense. Also removing a redundant error
message. Issue AST-15
2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell@digium.com>
* channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
(closes issue #12476) Reported by: davidw Patch by me
2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright@gmail.com>
* channels/chan_sip.c: Only complete the SIP channel name once for
'sip show channel <channel>'
2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher@digium.com>
* apps/app_setcallerid.c: For consistency sake, ensure that the
values that ${CALLINGPRES} returns are valid as an input to
SetCallingPres. (Closes issue #12472)
2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell@digium.com>
* main/autoservice.c: Remove redundant safety net. The check for
the autoservice channel list state accomplishes the same goal in
a better way. (issue #12470) Reported By: atis
2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Declaration of the peer channel in this scope
was making it so the peer variable defined in the outer scope was
never set properly, therefore making iterating through the
channel list always restart from the beginning. This bug would
have affected anyone who called chanspy without specifying a
first argument. (closes issue #12461) Reported by: stever28
* main/frame.c, include/asterisk/dsp.h: Add prototype for
ast_dsp_frame_freed. I'm not sure how this was compiling
before...
* main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
possible for a reference to a frame which was part of a freed DSP
to still be referenced, leading to memory corruption and eventual
crashes. This code change ensures that the dsp is freed when we
are finished with the frame. This change is very similar to a
change Russell made with translators back a month or so ago.
(closes issue #11999) Reported by: destiny6628 Patches:
11999.patch uploaded by putnopvut (license 60) Tested by:
destiny6628, victoryure
2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell@digium.com>
* Makefile: Fix the bininstall target to install from subdirs, as
well. (closes issue AST-8, patch from bmd at switchvox)
2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Use keepalives effectively in order diagnose
bug #12432.
2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c: Add special case for when the agi cannot be
executed, to comply with the documentation that we return failure
in that case. (closes issue #12462) Reported by: fmueller
Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
(license 14) Tested by: fmueller
2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright@gmail.com>
* apps/app_chanspy.c: Make sure we have enough room for the
recording's filename.
2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
initializing the structure to all zeroes in case it contains
fields that we don't write values into (which it does as of
Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher@digium.com>
* channels/chan_vpb.cc: Backport revisions for latest vpb drivers
to 1.4 (Closes issue #12457)
2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Fix "fallthrough" behavior here, so config
options in a previously configured user don't override settings
in general. (closes issue #12458) Reported by: tzafrir Patches:
chanzap_users_sections.diff uploaded by tzafrir (license 46)
2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Include the proper headers for using mkdir on
FreeBSD. (closes issue #12430) Reported by: ys Patches:
app_meetme.c.diff uploaded by ys (license 281)
2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Handle subscribe queues in all situations...
Thanks to festr_ on irc for telling me about this bug.
2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker@digium.com>
* contrib/scripts/autosupport: Update Digium autosupport script,
for more useful information. (closes issue #12452) Reported by:
angler Patches: autosupport.diff uploaded by angler (license 106)
* apps/app_queue.c: Allow autofill to work in the general section
of queues.conf. Additionally, don't try to (re)set options when
they have empty values in realtime (all unset columns would have
an empty value). (closes issue #12445) Reported by: atis Patches:
12445-autofill.diff uploaded by qwell (license 4)
* channels/chan_h323.c: The call_token on the pvt can occasionally
be NULL, causing a crash. If it is NULL, we can skip this
channel, since it can't the one we're looking for. (closes issue
#9299) Reported by: vazir
2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Increase the retry count when attempting to show
channels. This apparently cleared an issue someone was seeing
when attempting to show channels when the load was high. (closes
issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
by russell (license 2) Tested by: falves11
* apps/app_dial.c, apps/app_queue.c: If the datastore has been
moved to another channel due to a masquerade, then freeing the
datastore here causes an eventual double free when the new
channel hangs up. We should only free the datastore if we were
able to successfully remove it from the channel we are
referencing (i.e. the datastore was not moved). (closes issue
#12359) Reported by: pguido
* main/channel.c: Save a local copy of the generate callback prior
to unlocking the channel in case the generate callback goes NULL
on us after the channel is unlocked. Thanks to Russell for
pointing this need out to me.
2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: It is possible for the remote side to say
they want T38 but not give any capabilities. (closes issue
#12414) Reported by: MVF
* main/rtp.c: Don't change the SSRC when a new source comes into
play, this might happen quite often and depending on the remote
side... they might not like this. (closes issue #12353) Reported
by: dimas
2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson@digium.com>
* channels/chan_iax2.c: Several places in the code called
find_callno() (which releases the lock on the pvt structure) and
then immediately locked the call and did things with it.
Unfortunately, the call can disappear between the find_callno and
the lock, causing Bad Stuff(tm) to happen. Added
find_callno_locked() function to return the callno withtout
unlocking for instances that it is needed. (issue #12400)
Reported by: ztel
2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker@digium.com>
* main/pbx.c: It's possible that a channel can have an async goto
on the successful execution of an application as well. Closes
issue #12172.
2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson@digium.com>
* res/res_features.c: Fix a race condition that may happen between
a sip hangup and a "core show channel" command. This patch adds
locking to prevent the resulting crash. (closes issue #12155)
Reported by: tsearle Patches: show_channels_crash2.patch uploaded
by tsearle (license 373) Tested by: tsearle
* main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
LOW_MEMORY is enabled.
* channels/chan_sip.c: Be sure that we're not about to set
bridgepvt NULL prior to dereferencing it. (closes issue #11775)
Reported by: fujin
2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker@digium.com>
* main/file.c: Only try to prefix language if we are not using an
absolute path (suffix it otherwise).
en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
uploaded by qwell (license 4) Tested by: kuj, qwell
2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Forgot the 1.4 branch for russian language
fix. (closes issue #12404) Reported by: IgorG Patches:
voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
* apps/app_meetme.c: Create the directory where name recordings
will go if it does not exist. (closes issue #12311) Reported by:
rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
* channels/chan_sip.c: Don't add custom URI options if they don't
exist OR they are empty. (closes issue #12407) Reported by:
homesick Patches: uri_options-1.4.diff uploaded by homesick
(license 91)
2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: We need to set the persistant_route [sic]
parameter for the sip_pvt during the initial INVITE, no matter if
we're building the route set from an INVITE request or response.
(closes issue #12391) Reported by: benjaminbohlmann Tested by:
benjaminbohlmann
2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
not exist in cdr.conf, then an unload/load sequence is needed to
correct the problem. Track whether the load succeeded with a
variable, so we can fix this with a simple reload event, instead.
2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: If we receive an AUTHREQ from the remote
server and we are unable to reply (for example they have a secret
configured, but we do not) then queue a hangup frame on the
Asterisk channel. This will cause the channel to hangup and a
HANGUP to be sent via IAX2 to the remote side which is the proper
thing to do in this scenario. (closes issue #12385) Reported by:
viraptor
2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
a reinvite), then we should not send a BYE. (closes issue #12392)
Reported by: fnordian Patches: chan_sip.patch uploaded by
fnordian (license 110) with small modification from me
2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson@digium.com>
* channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
happy
2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson@digium.com>
* apps/app_parkandannounce.c: Fix potential buffer overflow that
could happen if more than 100 announce files were specified when
calling ParkAndAnnounce. This overflow is not exploitable
remotely and so there is no need for a security advisory. (closes
issue #12386) Reported by: davidw
2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Add a little more that is required for
previously added devices.
* channels/chan_skinny.c: Add support for several new(ish) devices
- most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
for providing me the required information.
* main/asterisk.c: Work around some silliness caused by
sys/capability.h - this should fix compile errors a number of
users have been experiencing.
2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/astgenkey.8: Add security note on astgenkey's
manpage. (closes issue #12373) Reported by: lmamane Patches:
20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
* channels/chan_sip.c: Move check for still-bridged channels out a
little further, to avoid possible deadlocks. (Closes issue
#12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
uploaded by Corydon76 (license 14) Tested by: callguy
2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp@digium.com>
* include/asterisk/slinfactory.h, main/slinfactory.c,
main/audiohook.c: If audio suddenly gets fed into one side of a
channel after a lapse of frames flush the other factory so that
old audio does not remain in the factory causing the sync code to
not execute. (closes issue #12296) Reported by: jvandal
2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
fixes a for loop (in realtime_peer) to check all the
ast_variables the loop was intending to test rather than just the
first one. The change exposed the problem of calling memcpy on a
NULL pointer, in this case the passed in sockaddr_in struct which
is now checked.
2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c, configs/skinny.conf.sample: Allow
playback with noanswer (and add earlyrtp option). (closes issue
#9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
(license 30) Tested by: pj, qwell, DEA, wedhorn
2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher@digium.com>
* funcs/func_strings.c: Force ast_mktime() to check for DST, since
strptime(3) does not. (Closes issue #12374)
2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson@digium.com>
* main/channel.c: This fix prevents a deadlock that was experienced
in chan_local. There was deadlock prevention in place in
chan_local, but it would not work in a specific case because the
channel was recursively locked. By unlocking the channel prior to
calling the generator's generate callback in
ast_read_generator_actions(), we prevent the recursive locking,
and therefore the deadlock. (closes issue #12307) Reported by:
callguy Patches: 12307.patch uploaded by putnopvut (license 60)
Tested by: callguy
2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
Reported by: vinsik Tested by: tecnoxarxa This one line change
makes an if inside a for loop (in realtime_peer) check all the
ast_variables the loop was intending to test rather than just the
first one.
2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Free newly allocated channel before
returning
* channels/chan_gtalk.c: Prevent call connections when codecs don't
match. (closes issue #10604) Reported by: keepitcool Patches:
branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
by: phsultan
2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp@digium.com>
* main/Makefile: Pass in the path to Zaptel for systems that
install Zaptel headers in a separate location.
* main/asterisk.c: One thing at a time... let's get 1.4 building.
2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard@digium.com>
* main/asterisk.c: add a Zaptel timer check to verify the timer is
responding when Zaptel support is compiled into Asterisk and
Zaptel drivers are loaded. This will help people not waste their
valuable time debugging side effects.
2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson@digium.com>
* channels/chan_zap.c: Fix the testing of the "res" variable so
that it is more logically correct and makes the correct warning
and debug messages print. (closes issue #12361) Reported by:
one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
(license 23)
* main/manager.c: Fix a race condition in the manager. It is
possible that a new manager event could be appended during a
brief time when the manager is not waiting for input. If an event
comes during this period, we need to set an indicator that there
is an event pending so that the manager doesn't attempt to wait
forever for an event that already happened. (closes issue #12354)
Reported by: bamby Patches: manager_race_condition.diff uploaded
by bamby (license 430) (comments added by me)
* apps/app_queue.c: Ensure that there is no timeout if none is
specified. (closes issue #12349) Reported by: johnlange
2008-04-01 Russell Bryant <russell@digium.com>
* Asterisk 1.4.19 released.
2008-03-28 Russell Bryant <russell@digium.com>
* Asterisk 1.4.19-rc4 released.
2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker@digium.com>
* formats/format_wav_gsm.c: The file size of WAV49 does not need to
be an even number. (closes issue #12128) Reported by: mdu113
Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
Tested by: qwell, mdu113
2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher@digium.com>
* doc/valgrind.txt: Update debugging text, since Valgrind
eliminated the --log-file-exactly option. (Closes issue #12320)
* main/acl.c: For FreeBSD, at least, the ifa_addr element could be
NULL. (closes issue #12300) Reported by: festr Patches:
acl.c.patch uploaded by festr (license 443)
2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf@digium.com>
* apps/app_playback.c, main/pbx.c: These small documentation
updates made in response to a query in asterisk-users, where a
user was using Playback, but needed the features of Background,
and had no idea that Background existed, or that it might provide
the features he needed. I thought the best way to avert these
kinds of queries was to provide "See Also" references in all
three of "Background", "Playback", "WaitExten". Perhaps a project
to do this with all related apps is in order.
* pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
#12302) Reported by: pj Tested by: murf These changes will set a
channel variable ~~EXTEN~~ just before generating code for a
switch, with the value of ${EXTEN}. The exten is marked as having
a switch, and ever after that, till the end of the exten, we
substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
arguments; (and the ${EXTEN: also). The reason for this, is that
because switches are coded using separate extensions to provide
pattern matching, and jumping to/from these switch extensions
messes up the ${EXTEN} value, which blows the minds of users.
2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker@digium.com>
* main/frame.c: Put this flag back so we don't change the API.
* main/frame.c: Remove excessive smoother optimization that was
causing audio glitches (small "pops") after (about 200ms later)
an "incorrectly" sized frame was received. While it would be very
nice to keep this as optimized as possible, it makes no sense for
the smoother to be dropping random bits of audio like this. Isn't
that the whole point of a smoother? Closes issue #12093.
2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp@digium.com>
* contrib/scripts/autosupport: Update autosupport script. (closes
issue #12310) Reported by: angler Patches: autosupport.diff
uploaded by angler (license 106)
2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming@digium.com>
* /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
2008) | 2 lines update UPGRADE notes to document usage of the
script ........
2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: This code change is made just for
clarification. It does exactly the same thing as before. It just
doesn't look as wrong.
* apps/app_voicemail.c: Add a lock to the vm_state structure and
use the lock around mail_open calls to prevent concurrent access
of the same mailstream. This, along with trunk's ability to
configure TCP timeouts for IMAP storage will help to prevent
crashes and hangs when using voicemail with IMAP storage. (closes
issue #10487) Reported by: ewilhelmsen
2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming@digium.com>
* codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
Merged revisions 111019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
2008) | 2 lines add a script to make getting the iLBC source code
simple for end users ........
2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: If we are requested to authenticate a
reinvite make sure that it contains T38 SDP if need be. (closes
issue #11995) Reported by: fall
* channels/chan_iax2.c: Make sure that full video frames are sent
whenever the 15 bit timestamp rolls over. (closes issue #11923)
Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
mihai (license 94)
2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming@digium.com>
* UPGRADE.txt: add note that the user will need to enable
codec_ilbc to get it to build
* codecs/ilbc/StateConstructW.h (removed),
codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
(removed), codecs/ilbc/getCBvec.c (removed),
codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
(removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
(removed), codecs/ilbc/getCBvec.h (removed),
codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
(removed), codecs/ilbc/FrameClassify.c (removed),
codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
(removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
(removed), codecs/ilbc/anaFilter.c (removed),
codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
(removed), codecs/ilbc/doCPLC.h (removed),
codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
(removed), codecs/ilbc/createCB.h (removed), CHANGES,
codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
(removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
(removed), codecs/ilbc/iCBSearch.h (removed),
codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
(removed), codecs/ilbc/hpOutput.h (removed),
codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
(removed), codecs/ilbc/iCBConstruct.c (removed),
codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
(removed), codecs/ilbc/syntFilter.h (removed),
codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
(removed): Merged revisions 110869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
2008) | 2 lines due to licensing restrictions, we cannot
distribute the source code for iLBC encoding and decoding... so
remove it, and add instructions on how the user can obtain it
themselves ........
2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker@digium.com>
* cdr/cdr_custom.c: Make file access in cdr_custom similar to
cdr_csv. Fixes issue #12268. Patch borrowed from r82344
2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler@digium.com>
* channels/chan_sip.c: This one line change makes an if inside a
for loop (in realtime_peer) check all the ast_variables the loop
was intending to test rather than just the first one.
2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: When reverting a commit, I accidentally left
in this bit which was an experiment to see what would happen. It
passed the compile test, and I didn't notice I had left this
change in too. So this is a revert of a revert...sort of.
2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp@digium.com>
* include/asterisk/options.h, main/asterisk.c, Makefile,
main/app.c: Add an option (transmit_silence) which transmits
silence during both Record() and DTMF generation. The reason this
is an option is that in order to transmit silence we have to
setup a translation path. This may not be needed/wanted in all
cases. (closes issue #10058) Reported by: tracinet
2008-03-24 19:17 +0000 [r110618] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: This is a revert for revision 108288. The
reason is that that revision was not for an actual bug fix per
se, and so it really should not have been in 1.4 in the first
place. Plus, people who compile with DO_CRASH are more likely to
encounter a crash due to this change. While I think the usage of
DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
beyond the scope of 1.4 and should be done instead in a developer
branch based on trunk so that all scheduler functions are fixed
at once. I also am reverting the change to trunk and 1.6 since
they also suffer from the DO_CRASH potential. (closes issue
#12272) Reported by: qq12345
2008-03-24 17:34 +0000 [r110614] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.
2008-03-21 14:32 +0000 [r110474] Jason Parker <jparker@digium.com>
* codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
mips platforms either. (closes issue #12270) Reported by:
zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
(license 33)
2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant <russell@digium.com>
* main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
to 50 ms in the autoservice thread. This really should not make a
difference except in very rare cases. That case would be that all
of the channels in autoservice are not generating any frames. In
that case, this change reduces the potential amount of time that
a thread waits in ast_autoservice_stop() for the autoservice
thread to wrap back around to the beginning of its loop. (closes
issue #12266, reported by dimas)
* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
110335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
| 6 lines Fix some very broken code that was introduced in 1.2.26
as a part of the security fix. The dnsmgr is not appropriate
here. The dnsmgr takes a pointer to an address structure that a
background thread continuously updates. However, in these cases,
a stack variable was passed. That means that the dnsmgr thread
would be continuously writing to bogus memory. ........
* apps/app_meetme.c: Fix a bug where when calls on the trunk side
hang up while on hold, the state is not properly reflected.
(closes issue #11990, reported by anakaoka, patched by me)
2008-03-19 20:33 +0000 [r110083] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Add a missing unlock in the case that memory
allocation fails in app_chanspy. Thanks to Russell for confirming
that this was an issue.
2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp <jcolp@digium.com>
* res/res_musiconhold.c: Add sanity checking for position resuming.
We *have* to make sure that the position does not exceed the
total number of files present, and we have to make sure that the
position's filename is the same as previous. These values can
change if a music class is reloaded and give unpredictable
behavior. (closes issue #11663) Reported by: junky
* main/rtp.c: Make sure that the mark bit does not incorrectly
cause video frame timestamps to be calculated as if they are
audio frames. (closes issue #11429) Reported by: sperreault
Patches: 11429-frametype.diff uploaded by qwell (license 4)
2008-03-19 17:12 +0000 [r109973] Jason Parker <jparker@digium.com>
* Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
(added): People report bugs about Asterisk crashing with DO_CRASH
enabled was getting a little silly... Now we only show certain
cflags when you run configure with --enable-dev-mode
(corresponding menuselect change to follow)
2008-03-19 15:41 +0000 [r109908] Steve Murphy <murf@digium.com>
* main/config.c: (closes issue #11442) Reported by: tzafrir
Patches: 11442.patch uploaded by murf (license 17) Tested by:
murf I didn't give tzafrir very much time to test this, but if he
does still have remaining issues, he is welcome to re-open this
bug, and we'll do what is called for. I reproduced the problem,
and tested the fix, so I hope I am not jumping by just going
ahead and committing the fix. The problem was with what file_save
does with templates; firstly, it tended to print out multiple
options: [my_category](!)(templateref) instead of
[my_category](!,templateref) which is fixed by this patch.
Nextly, the code to suppress output of duplicate declarations
that would occur because the reader copies inherited declarations
down the hierarchy, was not working. Thus: [master-template](!)
mastervar = bar [template](!,master-template) tvar = value
[cat](template) catvar = val would be rewritten as: ;! ;!
Automatically generated configuration file ;! Filename:
experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
[master-template](!) mastervar = bar
[template](!,master-template) mastervar = bar tvar = value
[cat](template) mastervar = bar tvar = value catvar = val This
has been fixed. Since the config reader 'explodes' inherited vars
into the category, users may, in certain circumstances, see
output different from what they originally entered, but it should
be both correct and equivalent.
2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant <russell@digium.com>
* main/utils.c: Tweak spacing in a recent change because I'm very
picky.
* apps/app_chanspy.c: Fix one place where the chanspy datastore
isn't removed from a channel. (issue #12243, reported by atis,
patch by me)
2008-03-18 20:52 +0000 [r109713] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This patch makes it so that all queue member
status changes are handled through device state code. This
removes several problems people were seeing where their queue
members would get into an "unknown" state. Huge props go to atis
on this one since he was the one who found the code section that
was causing the problem and proposed the solution. I just wrote
what he suggested :) (closes issue #12127) Reported by: atis
Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
by: atis, jvandal
2008-03-18 19:23 +0000 [r109648] Jason Parker <jparker@digium.com>
* codecs/log2comp.h: Allow codecs that use log2comp (g726) to
compile correctly on x86 with gcc4 optimizations. (closes issue
#12253) Reported by: fossil Patches: log2comp.patch uploaded by
fossil (license 140)
2008-03-18 17:58 +0000 [r109575] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Make sure an agent doesn't try to send
dtmf to a NULL channel closes issue #12242 Reported by Yourname
2008-03-18 Russell Bryant <russell@digium.com>
* Asterisk 1.4.19-rc3 released.
2008-03-18 16:25 +0000 [r109482] Terry Wilson <twilson@digium.com>
* include/asterisk/astobj.h: Fix character string being treated ad
format string
2008-03-18 15:10 +0000 [r109393] Jason Parker <jparker@digium.com>
* /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
3 lines Do not return with a successful authentication if the
From header ends up empty. (AST-2008-003) ........
2008-03-18 14:58 +0000 [r109386] Joshua Colp <jcolp@digium.com>
* main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
number of payloads accepted, and also make sure a given payload
does not exceed our maximum value. (AST-2008-002)
2008-03-18 06:37 +0000 [r109309] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ael-ntest23 (added),
pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
pbx/ael/ael-test/ael-ntest23/qq.ael (added),
pbx/ael/ael-test/ael-ntest23/t1 (added),
pbx/ael/ael-test/ael-ntest23/t2 (added),
pbx/ael/ael-test/ael-ntest23/t3 (added),
pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
issue #11903) Reported by: atis Many thanks to atis for spotting
this problem and reporting it. The fix was to straighten out how
items are placed on and removed from the file stack. Regressions
as well as the provided test case helped to straighten out all
code paths. valgrind was used to make sure all memory allocated
was freed. Sorry for not solving this earlier. I got distracted.
Added the ntest23 regression test, which is mainly a copy of
ntest22, but with a few juicy errors thrown in, to replicate the
kind of error that atis spotted.
2008-03-17 22:05 +0000 [r109226] Mark Michelson <mmichelson@digium.com>
* main/utils.c: Fix a logic flaw in the code that stores lock info
which is displayed via the "core show locks" command. The idea
behind this section of code was to remove the previous lock from
the list if it was a trylock that had failed. Unfortunately,
instead of checking the status of the previous lock, we were
referencing the index immediately following the previous lock in
the lock_info->locks array. The result of this problem, under the
right circumstances, was that the lock which we currently in the
process of attempting to acquire could "overwrite" the previous
lock which was acquired. While this does not in any way affect
typical operation, it *could* lead to misleading "core show
locks" output.
2008-03-17 17:55 +0000 [r109171] Michiel van Baak <michiel@vanbaak.info>
* channels/chan_skinny.c: Update the directory of placed calls on
skinny phones when dialing a channel that does not provide
progress (analog ZAP lines) The phone does handle the double
update on calls to channels that do provide progress and wont
insert duplicate items (closes issue #12239) Reported by: DEA
Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)
2008-03-17 16:24 +0000 [r109107] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: 200 OKs in response to a reinvite need to be
sent reliably. If the remote side does not receive one the dialog
will be torn down. (closes issue #12208) Reported by: atrash
2008-03-17 15:15 +0000 [r109057] Jason Parker <jparker@digium.com>
* main/file.c: Backport revision 106439 from trunk. I didn't
realize this was broken in 1.4 as well. Closes issue #12222.
2008-03-17 14:18 +0000 [r109012] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c: Make sure that we release the lock on the
spyee channel if the spyee or spy has hung up (closes issue
#12232) Reported by: atis
2008-03-16 21:47 +0000 [r108961] Michiel van Baak <michiel@vanbaak.info>
* main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
(closes issue #12228) Reported by: andrew Patches: SRC.patch
uploaded by andrew (license 240)
2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant <russell@digium.com>
* channels/chan_oss.c: Fix a channel name issue. chan_oss registers
the "Console" channel type, but it created channels with an "OSS"
prefix. (closes issue #12194, reported by davidw, patched by me)
* contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
start networking before asterisk, as well. (closes issue #12200,
reported by and change suggested by reinerotto)
2008-03-14 16:44 +0000 [r108737] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a race condition in the SIP packet
scheduler which could cause a crash. chan_sip uses the scheduler
API in order to schedule retransmission of reliable packets (such
as INVITES). If a retransmission of a packet is occurring, then
the packet is removed from the scheduler and retrans_pkt is
called. Meanwhile, if a response is received from the packet as
previously transmitted, then when we ACK the response, we will
remove the packet from the scheduler and free the packet. The
problem is that both the ACK function and retrans_pkt attempt to
acquire the same lock at the beginning of the function call. This
means that if the ACK function acquires the lock first, then it
will free the packet which retrans_pkt is about to read from and
write to. The result is a crash. The solution: 1. If the ACK
function fails to remove the packet from the scheduler and the
retransmit id of the packet is not -1 (meaning that we have not
reached the maximum number of retransmissions) then release the
lock and yield so that retrans_pkt may acquire the lock and
operate. 2. Make absolutely certain that the ACK function does
not recursively lock the lock in question. If it does, then
releasing the lock will do no good, since retrans_pkt will still
be unable to acquire the lock. (closes issue #12098) Reported by:
wegbert (closes issue #12089) Reported by: PTorres Patches:
12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
by: jvandal
2008-03-14 14:29 +0000 [r108682] Jason Parker <jparker@digium.com>
* res/res_musiconhold.c: Fix a potential segfault if chan (or
chan->music_state) is NULL. Closes issue #12210, credit to
edantie for pointing this out.
2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
Fix another issue that was causing crashes in chanspy. This
introduces a new datastore callback, called chan_fixup(). The
concept is exactly like the fixup callback that is used in the
channel technology interface. This callback gets called when the
owning channel changes due to a masquerade. Before this was
introduced, if a masquerade happened on a channel being spyed on,
the channel pointer in the datastore became invalid. (closes
issue #12187) (reported by, and lots of testing from atis) (props
to file for the help with ideas)
* channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
phones to blink when an extension that has been subscribed to
goes on hold. Otherwise, they just stay on like it does when an
extension is in use. (closes issue #11263) Reported by: russell
Patches: notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell
* apps/app_followme.c: Fix a couple uses of sprintf. The second one
could actually cause an overflow of a stack buffer. It's not a
security issue though, it only depends on your configuration.
2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
for autocongestion in chan_sip. The scheduler callback will
always return 0. This means that this id is never rescheduled, so
it makes no sense to loop trying to delete the id from the
scheduler queue. If we fail to remove the item from the queue
once, it will fail every single time. (Yes I realize that in this
case, the macro would exit early because the id is set to -1 in
the callback, but it still makes no sense to use that macro in
favor of calling ast_sched_del once and being done with it) This
is the first of potentially several such fixes.
* include/asterisk/sched.h: Added a large comment before the
AST_SCHED_DEL macro to explain its purpose as well as when it is
appropriate and when it is not appropriate to use it. I also
removed the part of the debug message that mentions that this is
probably a bug because there are some perfectly legitimate places
where ast_sched_del may fail to delete an entry (e.g. when the
scheduler callback manually reschedules with a new id instead of
returning non-zero to tell the scheduler to reschedule with the
same idea). I also raised the debug level of the debug message in
AST_SCHED_DEL since it seems like it could come up quite
frequently since the macro is probably being used in several
places where it shouldn't be. Also removed the redundant line,
file, and function information since that is provided by ast_log.
2008-03-12 19:57 +0000 [r108135] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c, main/channel.c: (closes issue #12187,
reported by atis, fixed by me after some brainstorming on the
issue with mmichelson) - Update copyright info on app_chanspy. -
Fix a race condition that caused app_chanspy to crash. The issue
was that the chanspy datastore magic that was used to ensure that
spyee channels did not disappear out from under the code did not
completely solve the problem. It was actually possible for
chanspy to acquire a channel reference out of its datastore to a
channel that was in the middle of being destroyed. That was
because datastore destruction in ast_channel_free() was done near
the end. So, this left the code in app_chanspy accessing a
channel that was partially, or completely invalid because it was
in the process of being free'd by another thread. The following
sort of shows the code path where the race occurred:
=============================================================================
Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
--------------------------------------||-------------------------------------
ast_channel_free() || - remove channel from channel list || -
lock/unlock the channel to ensure || that no references retrieved
from || the channel list exist. ||
--------------------------------------||-------------------------------------
|| channel_spy() - destroy some channel data || - Lock chanspy
datastore || - Retrieve reference to channel || - lock channel ||
- Unlock chanspy datastore
--------------------------------------||-------------------------------------
- destroy channel datastores || - call chanspy datastore d'tor ||
which NULL's out the ds' || - Operate on the channel ...
reference to the channel || || - free the channel || || || -
unlock the channel
--------------------------------------||-------------------------------------
=============================================================================
2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: if we receive an INVITE with a
Content-Length that is not a valid number, or is zero, then don't
process the rest of the message body looking for an SDP closes
issue #11475 Reported by: andrebarbosa
2008-03-12 18:26 +0000 [r108083] Joshua Colp <jcolp@digium.com>
* apps/app_mixmonitor.c, include/asterisk/audiohook.h,
main/audiohook.c: Add a trigger mode that triggers on both read
and write. The actual function that returns the combined audio
frame though will wait until both sides have fed in audio, or
until one side stops (such as the case when you call Wait).
(closes issue #11945) Reported by: xheliox
2008-03-12 16:59 +0000 [r108031] Russell Bryant <russell@digium.com>
* main/channel.c: Destroy the channel lock after the channel
datastores. (inspired by issue #12187)
2008-03-12 01:52 +0000 [r107877] Tilghman Lesher <tlesher@digium.com>
* contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
Document all of the possible realtime fields
2008-03-11 23:37 +0000 [r107714-107826] Jason Parker <jparker@digium.com>
* doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
ODBC voicemail. (closes issue #12186) Reported by: jsmith
Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
15)
* channels/chan_gtalk.c: Copy voicemail dependency logic for
res_adsi to chan_gtalk (for jabber). (closes issue #12014)
Reported by: junky
2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming <kpfleming@digium.com>
* Makefile.rules, channels/Makefile: get chan_vpb to build properly
in dev mode
2008-03-11 20:47 +0000 [r107712] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Add a newline on a log
2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp <jcolp@digium.com>
* res/res_features.c: Make sure the visible indication is on the
right channel so when the masquerade happens the proper
indication is enacted. (closes issue #11707) Reported by: iam
* apps/app_meetme.c: Add an additional check for setting conference
parameter when using the marked user options. It was possible for
it to return to a no listen/no talk state if a masquerade
happened. (closes issue #12136) Reported by: aragon
* apps/app_exec.c: Fix a minor spelling error. (closes issue
#12183) Reported by: darrylc
2008-03-11 Russell Bryant <russell@digium.com>
* Asterisk 1.4.19-rc2 released.
2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_rpt.c: backport a fix from trunk
* channels/misdn/isdn_lib.c, codecs/Makefile,
channels/chan_misdn.c: fix various other problems found by gcc
4.3
* configure, include/asterisk/autoconfig.h.in, configure.ac,
apps/app_sms.c: stop checking for mktime() in the configure
script... we don't use it, and the test is buggy under gcc 4.3
* configure, main/Makefile, configure.ac, makeopts.in: check for
compiler support for -fno-strict-overflow before using it (tested
with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
Reported by: Netview
* configure, configure.ac: fix small bug in IMAP toolkit testing
* main/udptl.c, utils/Makefile, main/Makefile,
main/editline/readline.c, pbx/Makefile: fix up various compiler
warnings found with gcc-4.3: - the output of flex includes a
static function called 'input' that is not used, so for the
moment we'll stop having the compiler tell us about unused
variables in the flex source files (a better fix would be to
improve our flex post-processing to remove the unused function) -
main/stdtime/localtime.c makes assumptions about signed integer
overflow, and gcc-4.3's improved optimizer tries to take
advantage of handling potential overflow conditions at compile
time; for now, suppress these optimizations until we can fiure
out if the code needs improvement - main/udptl.c has some
references to uninitialized variables; in one case there was no
bug, but in the other it was certainly possibly for unexpected
behavior to occur - main/editline/readline.c had an unused
variable
2008-03-11 00:59 +0000 [r107290] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c: If we fail to alloc a channel, we should
re-lock the pvt structure before returning.
2008-03-10 21:32 +0000 [r107230] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: Use non-global storage for eswitch
2008-03-10 20:27 +0000 [r107173] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Make sure to reenable echo can after a
"failed" (canceled, etc) three-way call. (closes issue #11335)
Reported by: rebuild
2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant <russell@digium.com>
* main/pbx.c: Fix another bug specifically related to asynchronous
call origination. Once the PBX is started on the channel using
ast_pbx_start(), then the ownership of the channel has been
passed on to another thread. We can no longer access it in this
code. If the channel gets hung up very quickly, it is possible
that we could access a channel that has been free'd. (inspired by
BE-386)
* main/pbx.c: Fix some bugs related to originating calls. If the
code failed to start a PBX on the channel (such as if you set a
call limit based on the system's load average), then there were
cases where a channel that has already been free'd using
ast_hangup() got accessed. This caused weird memory corruption
and crashes to occur. (fixes issue BE-386) (much debugging credit
goes to twilson, final patch written by me)
* main/channel.c: Resolve a compiler warning.
* main/channel.c: Fix a race condition where the generator can go
away (closes issue #12175, reported by edantie, patched by me)
2008-03-10 14:33 +0000 [r107016] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
unanswered CDRs are dropped to the CDR core, not everything uses
app_dial. (closes issue #11516) Reported by: ys Patches:
branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
anest, jcapp, dartvader
2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: don't generate D-Channel "up" and "down"
messages unless the channel state is actually changing; also,
generate the "up" message when an implicit "up" occurs due to
reception of a normal event when we thought the channel was
"down"
2008-03-07 22:51 +0000 [r106895] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Only start the SLA thread if SLA has actually
been configured.
2008-03-07 22:14 +0000 [r106842] Jason Parker <jparker@digium.com>
* main/editline/Makefile.in: Fix hardcoded grep in editline, were
GNU grep is required. (closes issue #12124) Reported by: dmartin
2008-03-07 19:32 +0000 [r106788] Joshua Colp <jcolp@digium.com>
* main/channel.c: Ignore source update control frame. (closes issue
#12168) Reported by: plack
2008-03-07 17:16 +0000 [r106704] Russell Bryant <russell@digium.com>
* include/asterisk/sched.h: Change a warning message to a debug
message. This is happening quite frequently, and it is not worth
spamming users with these messages unless we are pretty confident
that it should never happen. As it stands today, it _will_ and
_does_ happen and until that gets cleaned up a reasonable amount
on the development side, let's not spam the logs of everyone
else. (closes issue #12154)
2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Warn the user when a temporary greeting
exists (Closes issue #11409)
* main/rtp.c: Properly initialize rtp->schedid (Closes issue
#12154)
* apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
main/manager.c: Safely use the strncat() function. (closes issue
#11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
uploaded by Corydon76 (license 14)
2008-03-06 22:10 +0000 [r106437] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: Quell an annoying message that is likely to print
every single time that ast_pbx_outgoing_app is called. The reason
is that __ast_request_and_dial allocates the cdr for the channel,
so it should be expected that the channel will have a cdr on it.
Thanks to joetester on IRC for pointing this out
2008-03-06 04:40 +0000 [r106328] Tilghman Lesher <tlesher@digium.com>
* sounds/Makefile: Upgrade to the next release of sounds
2008-03-05 22:37 +0000 [r106237] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a potential deadlock and a few
different potential crashes. (closes issue #12145, reported by
thiagarcia, patched by me)
2008-03-05 22:32 +0000 [r106235] Joshua Colp <jcolp@digium.com>
* channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
apps/app_dial.c, main/channel.c, channels/chan_phone.c,
main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
channels/chan_alsa.c, apps/app_followme.c,
include/asterisk/frame.h: Add a control frame to indicate the
source of media has changed. Depending on the underlying
technology it may need to change some things. (closes issue
#12148) Reported by: jcomellas
2008-03-05 21:12 +0000 [r106178] Michiel van Baak <michiel@vanbaak.info>
* doc/realtime.txt: document var_metric so no bugreports will come
in when it's actually a configuration issue. (issue #12151)
Reported and patched by: caio1982 1.4 patch by me
2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: when a PRI call must be moved to a different
B channel at the request of the other endpoint, ensure that any
DSP active on the original channel is moved to the new one
(closes issue #11917) Reported by: mavetju Tested by: mavetju
2008-03-05 15:17 +0000 [r106015] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, include/asterisk/sched.h: Correctly
initialize retransid in SIP, and ensure that the warning when
failing to delete a schedule entry can actually hit the log.
(closes issue #12140) Reported by: slavon Patches: sch2.patch
uploaded by slavon (license 288) (Patch slightly modified by me)
2008-03-05 01:52 +0000 [r105932] Russell Bryant <russell@digium.com>
* main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
that I just noticed in the RTP code. The calculation for setting
the len field in an ast_frame of audio was wrong when G.722 is in
use. The len field represents the number of ms of audio that the
frame contains. It would have set the value to be twice what it
should be.
2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp <jcolp@digium.com>
* main/rtp.c: In addition to setting the marker bit let's change
our ssrc so they know for sure it is a different source.
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
new source of audio comes in (such as music on hold) make sure
the marker bit gets set. (closes issue #10355) Reported by:
wdecarne Patches: 10355.diff uploaded by file (license 11)
(closes issue #11491) Reported by: kanderson
2008-03-04 Russell Bryant <russell@digium.com>
* Asterisk 1.4.19-rc1 released.
2008-03-04 04:31 +0000 [r105591] Russell Bryant <russell@digium.com>
* main/pbx.c: Backport a minor bug fix from trunk that I found
while doing random code cleanup. Properly break out of the loop
when a context isn't found when verify that includes are valid.
2008-03-03 18:06 +0000 [r105572] Jason Parker <jparker@digium.com>
* res/snmp/agent.c: Fix type for astNumChannels. (closes issue
#12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
(license 192)
2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant <russell@digium.com>
* channels/chan_local.c: In the case of an ast_channel allocation
failure, take the local_pvt out of the pvt list before destroying
it.
* channels/chan_local.c: Fix a potential memory leak of the
local_pvt struct when ast_channel allocation fails. Also, in
passing, centralize the code necessary to destroy a local_pvt.
* main/autoservice.c: Update the copyright information for
autoservice. Most of the code in this file now is stuff that I
have written recently ...
* main/asterisk.c, main/channel.c, include/asterisk.h,
main/autoservice.c: Merge in some changes from
team/russell/autoservice-nochans-1.4 These changes fix up some
dubious code that I came across while auditing what happens in
the autoservice thread when there are no channels currently in
autoservice. 1) Change it so that autoservice thread doesn't keep
looping around calling ast_waitfor_n() on 0 channels twice a
second. Instead, use a thread condition so that the thread
properly goes to sleep and does not wake up until a channel is
put into autoservice. This actually fixes an interesting bug, as
well. If the autoservice thread is already running (almost always
is the case), then when the thread goes from having 0 channels to
have 1 channel to autoservice, that channel would have to wait
for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets
called with no channels and no fds to poll() on, such as was the
case with the previous code for the autoservice thread. In this
case, the code would call alloca(0), and pass the result as the
first argument to poll(). In this case, the 2nd argument to
poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it
explicit and ensures the pointers are NULL unless we have valid
data to put there. (related to issue #12116)
2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp <jcolp@digium.com>
* main/channel.c: It is possible for no audio to pass between the
current digit and next digit so expand logic that clears
emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
edgreenberg Patches: v1-11911.patch uploaded by dimas (license
88) Tested by: tbsky
* channels/chan_sip.c: Add a comment to describe some logic.
(closes issue #12120) Reported by: flefoll Patches:
chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
244)
2008-02-29 23:34 +0000 [r105409] Russell Bryant <russell@digium.com>
* main/autoservice.c: Fix a major bug in autoservice. There was a
race condition in the handling of the list of channels in
autoservice. The problem was that it was possible for a channel
to get removed from autoservice and destroyed, while the
autoservice thread was still messing with the channel. This led
to memory corruption, and caused crashes. This explains multiple
backtraces I have seen that have references to autoservice, but
do to the nature of the issue (memory corruption), could cause
crashes in a number of areas. (fixes the crash in BE-386) (closes
issue #11694) (closes issue #11940) The following issues could be
related. If you are the reporter of one of these, please update
to include this fix and try again. (potentially fixes issue
#11189) (potentially fixes issue #12107) (potentially fixes issue
#11573) (potentially fixes issue #12008) (potentially fixes issue
#11189) (potentially fixes issue #11993) (potentially fixes issue
#11791)
2008-02-29 14:47 +0000 [r105326] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Fix a potential memory leak
2008-02-29 14:34 +0000 [r105296] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: If the message file does not exist, just
return harmlessly, instead of crashing. (Closes issue #12108)
2008-02-29 13:48 +0000 [r105261] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Bump up the size of the uniqueid variable.
(closes issue #12107) Reported by: asgaroth
2008-02-29 13:05 +0000 [r105209] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Automatically create new buddy upon reception
of a presence stanza of type subscribed. (closes issue #12066)
Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
(license 73) Tested by: ffadaie, phsultan
2008-02-28 22:23 +0000 [r105116] Russell Bryant <russell@digium.com>
* main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
tracking code that was discovered by mmichelson. The issue is
that if the lock history array was full, then the functions to
mark a lock as acquired or not would adjust the stats for
whatever lock is at the end of the array, which may not be
itself. So, do a sanity check to make sure that we're updating
lock info for the proper lock. (This explains the bizarre stats
on lock #63 in BE-396, thanks Mark!)
2008-02-28 21:56 +0000 [r105113] Tilghman Lesher <tlesher@digium.com>
* contrib/init.d/rc.debian.asterisk: Update init script for LSB
compat (closes issue #9843) Reported by: ibc Patches:
rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
paravoid
2008-02-28 20:11 +0000 [r105059] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: When using autofill, members who are in use
should be counted towards the number of available members to call
if ringinuse is set to yes. Thanks to jmls who brought this issue
up on IRC
2008-02-28 19:20 +0000 [r104920-105005] Jason Parker <jparker@digium.com>
* main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
applications, if we get NULL. This protects against possible
segfaults in applications that may try to use data before
checking length (ast_strdupa'ing it, for example) (closes issue
#12100) Reported by: foxfire Patches: 12100-nullappargs.diff
uploaded by qwell (license 4)
* channels/chan_skinny.c: According to a video at www.cisco.com,
the 7921G supports 6 line appearances.
2008-02-28 00:05 +0000 [r104868] Tilghman Lesher <tlesher@digium.com>
* main/Makefile, build_tools/strip_nonapi: Compatibility fix for
PPC64 (closes issue #12081) Reported by: jcollie Patches:
asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
Tested by: jcollie, Corydon76
2008-02-27 21:49 +0000 [r104841] Mark Michelson <mmichelson@digium.com>
* main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
lockable list. This is because in some cases, the ast_dial may
exist in multiple threads due to asynchronous execution of its
application, and I found some cases where race conditions could
exist. 2. Check in ast_dial_join to be sure that the channel
still exists before attempting to lock it, since it could have
gotten hung up but the is_running_app flag on the
ast_dial_channel may not have been cleared yet. (closes issue
#12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
putnopvut (license 60) Tested by: jvandal
2008-02-27 20:56 +0000 [r104787] Joshua Colp <jcolp@digium.com>
* apps/app_chanspy.c: Don't loop around infinitely trying to spy on
our own channel, and don't forget to free/detach the datastore
upon hangup of the spy.
2008-02-27 20:36 +0000 [r104783] Mark Michelson <mmichelson@digium.com>
* main/file.c: Bump a couple of more buffers up by 2 so that
annoying warnings aren't generated like crazy on every
fileexists_core call.
2008-02-27 18:15 +0000 [r104704] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Ensure the session ID can't be 0.
2008-02-27 17:41 +0000 [r104665] Joshua Colp <jcolp@digium.com>
* main/file.c: Bump up the buffer by 2.
2008-02-27 17:33 +0000 [r104625] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
stuck in an infinite loop without being able to detect that the
calling channel hung up. (closes issue #12076, reported by junky,
patched by me)
2008-02-27 17:26 +0000 [r104598] Jason Parker <jparker@digium.com>
* res/res_features.c: Inherit language from the transfering channel
on a blind transfer. (closes issue #11682) Reported by: caio1982
Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
(license 22) Tested by: caio1982, victoryure
2008-02-27 17:07 +0000 [r104596] Joshua Colp <jcolp@digium.com>
* main/loader.c: Use the lock (which already existed, it just
wasn't used) on the updaters list to protect the contents instead
of the overall module list lock. (closes issue #12080) Reported
by: ChaseVenters
2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming <kpfleming@digium.com>
* main/file.c: fallback to standard English prompts properly when
using new prompt directory layout (closes issue #11831) Reported
by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
20) (modified by me to improve code and conform rest of function
to coding guidelines)
2008-02-27 16:45 +0000 [r104591] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: When we receive a known alarm, make sure
that the unknown alarm flag is not still set to make sure that
when we come back out of alarm, it gets reported in the log and
manager interface (after discussion with tzafrir on the -dev
list)
2008-02-27 15:52 +0000 [r104536] Joshua Colp <jcolp@digium.com>
* res/res_smdi.c: Only stop the MWI monitor thread if it was
actually started. (closes issue #12086) Reported by: francesco_r
2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c: Avoid some recursion in the cleanup code for
the chanspy datastore (closes issue #12076, reported by junky,
patched by me)
* channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
an alarm. However, Asterisk 1.4 does not know what to do with
these alarms. Only Asterisk 1.6 cares about it. So, if we get an
unknown alarm in chan_zap, don't generate confusing log messages
about it.
2008-02-26 18:26 +0000 [r104132-104141] Jason Parker <jparker@digium.com>
* Makefile: Add badshell to .PHONY target (thanks Kevin)
* Makefile: Since all shells aren't as awesome as bash, we have to
fail if somebody tries to use a literal "~" in DESTDIR.
* sounds/Makefile: Revert previous abspath change. ...abspath is
new in GNU make 3.81. I feel so...defeated. Must find new fix!
* sounds/Makefile: Fix a very bizarre issue we were seeing with our
buildbot when using a DESTDIR that wasn't an absolute path (such
as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
that some of the targets were being expanded to the full path, so
$@ ended up being /root/asterisk-1.4/[...]/ rather than
~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
in GNU make. (*cough*
http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
2008-02-26 00:25 +0000 [r104119] Russell Bryant <russell@digium.com>
* include/asterisk/smdi.h, apps/app_voicemail.c,
channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
Merge changes from team/russell/smdi-1.4 This commit brings in a
significant set of changes to the SMDI support in Asterisk. There
were a number of bugs in the current implementation, most notably
being that it was very likely on busy systems to pop off the
wrong message from the SMDI message queue. So, this set of
changes fixes the issues discovered as well as introducing some
new ways to use the SMDI support which are required to avoid the
bugs with grabbing the wrong message off of the queue. This code
introduces a new interface to SMDI, with two dialplan functions.
First, you get an SMDI message in the dialplan using
SMDI_MSG_RETRIEVE() and then you access details in the message
using the SMDI_MSG() function. A side benefit of this is that it
now supports more than just chan_zap. For example, with this
implementation, you can have some FXO lines being terminated on a
SIP gateway, but the SMDI link in Asterisk. Another issue with
the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the
same as the Asterisk voicemail box. There are now additional
directives in the smdi.conf configuration file which let you map
SMDI station IDs to Asterisk voicemail boxes. Yet another issue
with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change
when the change was made by someone calling into voicemail. If
the change was made by some other entity (such as with IMAP
storage, or with a web interface of some kind), then the MWI
change would never be sent. The SMDI module can now poll for MWI
changes if configured to do so. This work was inspired by and
primarily done for the University of Pennsylvania. (also related
to issue #9260)
2008-02-26 00:03 +0000 [r104111] Jason Parker <jparker@digium.com>
* channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
(closes issue #12050) Reported by: asgaroth Patches: 12050.patch
uploaded by putnopvut (license 60)
2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c: This patch fixes some pretty significant
problems with how app_chanspy handles pointers to channels that
are being spied upon. It was very likely that a crash would occur
if the channel being spied upon hung up. This was because the
current ast_channel handling _requires_ that the object is locked
or else it could disappear at any time (except in the owning
channel thread). So, this patch uses some channel datastore magic
on the spied upon channel to be able to detect if and when the
channel goes away. (closes issue #11877) (patch written by me,
but thanks to kpfleming for the idea, and to file for review)
* main/utils.c: Improve the lock tracking code a bit so that a
bunch of old locks that threads failed to lock don't sit around
in the history. When a lock is first locked, this checks to see
if the last lock in the list was one that was failed to be
locked. If it is, then that was a lock that we're no longer
sitting in a trylock loop trying to lock, so just remove it.
(inspired by issue #11712)
2008-02-25 21:37 +0000 [r104095] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make it so a users.conf user creates both a
SIP peer and a SIP user. The user will be used for inbound
authentication for the device, and peer will be used for placing
calls to the device. (closes issue #9044) Reported by: queuetue
Patches: sip-gui-friend.diff uploaded by qwell (license 4)
2008-02-25 21:31 +0000 [r104094] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: If the destination folder is full, don't
delete a message when exiting. (closes issue #12065) Reported by:
selsky Patch by: (myself)
2008-02-25 20:49 +0000 [r104092] Jason Parker <jparker@digium.com>
* main/config.c: Allow the use of #include and #exec in situations
where the max include depth was only 1. Specifically, this fixes
using #include and #exec in extconfig.conf. This was basically
caused because the config file itself raises the include level to
1. I opted not to raise the include limit, because recursion here
could cause very bizarre behavior. Pointed out, and tested by
jmls (closes issue #12064)
2008-02-25 18:38 +0000 [r104086] Russell Bryant <russell@digium.com>
* channels/chan_agent.c: Ensure that the channel doesn't disappear
in agent_logoff(). If it does, it could cause a crash. (fixes the
crash reported in BE-396)
2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: If a resubscription comes in for a dialog we
no longer know about tell the remote side that the dialog does
not exist so they subscribe again using a new dialog. (closes
issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
uploaded by file (license 11)
* channels/chan_sip.c: Due to recent changes tag will no longer be
NULL if not present so we have to use ast_strlen_zero to see if
it's actually blank. (closes issue #12061) Reported by: flefoll
Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
flefoll (license 244)
2008-02-22 22:45 +0000 [r104037] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Backwards debug message. (closes issue
#12052) Reported by: flefoll Patches:
chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
244)
2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson <mmichelson@digium.com>
* channels/chan_zap.c: And as a followup to revision 104026,
completely remove event-related calls from a section of code
where we know there was no event to handle or get.
* channels/chan_zap.c: Remove an incorrect debug message. It
reported that it had received a specific event and tried to
report which event was received. What actually was happening was
that it was reporting the number of bytes returned from a call to
read(). Thanks to Jared Smith for bringing the issue up on IRC
2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming <kpfleming@digium.com>
* main/manager.c: reduce the likelihood that HTTP Manager session
ids will consist of primarily '1' bits
2008-02-20 22:32 +0000 [r103956] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Clear up confusion when viewing the
QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
user's perspective, the queue does exist, we shouldn't tell them
we couldn't find the queue. Instead since it is a dead queue,
report a 0 waiting count This issue was brought up on IRC by jmls
2008-02-20 22:06 +0000 [r103953] Joshua Colp <jcolp@digium.com>
* channels/chan_zap.c: Don't wait for additional digits when
overlap dialing is enabled if the setup message contains the
sending_complete information element. (closes issue #11785)
Reported by: klaus3000 Patches:
sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
klaus3000 (license 65)
2008-02-20 21:40 +0000 [r103904] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: Fix a crash if the channel becomes NULL
while attempting to lock it. (closes issue #12039) Reported by:
danpwi
2008-02-20 17:53 +0000 [r103845] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Compat fix for Solaris (closes issue
#12022) Reported by: asgaroth Patches:
20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
Tested by: asgaroth
2008-02-19 20:28 +0000 [r103823] Joshua Colp <jcolp@digium.com>
* channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
(closes issue #11241) Reported by: tusar Patches:
h323id_as_callerid_name.patch uploaded by tusar (license 344)
2008-02-19 20:02 +0000 [r103821] Russell Bryant <russell@digium.com>
* channels/chan_local.c: Account for the fact that the "other"
channel can disappear while the local pvt is not locked. (fixes a
problem introduced in rev 100581) (closes issue #12012) Reported
by: stevedavies Patch by me
2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp <jcolp@digium.com>
* configure, configure.ac: Don't look for launchd when cross
compiling. (closes issue #12029) Reported by: ovi
* channels/chan_sip.c: Fix building of chan_sip.
2008-02-19 10:27 +0000 [r103806] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Make sure we send error replies correctly by
checking the via header.
2008-02-18 23:56 +0000 [r103801] Joshua Colp <jcolp@digium.com>
* main/channel.c: Ensure that emulated DTMFs do not get interrupted
by another begin frame. (closes issue #11740) Reported by: gserra
Patches: v1-11740.patch uploaded by dimas (license 88) (closes
issue #11955) Reported by: tsearle (closes issue #10530) Reported
by: xmarksthespot
2008-02-18 22:28 +0000 [r103790-103795] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Fix previous commit so that we actually
disable echocanbridged if echocancel is off.
* channels/chan_zap.c: Correct a message when echocancelwhenbridged
is on, but echocancel is not. Issue #12019
2008-02-18 20:52 +0000 [r103786] Mark Michelson <mmichelson@digium.com>
* main/app.c: There was an invalid assumption when calculating the
duration of a file that the filestream in question was created
properly. Unfortunately this led to a segfault in the situation
where an unknown format was specified in voicemail.conf and a
voicemail was recorded. Now, we first check to be sure that the
stream was written correctly or else assume a zero duration.
(closes issue #12021) Reported by: jakep Tested by: putnopvut
2008-02-18 17:31 +0000 [r103780] Tilghman Lesher <tlesher@digium.com>
* main/rtp.c, channels/chan_sip.c: When a SIP channel is being
auto-destroyed, it's possible for it to still be in bridge code.
When that happens, we crash. Delay the RTP destruction until the
bridge is ended. (closes issue #11960) Reported by: norman
Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
(license 14) Tested by: norman
2008-02-18 16:37 +0000 [r103770] Mark Michelson <mmichelson@digium.com>
* channels/chan_zap.c: Fix a linked list corruption that under the
right circumstances could lead to a looped list, meaning it will
traverse forever. (closes issue #11818) Reported by: michael-fig
Patches: 11818.patch uploaded by putnopvut (license 60) Tested
by: michael-fig
2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp <jcolp@digium.com>
* main/asterisk.c: Backport fix from issue #9325. (closes issue
#11980) Reported by: rbrunka
* channels/chan_sip.c: Don't care if the extension given doesn't
exist for subscription based MWI.
2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
condition (closes issue #11780) Reported by: guillecabeza
Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
380)
* main/loader.c: In the case that you try to directly reload a
module has returned AST_MODULE_LOAD_DECLINE, log a message
indicating that the module is not fully initialized and must be
initialized using "module load".
* main/loader.c: Don't attempt to execute the reload callback for a
module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
that was reported against chan_console in trunk. (closes issue
#11953, reported by junky, fixed by me)
2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson <mmichelson@digium.com>
* doc/imapstorage.txt, configure, configure.ac: Final round of
changes for configure script logic for IMAP Now if a directory is
specified, then we will search that directory for a source
installation of the IMAP toolkit. If none is found, then we will
use that directory as the basis for detecting a package
installation of the IMAP c-client. If that check fails, then
configure will fail.
* configure, configure.ac: Fix a bit of wrong logic in the
configure script that caused problems when trying to configure
without IMAP. Patch suggestion from phsultan, but I modified it
slightly. (closes issue #12003) Reported by: pj Tested by:
putnopvut
* doc/imapstorage.txt, configure, configure.ac: I apparently
misunderstood one of the requirements of this configure change.
Now, if a source directory is specified with the --with-imap
option, and a valid source installation is not detected there,
then configure will fail and will not check for a package
installation.
* doc/imapstorage.txt: Make a small clarification in the
documentation
* doc/imapstorage.txt: Update documentation regarding configuration
of IMAP
* apps/app_voicemail.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Change to the
configure logic regarding IMAP. Prior to this commit, if you
wished to configure Asterisk with IMAP support, you would use the
--with-imap configure switch in one of the following two ways:
--with-imap=/some/directory would look in the directory specified
for a UW IMAP source installation --with-imap would assume that
you had imap-2004g installed in .. relative to the Asterisk
source With this set of changes the two above options still work
the same, but there are two new behaviors, too.
--with-imap=system will assume that you have -libc-client.so
where you store your shared objects and will attempt to find
c-client headers in your include path either in the imap or
c-client directory. If either of the two original methods of
specifying the imap option should fail, then the check for
--with-imap =system will be performed in addition. It is only
after this "system" check that failure can happen.
* apps/app_voicemail.c: Fix build for non-IMAP builds
* apps/app_voicemail.c: Fix the new message count if delete=yes
when using IMAP storage. (closes issue #11406) Reported by:
jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
50) Tested by: jaroth
2008-02-14 19:51 +0000 [r103683-103684] Jason Parker <jparker@digium.com>
* funcs/func_cdr.c: swap location for this..
* funcs/func_cdr.c: Document the 'l' option to the CDR() function.
(Thanks voipgate for pointing out the option, and Leif for
providing text for it.) Closes issue #11695.
2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher <tlesher@digium.com>
* channels/chan_agent.c: We aren't talking to ourselves; we're
talking to someone else. (closes issue #11771) Reported by:
msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
(license 22) Tested by: caio1982, msetim
* apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
not loaded (which is a symbol dependency) (closes issue #11760)
Reported by: non-poster Patches: 20080114__bug11760.diff.txt
uploaded by Corydon76 (license 14) Tested by: Corydon76,
non-poster, jamesgolovich
2008-02-12 22:24 +0000 [r103503-103504] Jason Parker <jparker@digium.com>
* main/asterisk.c: revert accidental change from last commit. oops
* contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
that was impossible.
2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Even if no CallerID name or number has been
provided by the remote party still use the configured sip.conf
ones. (closes issue #11977) Reported by: pj
* apps/app_meetme.c: If entering a conference with the 'w' option
ensure that we can't listen or speak until the marked user
appears. (closes issue #11835) Reported by: alanmcmillan
2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming <kpfleming@digium.com>
* configs/zapata.conf.sample: improve 2BCT documentation a bit
(thanks Jared)
2008-02-09 06:23 +0000 [r103197] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Commit fix for being unable to send
voicemail from VoiceMailMain Reported by: William F Acker (via
the -users mailing list) Patch by: Corydon76 (license 14)
2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Prevent a potential three-thread deadlock. Also
added a comment block to explicitly state the locking order
necessary inside app_queue. (closes issue #11862) Reported by:
flujan Patches: 11862.patch uploaded by putnopvut (license 60)
Tested by: flujan
* channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
fails for Zaptel timing device. (closes issue #11891) Reported
by: tzafrir
2008-02-08 15:08 +0000 [r102968] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Make sure the presence of dbsecret is
factored into user scoring. (closes issue #11952) Reported by:
bbhoss
2008-02-07 19:53 +0000 [r102858] Jason Parker <jparker@digium.com>
* res/res_features.c: Specify which digit string was matched in
debug message. (closes issue #11949) Reported by: dimas Patches:
v1-feature-debug.patch uploaded by dimas (license 88)
2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming <kpfleming@digium.com>
* configs/zapata.conf.sample: document usage of 'transfer'
configuration option for ISDN PRI switch-side transfers
2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only consider a T.38-only INVITE compatible
if we have both a joint capability between us and them and if
they provided T.38.
* main/global_datastores.c: Add missing header file and
ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
snuffy
2008-02-06 15:19 +0000 [r102651] Russell Bryant <russell@digium.com>
* configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
that it gets inherited by outbound channels. (due to a discussion
between me and a user via email)
2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming <kpfleming@digium.com>
* pbx/Makefile, res/Makefile: ensure that all remaining
multi-object modules are built using their proper CFLAGS and
include directory paths
2008-02-06 00:26 +0000 [r102576] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Move around some defines to unbreak ODBC
storage. (closes issue #11932) Reported by: snuffy
2008-02-05 20:02 +0000 [r102453] Mark Michelson <mmichelson@digium.com>
* channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
issue #11919) Reported by: eferro Patches:
mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
Tested by: eferro
2008-02-05 19:52 +0000 [r102450] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: If a REGISTER attempt comes in that is a
retransmission of a previous REGISTER do not create a new nonce
value. (issue #BE-381)
2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming <kpfleming@digium.com>
* channels/Makefile: ensure that components of chan_misdn.so are
built using any special build options that the configure script
generated (reported by Philipp Kempgen on asterisk-dev)
2008-02-05 15:09 +0000 [r102378] Joshua Colp <jcolp@digium.com>
* res/res_clioriginate.c: Perform dialing asynchronously when using
the originate CLI command so the CLI does not appear to block.
(closes issue #11927) Reported by: bbhoss
2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, utils/muted.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
fix: OS X now deprecates the use of the daemon(3) API. (closes
issue #11908) Reported by: oej Patches:
20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
* funcs/func_strings.c: Missing braces. (closes issue #11912)
Reported by: dimas Patches: sprintf.patch uploaded by dimas
(license 88)
2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
(according to RFC 3261) (closes issue #9492) Reported by:
kryptolus Patches: bug9492.txt uploaded by oej (license 306)
Tested by: oej
* channels/chan_sip.c: Handle ACK and CANCEL in an invite
transaction - even if we get INFO transactions during the actual
call setup. (closes issue #10567) Reported by: jacksch Tested by:
oej Patch by: oej inspired by suggestions from neutrino88 in the
bug tracker
2008-02-01 23:06 +0000 [r101989] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
out that even though G.722 is 16 kHz, it is supposed to specified
as 8 kHz in the RTP, and RTP timestamps are supposed to be
calculated based on 8 kHz. (Apparently this is due to a bug in a
spec, but people follow it anyway, because it's the spec ...)
2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
voicemail, and fixed several other bugs while I'm in the area.
(closes issue #11615) Reported by: jamessan Patches:
20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, jamessan
* configure, include/asterisk/autoconfig.h.in, configure.ac,
acinclude.m4: Change detection of getifaddrs to use
AST_C_COMPILE_CHECK, backported from trunk (as suggested by
kpfleming)
2008-02-01 17:41 +0000 [r101822] Jason Parker <jparker@digium.com>
* apps/app_authenticate.c: Remove a needless (and incorrect) call
to feof() after fgets(). This would have exited the loop early if
you had an authentication file with no newline at the end.
2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant <russell@digium.com>
* apps/app_authenticate.c: off by one error
* apps/app_authenticate.c: Don't overwrite the last character of a
line if it's not a newline. This would happen if the last line in
the file doesn't have a newline. (pointed out by Qwell)
2008-02-01 15:55 +0000 [r101772] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/acl.c: Compatibility fix for OpenWRT (reported by Brian
Capouch via the mailing list)
2008-02-01 00:32 +0000 [r101693] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
strings for the case that no peer or hostname was provided, which
is the one part of the dial string that is absolutely required.
If it's not there, bail out. (closes issue #11897) Reported by
sokhapkin Patch by me
2008-02-01 00:06 +0000 [r101649] Mark Michelson <mmichelson@digium.com>
* apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
periods of no channel activity" (closes issue #9256) Reported by:
cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
rjain
2008-01-31 Russell Bryant <russell@digium.com>
* Asterisk 1.4.18 released.
2008-01-31 23:10 +0000 [r101601] Russell Bryant <russell@digium.com>
* main/translate.c, main/file.c: Fix a couple of places where
ast_frfree() was not called on a frame that came from a
translator. This showed itself by g729 decoders not getting
released. Since the flag inside the translator frame never got
unset by freeing the frame to indicate it was no longer in use,
the translators never got destroyed, and thus the g729 licenses
were not released. (closes issue #11892) Reported by: xrg
Patches: 11892.diff uploaded by russell (license 2) Tested by:
xrg, russell
2008-01-31 21:00 +0000 [r101531] Mark Michelson <mmichelson@digium.com>
* res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
beginning of an absolute pathname. 2. If ast_monitor_change_fname
is called and the new filename is the same as the old, then exit
early and don't set the filename_changed field in the monitor
structure. Setting it in this case was causing ast_monitor_stop
to erroneously delete them. (closes issue #11741) Reported by:
garlew Tested by: putnopvut
2008-01-31 19:52 +0000 [r101482] Jason Parker <jparker@digium.com>
* channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
for struct in_addr funkiness. Issue #11885, patch by snuffy.
2008-01-31 19:30 +0000 [r101480] Steve Murphy <murf@digium.com>
* main/pbx.c: closes issue #11845; that's the one where there's a
1004 byte cdr leak with every AMI Redirect to a zap channel
2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant <russell@digium.com>
* channels/chan_agent.c: Add more missing locking of the agents
list ...
* channels/chan_agent.c: Move the locking from find_agent() into
the agent dialplan function handler to ensure that the agent
doesn't disappear while we're looking at it.
* channels/chan_agent.c: Add missing locking to the find_agent()
function.
2008-01-30 15:41 +0000 [r101222] Joshua Colp <jcolp@digium.com>
* main/slinfactory.c: Fix an issue where if a frame of higher
sample size preceeded a frame of lower sample size and
ast_slinfactory_read was called with a sample size of the
combined values or higher a crash would happen. (closes issue
#11878) Reported by: stuarth
2008-01-30 15:34 +0000 [r101219] Jason Parker <jparker@digium.com>
* configs/extensions.conf.sample: Change default config to use
descending channel order of groups, rather than ascending. Fixes
a potential source of confusion in glare-type situations. Issue
11875, reported by JimVanM.
2008-01-30 15:23 +0000 [r101216] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a logic error with regards to autofill.
Prior to this change, it was possible for a caller to go out of
turn if autofill were enabled and callers ahead in the queue were
attempting to call a member. This change fixes this.
2008-01-30 11:20 +0000 [r101152] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Stop musiconhold on attended transfer.
(closes issue #11872) Reported by: gareth Patches:
svn-101018.patch uploaded by gareth (license 208)
2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard <dhubbard@digium.com>
* build_tools/make_version: updated build_tools to handle the
autotag directory structure changes; changes related to BE-353.
Patch by The Russell and reviewed by The Me.
2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Remove a memory leak from updating realtime
queues
* apps/app_queue.c: Fixing an erroneous return value returned when
attempting to pause or unpause a queue member fails. Fixes
BE-366, thanks to John Bigelow for writing the patch.
2008-01-29 17:57 +0000 [r100934] Joshua Colp <jcolp@digium.com>
* apps/app_mixmonitor.c: Don't forget to record the channel so we
know whether it is bridged or not later. (closes issue #11811)
Reported by: slavon
2008-01-29 17:43 +0000 [r100932] Russell Bryant <russell@digium.com>
* main/Makefile: Fix the last couple of issues related to building
from a path that contains spaces. (closes issue #11834)
2008-01-29 17:41 +0000 [r100930] Jason Parker <jparker@digium.com>
* channels/misdn_config.c: Initialize an array to 0s if config
option not specified. (closes issue #11860) Patches:
misdn_get_config.v1.diff uploaded by IgorG (license 20)
2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant <russell@digium.com>
* Makefile: Use GNU make magic instead of shell magic to escape
spaces in the working directory. (related to issue #11834)
* Makefile: Fix building Asterisk when the working path has spaces
in it. (closes issue #11834) Reported by: spendergrass Patched
by: me
2008-01-29 16:10 +0000 [r100835] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Allow zap groups above 30 to work properly.
(closes issue #11590) Reported by: tbsky
2008-01-29 10:36 +0000 [r100793] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: fixed potential segfault in misdn show
channels CLI command
2008-01-29 08:26 +0000 [r100740] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: (closes issue #11736) Reported by: MVF
Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
oej, MVF, revolution (russellb: This was the showstopper for the
release.)
2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
(went to 't' instead of 'T')
2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson@digium.com>
* channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
chan_vpb. It is alive and well.
2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
greeting files into the database like we do with the others.
Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
by dimas (license 88)
2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: For some reason, the use of this strdupa()
is leading to memory corruption on freebsd sparc64. This trivial
workaround fixes it. (closes issue #10300, closes issue #11857,
reported by mattias04 and Home-of-the-Brave)
* res/res_features.c: Fix a crash in ast_masq_park_call() (issue
#11342) Reported by: DEA Patches: res_features-park.txt uploaded
by DEA (license 3)
2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Correct a comment which made little/no
sense.
2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell@digium.com>
* main/channel.c, channels/chan_local.c,
include/asterisk/channel.h: Make some deadlock related fixes.
These bugs were discovered and reported internally at Digium by
Steve Pitts. - Fix up chan_local to ensure that the channel lock
is held before the local pvt lock. - Don't hold the channel lock
when executing the timing function, as it can cause a deadlock
when using chan_local. This actually changes the code back to be
how it was before the change for issue #10765. But, I added some
other locking that I think will prevent the problem reported
there, as well.
2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher@digium.com>
* main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
the scheduler, ignoring the return value could possibly cause
memory to be accessed after it is freed, which causes all sorts
of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our
taskid value). (closes issue #11386) Reported by: flujan Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson@digium.com>
* channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
now preferred that users of Voicetronix products use chan_zap in
combination with their zaptel drivers.
2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: This would have never been true, since we're
passing (sizeof(req.data) - 1) as the len to recvfrom().
2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/app.h: make these macros not assume that the
only other field in the structure is 'argc'... this is true when
someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
to define your own structure as long as it has the right fields
2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell@digium.com>
* main/asterisk.c: Update main Asterisk copyright info to 2008
2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker@digium.com>
* main/acl.c: Fix compilation on Solaris. (closes issue #11832)
Patches: bug-11832.diff uploaded by snuffy (license 35)
2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Second attempt. Don't change invitestate
when receiving 18x messages in CANCEL state. (issue #11736)
Reported by: MVF Patch by oej.
* channels/chan_sip.c: Make sure we don't cancel destruction on
calls in CANCEL state, even if we get 183 while waiting for
answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
bug11736.txt uploaded by oej (license 306) Tested by: MVF
2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson@digium.com>
* apps/app_externalivr.c: Fixing a typo.
2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell@digium.com>
* apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
beginning of a list of channels to potentially spy on. However,
if there were no matching channels, it would beep at you over and
over, which is pretty annoying. Now, it will only beep once in
the case that there are no channels to spy on, but it will still
beep again once it reaches the beginning of the channel list
again. (closes issue #11738, patched by me)
2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: These flag tests were illogical. They were
testing sip_peer flags on a sip_pvt. Thanks to Russell for
helping to get this odd problem figured out.
2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher@digium.com>
* apps/app_voicemail.c: When we reset the password via an external
command, we should also reset the password stored in the
in-memory list, too (otherwise it doesn't really take effect).
(closes issue #11809) Reported by: davetroy Patches:
fix_externpass.diff uploaded by davetroy (license 384)
* res/res_odbc.c: Oops, should have checked for a NULL obj, here,
too
* main/acl.c: Just confirmed that all current platforms need this
header file
2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Thanks to Russell's education I realize that
BUFSIZ has changed since I learned the C language over 20 years
ago... Resetting chan_sip to the size of BUFSIZ that I expected
in my old head to avoid to heavy memory allocations on some
systems.
2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher@digium.com>
* main/acl.c: Fix the defines for OS X (and Solaris, too)
2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej@edvina.net>
* channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
apps/app_followme.c: Add more dependencies on chan_local and add
a note to the description of chan_local so that people don't
disable it in menuselect just to clean up.
* apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
still runs without chan_local, but will be missing forwarding
functionality.
2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher@digium.com>
* main/acl.c: Ensure that we can get an address even when we don't
have a default route. (closes issue #9225) Reported by: junky
Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
(license 14) Tested by: oej, loloski, sergee
2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Cleaning up some documentation that led to
confusion in a bug report
2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: Fixing an issue wherein monitoring local
channels was not possible. During a channel masquerade, the
monitors on the two channels involved are swapped. In 99% of the
cases this results in the desired effect. However, if monitoring
a local channel, this caused the monitor which was on the local
channel to get moved onto a channel which is immediately hung up
after the masquerade has completed. By swapping the monitors
prior to the masquerade, we avoid the problem by tricking the
masquerade into placing the monitor back onto the channel where
we want it. During the investigation of the issue, the channel's
monitor was the only thing that was swapped in such a manner
which did not make sense to have done. All other variable
swapping made sense.
2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c, configs/res_odbc.conf.sample,
include/asterisk/res_odbc.h: Permit the user to specify number of
seconds that a connection may remain idle, which fixes a crash on
reconnect with the MyODBC driver. (closes issue #11798) Reported
by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
uploaded by Corydon76 (license 14) Tested by: mvanbaak
2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Bump the buffer size for Via headers up to
512. There are some exceptionally large Via headers out there.
(closes issue #11783) Reported by: ofirroval
2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell@digium.com>
* main/slinfactory.c: Fix a couple of memory leaks with frame
handling. Specifically, ast_frame_free() needed to be called on
the frame that came from the translator to signed linear.
2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp@digium.com>
* include/asterisk/channel.h: Remove the __ in front of the unused
variable. This causes some compilers to freak out.
2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell@digium.com>
* include/asterisk/translate.h, main/frame.c: Revert adding the
packed attribute, as it really doesn't make sense why that would
do any good. Fix the real bug, which is to do the check to see if
the frame came from a translator at the beginning of
ast_frame_free(), instead of at the end. This ensures that it
always gets checked, even if none of the parts of the frame are
malloc'd, and also ensures that we aren't looking at free'd
memory in the case that it is a malloc'd frame. (closes issue
#11792, reported by explidous, patched by me)
* include/asterisk/translate.h: Since we're relying on the offset
between the frame and the beginning of the translator pvt struct,
set the packed attribute to make sure we get to the right place.
(potential fix for issue #11792)
2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson@digium.com>
* res/res_features.c: This should at least temporarily fix a
problem where the 't' Dial option is incorrectly passed to the
transferee when built-in attended transfers are used. There is
still a problem with 'T', but better to fix some problems than no
problems while we work on it. (closes issue #7904) Reported by:
k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
(license 138) Tested by: sergee, otherwiseguy
2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand@digium.com>
* configs/cdr.conf.sample: doh! revert a revert of a revert
(changed by mistake in 99010)
* main/manager.c, configs/cdr.conf.sample: missed that one while
reverting
* main/manager.c: reverting 99001 - We need the Max-Age for
extending the life of cookie mansession_id
2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell@digium.com>
* main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
Have IAX2 optimize the codec translation path just like chan_sip
does it. If the caller's codec is in our codec list, move it to
the top to avoid transcoding. (closes issue #10500) Reported by:
stevedavies Patches: iax-prefer-current-codec.patch uploaded by
stevedavies (license 184) iax-prefer-current-codec.1.4.patch
uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
sheldonh
2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming@digium.com>
* main/manager.c: we should only send the Set-Cookie header to the
browser on the first response after creating a manager session,
not on every response (doing so causes the browser to clear any
local cookies it may have associated with the session)
2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker@digium.com>
* configs/zapata.conf.sample: Add a clarification about the
immediate= option of zapata.conf Issue 11784, patch by klaus3000.
2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell@digium.com>
* .cleancount, include/asterisk/channel.h: Add an unused pointer to
the ast_channel struct. This makes the ast_channel structure
retain the same size as it had in previous 1.4 releases. Also,
all of the offsets for members in the structure are still the
same (except for the two pointers that got replaced for the new
spy/whisper architecture.)
2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp@digium.com>
* .cleancount: Bump up cleancount due to previous commit that
changed the channel structure.
* apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
(added), main/Makefile, include/asterisk/chanspy.h (removed),
include/asterisk/channel.h, main/audiohook.c (added): Replace
current spy architecture with backport of audiohooks. This should
take care of current known spy issues.
* channels/chan_iax2.c: Add missing NULLs at end of two
ast_load_realtimes. (closes issue #11769) Reported by: tequ
Patches: chaniax.patch uploaded by dimas (license 88)
2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson@digium.com>
* channels/chan_local.c: Fix a deadlock in chan_local in
local_hangup. There was contention because the local_pvt was held
and it was attempting to lock a channel, which is the incorrect
locking order. (closes issue #11730) Reported by: UDI-Doug
Patches: 11730.patch uploaded by putnopvut (license 60) Tested
by: UDI-Doug
2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp@digium.com>
* main/dial.c: Introduce a lock into the dialing API that protects
it when destroying the structure. (closes issue #11687) Reported
by: callguy Patches: 11687.diff uploaded by file (license 11)
* main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
issue #11777) Reported by: tootai
* channels/chan_sip.c: Don't drop the old record route information
when dealing with packets related to a reinvite. (closes issue
#11545) Reported by: kebl0155 Patches: reinvite-patch.txt
uploaded by kebl0155 (license 356)
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
versions use a separate library for some things so we need to use
it if present in codec_speex. (closes issue #11693) Reported by:
yzg
2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Change a buffer in check_auth() to be a
thread local dynamically allocated buffer, instead of a massive
buffer on the stack. This fixes a crash reported by Qwell due to
running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in
chan_sip is arbitrary and careless. BUFSIZ is a system specific
define. On my machine, it is 8192, but by definition (according
to google) could be as small as 256. So, this buffer in
check_auth was 16 kB. We don't even support SIP messages larger
than 4 kB! Further usage of this define should be avoided, unless
it is used in the proper context.
* main/rtp.c, include/asterisk/translate.h, main/frame.c,
main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
some memory access errors pointed out by the valgrind2.txt output
on issue #11698. The issue here is that it is possible for an
instance of a translator to get destroyed while the frame
allocated as a part of the translator is still being processed.
Specifically, this is possible anywhere between a call to
ast_read() and ast_frame_free(), which is _a lot_ of places in
the code. The reason this happens is that the channel might get
masqueraded during this time. During a masquerade, existing
translation paths get destroyed. So, this patch fixes the issue
in an API and ABI compatible way. (This one is for you,
paravoid!) It changes an int in ast_frame to be used as flag
bits. The 1 bit is still used to indicate that the frame contains
timing information. Also, a second flag has been added to
indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called
in translate.c to let it know that this frame is doing being
processed. At this point, the flag gets cleared. Also, if the
translator was requested to be destroyed while its internal frame
still had this flag set, its destruction has been deffered until
it finds out that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue,
and I was not able to think of a better solution ...
2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Based on the boundary found move over the
correct amount. (closes issue #11750) Reported by: tasker
* channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
in the multipart mixed content type. (closes issue #11750)
Reported by: tasker
2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Adding in appropriate unlocks for the locks
I added. Thanks to joetester on IRC for pointing this out.
2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell@digium.com>
* main/translate.c: Revert a change that introduces an unacceptable
performance hit and is causing memory leaks ... (from rev 97973)
2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixing another compilation error. I'm a bit off
today :(
* apps/app_queue.c: Oops. Last commit had compilation error.
* apps/app_queue.c: Adding explicit defaults for missing options to
init_queue. This is necessary because if a user either removes or
comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the
value from prior to the reload. Thanks to John Bigelow for
pointing this error out to me.
2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Add a connection timeout attribute, as that was
what was intended with the login timeout, but ODBC divides it up
into 2 different timeouts. (Closes issue #11745)
2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
(closes issue #11646) Reported by: caio1982 Patches:
dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
by: caio1982, mvanbaak
2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand@digium.com>
* main/http.c: Comment explaining how to force browser to always
read some html files from server.
2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp@digium.com>
* main/rtp.c: If the incoming RTP stream changes codec force the
bridge to break if the other side does not support it. (closes
issue #11729) Reported by: tsearle Patches:
new_codec_patch_udiff.patch uploaded by tsearle (license 373)
* res/res_agi.c: If the channel is hungup during RECORD FILE send a
result code of -1 to be uniform with everything else. (closes
issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
by davevg (license 209)
2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Properly report the hangup cause as no answer
when someone does not answer (closes issue #10574, reported by
boch, patched by moy)
2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher@digium.com>
* codecs/gsm/Makefile: Add another exception (which doesn't work)
for -march optimization flag. Reported by: thomasmebes Patch by:
tilghman (Closes issue #11563)
2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell@digium.com>
* doc/security.txt, main/asterisk.c, configure,
include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
makeopts.in: Backport the ability to set the ToS bits on Linux
when not running as root. Normally, we would not backport
features into 1.4, but, I was convinced by the justification
supplied by the supplier of this patch. He pointed out that this
patch removes a requirement for running as root, thus reducing
the potential impacts of security issues. (closes issue #11742)
Reported by: paravoid Patches: libcap.diff uploaded by paravoid
(license 200)
2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp@digium.com>
* apps/app_followme.c: Ensure the return value of ast_bridge_call
is passed back up as the application return value. This is needed
for transfers to function so the PBX core knows to continue
execution. (closes issue #10327) Reported by: kkiely
2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Back out changes from revision 97077, since
it wasn't perfect
2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell@digium.com>
* main/frame.c: Fix samples vs. length calculations for g722
* main/translate.c: Simplify this code with a suggestion from Luigi
on the asterisk-dev list. Instead of using is16kHz(), implement a
format_rate() function.
* main/translate.c: Fix various timing calculations that made
assumptions that the audio being processed was at a sample rate
of 8 kHz.
2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c, main/translate.c: 1) When we get a
translated frame out, clone it, because if the translator pvt is
freed before we use the frame, bad things happen. 2) Getting a
failure from ast_sched_delete means that the schedule ID is
currently running. Don't just ignore it. (Closes issue #11698)
2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Let us leave a voicemail for ourself if we
have logged into VoiceMailMain and chosen to leave a message.
(closes issue #11735, reported and patched by jamessan)
2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf@digium.com>
* pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
overrode the normally generate yyfree func with our own version
that checks the pointer for non-null before passing to free().
Also takes care of a little problem with 2.5.33 and the use of
the __STDC_VERSION__ macro.
* main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
fix for 2 things: a problem Terry was having in OSX with null
pointers, which was my fault, as I probably forgot to run the sed
script last time I made mods. So, I moved the fix into the flex
input itself. Then, I found when I used flex 2.5.33, that it was
using __STDC_VERSION__, and that's not real good; so I added back
in a DIFFERENT sed script to fix that little mess. Tested
everything, a couple different ways. Hope I did no harm, at the
least.
2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker@digium.com>
* include/asterisk/frame.h: Fix a comment that is no longer true.
2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell@digium.com>
* pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
pbx/kdeconsole_main.cc (removed): Remove other remnants of
pbx_kdeconsole
* pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
configure, include/asterisk/autoconfig.h.in, configure.ac,
makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
worked in ages, and nobody has complained. (closes issue #11706,
reported by caio1982)
2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp@digium.com>
* funcs/func_groupcount.c: Don't try to copy the category from the
group if no category exists. (closes issue #11724) Reported by:
IgorG Patches: group_count.v1.patch uploaded by IgorG (license
20)
2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell@digium.com>
* pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
messages
* pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
actually load on my system still (related to issue #11706)
2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker@digium.com>
* main/cli.c: Correctly display a message if a command could not be
found. Also fix a comment which may have led to this happening.
Issue 11718, reported by kshumard.
* main/cli.c: Fix some locking and return value funkiness. We
really shouldn't be unlocking this lock inside of a function,
unless we locked it there too.
2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
smaller functions this time
2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell@digium.com>
* res/res_features.c: Fix saying the parking space number to the
caller doing the parking ...
2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming@digium.com>
* codecs/codec_zap.c: report the same message whether Zaptel does
not have transcoder support loaded or no transcoders were found
2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
function. As underlined in issue #10437 by Josh, we need to
prevent a possible memory leak. We only set the name part of the
caller id, the number part is not relevant when dealing with
JIDs. Closes issue #11549.
2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Don't do conferencing totally in Zaptel if
Monitor is running on the channel. (closes issue #11709) Reported
by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
(license 371)
2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: pass the right variable to get an error
string... oops
* channels/chan_zap.c: add error number output to ioctl failure
messages to help with debugging
2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher@digium.com>
* main/cli.c, main/editline/readline.c: Allow filename completion
on zero-length modules, remove a memory leak, remove a file
descriptor leak, and make filename completion thread-safe.
Patched and tested by tilghman. (Closes issue #11681)
2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: use the \retval doxygen command properly
* apps/app_queue.c: Part 1 of N of adding doxygen comments to
app_queue. I picked some of the most common functions used (which
also happen to be some the biggest/ugliest functions too) to
document first. I'm pretty new to doxygen so criticism is
welcome.
* apps/app_queue.c: Some coding guidelines-related cleanup
2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp@digium.com>
* channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
(closes issue #11443) Reported by: eferro Patches:
dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
337)
2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher@digium.com>
* main/autoservice.c, main/utils.c: Increase constants to where
we're less likely to hit them while debugging. (Closes issue
#11694)
2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Making some changes designed to not allow
for a corrupted mailstream for a vm_state. 1. Add locking to the
vm_state retrieval functions so that no linked list corruption
occurs. 2. Make sure to always grab the persistent vm_state when
mailstream access is necessary. 3. Correct an incorrect return
value in the init_mailstream function. (closes issue #11304,
reported by dwhite)
2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp@digium.com>
* funcs/func_groupcount.c: If no group has been provided to the
GROUP_COUNT dialplan function then use the first one specific to
the channel. (closes issue #11077) Reported by: m4him
* apps/app_queue.c: Make app_queue calls work with directed pickup.
(closes issue #11700) Reported by: jbauer
2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
found in issue #11386, but not completely closing that issue.
2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 96931 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
2 lines Change misery.digium.com to pbx.digium.com ........
* res/res_smdi.c: Don't crash if something happens when setting up
an SMDI interface and it gets destroyed before the SMDI port
handling thread gets created.
2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Indentation fix, makes the code easier to read
* res/res_jabber.c: Compute the base64 value over the
[authzid]\0authcid\0password string, thus excluding the trailing
NULL byte. This change has already been committed to trunk, see
#11644.
2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell@digium.com>
* main/devicestate.c: Don't pass an empty string as the device
name.
2008-01-04 23:03 +0000 [r96575] Tilghman Lesher <tlesher@digium.com>
* main/devicestate.c: Fix the problem of notification of a device
state change to a device with a '-' in the name. Could probably
do with a better fix in trunk, but this bug has been open way too
long without a better solution. Reported by: stevedavies Patch
by: tilghman (Closes issue #9668)
2008-01-04 22:55 +0000 [r96573] Jason Parker <jparker@digium.com>
* res/res_features.c: Properly continue in the dialplan if using
PARKINGEXTEN and the slot is full. Issue 11237, patch by me.
2008-01-04 19:27 +0000 [r96525] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: If you change the bindaddr in sip.conf to a
non-bound address and reload, sip goes kablooie. Reported and
patched by: one47 (Closes issue #11535)
2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: Make use of the temporary channel pointer
while the pvt is unlocked. (closes issue #11675) Reported by:
flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
uploaded by flefoll (license 244)
* channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
been destroyed before we get to this point (closes issue #11672,
reported by snuffy, patched by me)
2008-01-03 21:37 +0000 [r96318] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c: Missed initialization caused crash.
Reported and fixed by: tiziano (Closes issue #11671)
2008-01-03 12:12 +0000 [r96198-96199] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: make sure frame is completely clean,
before we send it to asterisk as DTMF. If we don't make it clean,
it happens that one way audio occurs..
* channels/chan_misdn.c: when overlapdial was used and no number
was dialed, the call was dropped, now we just jump into the s
extension, which makes a lot more sense.
2008-01-02 23:46 +0000 [r96102] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: We need to reset the membername to NULL on each
iteration of this loop, otherwise the result is that multiple
members can have the same name, since the variable was not reset
on each iteration of the loop.
2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant <russell@digium.com>
* pbx/pbx_config.c: Convert locks of the contexts list in
pbx_config to the appropriate rdlock or wrlock
* pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
list.
* apps/app_macro.c: app_macro only needs a rdlock on the contexts
list.
2008-01-02 Russell Bryant <russell@digium.com>
* Asterisk 1.4.17 released.
2008-01-02 20:24 +0000 [r95946] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Allocate a SIP refer structure when
performing a transfer using BYE with Also so that the transfer
information is properly stored. (AST-2008-028) (closes issue
#11637) Reported by: greyvoip
2008-01-02 17:51 +0000 [r95890] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: A change to improve the accuracy of queue
logging in the case where a member does not answer during the
specified timeout period. Prior to this change, there was a small
chance that the member name recorded in this case would be blank.
Also prior to this change, if using the ringall strategy, if no
one answered the call during the specified timeout, the member
name listed in the queue log would randomly be one of the members
that was rung. (closes issue #11498, reported and tested by
hloubser, patched by me)
2007-12-31 23:43 +0000 [r95577] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: Avoiding a potentially bad locking situation.
ast_merge_contexts_and_delete writelocks the conlock, then calls
ast_hint_extension, which attempts to readlock the same lock.
Recursion with read-write locks is dangerous, so the inner lock
needs to be removed. I did this by copying the "guts" of
ast_hint_extension into ast_merge_contexts_and_delete (sans the
extra lock). (this change is inspired by the locking problems
seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
2007-12-31 20:27 +0000 [r95470] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Allow the default "0" to be returned if the
STAT fails (Closes issue #11659)
2007-12-28 18:24 +0000 [r95191] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove duplicate increment of the header
count in the add_header() function. (closes issue #11648)
Reported by: makoto Patch provided by sergee, committed patch by
me, inspired by comments from putnopvut
2007-12-28 00:16 +0000 [r95095] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: I found a bug while browsing the queue code and
managed to reproduce it in a small setup. If a queue uses the
ringall strategy, it was possible through unfortunate coincidence
for a single member at a given penalty level to make app_queue
think that all members at that penalty level were unavailable and
cause the members at the next penalty level to be rung. With this
patch, we will only move to the next penalty level if ALL the
members at a given penalty level are unreachable.
2007-12-27 21:40 +0000 [r95024] Russell Bryant <russell@digium.com>
* main/channel.c: Don't report a syntax error when an empty string
is passed to ast_get_group. Just return 0. (closes issue #11540)
Reported by: tzafrir Patches: group_empty.diff uploaded by
tzafrir (license 46) -- slightly changed by me
2007-12-27 20:09 +0000 [r94977] Mark Michelson <mmichelson@digium.com>
* main/io.c: Fixing a typo in a comment.
2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp <jcolp@digium.com>
* channels/chan_h323.c: Include types.h in chan_h323 as without it
it can not be compiled on some operating systems like FreeBSD to
name one. (closes issue #11585) Reported by: sobomax Patches:
chan_h323.c.diff uploaded by sobomax (license 359)
* channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
set or not on the dialog. It is possible for it to be a pointer
to NULL. (closes issue #11557) Reported by: FuriousGeorge
2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant <russell@digium.com>
* main/pbx.c: Now that the contexts lock is a read/write lock, it
should not be locked here in ast_hint_state_changed(). This makes
it get locked recursively which now causes a deadlock. (closes
issue #11080, thanks to callguy for the access to a deadlocked
machine)
* include/asterisk/translate.h, main/translate.c: Use the constant
that I really meant to use here ...
* main/translate.c: Change ast_translator_best_choice() to only pay
attention to audio formats. This fixes a problem where Asterisk
claims that a translation path can not be found for channels
involving video. (closes issue #11638) Reported by: cwhuang
Tested by: cwhuang Patch suggested by cwhuang, with some
additional changes by me.
2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming <kpfleming@digium.com>
* main/manager.c: make this comment explain the situation in an
even more explicit fashion
2007-12-26 20:43 +0000 [r94808] Tilghman Lesher <tlesher@digium.com>
* main/manager.c: Workaround for what is probably a glibc bug (but
we'll see this crop up again and again, if we don't add the
workaround). Reported by: rolek Patch by: tilghman (Closes issue
#11601, closes issue #11426)
2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant <russell@digium.com>
* main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
was already set before starting autoservice, remember it and
ensure that the channel has the same setting when autoservice
gets stopped. (pointed out by d1mas, patched up by me)
* main/autoservice.c: When a channel is in autoservice, mark a flag
on the channel that says that we only care about the END of a
digit. That way, no magic digit emulation stuff will happen when
all we're doing is queueing up END frames.
* res/res_features.c: Don't try to send a parked call back to
itself. (closes issue #11622, reported by djrodman, patched by
me)
* main/autoservice.c: Don't store DTMF BEGIN frames while a channel
is in autoservice. It's just going to make ast_read() do a lot of
extra work when the channel comes back out of autoservice.
(closes issue #11628, patched by me)
* Makefile: List include/asterisk/version.h as a .PHONY target
because we want the commands listed for this target to be
executed regardless of whether the file exists or not. This fixes
having the version not up to date when running from svn. (closes
issue #11619, reported by plack, fixed by me)
2007-12-25 02:27 +0000 [r94769] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: file says... build on the builders.
2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher <tlesher@digium.com>
* main/channel.c: Race: we need to wait to queue a NewChannel event
until after the channel is inserted into the channel list. The
reason is because some manager users immediately queue requests
from the channel when they see that event and are confused when
Asterisk reports no such channel. (Closes issue #11632)
* channels/chan_sip.c: More deadlock avoidance code (this time
between sip_monitor and sip_hangup) Reported by: apsaras Patch
by: tilghman (Closes issue #11413)
* channels/chan_sip.c: Another bit of bad logic in realtime_peer
Reported by: dimas Patch by: dimas (Closes issue #11631)
2007-12-23 01:21 +0000 [r94660] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Argh... I suppose third time's the charm.
2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Bunch of coding guidelines cleanup
* apps/app_voicemail.c: Better quota support for using IMAP storage
voicemail (closes issue #11415, reported by jaroth) (closes issue
#11152, reported by selsky) Patch provided by jaroth
* apps/app_voicemail.c: The mail_copy c-client function does not
expect a full imap mailbox string, just the name of the mailbox.
(closes issue #11419, reported and patched by jaroth, with
additional patchwork from me)
* main/dial.c: Since we are freeing list elements within a list
traversal, we need to use the safe traversal and remove the item
from the list before freeing it. (closes issue 11612, reported by
dtyoo)
2007-12-21 16:37 +0000 [r94466] Russell Bryant <russell@digium.com>
* main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
a read/write lock to resolve a deadlock. This has a nice side
benefit of improving performance. :) (closes issue #11609)
(closes issue #11080)
2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Removing a debug message I accidentally just
committed
* main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
dates and times. Also some coding guidelines cleanup. (closes
issue #11599, reported and patched by caio1982, coding guidelines
cleanup by me)
2007-12-21 15:07 +0000 [r94418] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: Fix for restart-as-user problem reported via the
-dev list
2007-12-20 Russell Bryant <russell@digium.com>
* Asterisk 1.4.16.2 released.
2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
5 lines Fix another potential seg fault ... (closes issue #11606)
Reported by: dimas ........
* channels/chan_zap.c: Fix a deadlock in d-channel handling in
chan_zap. This deadlock was introduced by the fix to ensure that
channels are properly locked when handling channel variables.
There were sections of this code where the channel pvt was locked
before the channel lock, when in fact it _must_ be the other way
around. (closes issue #11582) Reported by: bugi
2007-12-19 23:02 +0000 [r94122] Mark Michelson <mmichelson@digium.com>
* res/res_monitor.c: Sox versions 13.0.0 and newer do not have
"soxmix" and instead use sox -m. res_monitor needs to use this if
the user does not have soxmix. (closes issue #11589, reported by
amessina, patch inspired by amessina but with a flourish from me)
2007-12-19 22:48 +0000 [r94077] Russell Bryant <russell@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: Check
for the existence of the soxmix application on the target
platform and have the result available in autoconfig.h. (part of
issue #11589)
2007-12-19 Russell Bryant <russell@digium.com>
* Asterisk 1.4.16.1 released.
2007-12-19 17:29 +0000 [r93955] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
NULL.
2007-12-19 17:04 +0000 [r93949] Tilghman Lesher <tlesher@digium.com>
* channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
defined (Closes issue #11602)
2007-12-18 22:42 +0000 [r93764] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: FreeBSD also does not have byte swap
functions. Issue 11586, patch by sobomax.
2007-12-18 Russell Bryant <russell@digium.com>
* Asterisk 1.4.16 released.
2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
93667 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
| 2 lines Fixing AST-2007-027 (Closes issue #11119) ........
2007-12-18 17:02 +0000 [r93625] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Rework deadlock avoidance used in ast_write,
since it meant that agent channels which were being monitored had
one audio file recorded and one empty audio file saved. (closes
issue #11529, reported by atis patched by me)
2007-12-17 22:56 +0000 [r93381-93420] Jason Parker <jparker@digium.com>
* main/translate.c: What was I thinking when I wrote this
masterpiece? -1 + 1 = 0.. who woulda thunk it?.
2007-12-17 22:28 +0000 [r93377] Joshua Colp <jcolp@digium.com>
* main/utils.c: Do not try to access information about a lock when
printing out a trylock attempt. It is possible for the lock that
it references to no longer be valid. This would have caused
segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
Reported by: callguy (closes issue #11100) Reported by: callguy
2007-12-17 21:12 +0000 [r93336] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/time.h: Today is tomorrow's yesterday, and
yesterday's tomorrow is today, and tomorrow's tomorrow is the day
after tomorrow, so who cares if you recycle anyway? If this
confuses you, that's nothing compared to what this fixes. ;-)
2007-12-17 19:53 +0000 [r93291] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: We need to create the directory for a
voicemail user even if they are using IMAP storage since
greetings are stored in the filesystem. (closes issue #11388,
reported by spditner, patch by me inspired by a patch by
spditner)
2007-12-17 18:05 +0000 [r93250] Joshua Colp <jcolp@digium.com>
* channels/chan_zap.c: If a call is received with a called number
IE containing nothing go to the 's' extension. (closes issue
#9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
uploaded by Corydon76 (license 14)
2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming <kpfleming@digium.com>
* funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
res/Makefile, channels/Makefile, formats/Makefile: fix some
copy-and-paste leftovers
2007-12-17 07:15 +0000 [r93182] Olle Johansson <oej@edvina.net>
* channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
dependencies on res_monitor and res_features. I wonder if
Asterisk can run at all without res_features. My guess is that
there's propably a lot of more modules and the core that depends
on it. Reported by: caio1982 (closes issue #11574)
2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming <kpfleming@digium.com>
* formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
(removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata
required for menuselect and the rest of the build system is
extracted from the source files. Since I had a few hours to kill
on an airplane today, I decided to improve this situation... so
now the system caches the extracted metadata and uses it to build
the menuselect 'tree' as much as it can. The result of this is
that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from
the cache files. I also reduced the number of forked processes
required to do the metadata extraction; it was actually possible
to do most of what we needed in the Makefiles themselves without
using any shell scripts at all! On my laptop, these changes
resulted in an 80% decrease in the time required for the
'menuselect.makeopts' automatic check to occur after editing a
single source file. While doing this work I also cleaned up a few
minor things in the Makefiles, adding a check for 'awk' to the
configure script and changed all remaining places we use 'grep'
or 'awk' to use the ones found by the configure script, and
changed the 'prep_tarball' script to build the menuselect
metadata so that tarballs of Asterisk will include it and won't
require the user to wait while it is extracted after unpacking.
2007-12-14 17:36 +0000 [r93000] Russell Bryant <russell@digium.com>
* main/config.c: There are a lot of existing systems that #include
non-existent files. So, to make the transition to treating this
as an error a bit less painless, just issue a huge error message
for now. Then, later, we can reinstate the code that treats it as
a failure. (Thanks to philippel for the feedback)
2007-12-14 15:16 +0000 [r92937] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Up the length of the format on the SIP
channel since it can now be rather long. (closes issue #11552)
Reported by: francesco_r
2007-12-14 15:05 +0000 [r92934] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
state setting and overlap_task thread starting.
2007-12-14 15:01 +0000 [r92933] Tilghman Lesher <tlesher@digium.com>
* res/res_agi.c: Change help documentation to match actual behavior
(FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
(Closes issue #11548)
2007-12-14 01:24 +0000 [r92875] Mark Michelson <mmichelson@digium.com>
* include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
don't spam the CLI with messages about possible deadlocks.
Instead just print the intended single message every five
seconds. (closes issue 11537, reported and patched by dimas)
2007-12-13 21:28 +0000 [r92815] Tilghman Lesher <tlesher@digium.com>
* channels/chan_zap.c: Properly initialize polarity statuses, so
that they are detected properly. Reported by: julianjm Patch by:
julianjm (Closes issue #10238)
2007-12-13 20:13 +0000 [r92809] Jason Parker <jparker@digium.com>
* main/pbx.c: Make application help text a little more clear about
the use of extensions in a filename.
2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Prevent another potential fd leak
* apps/app_voicemail.c: Prevent a possible fd leak.
2007-12-13 00:11 +0000 [r92696] Jason Parker <jparker@digium.com>
* main/config.c, channels/chan_sip.c, channels/chan_h323.c,
channels/chan_iax2.c: If a typo is found in a config file, we
previous continued on with what was already loaded. We do not
want to do this (see bug below for details). This makes it so
that if a [ is found without a ], the entire config will fail,
and nothing in it will be loaded. Isue #10690.
2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming <kpfleming@digium.com>
* codecs/codec_zap.c: emit a warning message when we drop a G.729B
CNG frame destined for the transcoder
2007-12-12 21:15 +0000 [r92617] Jason Parker <jparker@digium.com>
* apps/app_meetme.c: Don't increment user count until after name
has been recorded (if enabled). Issue 11048, tested by pep.
2007-12-12 19:40 +0000 [r92556] Russell Bryant <russell@digium.com>
* res/res_features.c: resolve compiler warning
2007-12-12 17:46 +0000 [r92510] Mark Michelson <mmichelson@digium.com>
* res/res_features.c: Correctly detect where a dynamic feature was
activated. Before this patch, the channel which initiated the
bridge was always assumed to have been the one which activated
the dynamic feature. This patch corrects this. (closes issue
#11529, reported and patched by nic_bellamy)
2007-12-12 16:52 +0000 [r92463] Tilghman Lesher <tlesher@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: Test
directly for the API that fixed AST-2007-026, to ensure that
older versions of PostgreSQL are no longer acceptable. (Closes
issue #11526)
2007-12-12 16:08 +0000 [r92443] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Removing an unused variable.
2007-12-11 19:51 +0000 [r92363] Joshua Colp <jcolp@digium.com>
* main/global_datastores.c: Fix potential memory leak with the
dialed interfaces list if another memory allocation fails.
(closes issue #11507) Reported by: eliel Patches:
global_datastores.c.patch uploaded by eliel (license 64)
2007-12-11 17:42 +0000 [r92323] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixing autofill to be more accurate.
Specifically, if calls ahead of the current caller were ringing
members (but not yet bridged) there could be available members
and waiting callers who would not get matched up. The member
availability checker was correctly determining the number of
available members in this scenario, but the queue itself did not
parallelly reflect this status on the pending calls. This commit
corrects the issue. (closes issue #11459, reported by
equissoftware, patched by me)
2007-12-10 16:36 +0000 [r92204] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Add G729A as another possible payload name for G729.
Some devices use this instead of G729, which is perfectly normal
since the payload number itself is defined and can't be used by
anything else so the name doesn't matter that much. (closes issue
#11483) Reported by: revolution Patches: rtp.diff uploaded by
revolution (license 346)
2007-12-10 16:29 +0000 [r92202] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: If there are no members in a queue, then the
loop where the datastore for detecting duplicate dialed numbers
will be skipped, meaning the datastore isn't created. This means
that when we try to free it, there's a crash. This stops that
crash from occurring. (closes issue #11499, reported by slavon,
patched by eliel)
2007-12-10 16:13 +0000 [r92200] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: It is possible for nativeformats to contain
more then one codec, so print out multiple ones. (closes issue
#11366) Reported by: ovi
2007-12-10 14:04 +0000 [r92158] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Avoid reinvite race situations with two
Asterisks trying to reinvite each other in 1.4 and trunk. This
patch implements support for the 491 error code that Asterisk 1.4
generates on situations where we get an incoming INVITE and
already has one in progress. Thanks to mavetju for reporting and
to Raj Jain for an excellent explanation of the problem. Patch by
myself. Tested with 8 Asterisk servers connected to each other in
a training network. Closes issue #10481
2007-12-07 23:29 +0000 [r91890] Jason Parker <jparker@digium.com>
* main/dsp.c: We need to make sure we free the input frame if we
return a different frame in ast_dsp_process. Issue 11273, pointed
out by dimas, with a patch by eliel.
2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming <kpfleming@digium.com>
* codecs/codec_zap.c: even though Asterisk explicitly requests that
endpoints using G.729 do *not* use Annex B (silence detection and
comfort noise generation) some do anyway; the transcoder card
interface does not currently work properly with CNG frames, so
trim off the CNG before sending the data
2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant <russell@digium.com>
* main/utils.c: Make the lock protecting each thread's list of
locks it currently holds recursive. I think that this will fix
the situation where some people have said that "core show locks"
locks up the CLI. (related to issue #11080)
* include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
code. The ast_mutex_init() function had the mutex attribute
object marked as static. This means that multiple threads
initializing locks at the same time could step on each other and
end up with improperly initialized locks. (found when tracking
down locking issues related to issue #11080)
* include/asterisk/lock.h: I love fixing lock related errors in the
lock debugging code. That's about as ironic as it gets in
Asterisk programming land. Anyway, I spotted this bug while
trying to track down why systems are locking up and acting weird
in issue #11080. The mutex attribute object was marked as static
in this function when it should not have been.
* apps/app_dial.c: * Add channel locking around datastore
operations that expect the channel to be locked. * Document why
we don't record Local channels in the dialed interfaces list. *
Remove the dialed variable as it isn't needed. * Restructure some
code for clarity and coding guidelines stuff
* apps/app_queue.c: * Add channel locking around datastore
operations that expect the channel to be locked. * Document why
we don't record Local channels in the dialed interfaces list. *
Handle memory allocation failure. * Remove the dialed variable,
as it wasn't actually needed. * Tweak some formatting to conform
to coding guidelines.
* main/autoservice.c: * Add a bit more of a verbose comment as to
why a hangup frame needs to be queued up if autoservice gets a
NULL return from ast_read(). * Make the process of queueing the
hangup frame more efficient by putting the frame where it is
going to end up and avoiding some locking and extra memory
allocations and freeing.
2007-12-07 15:39 +0000 [r91737] Mark Michelson <mmichelson@digium.com>
* main/autoservice.c: Hangups that happen during autoservice were
not processed appropriately. This is because a hangup actually
causes a NULL frame to be received, not a hangup frame. Queueing
a hangup if we receive a NULL frame during autoservice corrects
this problem (closes issue #11467, reported by jmls, patched by
me)
2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant <russell@digium.com>
* apps/app_dial.c: Don't unlock the dialed_interfaces list until
we're done messing with the iterator.
* apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
from Queue() and Dial() again. There was a slight flaw in the
code to prevent call forwards from looping that caused this
problem. (related to issue #11486)
* apps/app_queue.c: Fix in an issue in the call forwarding handling
code that was causing crashes on every call into a queue. I'm not
entirely sure about the logic in this part of the code, so I want
to look at it some more tomorrow. However, this makes it safe and
keeps it from crashing. (closes issue #11486, reported by adamg,
patched by me)
2007-12-07 00:52 +0000 [r91637] Tilghman Lesher <tlesher@digium.com>
* main/rtp.c: At the end of a call, when we're reporting, RTCP may
already be partially torn down, so check for NULL dereference
Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
2007-12-06 20:25 +0000 [r91541] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: IMAP storage did not honor the maxmsg
setting in voicemail.conf, and it also had the possibility of
crashing if a user had more than 256 messages in their voicemail.
This patch kills two birds with one stone by adding maxmsg
support and also setting a hard limit on the number of messages
at 255 so that the crashes cannot happen. (closes issue #11101,
reported by Skavin, patched by me)
2007-12-06 19:11 +0000 [r91501] Russell Bryant <russell@digium.com>
* main/loader.c, include/asterisk/module.h: Add a new module flag
to indicate that a build sum is present. Modules built against
older Asterisk 1.4 headers will now load properly with just a
warning indicating that they are old and may cause problems.
(patch by paravoid)
2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp <jcolp@digium.com>
* main/udptl.c: Fix various in the udptl implementation. It could
return empty modem frames, have an incorrect sequence number on
packets, and display the wrong sequence number in the debug
messages. (closes issue #11228) Reported by: Cache Patches:
udptl-4.patch uploaded by dimas (license 88)
* channels/chan_sip.c: Add support for accepting and sending T.38
in the initial INVITE. (closes issue #9402) Reported by: thdei
2007-12-06 12:54 +0000 [r91366] Olle Johansson <oej@edvina.net>
* main/loader.c, include/asterisk/logger.h, main/logger.c: Make
sure logger is reloaded at general reload in the cli. (Discovered
during Asterisk training in Portugal)
2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Reverting extra stuff I didn't mean to
commit
* apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
did not properly handle the case where only a label was provided.
This was due to the fact that the answering channel did not have
an extension set, so ast_parseable_goto would fail. This fix
eliminates the call to ast_parseable_goto on the answering
channel since it is a wasteful call. The answering channel and
the calling channel are both directed to the same extension and
context, just different priorities, so we can just copy the
values from the calling channel to the answering channel and
increment the answering channel's priority. (closes issue #11382,
reported by jon, patch by me with correction by jon)
2007-12-05 21:38 +0000 [r91237] Tilghman Lesher <tlesher@digium.com>
* sounds/Makefile: Upgrade to the latest version of extra sounds
2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant <russell@digium.com>
* main/threadstorage.c: Make the lock in the threadstorage
debugging code untracked to avoid a deadlock on thread
destruction. (closes issue #11207) Reported by: ys Patches:
threadstorage.c.diff uploaded by ys (license 281) Also fixes an
open bug report: (closes issue #11446)
* main/utils.c: When DEBUG_THREADS is enabled, we only have the
details about who is holding a lock that we are waiting on for a
mutex, not rwlocks. This should fix the problem where people have
reported "core show locks" crashing sometimes.
* include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
reported as happening on Mac systems. It turns out that the
problem was the Mac version of the ast_atomic_fetchadd_int()
function. The Mac atomic add function returns the _new_ value,
while this function is supposed to return the old value. So, the
crashes happened on unreferencing objects. If the reference count
was decreased to 1, ao2_ref() thought that it had been decreased
to zero, and called the destructor. However, there was still an
outstanding reference around. (closes issue #11176) (closes issue
#11289)
* include/asterisk/file.h, configure,
include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/compiler.h: Modify file.h to maintain API
compatibility with earlier versions. If a recent compiler is
being used, then a warning will show up for any modules still
using the old name "private" instead of "_private". (patch
suggested by paravoid)
* main/pbx.c: Make some changes to some additions I made recently
for doing channel autoservice when looking up extensions. This
code was added to handle the case where a dialplan switch was in
use that could block for a long time. However, the way that I
added it, it did this for all extension lookups. However, lookups
in the in-memory tree of extensions should _not_ take long enough
to matter. So, move the autoservice stuff to be only around
executing a switch.
2007-12-04 17:28 +0000 [r90876] Jason Parker <jparker@digium.com>
* main/channel.c: If we fail to create a channel after allocating a
timing fd, we need to make sure to close it. Issue 11454, patch
by eliel.
2007-12-04 05:29 +0000 [r90798] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c: Fix build issue on the build cluster.
2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/compat.h: Solaris requires the inclusion of
sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
snuffy,tilghman (Closes issue #11430)
* res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
NULL deref could result Reported by: xrg Patch by: tilghman
(Closes issue #11387)
2007-12-03 23:12 +0000 [r90735] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, main/channel.c, main/global_datastores.c
(added), channels/chan_local.c, main/Makefile,
include/asterisk/channel.h, include/asterisk/global_datastores.h
(added), apps/app_queue.c: A big one... This is the merge of the
forward-loop branch. The main change here is that call-forwards
can no longer loop. This is accomplished by creating a datastore
on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is
created inherits the datastore. If, through this progression of
forwards and datastore inheritance, a device is attempted to be
dialed a second time, it will simply be skipped and a warning
message will be printed to the CLI. After the dialing has been
completed, the datastore is detached from the channel and
destroyed. This change also introduces some side effects to the
code which I shall enumerate here: 1. Datastore inheritance has
been backported from trunk into 1.4 2. A large chunk of code has
been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been
requested but before it has been called. This was removed because
call-forwarding still works fine without it, it makes the code
less error-prone should it need changing, and it made this set of
changes much less painful to just have the forwarding handled in
one place in each module. 3. Two new files, global_datastores.h
and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in
either app_dial or app_queue, so they need a common place to find
the datastore info. This approach was taken in case similar
datastores are needed in the future, there will be a common place
to add them.
2007-12-03 22:06 +0000 [r90696] Jason Parker <jparker@digium.com>
* apps/app_meetme.c: Make sure we always close the conference fd if
we have an open one. Issue 11383, reported by markmhy, patch by
eliel.
2007-12-03 20:59 +0000 [r90639] Mark Michelson <mmichelson@digium.com>
* channels/chan_mgcp.c: Changing some bad logic when calculating
the interdigit timeout. (closes issue #11402, reported and
patched by eferro)
2007-12-03 20:51 +0000 [r90607] Jason Parker <jparker@digium.com>
* res/res_features.c: Fix crash in ParkAndAnnounce application.
Issue #11436, reported by lytledd, patch by eliel.
2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Do not create a smoother for G723.1 frames, they need
to be left alone to their native 20/24 byte size.
* .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
the indication currently playing on a channel when a masquerade
operation happens. (issue #BE-88)
2007-12-03 18:20 +0000 [r90546] Jason Parker <jparker@digium.com>
* channels/chan_iax2.c: Only log debug messages if debug is
enabled. Closes issue #11416, patch by casper.
2007-12-02 18:18 +0000 [r90470] Russell Bryant <russell@digium.com>
* apps/app_queue.c: The other day when I went through making
changes as a result of the ao2_link() change, I added some code
to set pointers to NULL after they were unreferenced. This
pointed out that in this place, the object was unreferenced
before the code was done using it. So, move the unref down a
little bit. (crash reported by jmls on IRC)
2007-12-02 09:34 +0000 [r90432] Tilghman Lesher <tlesher@digium.com>
* main/autoservice.c: Clarify the return value on autoservice.
Specifically, if you started autoservice and autoservice was
already on, it would erroneously return an error. Reported by:
adiemus Patch by: dimas (Closes issue #11433)
2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant <russell@digium.com>
* main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
ao2_link(). Previously, in inherited a reference. Now, it
automatically increases the reference count to reflect the
reference that is now held by the container. This was done to be
more consistent with ao2_unlink(), which automatically releases
the reference held by the container. It also makes it so it is no
longer possible for a pointer to be invalid after ao2_link()
returns.
* include/asterisk/astobj2.h: Add some notes on the behavior of
ao2_unlink() after a discussion with Tilghman
2007-11-30 14:43 +0000 [r90269] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix locking issues under one legged replaces
scenarios. (closes issue #11420) Reported by: irroot Patches:
chan_sip_oneleg.patch uploaded by irroot (license 52)
2007-11-30 00:16 +0000 [r90231] Mark Michelson <mmichelson@digium.com>
* channels/chan_mgcp.c: Clear the DTMF buffer if the call times
out. (closes issue #11418, reported and patched by eferro)
2007-11-29 Russell Bryant <russell@digium.com>
* Asterisk 1.4.15 released.
2007-11-29 19:48 +0000 [r90166] Tilghman Lesher <tlesher@digium.com>
* cdr/cdr_pgsql.c: Properly escape cdr->src and cdr->dst and ensure
we use thread-safe escaping (Fixes AST-2007-026)
2007-11-29 19:38 +0000 [r90163] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This patch handles the case where a queue
member with a negative penalty is added via the manager. If a
negative value is submitted for a member penalty, we set it to 0.
(closes issue #11411, reported and patched by Laureano)
2007-11-29 19:24 +0000 [r90154-90160] Tilghman Lesher <tlesher@digium.com>
* res/res_config_pgsql.c: Properly escape input buffers (Fixes
AST-2007-025)
* formats/format_g726.c, include/asterisk/file.h,
formats/format_wav.c, formats/format_pcm.c,
formats/format_ogg_vorbis.c, main/file.c, formats/format_h263.c,
formats/format_h264.c, formats/format_wav_gsm.c: Use of "private"
as a field name in a header file messes with C++ projects
Reported by: chewbacca Patch by: casper (Closes issue #11401)
* sounds/Makefile: Upgrade the core sounds release version
2007-11-29 00:36 +0000 [r90142-90147] Russell Bryant <russell@digium.com>
* funcs/func_callerid.c: fix some formatting i accidentally changed
* funcs/func_callerid.c, main/channel.c,
include/asterisk/channel.h: This set of changes is to make some
callerID handling thread-safe. The ast_set_callerid() function
needed to lock the channel. Also, the handlers for the CALLERID()
dialplan function needed to lock the channel when reading or
writing callerid values directly on the channel structure.
* include/asterisk/file.h, main/file.c: Merge a change from
team/russell/chan_refcount ... This makes ast_stopstream()
thread-safe.
2007-11-28 22:59 +0000 [r90101] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c: Fix a few memory leaks. (closes issue #11405)
Reported by: eliel Patches: load_realtime.patch uploaded by eliel
(license 64)
2007-11-28 22:30 +0000 [r90098] Kevin P. Fleming <kpfleming@digium.com>
* configs/users.conf.sample, main/manager.c: it is impossible to
set permissions for manager accounts created by users.conf
(reported internally, patched by me)
2007-11-28 22:08 +0000 [r89999-90059] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: Removing some seemingly pointless code. This sets a
channel variable for every priority executed in the dialplan if
you have debug set to anything non-zero. This seems pointless due
to the fact that these channel variables are not referenced
anywhere else in the code and their names are esoteric enough
that they would not be practical to reference in the dialplan.
Plus the fact that this behavior isn't documented anywhere means
that the change is not likely to cause any disruption. If
anything, this may actually cause a slight performance increase
if running with debug on. The motivating influence for this code
change is the eventwhencalled option for queues. If set to vars,
all channel variables will be output to the manager. These
unnecessary channel variables make the output a lot more
difficult to deal with.
* apps/app_voicemail.c: Recording greetings when using IMAP storage
was causing zero-length files to be stored. Since greetings are
not retrieved from IMAP anyway, it is pointless to attempt
storing them there. (closes issue #11359, reported by spditner,
patched by me)
2007-11-28 00:20 +0000 [r89839-89893] Russell Bryant <russell@digium.com>
* main/pbx.c, include/asterisk/pbx.h: - update documentation for
some of the goto functions to note that they handle locking the
channel as needed - update ast_explicit_goto() to lock the
channel as needed
* main/autoservice.c: Don't do frame processing if ast_read()
returned NULL.
* apps/app_queue.c: Instead of depending on the return value of
ast_true(), explicitly set the eventwhencalled variable to 1.
* main/pbx.c: Don't start/stop autoservice in
pbx_extension_helper() unless a channel exists
2007-11-27 23:10 +0000 [r89837] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Two changes with regards to the
'eventwhencalled' option of queues.conf 1) Due to some signed vs.
unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes'
did exactly the same thing. Thus the sign change of the ast_true
call. 2) The vars2manager function overwrote a \n for every
channel variable it parsed, resulting in bizarre output for the
channel variables. This patch remedies this. (related to issue
#11385, however I'm not sure if this will actually be enough to
close it)
2007-11-27 21:45 +0000 [r89790] Russell Bryant <russell@digium.com>
* main/autoservice.c, main/pbx.c: Merge changes from
team/russell/autoservice_1.4 This set of changes fixes an issue
that was reported to me on IRC yesterday. The user, d1mas, was
using chan_zap for incoming calls and was having DTMF recognition
issues in some situations. Specifically, he noticed that the
problem occurred when using DISA or WaitExten. He also noticed
that when using Read, the problem did not occur. His system also
used DUNDi for dialplan lookups. So, he theorized that if the
DUNDi lookups blocked for some period of time, that audio from
the zap channel could get lost. If the audio got lost, then it
wouldn't be run through the DTMF detector, and digits could get
lost. He was correct, and the following set of changes fixes the
problem. However, the changes go a little bit further than what
was necessary to fix this exact problem. 1) I updated
pbx_extension_helper() to autoservice the associated channel to
handle cases where extension lookups may take a long time. This
would normally be a dialplan switch that does some lookup over
the network, such as the DUNDi or IAX2 switches. This ensures
that even while a DUNDi lookup is blocking, the channel will be
continuously serviced. 2) I made a change to the autoservice
code. This is actually something that has bothered me for a long
time. When a channel is in autoservice, _all_ frames get thrown
away. However, some frames really shouldn't be thrown away. The
most notable examples are signalling (CONTROL) frames, and DTMF.
So, this patch queues up important frames while a channel is in
autoservice. When autoservice is stopped on the channel, the
queued up frames get stuck back on the channel so that they can
get processed instead of thrown away. 3) I made another change to
the autoservice code to handle the case where autoservice is
started on channels recursively. Previously, you could call
ast_autoservice_start() multiple times on a channel, and it would
stop the first time ast_autoservice_stop() gets called. Now, it
will ensure that autoservice doesn't actually stop until the
final call to ast_autoservice_stop().
2007-11-27 20:22 +0000 [r89727] Mark Michelson <mmichelson@digium.com>
* res/res_config_pgsql.c: Changing some calls from free() to
ast_free() since they were allocated with ast_calloc(). (closes
issue #11390, reported and patched by Laureano)
2007-11-27 20:16 +0000 [r89701-89709] Kevin P. Fleming <kpfleming@digium.com>
* main/app.c: on second thought... revert all the other changes
i've made in app options parsing leaving only one: if an empty
argument is supplied for an option, set that argument pointer to
point to an empty string rather than NULL, so that the
application can do normal checks on it without worrying about it
being NULL
* main/app.c: generate a warning when an application option that
requires an argument is ignored due to lack of an argument
2007-11-27 16:12 +0000 [r89634] Russell Bryant <russell@digium.com>
* configs/voicemail.conf.sample: Add a note to the sample voicemail
config noting that when using IMAP storage, only the first format
specified will be attached to the message.
2007-11-27 15:38 +0000 [r89631] Tilghman Lesher <tlesher@digium.com>
* funcs/func_env.c: Default result of STAT should be "0" not "".
Reported via the -users mailing list, fixed by me.
2007-11-27 15:23 +0000 [r89624-89630] Olle Johansson <oej@edvina.net>
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: If we
get a codec offer using a well-known payload type, but using it
for another codec that we don't know, Asterisk did not remove
that codec from the list. With this patch, we remove the codec
from audio and video rtp objects and deny it ever existed. Thanks
to lasse for testing. (closes issue #11376) Reported by: lasse
Patches: bug11376.txt uploaded by oej (license 306) Tested by:
lasse
* configs/sip.conf.sample: Clarify limitonpeers=yes (closes issue
#11304) Reported by: pj
2007-11-27 06:24 +0000 [r89622] Steve Murphy <murf@digium.com>
* apps/app_dial.c, main/cdr.c, configs/cdr.conf.sample,
include/asterisk/cdr.h: closes issue #11379; OK, this is an
attempt to make both sides happy. To the cdr.conf file, I added
the option 'unanswered', which defaults to 'no'. In this mode,
you will see a cdr for a call, whether it was answered or not.
The disposition will be NO ANSWER or ANSWERED, as appropriate.
The src is as you'd expect, the destination channel will be one
of the channels from the Dial() call, usually the last in the
list if more than one chan was specified. With unanswered set to
'yes', you will still see this cdr entry in both cases. But in
the case where the dial timed out, you will also see a cdr for
each line attempted, marked NO ANSWER, with no destination
channel name. The new option defaults to 'no', so you don't see
the pesky extra cdr's by default, and you will not see the
irritating 'not posted' messages.
2007-11-26 23:10 +0000 [r89616-89618] Mark Michelson <mmichelson@digium.com>
* apps/app_playback.c: After issuing a "say load new", if a caller
hangs up during the middle of playback of a number, app_playback
will continue to try to play the remaining files. With this
change, no more files will be played back upon hangup. (closes
issue #11345, reported and patched by IgorG)
* apps/app_playback.c: After issuing a "say load new" tons of
warning messages are printed out to the CLI every time do_say in
app_playback is called. Removing these warnings
2007-11-26 21:10 +0000 [r89599-89610] Joshua Colp <jcolp@digium.com>
* main/dial.c: Fix issues with async dialing with an application
executing. The application has to be terminated and control
returned to the thread before hanging things up. (issue #BE-252)
* res/res_features.c: Add module counting removal for error
conditions. (closes issue #11333) Reported by: Laureano Patches:
res_features_v2.c.patch uploaded by Laureano (license 265)
2007-11-26 17:41 +0000 [r89594] Russell Bryant <russell@digium.com>
* main/pbx.c: Add channel locking to a function that needed to be
doing it. This is just a little something I noticed while working
on a completely unrelated issue.
2007-11-26 17:36 +0000 [r89587-89592] Joshua Colp <jcolp@digium.com>
* pbx/pbx_config.c: Use ast_free to free memory, or else we shall
implode if MALLOC_DEBUG is enabled. (closes issue #11347)
Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys
(license 281)
* apps/app_mixmonitor.c: Close the audio file before sending it to
the post processing application. (closes issue #11357) Reported
by: reformed Patches: mixmonitor.patch uploaded by reformed
(license 330)
2007-11-26 17:20 +0000 [r89586] Kevin P. Fleming <kpfleming@digium.com>
* main/app.c: when parsing application options that take arguments,
don't indicate that the option was supplied unless a
non-zero-length argument was found for it
2007-11-26 15:48 +0000 [r89580] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Revert vmu->email back to an empty string
if it was empty when imap_store_file was called. This prevents
sending a duplicate e-mail. (closes issue #11204, reported by
spditner, patched by me)
2007-11-26 15:34 +0000 [r89571-89577] Joshua Colp <jcolp@digium.com>
* main/channel.c: If channel allocation fails because the alert
pipe could not be created also free the scheduler context.
(closes issue #11355) Reported by: eliel Patches:
main.channel.c.patch uploaded by eliel (license 64)
* apps/app_meetme.c: When unloading app_meetme destroy any auto
created contexts created by SLA. (closes issue #11367) Reported
by: eliel
2007-11-25 17:17 +0000 [r89559] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c, configs/res_odbc.conf.sample,
include/asterisk/res_odbc.h, res/res_config_odbc.c: We previously
attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all
databases, since on databases which natively use the backslash as
a delimiter, the backslash itself needs to be delimited, but on
other databases that have no delimiter, backslashing the
backslash causes an error. So the only solution that I can come
up with is to create an option in res_odbc that explicitly
specifies whether or not backslash is a native delimiter. If it
is, we use it natively; if not, we use the ESCAPE clause to make
it one. Reported by: elguero Patch by: tilghman (Closes issue
#11364)
2007-11-24 16:59 +0000 [r89534-89545] Tilghman Lesher <tlesher@digium.com>
* res/res_adsi.c: Free some frames that would otherwise leak on
error. Reported by: Laureano Patch by: Laureano,tilghman (Closes
issue #11351)
* apps/app_voicemail.c, main/app.c: Currently, zero-length
voicemail messages cause a hangup in VoicemailMain. This change
fixes the problem, with a multi-faceted approach. First, we do
our best to avoid these messages from being created in the first
place, and second, if that fails, we detect when the voicemail
message is zero-length and avoid exiting at that point. Reported
by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083)
* main/manager.c: Up until this point, the XML output of the
manager has been technically invalid, due to the repetition of
certain parameters in a single event. This caused various issues
for XML parsers, some of which refused to parse at all, given the
invalidity of the rendered XML. So this commit fixes the XML
output, ensuring that each entity parameter has a unique name,
thus ensuring valid XML. Reported by: msetim Patch by: tilghman
(Closes issue #10220)
* res/res_config_odbc.c: Use ESCAPE clause for the first parameter,
not just 2nd-Nth parameters. Reported by: apsaras Patch by:
tilghman (Closes issue #11353)
2007-11-22 17:29 +0000 [r89527] Russell Bryant <russell@digium.com>
* configs/agents.conf.sample: mvanbaak pointed out a spelling error
in this sample configuration file. While I was at it, I went
ahead and tweaked it a little bit more.
2007-11-21 19:27 +0000 [r89493-89495] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fix a small error I made in my previous commit
* apps/app_queue.c: Changing an inaccurate debug message to be less
inaccurate. Under the circumstances, this message would always
report that there were 0 members available, even though that may
not be true.
2007-11-21 18:59 +0000 [r89491] Terry Wilson <twilson@digium.com>
* res/res_features.c: If a channel gets masqueraded in the middle
of a park, don't play the announcement to the masqueraded
channel, and dial back to the original channel on timeout.
2007-11-20 19:16 +0000 [r89461-89462] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/module.h: re-doxygen some comments
* main/loader.c, include/asterisk/module.h,
build_tools/make_buildopts_h: bring back compile-option checking
when loading modules, only this time use a string-based storage
and comparison mechanism because it is easier to support on other
platforms
2007-11-20 17:50 +0000 [r89457] Mark Michelson <mmichelson@digium.com>
* main/pbx.c: According to comments in main/pbx.c, it is essential
that if we are going to lock the conlock as well as the hints
lock, it must be locked in that respective order. In order to
prevent a potential deadlock, we need to lock the conlock prior
to locking the hints lock in ast_hint_state_changed (see the call
stack example on issue #11323 for how this can happen). (closes
issue #11323, reported by eelcob, suggestion for patch by eelcob,
patch by me)
2007-11-20 15:22 +0000 [r89450] Steve Murphy <murf@digium.com>
* doc/queues-with-callback-members.txt: closes issue #11324; break
statements missing in switch cases.
2007-11-20 13:40 +0000 [r89445] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: added RR patch from iroot #10908, thanks.
2007-11-19 15:53 +0000 [r89416-89419] Joshua Colp <jcolp@digium.com>
* res/res_features.c: Print out the correct filename
(features.conf) in the log message when parkpos options are
incorrect. (closes issue #11295) Reported by: Laureano Patches:
res_features.c.patch uploaded by Laureano (license 265)
* doc/localchannel.txt: Clarify documentation a bit, include that a
frame has to pass through the core in order for the Local channel
optimization to happen. (closes issue #11246) Reported by: jon
2007-11-16 Russell Bryant <russell@digium.com>
* Asterisk 1.4.14 released.
2007-11-16 22:26 +0000 [r89339] Russell Bryant <russell@digium.com>
* main/loader.c, include/asterisk/module.h,
build_tools/make_buildopts_h: Temporarily revert revision 89325,
which added md5 magic for keeping track of what build options
were used. We agreed that we should remove this before making a
1.4 release, and then we can put it back in. Then, we can take a
month or so to play around with it to get it how we want it.
2007-11-16 16:47 +0000 [r89325] Kevin P. Fleming <kpfleming@digium.com>
* main/loader.c, include/asterisk/module.h,
build_tools/make_buildopts_h: To help combat problems where
people build external modules (asterisk-addons or others) and
then change the build options of the Asterisk build in a way that
makes the incompatible without warning, this commit introduces an
MD5 signature of the important build-time options and includes
that signature into modules when they are built. When the loader
loads one of these modules and notices the problem, it will emit
a warning to console and refuse to initialize the module, as
doing so could cause the system to be unstable or even crash. If
you upgrade to this version of Asterisk, you must rebuild *all*
of your modules that came from other sources before trying to run
this version. If you are using Digium's G.729 binary codec
module, you will need v33 or newer.
2007-11-16 15:28 +0000 [r89323] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Make realtime queues accessible from the
QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and
patched by atis, with small modifications from me)
2007-11-15 18:37 +0000 [r89298-89302] Tilghman Lesher <tlesher@digium.com>
* Makefile: Start Asterisk in Debian at a more reasonable time
(since zaptel is at level 20)
* channels/misdn/isdn_lib.c: Fix an uninitialized memory read found
by valgrind
* channels/chan_iax2.c: Yet another memory corruption issue.
Reported by: atis Patch by: tilghman Fixes issue #10923
2007-11-15 17:19 +0000 [r89296] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Update the SLAStation application to account
for the case where the SLA thread has a call out to the station,
but the user has pressed a line button to answer the call instead
of picking up the handset. If they do, the phone sends out a new
INVITE. So, the SLAStation app must check to see if it is picking
up a ringing trunk, and ensure that the other stations stop
ringing. (reported internally, patched by me, tested by mogorman)
2007-11-15 14:57 +0000 [r89286-89288] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Undoing previous commit since I realize it was
wrong
* main/manager.c: Adding a missing mutex unlock. (closes issue
11256, reported and patched by ys)
2007-11-15 11:26 +0000 [r89280-89281] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't send re-invites during pending INVITE
transactions. Patch by one47 - thanks! Closes issue #9305
* channels/chan_sip.c: Improve support for multipart messages. Code
by gasparz, changes by me (mostly formatting). Thanks, gasparz!
Closes issue #10947
2007-11-14 23:23 +0000 [r89275] Tilghman Lesher <tlesher@digium.com>
* main/app.c: When a recording ends with '#', we are improperly
trimming an extra 200ms from the recording. Reported by: sim
Patch by: tilghman Closes issue #11247
2007-11-14 01:15 +0000 [r89260] Joshua Colp <jcolp@digium.com>
* main/srv.c: Return the proper value when the srv_callback
function executes properly. (closes issue #11240) Reported by:
jtodd
2007-11-13 21:07 +0000 [r89248-89254] Jason Parker <jparker@digium.com>
* channels/chan_zap.c, channels/chan_iax2.c: Fix building on newer
systems which require a third arg to open() when using O_CREAT.
Issue 11238, reported by puzzled.
* res/res_features.c: Revert change from revision 67064. It is
documented behavior that if a parking extension already exists
while using PARKINGEXTEN, dialplan execution will continue. If
blind transferring to a Park with PARKINGEXTEN, you must keep
this in mind, and handle the failure yourself. Issue 11237,
reported by jon.
2007-11-13 17:34 +0000 [r89246] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: If we set a value for qualify, we should
actually pay attention to it, instead of overriding the value
2007-11-13 16:02 +0000 [r89241] Mark Michelson <mmichelson@digium.com>
* apps/app_mixmonitor.c: Reverting commit made in revision 89205
since it is unnecessary. Thanks to Kevin for pointing this out
2007-11-13 13:51 +0000 [r89239] Tilghman Lesher <tlesher@digium.com>
* main/utils.c: Debugging is running into the 16-lock limit.
Increase to avoid. (This define is only effective when debugging
is turned on, so there's no effect for most installations.)
2007-11-13 00:56 +0000 [r89205] Mark Michelson <mmichelson@digium.com>
* apps/app_mixmonitor.c: Some sanity checking for MixMonitor. If
only 1 argument is given, then the args.options and
args.post_process strings are uninitialized and could contain
garbage. This change handles this situation properly by only
using arguments that we have parsed.
2007-11-12 20:46 +0000 [r89194] Jason Parker <jparker@digium.com>
* main/pbx.c: Fix a typo pointed out by De_Mon on #asterisk-dev
2007-11-12 20:16 +0000 [r89184-89191] Tilghman Lesher <tlesher@digium.com>
* main/config.c: If two config writes collide, file corruption
could result. Use a mkstemp() file, instead. Reported by:
paravoid Patch by: tilghman Closes issue #10781
* main/channel.c, channels/chan_sip.c: Fix two cases of memory
corruption caused by background threads. Reported by: atis Patch
by: tilghman Fixes issue #10923
2007-11-12 11:26 +0000 [r89169-89173] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, configs/misdn.conf.sample: if we're NT and
no number was dialed and overlapdial is set, we wait for the ISDN
timeout instead of starting our own timer. added a comment for
the misdn.conf.sample for the overlapdial config option.
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c: added
restart all interfaces Restart_Indicator, to automatically send a
RESTART after the L2 of a PTP Port comes up. Also fixed some
places where we have send a RELEASE without need for it.
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
state/event issue with overlapdial=yes when no extension matched.
removed the general sending of a RELEASE_COMPLETE when we receive
a RELEASE, this is done by mISDNuser/mISDN. This makes it
possible to use asterisk-1.4 with mISDN trunk, but requires users
of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6
(when using the NT mode at all)
* channels/misdn/isdn_lib.c: fixed the support for CW and therefore
for the reject_cause option.
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/misdn/isdn_lib.h, channels/chan_misdn.c,
channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
aded ntkeepcalls option, to avoid droÃpping calls when the L2
goes down on a PTP link. There are some pbx which do turn off the
L1 for a very short while and restart it immediately. normally
T310 should be started and after 10 seconds or so the calls
should be dropped, this is a simple fix wihtout this timer.
2007-11-08 23:52 +0000 [r89125] Jason Parker <jparker@digium.com>
* main/say.c: Properly say the seconds here.. Issue 11203, fix
described by vma.
2007-11-08 21:00 +0000 [r89119] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Rework of the commit I made yesterday to use
the already built-in ast_uri_decode function as opposed to my
home-rolled one. Also added comments. Thanks to oej for pointing
me in the right direction
2007-11-08 18:45 +0000 [r89115] Jason Parker <jparker@digium.com>
* configs/res_odbc.conf.sample: Avoid warnings on load when using
sample configuration files. Issue 11195, patch by eliel.
2007-11-08 16:47 +0000 [r89111] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: I made this same adjustment in trunk to fix
a bug, and it makes sense to do it in 1.4 as well. If an
imapfolder is specified in voicemail.conf, don't ever explicitly
connect to INBOX since it may not exist.
2007-11-08 05:26 +0000 [r89105] Kevin P. Fleming <kpfleming@digium.com>
* main/srv.c: fix a glaring bug in the new SRV record handling that
would cause incorrect weight sorting
2007-11-08 04:55 +0000 [r89103] Tilghman Lesher <tlesher@digium.com>
* doc/valgrind.txt: Typo
2007-11-08 02:26 +0000 [r89095-89101] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Do not add a sip: to the beginning of the To
URI unless needed. (closes issue #10756) Reported by: goestelecom
* channels/chan_sip.c: Improve the devicestate logic for multiple
devices. If any are available then the extension is considered
available. (closes issue #10164) Reported by: nic_bellamy
Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic
(license 299)
* channels/chan_sip.c: Add support for allowing one outgoing
transaction. This means if a response comes back out of order
chan_sip will still handle it. I dream of a chan_sip with real
transaction support. (closes issue #10946) Reported by: flefoll
(closes issue #10915) Reported by: ramonpeek (closes issue #9567)
Reported by: atca_pres
* channels/chan_sip.c: If callerid is configured in sip.conf use
that for checking the presence of an extension in the dialplan.
(closes issue #11185) Reported by: spditner
2007-11-07 23:39 +0000 [r89093] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: The member refcount must be incremented, to
avoid using it after deallocation. A huge thanks go to lvl- for
patiently providing the necessary valgrind output that was
necessary to finding this problem of memory corruption. Reported
by: lvl- Patch by: tilghman Closes issue #11174
2007-11-07 22:40 +0000 [r89090] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: This patch makes it possible for SIP phones
to dial extensions defined with '#' characters in extensions.conf
AND maintain their escaped characters when forming URI's (closes
issue #10681, reported by cahen, patched by me, code review by
file)
2007-11-07 21:40 +0000 [r89088] Steve Murphy <murf@digium.com>
* cdr/cdr_tds.c, pbx/pbx_ael.c, res/res_jabber.c: In response to
10578, I just ran 1.4 thru valgrind; some of the config leakage
I've already fixed, but it doesn't hurt to double check. I found
and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major,
tho.
2007-11-07 15:56 +0000 [r89085] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Fixing a segfault in the manager "core show
channels concise" command. (closes issue #11183, reported by arnd
and patched by ys)
2007-11-07 04:07 +0000 [r89079] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.ael.sample: Suppress AEL warnings on load.
Reported by: eliel Patch by: eliel Closes issue #11178
2007-11-06 20:18 +0000 [r89053] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Fix init_classes() so that classes that
actually do have files loaded aren't treated as empty, and
immediately destroyed ...
2007-11-06 19:09 +0000 [r89046] Jason Parker <jparker@digium.com>
* codecs/codec_zap.c: Correctly set the total number of channels
from a zaptel transcoder board. SPD-49, patch by Matthew
Nicholson.
2007-11-06 19:09 +0000 [r89045] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: We went to the trouble of creating a
method of tracking failed trylocks, then never turned it on
(oops).
2007-11-06 18:53 +0000 [r89042] Olle Johansson <oej@edvina.net>
* main/tdd.c: Bug fixes to tdd support in zaptel.
2007-11-06 18:20 +0000 [r89037] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: If someone were to delete the files used
by an existing MOH class, and then issue a reload, further use of
that class could result in a crash due to dividing by zero. This
set of changes fixes up some places to prevent this from
happening. (closes issue #10948) Reported by: jcomellas Patches:
res_musiconhold_division_by_zero.patch uploaded by jcomellas
(license 282) Additional changes added by me.
2007-11-06 17:52 +0000 [r89036] Steve Murphy <murf@digium.com>
* main/config.c: closes issue #8786 - where the [catname](!) and
[catname](othercat1,othercat2,...) notation gets dropped across a
ConfigUpdate (or any other thing that would cause a config file
to be written). While I was at it, I also cleaned up some of the
destroy routines to free up comments, which was not being done.
Made sure the new struct I introduced is also cleaned up properly
at destruction time. My code handles multiple template
inclusions. Many thanks to ssokol for his patch, which, while not
literally used in the final merge, served as a foundation for the
fix.
2007-11-06 17:08 +0000 [r88994-89032] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make it so that if a peer is determined to
be unreachable using qualify their devicestate will report back
unavailable. (closes issue #11006) Reported by: pj
* channels/chan_zap.c: Fix improbable but possible memory leaks in
chan_zap. (closes issue #11166) Reported by: eliel Patches:
chan_zap.c.patch uploaded by eliel (license 64)
2007-11-06 13:50 +0000 [r88931] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: Remove some checks to see if locks are
initialized from the non-DEBUG_THREADS versions of the lock
routines. These are incorrect for a number of reasons: - It
breaks the build on mac. - If there is a problem with locks not
getting initialized, then the proper fix is to find that place
and fix the code so that it does get initialized. - If additional
debug code is needed to help find the problem areas, then this
type of things should _only_ be put in the DEBUG_THREADS
wrappers.
2007-11-06 02:52 +0000 [r88862] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/srv.h: update comment to match the state of the
code
2007-11-05 23:29 +0000 [r88826] Mark Michelson <mmichelson@digium.com>
* main/channel.c: Reworked deadlock avoidance in __ast_read.
Restored audio to callback agents. (closes issue #11071, reported
by callguy, patched by me, tested by callguy and Ted Brown)
2007-11-05 22:07 +0000 [r88709-88805] Russell Bryant <russell@digium.com>
* main/pbx.c, include/asterisk/pbx.h: After seeing crashes related
to channel variables, I went looking around at the ways that
channel variables are handled. In general, they were not handled
in a thread-safe way. The channel _must_ be locked when reading
or writing from/to the channel variable list. What I have done to
improve this situation is to make pbx_builtin_setvar_helper() and
friends lock the channel when doing their thing. Asterisk API
calls almost all lock the channel for you as necessary, but this
family of functions did not. (closes issue #10923, reported by
atis) (closes issue #11159, reported by 850t)
* channels/chan_sip.c: When traversing the list of channel
variables here in transmit_invite(), the asterisk channel must be
locked, as this data may change at any time. (I have seen
numerous reports of crashes related to the handling of channel
variables. There are a couple of issues on the bug tracker
related to it, but it has also been noted on IRC and mailing
lists. So, I am finding and fixing some places where channel
variables are handled improperly.)
* channels/chan_sip.c: Fix up some indentation.
* main/srv.c, include/asterisk/srv.h: Merge changes from
asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV
record support in Asterisk was broken. There was no guarantee on
what record Asterisk would choose to actually use. This set of
changes improves the situation by ensuring that Asterisk will
choose the highest priority record.
* main/channel.c: Merge the last bit of changes from
asterisk/team/russell/readq-1.4 The issue here is that the
channel frame readq handling got broken when the code was
converted to use the linked list macros. It caused corruption of
the list head and tail pointers. So, I fixed up the usage of the
linked list macros and in passing, simplified the code. I also
documented what the code is doing, as it was a bit difficult to
figure out at first. This bug showed itself with crashes showing
messed up head/tail pointers for the readq. However, there are a
couple of crashes that aren't quite as obvious, but I think may
be related. So, if your bug gets closed by this commit, but you
still have a problem, please reopen or create a new bug report.
(closes issue #10936) (closes issue #10595) (closes issue #10368)
(closes issue #11084) (closes issue #10040) (closes issue #10840)
2007-11-05 18:47 +0000 [r88671] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: If a SIP channel is put on hold multiple
times do not keep incrementing the onHold value. (closes issue
#11085) Reported by: francesco_r Tested by: blitzrage (closes
issue #10474) Reported by: acennami
2007-11-05 17:46 +0000 [r88624] Russell Bryant <russell@digium.com>
* main/channel.c: Fix up datastore handling in ast_do_masquerade().
The code is intended to move any channel datastores from the old
channel to the new one. However, it did not use the linked list
macros properly to accomplish the task. The existing code would
only work if there was only a single datastore on the old
channel.
2007-11-05 17:19 +0000 [r88585] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Make sure we destroy the config structure on
configuration failure. Issue 11163, patch by eliel.
2007-11-05 16:20 +0000 [r88539] Tilghman Lesher <tlesher@digium.com>
* res/res_odbc.c: Don't check used pooled connections for
connection status, as it will cause issues for prepared queries.
Reported by: Nick Gorham (via -dev list) Patch by: tilghman
2007-11-04 22:38 +0000 [r88471] Luigi Rizzo <rizzo@icir.org>
* include/asterisk/stringfields.h, main/channel.c,
apps/app_meetme.c, channels/chan_sip.c, channels/chan_iax2.c:
Rename ast_string_field_free_pool to
ast_string_field_free_memory, and ast_string_field_free_all to
ast_string_field_reset_all to avoid misuse (due to too similar
names and an error in documentation). Fix two related memory
leaks in app_meetme. No need to merge to trunk, different fix
already applied there. Not applicable to 1.2
2007-11-02 20:49 +0000 [r88328-88366] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make subscribecontext behave as advertised.
It will now look for the presence of a hint in the given context
(be it subscribecontext or context). (closes issue #10702)
Reported by: slavon
* channels/chan_sip.c: If an INFO request within a dialog is
received with a content length of 0 simply send back a 200 OK. It
is valid to do this and the remote side is probably using it to
make sure the signalling is still alive. (closes issue #5747)
Reported by: chandi Patches: infofix-81430-1.patch uploaded by
IgorG (license 20)
2007-11-02 16:51 +0000 [r88283] Jason Parker <jparker@digium.com>
* main/say.c: We need to make sure to specify a language to
ast_fileexists, otherwise it may fail for anything besides en
Issue 11147, fix discovered by both citats and myself
(independently), with input from Corydon76
2007-11-02 13:03 +0000 [r88116-88210] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Fix build on Solaris Reported by: snuffy
Patch by: ys Closes issue #11143
* doc/valgrind.txt (added): Add some notes on using valgrind
2007-11-01 16:21 +0000 [r88078] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Make sure we set the poll fds to NULL after
free()ing it. Part of issue 11017, patch by tzafrir.
2007-11-01 13:27 +0000 [r87970-88026] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Fix up commit for my Zap channel with spies in
Meetme fix. (thanks Tony Mountifield!)
* apps/app_meetme.c: If a Zap channel contains a spy or a spy is
added take it out of the conference in kernel space and make it
go through Asterisk so the spy gets audio from both sides.
(closes issue #10060) Reported by: mparker
2007-10-31 21:23 +0000 [r87906-87908] Jason Parker <jparker@digium.com>
* res/res_jabber.c: Make sure we free some allocated memory before
returning. Issue 11131, patch by eliel.
* channels/chan_gtalk.c: Don't try to allocate memory that we're
just going to re-allocate later anyways. Issue 11130, patch by
eliel.
2007-10-31 18:03 +0000 [r87852] Tilghman Lesher <tlesher@digium.com>
* Makefile: Create samples for ALL of the available options in
asterisk.conf
2007-10-31 17:49 +0000 [r87775-87849] Steve Murphy <murf@digium.com>
* pbx/pbx_config.c: closes issue #11108 -- where the 'dialplan
save' cli command saves a file where the semicolon is not
escaped. Fixed this; User also wanted comments to be preserved
across dialplan save, but this is impossible at this point in
time, because comments are not stored in the dialplan. They are
'compiled' out of extensions.conf. The only way to preserve those
comments is to use the config file reader/writer that the GUI
uses to allow online user edits. extensions.conf is first and
foremost, a config file, and is read in by the normal config-file
reading routines. Then, it is processed into a dialplan
(context/exten structs).
* pbx/pbx_ael.c: Included some verbage in the check_includes func,
to inform the user that included contexts that have no match in
the AEL, might be OK, as AEL cannot check in the extensions.conf
or the in-memory contexts, as they may not be there at the time
of the check.
2007-10-30 23:02 +0000 [r87739] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Fix for uninitialized mutexes on *BSD
Reported by: ys Fixed by: ys Closes issue #11116
2007-10-30 21:19 +0000 [r87686] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Merge the changes from
team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a
race condition related to the handling of POKEing peers.
Essentially, a reference to a peer is held by the scheduler when
there are pending callbacks, but the reference count didn't
reflect it. So, it was possible for a peer to hit a reference
count of zero and have its destructor begin to be called at the
same time that the scheduler thread ran a POKE related callback.
If that happened, a crash would likely occur. (closes issue
#11082, closes issue #11094)
2007-10-30 20:29 +0000 [r87650] Jason Parker <jparker@digium.com>
* channels/Makefile: Only try to clean out h323/ if the
h323/Makefile exists.
2007-10-30 16:13 +0000 [r87571] Joshua Colp <jcolp@digium.com>
* res/res_features.c: Add two more checks before printing out a
warning message about bridging. If either channel has hungup of
course the bridge will have failed. (closes issue #10009)
Reported by: dimas
2007-10-30 15:45 +0000 [r87567] Jason Parker <jparker@digium.com>
* main/editline/np/vis.c: Fix build of editline on Solaris. Issue
11113, patch by snuffy.
2007-10-30 15:10 +0000 [r87534] Joshua Colp <jcolp@digium.com>
* apps/app_followme.c: Return 1.4 to a state where it builds.
Changing the arguments to a function and not changing where they
are used is bad, mmmk?
2007-10-30 14:31 +0000 [r87514] BJ Weschke <bweschke@btwtech.com>
* apps/app_followme.c: Fix issue where the recorded name wasn't
getting removed correctly. (closes issue #11115) Reported by:
davevg Patches: followme-v3.diff
2007-10-29 22:13 +0000 [r87460-87465] Kevin P. Fleming <kpfleming@digium.com>
* codecs/gsm: missed one directory
* codecs/ilbc, formats, utils/Makefile, agi/Makefile, funcs,
codecs/lpc10, main/db1-ast, main/editline, main,
codecs/ilbc/Makefile, pbx, res, channels, main/db1-ast/Makefile,
codecs/lpc10/Makefile, utils, codecs, agi,
main/editline/Makefile.in, apps, Makefile.moddir_rules, cdr:
clean up (and ignore) assembler and preprocessor intermediate
files if any are created during the build
* Makefile: don't put '-pipe' into ASTCFLAGS if '-save-temps' is
already there (used when debugging preprocessor issues) because
the compiler will whine about each compile command
2007-10-29 21:06 +0000 [r87427] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Removing a completely unnecessary quota
check from IMAP code.
2007-10-29 20:22 +0000 [r87373-87396] Russell Bryant <russell@digium.com>
* main/utils.c, include/asterisk/lock.h: Add some more details to
the output of "core show locks". When a thread is waiting for a
lock, this will now show the details about who currently has it
locked. (inspired by issue #11100)
* main/astmm.c: Remove a lock that doesn't make any sense. The
regions lock needs to be held when traversing the list of
allocated chunks so that they can be printed out to the CLI.
(Thanks to eliel on #asterisk-dev for pointing this out!)
2007-10-29 17:20 +0000 [r87342] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix issue where if both sides of the dialog
cancelled the dialog at the same time chan_sip could kepe
retransmitting a response for no reason. (closes issue #9566)
Reported by: atca_pres Patches: bug9566.patch uploaded by oej
2007-10-29 17:13 +0000 [r87340] Jason Parker <jparker@digium.com>
* funcs/func_realtime.c, funcs/func_cut.c: Allow some function
modules to compile under dev mode. Issue 11104, patch by andrew.
2007-10-29 14:23 +0000 [r87294] Joshua Colp <jcolp@digium.com>
* main/utils.c: Fix issue with ast_unescape_semicolon going into an
endless loop. (closes issue #10550) Reported by: ramonpeek
Patches: unescape-85177-1.patch uploaded by IgorG (license 20)
2007-10-28 13:46 +0000 [r87262] Tilghman Lesher <tlesher@digium.com>
* funcs/func_realtime.c, funcs/func_odbc.c, funcs/func_strings.c,
funcs/func_cut.c: Add autoservice to several more functions which
might delay in their responses. Also, make sure that func_odbc
functions have a channel on which to set variables. Reported by
russell Fixed by tilghman Closes issue #11099
2007-10-26 16:34 +0000 [r87168] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael.tab.c,
pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c,
include/asterisk/ael_structs.h, pbx/ael/ael.tab.h,
utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16,
pbx/ael/ael.flex: closes issue #11086 where a user complains that
references to following contexts report a problem; The problem
was REALLy that he was referring to empty contexts, which were
being ignored. Reporter stated that empty contexts should be OK.
I checked it out against extensions.conf, and sure enough, empty
contexts ARE ok. So, I removed the restriction from AEL. This,
though, highlighted a problem with multiple contexts of the same
name. This should be OK, also. So, I added the extend keyword to
AEL, and it can preceed the 'context' keyword (mixed with
'abstract', if nec.). This will turn off the warnings in AEL if
the same context name is used 2 or more times. Also, I now call
ast_context_find_or_create for contexts now, instead of just
ast_context_create; I did this because pbx_config does this. The
'extend' keyword thus becomes a statement of intent. AEL can now
duplicate the behavior of pbx_config,
2007-10-26 13:54 +0000 [r87120] Tilghman Lesher <tlesher@digium.com>
* funcs/func_curl.c: The addition of autoservice to func_curl
additionally made func_curl dependent on the existence of a
channel, with no real reason. This should make func_curl once
again work without a channel. Reported by jmls. Fixed by
tilghman. Closes issue #11090
2007-10-25 23:03 +0000 [r87069] Kevin P. Fleming <kpfleming@digium.com>
* main/channel.c, include/asterisk/linkedlists.h: appending one
list to another should leave the first list empty, and not
require the user to do that
2007-10-25 22:53 +0000 [r87067] Tilghman Lesher <tlesher@digium.com>
* funcs/func_cut.c: Backport alternate encoding of newline
delimiters from trunk to 1.4, as approved by Russell Reported by
blitzrage Closes issue #10903
2007-10-24 20:56 +0000 [r86982] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: Correctly respect hidecalleridname
configuration option. Simplify code slightly in the process.
Issue 11079, reported by ddv2005
2007-10-24 04:14 +0000 [r86880-86936] Steve Murphy <murf@digium.com>
* pbx/ael/ael.tab.c, pbx/ael/ael.y: closes issue #11037 -- unable
to specify app:spec in hint arguments
* funcs/func_logic.c: closes issue #11052 -- where nothing after
the ? will allow un-initialized variable values to corrupt and
crash asterisk on 64-bit platforms
* main/Makefile: this update to Makefile corrects how ast_expr2f.c
should be generated
* main/ast_expr2f.c: This should get rid of a really, really
irritating warning generated by some 64-bit platforms from libc,
where free(0) is frowned upon
2007-10-22 21:36 +0000 [r86836] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: If lock tracking is not enabled, then we
can not attempt to log any mutex failures. If so, we could end up
in infinite recursion. The only lock that is affected by this is
a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes
issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys
(license 281)
2007-10-22 17:38 +0000 [r86787] Tilghman Lesher <tlesher@digium.com>
* main/astmm.c: Minor FreeBSD build fix
2007-10-22 16:35 +0000 [r86754-86756] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: After reading online I have confirmed that
Record-Route headers should be copied to 1xx responses as well.
(closes issue #10113) Reported by: makoto
* apps/app_controlplayback.c: Make sure res is a positive value
before performing the check to determine whether the user stopped
it or not. (closes issue #11023) Reported by: cfc
2007-10-22 15:52 +0000 [r86726-86750] Russell Bryant <russell@digium.com>
* main/channel.c: Don't leak a frame in the case that an END frame
is received and the time since the BEGIN is less than that of the
defined minimum DTMF duration. (closes issue #11051) Reported by:
casper Patches: channel.c.86664.diff uploaded by casper (license
55)
* include/asterisk/lock.h: Update the static mutex initializer to
include the initialization of the internal mutex used to protect
the lock debugging data. (closes issue #11044, patch suggested by
Ivan)
2007-10-22 14:48 +0000 [r86694] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Account for the fact that sometimes headers
may be terminated with \r\n instead of just \n (closes issue
#11043, reported by yehavi)
2007-10-22 14:27 +0000 [r86630-86663] Joshua Colp <jcolp@digium.com>
* main/channel.c: Move log message to before the frame it
references is freed. (closes issue #11050) Reported by: slavon
Patches: channel.c.86662.diff uploaded by casper (license 55)
* pbx/pbx_dundi.c: Fix tab completion for dundi show peer. (closes
issue #11041) Reported by: jsmith Patches:
asterisk-dundicomplete.diff.txt uploaded by jamesgolovich
(license 176)
* main/loader.c: Fixes for building under OpenSolaris. (closes
issue #11047) Reported by: snuffy Patches: 11047-fixes.diff
uploaded by snuffy (license 35)
2007-10-22 09:21 +0000 [r86598] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: we send
DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan
does not match after an overlap call. Also added out_cause=1
2007-10-19 16:38 +0000 [r86469-86502] Joshua Colp <jcolp@digium.com>
* main/app.c: When returning a DTMF digit from
ast_control_streamfile cast it as a char so that 0 does not
overlap with the success return code. (closes issue #11023)
Reported by: cfc
* channels/chan_sip.c: Fix two issues with domains and transfers.
If a port was given in the hostname it was treated as part of the
hostname. If domains were configured but external domains were
not enabled all transfers would be considered remote. (closes
issue #11027) Reported by: ramonpeek Patches: 11027-1.diff
uploaded by ramonpeek (license 266)
* channels/chan_sip.c: Set port number in received as information
for registrations as well. (closes issue #11028) Reported by:
brad-x
2007-10-19 01:45 +0000 [r86438] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Fixed OSP module did not report
source/devinfo IP in correct format.
2007-10-18 22:01 +0000 [r86405-86406] Jason Parker <jparker@digium.com>
* Makefile: Correct documentation. I removed the wrong line..
* Makefile: Add documentation for options in asterisk.conf Issue
11029, patch by eserra
2007-10-18 21:16 +0000 [r86330-86372] Russell Bryant <russell@digium.com>
* configs/iax.conf.sample, channels/chan_iax2.c: Revert erroneous
commit.
* configs/iax.conf.sample, channels/chan_iax2.c: Add support for
setting the maximum trunk size for IAX2 trunking
* main/channel.c, include/asterisk/channel.h: The channel needs to
stay locked while running timer callbacks, as they access and
modify channel data that may change elsewhere. I went through
every timer callback in the source tree to make sure that none of
them did any additional locking that could introduce deadlocks,
and all is well. (closes issue #10765) Reported by: Ivan Patches:
ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license
229)
2007-10-18 17:38 +0000 [r86328] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: If a non-existent file is specified to be
played either as a periodic announcement or as a hold/position
announcement, the caller would be kicked out of the queue. No
longer does this happen.
2007-10-18 15:45 +0000 [r86237-86296] Russell Bryant <russell@digium.com>
* codecs/codec_zap.c: Execute the RELEASE operation on transcoder
channels in the destroy callback. (patch from jsloan)
* main/utils.c: Revert a change that I made for issue #10979 which,
as has been pointed out to me in issue #11018, doesn't really
make sense. There is no reason to have the base64 decode function
force a '\0' terminated buffer, when the result is almost always
binary, anyway. In fact, this caused some breakage, as some code
in res_crypto passed in a buffer exactly the right size to get
its binary result, which got stomped on by this patch. (closes
issue #11018, reported by dimas)
2007-10-17 21:39 +0000 [r86202] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Changing the strategy field of the call_queue
struct to be signed instead of unsigned, since the code attempts
to set the strategy to -1 if you specify a bogus strategy. While
this isn't a huge issue in 1.4, it could be a problem for someone
who, say, tries to use the roundrobin strategy in trunk (despite
all the deprecation warnings in 1.4).
2007-10-17 17:57 +0000 [r86149] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: If Asterisk is in the middle of shutting
down, respond to OPTIONS with 503 Unavailable. (closes issue
#10994) Reported by: eserra Patches: sip-options-503.patch
uploaded by eserra (license 45)
2007-10-17 16:58 +0000 [r86117] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Whoops, forgot to remove the original
sip_scheddestroy. (closes issue #11010) Reported by: vadim
2007-10-17 15:23 +0000 [r86066] Tilghman Lesher <tlesher@digium.com>
* main/asterisk.c: When runuser/rungroup is specified, a remote
console could only be attained by root (Closes issue #9999)
2007-10-17 15:06 +0000 [r86063] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't schedule dialog destruction if a
MESSAGE is received using an existing dialog. (closes issue
#11010) Reported by: vadim
2007-10-16 23:35 +0000 [r86028-86032] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample: Since monitor-join is deprecated now,
remove the example from the sample queues.conf file
* UPGRADE.txt: Updating UPGRADE.txt to reflect the deprecation of
the monitor-join queue option
* apps/app_queue.c: Adding deprecated warning to monitor-join
option, since the plan is to no longer support this in favor of
monitor-type = mixmonitor (related to issue #10885)
2007-10-16 22:36 +0000 [r85994-85997] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: really picky formatting tweak ...
* include/asterisk/lock.h: Some locking errors exposed the fact
that the lock debugging code itself was not thread safe. How
ironic! Anyway, these changes ensure that the code that is
accessing the lock debugging data is thread-safe. Many thanks to
Ivan for finding and fixing the core issue here, and also thanks
to those that tested the patch and provided test results. (closes
issue #10571) (closes issue #10886) (closes issue #10875) (might
close some others, as well ...) Patches: (from issue #10571)
ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license
229) - a few small changes by me
2007-10-16 21:14 +0000 [r85958] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Trying to remove a non-dynamic queue member via
dynamic means can lead to some interesting (read nasty)
situations. This patch clears up the issue by making only dynamic
queue members removable via dynamic methods.
2007-10-16 19:41 +0000 [r85921] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/localtime.c: Also set up gmtoff (this is used in the
%z gnu extension to strftime) Reported and fixed by jcmoore
Closes issue #11002
2007-10-16 19:10 +0000 [r85896] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Remove a pointless lock.
2007-10-16 15:21 +0000 [r85852] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixing a double free which happens in the
statechange thread. (closes issue #10987, reported by andrew)
2007-10-16 14:52 +0000 [r85818-85850] Joshua Colp <jcolp@digium.com>
* apps/app_hasnewvoicemail.c: Check to make sure a value has been
given to the VMCOUNT dialplan function. (closes issue #10996)
Reported by: marsosa
* main/threadstorage.c: Fix memory allocation issue in
threadstorage. (closes issue #10995) Reported by: snuffy Patches:
new-patch.diff uploaded by snuffy (license 35)
2007-10-16 10:46 +0000 [r85800] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Fix the output for this channel help CLI
command
2007-10-15 21:10 +0000 [r85717-85720] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Ensure that no pending state changes are leaked
when the device state change thread gets stopped on module
unload.
* apps/app_queue.c: Previously, app_queue created a thread to
handle every single device state change. I changed this a while
ago in trunk for performance reasons. However, bug 8407 points
out that it is actually a race condition, causing device state
changes to get processed in random order. So, I backported my
changes from trunk to 1.4. (closes issue #8407, patch provided by
tim_ringenbach, committed patch by me)
2007-10-15 20:29 +0000 [r85687] Tilghman Lesher <tlesher@digium.com>
* apps/app_stack.c: Don't execute a gosub if the arguments is
zero-len (not just NULL) Reported by davevg Fixed by me Closes
issue #10985
2007-10-15 20:21 +0000 [r85686] Russell Bryant <russell@digium.com>
* main/say.c: Add a small fix for the tw version of saying dates.
(closes issue #7827) Reported by: sharkey Patches: say.nits.patch
uploaded by sharkey (license 172)
2007-10-15 20:15 +0000 [r85684] Jason Parker <jparker@digium.com>
* Makefile: Properly use DESTDIR in 'config' target. Do not try to
run chkconfig or similar if using DESTDIR. Issue 10938, patch by
cabal95.
2007-10-15 19:22 +0000 [r85604-85649] Russell Bryant <russell@digium.com>
* main/utils.c: Be pedantic about handling memory allocation
failure.
* main/utils.c: The loop in the handler for the "core show locks"
could potentially block for some amount of time. Be a little bit
more careful and prepare all of the output in an intermediary
buffer while holding a global resource. Then, after releasing it,
send the output to ast_cli().
* channels/chan_sip.c: Make the default for the srvlookup option to
be yes. It doesn't really make sense for it to default to off.
The default configuration file has it on, and proper RFC
behavior, as indicated by a comment in the code, is for it to be
on. So, let's have it on by default to make lives easier. (closes
issue #10954, suggested by jtodd)
2007-10-15 16:39 +0000 [r85571] Joshua Colp <jcolp@digium.com>
* configs/features.conf.sample: Document that DTMF based features
only work when two channels are bridged together. (closes issue
#10773) Reported by: pbayley
2007-10-15 16:34 +0000 [r85561] Russell Bryant <russell@digium.com>
* include/asterisk/strings.h: Make a few changes so that characters
in the upper half of the ISO-8859-1 character set don't get
stripped when reading configuration. (closes issue #10982,
dandre)
2007-10-15 16:22 +0000 [r85559] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Bring both DTMF begin and end frames up through to
the core for DTMF feature handling. (closes issue #10826)
Reported by: dimas
2007-10-15 15:40 +0000 [r85556] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: Ensure the buffer passed to
ast_canmatch_extension() is properly initialized so that it is
null terminated. (issue #10977) Reported by: dimas Patches:
pbxdundi.patch uploaded by dimas (license 88) - small mods by me
2007-10-15 14:55 +0000 [r85552] Joshua Colp <jcolp@digium.com>
* main/rtp.c: If Monitor or a spy was added to a P2P or native
bridged channel bring the channel back to the generic bridging
core so the monitor or spy operations work. (closes issue #10943)
Reported by: julianjm
2007-10-15 13:16 +0000 [r85540-85548] Russell Bryant <russell@digium.com>
* main/db.c: Suppress a LOG_DEBUG message if debug is not enabled.
(closes issue #10980) Reported by: casper Patches:
db.c.84633.diff uploaded by casper (license 55)
* main/asterisk.c: Make sure remote consoles unmute themselves
again after reconnecting. (closes issue #10847) Reported by: atis
Patches: console_unmute_on_reconnect.patch uploaded by atis
(license 242)
* main/utils.c: Make sure that the base64 decoder returns a
terminated string. (closes issue #10979) Reported by: ys Patches:
util.c.diff uploaded by ys (license 281) - small mods by me
* pbx/pbx_config.c: Don't create the context for users in
users.conf until we know at least one user exists. (closes issue
#10971) Reported by: dimas Patches: pbxconfig.patch uploaded by
dimas (license 88)
2007-10-13 15:26 +0000 [r85536] Tilghman Lesher <tlesher@digium.com>
* configs/extensions.ael.sample: Remove deprecated syntax from
sample ael file Reported and patched by: dimas Closes issue
#10967
2007-10-13 05:48 +0000 [r85532-85533] Russell Bryant <russell@digium.com>
* main/asterisk.c, main/cli.c, include/asterisk/logger.h: Fix an
issue with console verbosity when running asterisk -rx to execute
a command and retrieve its output. The issue was that there was
no way for the main Asterisk process to know that the remote
console was connecting in the -rx mode. The way that James has
fixed this is to have all remote consoles muted by default. Then,
regular remote consoles automatically execute a CLI command to
unmute themselves when they first start up. (closes issue #10847)
Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded
by jamesgolovich (license 176)
* main/asterisk.c, main/cli.c, include/asterisk/cli.h: Properly
handle the case where read() may return the text for more than
one CLI command at once for a remote console. (closes issue
#10888) Reported by: jamesgolovich Patches:
asterisk-climultiple.diff.txt uploaded by jamesgolovich (license
176)
2007-10-12 18:30 +0000 [r85523] Tilghman Lesher <tlesher@digium.com>
* doc/asterisk-mib.txt, doc/PEERING, LICENSE: Change Digium address
2007-10-12 15:45 +0000 [r85515-85517] Russell Bryant <russell@digium.com>
* res/res_smdi.c: Fix a spelling error in a log message. SMDI, not
SDMI. (closes issue #10959)
* pbx/pbx_realtime.c: Fix the potential use of an uninitialized
buffer in a log message. (closes issue #10958) Reported by: dimas
Patches: realtime.patch uploaded by dimas (license 88)
2007-10-11 15:26 +0000 [r85397] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: When creating a new packet don't try to stop
retransmission of it. It was just allocated/created so it's
impossible for it to have already been scheduled. (closes issue
#10945) Reported by: flefoll Patches:
chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll
(license 244)
2007-10-11 04:35 +0000 [r85356] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c: A dollar sign by itself, not indicating a start of a
variable or expression prematurely ends substitution (closes
issue #10939)
2007-10-10 Russell Bryant <russell@digium.com>
* Asterisk 1.4.13 released.
2007-10-10 15:56 +0000 [r85316] Russell Bryant <russell@digium.com>
* include/asterisk/file.h: I introduced a new member to the
ast_filestream struct in 1.4.12, but put it in the middle of the
struct, instead of at the end. One of the Debian folks, paravoid,
pointed out that this breaks binary compatability with modules
compiled against older headers. So, I'm moving the new member to
the end of the struct to resolve the situation.
2007-10-10 15:51 +0000 [r85315] Mark Michelson <mmichelson@digium.com>
* main/utils.c: The thread ID should be unsigned.
2007-10-10 14:42 +0000 [r85277-85280] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: If devicestate is passed a port number strip
it out. (closes issue #10930) Reported by: ibc
* channels/chan_sip.c: Add support for handling a 182 Queued
response. (closes issue #10924) Reported by: ramonpeek Patches:
queued-182.diff uploaded by ramonpeek (license 266)
2007-10-10 14:26 +0000 [r85276] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: A bunch of changes from sprintf to
snprintf. See security advisory AST-2002-022
2007-10-10 14:14 +0000 [r85242] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Close voicemail message description file if
duration did not meet the minimum, or else we will eventually run
out of file descriptors. (closes issue #10918) Reported by:
brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license
279)
2007-10-10 06:24 +0000 [r85195] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/frame.h: use a macro instead of an inline
function, so that backtraces will report the caller of
ast_frame_free() properly
2007-10-09 21:55 +0000 [r85158] Tilghman Lesher <tlesher@digium.com>
* main/channel.c, main/utils.c, include/asterisk/lock.h: This
commit fixes the following issues: - Deadlock in ast_write (issue
#10406) - Deadlock in ast_read (issue #10406) - Possible mutex
initialization error in lock.h (issue #10571)
2007-10-09 14:30 +0000 [r84990-85093] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't perform a reinvite if a transfer is in
progress. (issue #10915) Reported by: ramonpeek
* main/rtp.c: Only update codec information if the channel has a
technology private structure. (issue #10915) Reported by:
ramonpeek
* main/rtp.c: Update codec information as well as address when
doing hold reinvites. (issue #10868) Reported by: mavince
* main/channel.c: Don't keep trying to native bridge if either of
the channels are involved in a masquerade operation to be done.
(closes issue #10696) Reported by: tbelder
2007-10-08 03:28 +0000 [r84957] Russell Bryant <russell@digium.com>
* Makefile.rules: Enable file dependency tracking for _all_ builds,
and not just for builds with dev-mode enabled. I have seen enough
problems caused by this that I don't think it's worth keeping. I
want to continue to encourage anybody that is interested to
continue to run Asterisk from svn. Furthermore, I do not want
their systems to break when we change a structure definition in a
header file. :)
2007-10-07 16:15 +0000 [r84890-84902] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: Presence packets from a client who's connected
with our Jabber ID are valid, therefore, those clients must be
considered as buddies. The resource string helps us make the
distinction between clients. Closes issue #10707, reported by
yusufmotiwala.
* res/res_jabber.c: Prevent Asterisk from crashing when receiving a
presence packet without resource from a buddy that is known to
have a resource list. Revert a change I previously made, where
Asterisk could point to a freed memory location.
2007-10-05 19:42 +0000 [r84851] Tilghman Lesher <tlesher@digium.com>
* main/db.c: Log exactly why we can't open the database, if we fail
(closes issue #10887)
2007-10-05 18:55 +0000 [r84818] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Update the remembered RTP peer information when
putting an endpoint on hold or taking it off hold so that the RTP
stack does not initiate a needless reinvite. (closes issue
#10868) Reported by: mavince
2007-10-05 16:44 +0000 [r84783] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: Do deadlock avoidance in a couple more
places. You can't lock two channels at the same time without
doing extra work to make sure it succeeds. (closes issue #10895,
patch by me)
2007-10-05 Russell Bryant <russell@digium.com>
* Asterisk 1.4.12.1 released. (This is mainly to include the
app_queue fix for a memory leak on reload, but includes a couple
of other bug fixes, as well.)
2007-10-05 01:39 +0000 [r84742] Russell Bryant <russell@digium.com>
* main/manager.c: Fix a copy/paste error in the description of
UpdateConfig that was pointed out by JerJer on #asterisk-dev
2007-10-04 21:57 +0000 [r84692] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Don't allocate space for queue members unless
it's needed. You end up deleting dynamic members on a reload. Not
good. closes issue (#10879, reported by dazza76, patched by me)
2007-10-04 21:36 +0000 [r84690] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: callers of sig2str already add the word
'signalling' in the appropriate place, so don't duplicate it
2007-10-04 14:51 +0000 [r84637] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c: Create a duplicate of the channel's member name
as the tab completion stuff will free it. (closes issue #10884)
Reported by: adamg
2007-10-03 22:59 +0000 [r84581] Tilghman Lesher <tlesher@digium.com>
* main/rtp.c: When an RFC 2833 event is sent that we don't
recognize, ignore it, don't queue a NULL digit (closes issue
#10877)
2007-10-03 18:20 +0000 [r84511-84544] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: closes issue #10870 ; where a CUT() function call
in a switch expr doesn't execute correctly, because the commas in
the function args are not converted to vertbars before the func
is called. I modified just the switch code to convert the commas
to vertbars if there, but if more of these sort of probs are
found, I may have to resort to something a little more
fundamental. We'll see, I guess.
* pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
pbx/ael/ael-test/ref.ael-vtest13,
pbx/ael/ael-test/ref.ael-vtest17,
pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c,
pbx/ael/ael-test/ref.ael-test5: closes issue #10834 ; where a
null input to a switch statement results in a hangup; since
switch is implemented with extensions, and the default case is
implemented with a '.', and the '.' matches 1 or more remaining
characters, the case where 0 characters exist isn't matched, and
the extension isn't matched, and the goto fails, and a hangup
occurs. Now, when a default case is generated, it also generates
a single fixed extension that will match a null input. That
extension just does a goto to the default extension for that
switch. I played with an alternate solution, where I just tack an
extra char onto all the patterns and the goto, but not the
default case's pattern. Then even a null input will still have at
least one char in it. But it made me nervous, having that extra
char in , even if that's a pretty secret and low-level issue.
2007-10-02 Russell Bryant <russell@digium.com>
* Asterisk 1.4.12 released.
2007-10-02 20:06 +0000 [r84474] Russell Bryant <russell@digium.com>
* Makefile, build_tools/prep_tarball: * Don't build the
menuselect-tree for the tarball, as it requires running the
configure script first * Change the Makefile to note that
menuselect-tree depends on the configure script.
2007-10-02 19:01 +0000 [r84410-84437] Jason Parker <jparker@digium.com>
* res/res_features.c: Fix some odd formatting I missed..
* res/res_features.c: Finish up on transferee channel before return
on failure. Issue 10821, patch by Ivan
2007-10-02 14:12 +0000 [r84370] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Use snprintf instead of sprintf in one
place. There is no vulnerability here due to various buffer sizes
around the code, but I still didn't like seeing a non
length-limited copy of data coming off of the wire into a stack
buffer, as this would be a problem in the future if buffer sizes
elsewhere got changed or size limitations removed ...
2007-10-02 09:48 +0000 [r84345] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: terminate USERUSER String with 0
2007-10-01 21:52 +0000 [r84291] Jason Parker <jparker@digium.com>
* Makefile, Makefile.rules, channels/Makefile: Add dist-clean
support for subdirs. Change h323 to only remove the Makefile on a
dist-clean, rather than a clean. This fixes a bug I found with
trying to run make after a make clean
2007-10-01 21:25 +0000 [r84274] Dwayne M. Hubbard <dhubbard@digium.com>
* main/channel.c, main/manager.c, channels/chan_agent.c: moved
get_base_channel() code from action_redirect to
ast_channel_masquerade() for issue 7706 and BE-160
2007-10-01 21:18 +0000 [r84273] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: Anything to keep gcc 4.2 happy...
2007-10-01 21:07 +0000 [r84271] Russell Bryant <russell@digium.com>
* main/utils.c, include/asterisk/lock.h: Fulfull a feature request
from Qwell on the "core show locks" output. It will now note the
lock type for each lock that a thread holds. (mutex, rdlock, or
wrlock)
2007-10-01 20:27 +0000 [r84239] Steve Murphy <murf@digium.com>
* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: closes issue
#10777 -- by returning a null for the parse tree when there's
really nothing there, and making sure we don't try to do checking
on a null tree.
2007-10-01 19:56 +0000 [r84166-84236] Russell Bryant <russell@digium.com>
* res/res_agi.c: Add another sanity check in the AGI read loop. We
really don't care about EAGAIN unless we didn't read an entire
line. If there is a newline at the end if the read buffer, break,
because we got the whole thing. (reported and patched by bmd)
* include/asterisk/lock.h: Show rwlocks in the "core show locks"
output. Before, it only showed mutexes.
* channels/Makefile: Remove another file in "make clean". (closes
issue #10814, paravoid)
* apps/app_dial.c: Simplify the CAN_EARLY_BRIDGE macro a bit.
2007-10-01 14:10 +0000 [r84158-84163] Joshua Colp <jcolp@digium.com>
* configs/usbradio.conf.sample (removed): Remove chan_usbradio
config file from tree, it is not present in here. (closes issue
#10839) Reported by: casper
* res/res_musiconhold.c: Fix randomness. save_pos was being set to
0 initially instead of -1, causing it to jump to position 0 when
moh started. (closes issue #10859) Reported by: jamesgolovich
Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich
(license 176)
* apps/app_dial.c: Only attempt early bridging if the options given
to Dial() permit it. (closes issue #10861) Reported by: peekyb
2007-09-30 20:02 +0000 [r84146] Russell Bryant <russell@digium.com>
* include/asterisk/module.h: Fix the AST_MODULE_INFO macro for C++
modules. The load and reload parameters were in the wrong place.
(closes issue #10846, alebm)
2007-09-29 23:00 +0000 [r84133-84135] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ael-ntest22/t1/a.ael (added),
pbx/ael/ael-test/ael-ntest22/t1/b.ael (added),
pbx/ael/ael-test/ael-ntest22/t1/c.ael (added),
pbx/ael/ael-test/ael-ntest22/t2/d.ael (added),
pbx/ael/ael-test/ael-ntest22/t2/e.ael (added),
pbx/ael/ael-test/ael-ntest22/t2/f.ael (added),
pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22
(added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added),
pbx/ael/ael-test/ref.ael-test3,
pbx/ael/ael-test/ael-ntest22/t3/h.ael (added),
pbx/ael/ael-test/ref.ael-test4,
pbx/ael/ael-test/ael-ntest22/t3/i.ael (added),
pbx/ael/ael-test/ael-ntest22/t3/j.ael (added),
pbx/ael/ael-test/ael-ntest22/qq.ael (added),
pbx/ael/ael-test/ael-ntest22/t1 (added),
pbx/ael/ael-test/ael-ntest22/t2 (added),
pbx/ael/ael-test/ael-ntest22/t3 (added),
pbx/ael/ael-test/ael-ntest22/extensions.ael (added),
pbx/ael/ael-test/ael-ntest22 (added): This is a regression update
that matches what I did in 84134 for AEL regressions.
* pbx/ael/ael_lex.c, pbx/ael/ael.flex: This issue sort of closes
10786; All config files support #include with globbing (you know,
*,[chars],?,{list,list},etc), so I've updated the AEL system to
support this also.
2007-09-28 14:13 +0000 [r84049-84078] Tilghman Lesher <tlesher@digium.com>
* main/say.c: Correct pronunciations of numbers for .nl (Closes
issue #10837)
* main/channel.c: Avoid a deadlock with ALL of the locks in the
masquerade function, not just the pairs of channels. (Closes
issue #10406)
2007-09-27 23:12 +0000 [r84018] Dwayne M. Hubbard <dhubbard@digium.com>
* main/manager.c, channels/chan_agent.c,
include/asterisk/channel.h: if an Agent is redirected, the base
channel should actually be redirected. This was causing multiple
issues, especially issue 7706 and BE-160
2007-09-27 00:01 +0000 [r83976] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: remove a todo item that has been completed
2007-09-26 23:53 +0000 [r83974] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_alsa.c: avoid the weird usage of assert() in the
ALSA header files that gcc 4.2 wants to complain about
2007-09-26 21:35 +0000 [r83910-83943] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: I changed my mind ... I think this should be
a LOG_NOTICE.
* channels/chan_sip.c: Add a log message that was requested by the
masses in the developer tutorial session at Astricon. chan_sip
did not output any message when a call was rejected because the
extension was not found. This adds a verbose message (at verbose
level 3) to note when this happens.
* channels/chan_misdn.c: Fix building chan_misdn under dev-mode.
(please run the configure script with --enable-dev-mode so this
doesn't happen again ...)
2007-09-26 18:35 +0000 [r83879] Tilghman Lesher <tlesher@digium.com>
* channels/chan_zap.c: Remove unused 4k of memory on the program
stack (closes issue #10827)
2007-09-25 14:13 +0000 [r83637-83773] Tilghman Lesher <tlesher@digium.com>
* main/app.c: jmls pointed out that unsetting the group and setting
the group to the blank string aren't quite the same.
* build_tools/make_defaults_h: In the source, keys are relative to
the datadir, not varlib (which is the same in most cases, but
it's good to be accurate). Closes issue #10811
* doc/realtime.txt: Oops. Removed the unworkable workaround. This
note should never have been in the release.
* main/app.c: Making change to group splitting, as discussed on the
-dev list. The main effect of this will be to permit
Set(GROUP([cat])=), i.e. unsetting a group.
2007-09-24 07:54 +0000 [r83620] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: fixed round_robin group dial method, this
never worked well on BRI Ports (2 channels)
2007-09-22 19:39 +0000 [r83558-83589] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This closes issue #10788 -- The exact same fixes
are made here for the first arg in the for(arg1; arg2; arg3) {}
statement, as were done for the 3rd arg. It can now be an
assignment that will embedded in a Set() app, or a macro call, or
an app call.
* pbx/pbx_ael.c: This closes issue #10788 -- the 3rd arg in the for
statement is now wrapped in Set() only if there's an '=' in that
string. Otherwise, if it begins with '&', then a Macro call is
generated; otherwise it is made into an app call. A bit more
accomodating, keeps the new guys happy, and the guys with ael-1
code should be happy, too
2007-09-21 14:37 +0000 [r83432] Russell Bryant <russell@digium.com>
* main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c,
channels/chan_misdn.c, pbx/ael/ael.tab.c, main/ast_expr2f.c,
main/file.c, include/asterisk/sched.h, channels/chan_h323.c,
pbx/pbx_dundi.c, utils/ael_main.c, main/ast_expr2.fl,
channels/chan_mgcp.c, main/sched.c, res/res_config_pgsql.c,
main/dnsmgr.c, channels/chan_sip.c, pbx/ael/ael.y,
main/db1-ast/hash/hash.c, include/asterisk/channel.h,
channels/chan_iax2.c: gcc 4.2 has a new set of warnings dealing
with cosnt pointers. This set of changes gets all of Asterisk
(minus chan_alsa for now) to compile with gcc 4.2. (closes issue
#10774, patch from qwell)
2007-09-21 13:34 +0000 [r83400] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix video under certain circumstances. It
would have been possible for the formats on the channel to not
contain the video format. (closes issue #10782) Reported by:
cwhuang
2007-09-20 21:16 +0000 [r83316-83348] Russell Bryant <russell@digium.com>
* main/asterisk.c: When daemonizing, don't change working directory
to "/". It makes it not be able to do a core dump when not
running as uid=root. (closes issue #10766, xrg)
* contrib/scripts/safe_asterisk: Change safe_asterisk to explicitly
ask for /bin/bash, as it uses bashisms. (closes issue #10772,
reported by culrich)
2007-09-20 17:09 +0000 [r83246] Jason Parker <jparker@digium.com>
* apps/app_disa.c: If # is pressed after dialing an extension in
DISA, stop trying to collect more digits. (issue #10754) Reported
by: atis Patches: app_disa.c.branch.patch uploaded by atis
(license 242) app_disa.c.trunk.patch uploaded by atis (license
242)
2007-09-20 16:25 +0000 [r83230-83232] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make sure the minimum T1 timer value is
obeyed in all cases. (closes issue #10768) Reported by: flefoll
Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll
(license 244) chan_sip.c.br14.83070.retrans-patch uploaded by
flefoll (license 244)
* channels/chan_sip.c: Fix a minor spelling error. (closes issue
#10769) Reported by: flefoll Patches:
chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license
244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll
(license 244)
2007-09-19 19:50 +0000 [r83121-83179] Russell Bryant <russell@digium.com>
* apps/app_system.c: The System() and TrySystem() applications can
take a substantial amount of time to execute while not servicing
the channel. So, put the channel in autoservice while the command
is being executed. (closes issue #10726, reported by mnicholson)
* funcs/func_curl.c: Using curl can take a substantial amount of
time, so the channel should be autoserviced while waiting for it
to complete. (closes issue #10725, reported by mnicholson)
* channels/chan_iax2.c: When handling a reload of chan_iax2, don't
use an ao2_callback() to POKE all peers. Instead, use an
iterator. By using an iterator, the peers container is not locked
while the POKE is being done. It can cause a deadlock if the
peers container is locked because poking a peer will try to lock
pvt structs, while there is a lot of other code that will hold a
pvt lock when trying to go lock the peers container. (reported to
me directly by Loic Didelot. Thank you for the debug info!)
* main/manager.c: Fix up another potential race condition. Do the
loop decrementing use count on events with the eventq protected
from being changed. (reported on IRC by Ivan)
2007-09-19 13:47 +0000 [r83070-83074] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c: Protect the CDR record from modification by
pbx_exec so that the application data contains the Queue data.
(closes issue #10761) Reported by: snar Patches:
app-queue-mixmonitor.patch uploaded by snar (license 245)
* channels/chan_sip.c: (closes issue #10760) Reported by: dimas
Patches: chan_sip.patch uploaded by dimas (license 88) Read in
subscribecontext option in general to be the default.
2007-09-19 09:32 +0000 [r83023-83024] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: removed comment which violates the coding
guidelines.
* channels/misdn_config.c, channels/chan_misdn.c,
channels/misdn/chan_misdn_config.h: added 'astdtmf' option to
allow configuring the asterisk dtmf detector instead of the
mISDN_dsp ones. also added the patch from irroot #10190, so that
dtmf tones detected by the asterisk detector are passed outofband
to asterisk, to make any use of dtmf tones at all.
2007-09-19 00:19 +0000 [r82992] Russell Bryant <russell@digium.com>
* apps/app_flash.c: Change the description of app_flash to note how
it can be a useful tool instead of just saying that it is
generally a worthless feature. (Thanks to Jim Van Meggelen for
pointing it out and providing the proposed text)
2007-09-18 23:41 +0000 [r82961] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c: Initialize a variable to NULL to make the world
happy.
2007-09-18 22:42 +0000 [r82929] Russell Bryant <russell@digium.com>
* include/asterisk/agi.h, res/res_agi.c: Add a new patch to handle
interrupting the fgets() call when using FastAGI. This version of
the patch maintains the original behavior of the code when not
using FastAGI. (closes issue #10553) Reported by: juggie Patches:
res_agi_fgets-4.patch uploaded by juggie (license 24)
res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight
mods by me Tested by: juggie, festr
2007-09-18 21:49 +0000 [r82887-82913] Doug Bailey <dbailey@digium.com>
* main/manager.c: Corrected patch applied in revision r82887.
* main/manager.c: Fixed a bug where http manager sessions prevented
the eventq from being cleaned out because http manager sessions
do not have a valid file descriptor.
2007-09-18 20:56 +0000 [r82867] Russell Bryant <russell@digium.com>
* main/manager.c: Fix a memory leak that can occur on systems under
higher load. The issue is that when events are appended to the
master event queue, they use the number of active sessions as a
use count so it will know when all active sessions at the time
the event happened have consumed it. However, the handling of the
number of sessions was not properly synchronized, so the use
count was not always correct, causing an event to disappear
early, or get stuck in the event queue for forever. (closes issue
#9238, reported by bweschke, patch from Ivan, modified by me)
2007-09-18 20:09 +0000 [r82865] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Moving the logic for handling an empty
membername to the create_member function so that there is a
common place where this occurs instead of being spread out to
several different places.
2007-09-18 18:59 +0000 [r82834] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: there is no need for conditional logic to
select ->interface or ->membername, snince ->membername will
always be populated
2007-09-18 16:31 +0000 [r82802] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: When copying the contents from the wildcard
peer, do a deep copy instead of shallow copy so that it doesn't
crash when beging destroyed. (closes issue #10546, patch by me)
2007-09-18 15:28 +0000 [r82751] Jason Parker <jparker@digium.com>
* configs/sip.conf.sample: Correct the allowexternaldomains option
in SIP sample config. Issue 10753
2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c, main/stdtime/localtime.c: Put a memset in
ast_localtime() instead of a couple places in app_voicemail to
prevent the problem everywhere instead of just a couple of
places. (related to issue #10746)
* apps/app_voicemail.c: Initialize some memory to fix crashes when
leaving voicemail. This problem was fixed by running Asterisk
under valgrind. (closes issue #10746, reported by arcivanov,
patched by me) *** IMPORTANT NOTE: We need to check to see if
this same bug exists elsewhere.
* res/res_features.c: Handle the case where there are multiple
dynamic features with the same digit mapping, but won't always
match the activated on/by access controls. In that case, the code
needs to keep trying features for a match. (reported by Atis on
the asterisk-dev list, patched by me)
2007-09-17 16:40 +0000 [r82590-82592] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: revert a change that wasn't supposed to be
committed... doh!
* apps/app_queue.c, channels/chan_iax2.c: fix a couple of places
where a logical member name (if specified) was not used, but
instead the direct interface was listed
2007-09-17 02:00 +0000 [r82514] Joshua Colp <jcolp@digium.com>
* main/pbx.c: (closes issue #10734) Reported by: asgaroth Instead
of passing a NULL pointer into snprintf pass "". It makes Solaris
much happier.
2007-09-14 21:19 +0000 [r82444] Steve Murphy <murf@digium.com>
* main/cdr.c: closes issue #10668; thanks to arkadia for his patch;
had to leave out the bit about ending the previous cdr in the
fork; it would destroy current implementations.
2007-09-14 21:17 +0000 [r82435] Russell Bryant <russell@digium.com>
* configs/zapata.conf.sample: Add a note to help clarify the value
set with the echocancel option. (inspired by Malcolm's blog post
on blogs.digium.com about HPEC)
2007-09-14 18:35 +0000 [r82396-82398] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Crap, I broke the build. Fixed.
* apps/app_queue.c: Adding member name field to manager events
where they were missing before (closes issue #10721, reported by
snar)
2007-09-14 17:48 +0000 [r82394] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: If a channel does not have an owner, do not
try to set a channel variable. This will end up making the
channel variable global, which is not right. Closes issue #10720,
patch by flefoll.
2007-09-14 15:50 +0000 [r82382-82385] Russell Bryant <russell@digium.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
checking for libusb here, so nobody has to deal with conflicts in
the chan_usbradio-1.4 branch every time the configure script gets
changed
* channels/chan_usbradio.c (removed), channels/xpmr (removed),
channels/Makefile: Remove chan_usbradio from the main 1.4 branch.
It can't live here because we have a strict policy to not include
new features in release branches. However, I'm going to merge it
into trunk, and I also have a special 1.4 based branch that
includes this module. svn co
http://svn.digium.com/svn/asterisk/team/jdixon/chan_usbradio-1.4
2007-09-14 14:42 +0000 [r82376] Mark Michelson <mmichelson@digium.com>
* doc/CODING-GUIDELINES: Fixing a typo in the coding guidelines
(closes issue #10717, reported and patched by leedm777)
2007-09-14 01:24 +0000 [r82368] Jim Dixon <telesistant@hotmail.com>
* apps/app_rpt.c: Fixed problem with changes made to cdr
functionality
2007-09-14 00:52 +0000 [r82367] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_usbradio.c: this new driver may not live in this
branch for long (since it is a new feature), but it definitely
should not be built by default
2007-09-14 00:34 +0000 [r82366] Jim Dixon <telesistant@hotmail.com>
* apps/app_rpt.c, channels/xpmr/xpmr_coef.h (added),
channels/chan_usbradio.c (added), channels/xpmr/xpmr.h (added),
channels/xpmr (added), channels/xpmr/LICENSE (added),
channels/xpmr/sinetabx.h (added), configs/usbradio.conf.sample
(added), channels/Makefile, channels/xpmr/xpmr.c (added): Added
channel driver for USB Radio device and support thereof.
2007-09-13 23:11 +0000 [r82358] Jason Parker <jparker@digium.com>
* pbx/pbx_spool.c: Fix a small typo. retrytime > waittime
2007-09-13 20:16 +0000 [r82346] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Preemptively fixing a possible segfault. It is
possible that queuename is NULL (meaning pause ALL queues), so
use q->name instead.
2007-09-13 20:11 +0000 [r82344] Jason Parker <jparker@digium.com>
* cdr/cdr_csv.c: Fix a crash that could occur in cdr_csv when
mutliple threads tried to close the same file. Do we actually
need the locking here? What happens if you open the same file
twice, and two threads try to write to it at the same time? Is
fputs() going to write out the entire line at once? I suspect
that it could be possible for the second fopen to run during the
first fputs, so the position could be in the middle of the
previously written line... Issue 10347, initial patch by
explidous (but I removed all of the paranoia stuff..)
2007-09-13 18:57 +0000 [r82337-82339] Russell Bryant <russell@digium.com>
* main/astobj2.c: resolve a warning when not building under dev
mode
* main/astobj2.c, main/asterisk.c, include/asterisk.h: Only compile
in tracking astobj2 statistics if dev-mode is enabled. Also, when
dev mode is enabled, register the CLI command that can be used to
run the astobj2 test and print out statistics.
2007-09-13 18:12 +0000 [r82335] Kevin P. Fleming <kpfleming@digium.com>
* /, LICENSE: Merged revisions 82334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007)
| 2 lines clarify the OpenSSL and OpenH323 license exceptions
........
2007-09-13 16:25 +0000 [r82326] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Added logic to handle the unlikely case that
someone has two queues with the same name. Asterisk will log a
warning message letting the user know that one was already
defined with that name and is it skipping all further instances.
This also will work for realtime queues but in order for that to
happen, the user would have to trigger a perfectly timed reload
as a realtime queue is being looked up, which is highly unlikely
(but taken care of nonetheless).
2007-09-13 11:47 +0000 [r82309] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Closes issue #9401, reported and patched
by irrot, with slight modifications by me. Handle DTMF sent by
Asterisk properly.
2007-09-12 21:56 +0000 [r82296] Russell Bryant <russell@digium.com>
* res/res_agi.c: Fix a check of the wrong pointer, as pointed out
by an XXX comment left in the code. The problem was harmless,
however.
2007-09-12 21:28 +0000 [r82291] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/tzfile.h: Oops, wrong location for FreeBSD zone
files
2007-09-12 20:24 +0000 [r82286] Dwayne M. Hubbard <dhubbard@digium.com>
* apps/app_meetme.c: remove a race condition for the creation of
recordthread's, and fix a small memory leak. This closes issue#
10636
2007-09-12 20:12 +0000 [r82285] Tilghman Lesher <tlesher@digium.com>
* main/stdtime/private.h, main/stdtime/tzfile.h,
include/asterisk/localtime.h, main/stdtime/localtime.c: Working
on issue #10531 exposed a rather nasty 64-bit issue on
ast_mktime, so we updated the localtime.c file from source. Next
we'll have to write ast_strptime to match.
2007-09-12 15:16 +0000 [r82278-82280] Russell Bryant <russell@digium.com>
* main/asterisk.c: Clean up the output of "asterisk -h". This
tweaks the wording and wraps lines at 80 characters. (closes
issue #10699, seanbright)
* res/res_agi.c: revert patch from issue #10553, as someone not
using fastagi reported that this broke their system.
2007-09-12 14:30 +0000 [r82274-82276] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Accidentally committed changes to
app_voicemail which do NOT need to be in the 1.4 branch yet.
reverting...
* apps/app_voicemail.c, apps/app_queue.c: We should only initialize
a realtime queue when it is allocated, not every time we access
it. This prevents the members ao2_container from being
reallocated every time the queue is accessed. I also removed a
debug message I had accidentally left in on a previous commit.
2007-09-11 22:37 +0000 [r82267] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Fix incorrect uses of ao2_find(). Every one of
these calls was reading bogus memory ...
2007-09-11 21:41 +0000 [r82265] Joshua Colp <jcolp@digium.com>
* codecs/gsm/src/lpc.c, codecs/gsm/src/long_term.c: (closes issue
#10679) Reported by: andrew Build under dev mode when K6OPTS is
enabled.
2007-09-11 20:49 +0000 [r82263] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Fix another missing unref of member objects.
This one was pointed out by Marta. When building the outgoing
list in try_calling(), a member reference is stored in each
outgoing entry. However, when this list got destroyed, the
reference was not released.
2007-09-11 20:36 +0000 [r82261] Steve Murphy <murf@digium.com>
* main/cdr.c: this change should fix issue # 10659 -- what I worry
about is how many other bug reports it may generate. Hopefully,
we can please the/a majority. Hopefully. We shall see. Calls not
marked ANSWERED and with only one channel name will not be
posted. This should eliminate the double CDR's.
2007-09-11 16:05 +0000 [r82252] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: All instances of ao2_iterators which were just
named 'i' have been renamed to 'mem_iter' so that when refcounted
queues are merged into trunk, there will be little confusion
regarding iterator names, especially when a queue and member
iterator are used in the same function.
2007-09-11 16:03 +0000 [r82250] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: The sample dundi.conf claims support for a
wildcard peer entry - [*], but the code did not support it. This
patch makes it work. (closes issue #10546, patch by dds, with
some changes by me)
2007-09-11 16:01 +0000 [r82249] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed a
hold/retrieve issue.
2007-09-11 15:26 +0000 [r82245] Russell Bryant <russell@digium.com>
* res/res_agi.c: (closes issue #10553) Reported by: juggie Patches:
res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by:
juggie When using fastagi, fgets() can return before a full line
is read. Add explicit handling for the case where it gets
interrupted.
2007-09-11 14:56 +0000 [r82243] Joshua Colp <jcolp@digium.com>
* pbx/pbx_dundi.c: (closes issue #10577) Reported by: jamesgolovich
Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich
(license 176) Don't leak memory when unloading DUNDi.
2007-09-11 14:34 +0000 [r82198-82240] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Add a couple more missing unrefs of queue
member objects
* apps/app_queue.c: Add a missing unref of a queue member in an
error handling block
* apps/app_queue.c: Document why membercount can not simply be
replaced by ao2_container_count()
* main/astobj2.c: backport astobj2 race condition fix. This
function is the exact same as trunk so it applies here as well.
2007-09-10 18:02 +0000 [r82155] Tilghman Lesher <tlesher@digium.com>
* apps/app_queue.c: Convert struct member to use refcounts (closes
issue #10199)
2007-09-10 15:02 +0000 [r82091] Mark Michelson <mmichelson@digium.com>
* configs/misdn.conf.sample: Removing non-existent options from
misdn configuration sample. (closes issue #10678, reported and
patched by IgorG)
2007-09-09 02:35 +0000 [r82028] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Fix inline compiles on really old
compilers (who uses gcc 2.7 anymore, really?)
2007-09-08 18:41 +0000 [r81952-81997] Russell Bryant <russell@digium.com>
* main/asterisk.c: Fix a small memory leak. ast_unregister_atexit()
did not free the entry it removed.
* .cleancount: (closes issue #10672) Bump the cleancount so that a
"make clean" will be forced. This is needed because my fix in
revision 81599 made a change to a data structure in file.h, and
since file dependency tracking is only on with dev-mode enabled,
file format modules that don't get rebuilt may crash, as is the
case with this issue. This makes me wonder - how much faster does
the code build without the file dependency tracking enabled? If
it doesn't make much of a difference, then it may be worth just
keeping it on all of the time, or perhaps just not in release
tarballs, so that this type of issue is avoided.
2007-09-07 19:48 +0000 [r81923] Jason Parker <jparker@digium.com>
* apps/app_queue.c: Allow the MEMBERINTERFACE variable to be used
as the mixmonitor filename. This moves the setting of the
MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch
by sim.
2007-09-07 15:25 +0000 [r81886] Mark Michelson <mmichelson@digium.com>
* configs/queues.conf.sample: Moving the explanation for joinempty
to a more appropriate place
2007-09-06 22:28 +0000 [r81832] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: (closes issue #9724, closes issue #10374)
Reported by: kenw Patches: 9724.txt uploaded by russell (license
2) Tested by: kenw, russell Resolve a deadlock that occurs when
doing a SIP transfer to parking. I come across this type of
deadlock fairly often it seems. It is very important to mind the
boundary between the channel driver and the core in respect to
the channel lock and the channel-pvt lock. Channel drivers lock
to lock the pvt and then the channel once it calls into the core,
while the core will do it in the opposite order. The way this is
avoided is by having channel drivers either release their pvt
lock while calling into the core, or such as in this case,
unlocking the pvt just long enough to acquire the channel lock.
2007-09-06 22:05 +0000 [r81778-81826] Jason Parker <jparker@digium.com>
* Makefile: We added COPTS for ASTCFLAGS additions, but not LDOPTS
for ASTLDFLAGS. This adds LDOPTS
* include/asterisk/astobj2.h: This should fix a build issue that
people building against uClibc were seeing with the addition of
astobj2
2007-09-06 19:40 +0000 [r81776] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: (closes issue #10122) Reported by:
stevefeinstein Patches: meetme-unmute-manager.diff uploaded by
qwell (license 4) Tested by: stevefeinstein After looking over
the code I agree with Qwell. Setting the file descriptor to
conference each time just causes a fight back and forth.
2007-09-06 16:56 +0000 [r81743] Philippe Sultan <philippe.sultan@gmail.com>
* include/asterisk/jabber.h, channels/chan_gtalk.c: Various string
length fixes. Removed an unused variable in aji_client structure
(context)
2007-09-06 16:25 +0000 [r81682-81713] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixes an issue where valid DTMF had to be
pressed twice to exit a queue if a member's phone was ringing.
(closes issue #10655, reported by strider2k, patched by me)
* res/res_features.c: Fixes a memory leak (closes issue #10658,
reported and patched by Ivan)
2007-09-06 14:20 +0000 [r81650] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: According to both RFC 3920 - section 9.1.2 -
and Google's XMPP server complaint, if set, the 'from' attribute
must be set to the user's full JID.
2007-09-05 20:53 +0000 [r81599] Russell Bryant <russell@digium.com>
* include/asterisk/file.h, main/say.c, res/res_features.c,
main/file.c, include/asterisk/channel.h: Fix an issue that can
occur when you do an attended transfer to parking. If you
complete the transfer before the announcement of the parking spot
finishes, then the channel being parked will hear the remainder
of the announcement. These changes make it so that will not
happen anymore. Basically, res_features sets a flag on the
channel is playing the announcement to so that the file streaming
core knows that it needs to watch out for a channel masquerade,
and if it occurs, to abort the announcement. (closes BE-182)
2007-09-05 17:18 +0000 [r81569] Tilghman Lesher <tlesher@digium.com>
* include/asterisk/lock.h: Solaris x86 compatibility fix
2007-09-05 15:19 +0000 [r81525] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixing the build...
2007-09-05 15:14 +0000 [r81523] Jason Parker <jparker@digium.com>
* channels/chan_phone.c: Do not try to unregister a NULL channel
tech. Also changed load_module function to use defines rather
than numbers for return values. Issue 10651, patch by
rbraun_proformatique, with additions by me.
2007-09-05 15:03 +0000 [r81520] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Reverting behavior of QUEUE_MEMBER_COUNT to
only count members who are logged in and available. (related to
issue #10652, reported by wuwu)
2007-09-05 13:11 +0000 [r81492] Joshua Colp <jcolp@digium.com>
* main/channel.c: (closes issue #10650) Reported by: tacvbo Only
print out that the spy was removed while holding the spy lock.
2007-09-04 20:54 +0000 [r81453-81455] Jason Parker <jparker@digium.com>
* apps/app_followme.c: Rather than attempt to play a file, we can
just check whether it exists. Issue 10634, patch by me, testing
by pabelanger, sanity checked by bweschke
* configs/followme.conf.sample: Change default followme config file
to point to the correct files. Issue 10644, patch by pabelanger
2007-09-04 18:37 +0000 [r81448] Russell Bryant <russell@digium.com>
* main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
Remove the typedefs on ao2_container and ao2_iterator. This is
simply because we don't typedef objects anywhere else in
Asterisk, so we might as well make this follow the same
convention.
2007-09-04 16:40 +0000 [r81442] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: there is no point in sending 401
Unauthorized to a UAS that sent us a properly-formatted
Authentication header with the expected username and nonce but an
incorrect response (which indicates the shared secret does not
match)... instead, let's send 403 Forbidden so that the UAS
doesn't retry with the same authentication credentials repeatedly
2007-09-04 14:23 +0000 [r81435-81439] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: (closes issue #10632) Reported by:
jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded
by jamesgolovich (license 176) Fix memory leak when unloading
chan_iax2. The firmware files were not being freed.
* main/channel.c: (closes issue #10476) Reported by: mdu113 Only
look for the end of a digit when waiting for a digit. This in
turn disables emulation in the core.
* main/dns.c: (closes issue #10610) Reported by: john Patches:
dns.c.patch uploaded by john (license 218) Tested by: mvanbaak
Don't return a match if no SRV record actually exists.
2007-09-03 18:57 +0000 [r81433] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Remove a couple of calls to
ast_string_field_free_pools() on peers in error handling blocks
in the code for building peers. The peer object destructor does
this and doing it twice will cause a crash. (closes issue #10625,
reported by and patched by pnlarsson)
2007-09-01 15:57 +0000 [r81426-81428] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Changed a comment to be more accurate. (really
this is just a test to make sure I can commit properly from home)
* main/astobj2.c, include/asterisk/astobj2.h: Making match_by_addr
into ao2_match_by_addr and making it available everywhere since
it could be a handy callback to have
2007-08-31 21:27 +0000 [r81418] Russell Bryant <russell@digium.com>
* include/asterisk/astobj2.h: Remove references to a debugging
parameter that does not exist
2007-08-31 19:48 +0000 [r81416] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixed broken behavior of a reload on realtime
queues. Prior to this patch, if a reload was issued and a
realtime queue had callers waiting in it, then the queue would be
removed from the queue list, but it would not actually be freed
(in fact, a debug message warning about a memory leak would come
up). With this patch, reloads do not touch realtime queues at
all.
2007-08-31 19:16 +0000 [r81415] Tilghman Lesher <tlesher@digium.com>
* funcs/func_logic.c: The IF() function was not allowing true
values that had embedded colons (closes issue #10613)
2007-08-31 18:44 +0000 [r81412] Jason Parker <jparker@digium.com>
* apps/app_dial.c: Re-order dial options to be in line with the
existing alpha order. Issue 10621, initial patch by junky
2007-08-31 17:38 +0000 [r81410] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Make the 'gtalk show channels' CLI command
available. Closes issue 10548, reported by keepitcool.
2007-08-31 15:53 +0000 [r81406] Joshua Colp <jcolp@digium.com>
* res/res_speech.c: Make it the engine's responsible to check for
the presence of results.
2007-08-31 15:51 +0000 [r81405] Kevin P. Fleming <kpfleming@digium.com>
* codecs/codec_zap.c: add missing "transcoder show" (and deprecated
"show transcoder") CLI commands that were in 1.2 but never added
to 1.4
2007-08-31 14:38 +0000 [r81401-81403] Joshua Colp <jcolp@digium.com>
* res/res_features.c: (closes issue #10618) Reported by: dimas
Don't pass through the stopped sounds frame.... just drop it.
* res/res_features.c: (closes issue #10009) Reported by: dimas
Don't output a bridge failed warning message if it failed because
one of the channels was part of the masquerade process. That is
perfectly normal.
2007-08-30 22:05 +0000 [r81397] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Removing an extraneous (and possibly
misleading) log message. Firstly, if the announce file isn't
found, the streaming functions will report it. Secondly, not all
non-zero returns from play_file mean that the announce file
wasn't found. Positive return values simply mean that a digit was
pressed (most likely to skip through the announcement). (closes
issue #10612, reported and patched by dimas)
2007-08-30 21:23 +0000 [r81395] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: (closes issue #10514) Reported by: casper
Patches: chan_sip.c.80129.diff uploaded by casper (license 55)
Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible
for it to ever be that value.
2007-08-30 21:11 +0000 [r81392] Steve Murphy <murf@digium.com>
* main/cdr.c: via issue 10599, where 'CDR already initialized'
messages are being generated. Since all channels will have an
init'd CDR attached at creation time, this message is now
particularly useless. Removed.
2007-08-30 15:38 +0000 [r81383] Russell Bryant <russell@digium.com>
* channels/h323/ast_h323.cxx: Add missing checks for the PTRACING
define. (closes issue #10559, paravoid)
2007-08-30 15:35 +0000 [r81381] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Changed some manager event messages to reflect
whether a queue member is a realtime member or not
2007-08-30 15:33 +0000 [r81379] Russell Bryant <russell@digium.com>
* configs/modem.conf.sample (removed), configs/enum.conf.sample,
configs/extensions.ael.sample: Fix a typo, update a reload
command, and remove an unused configuration file. (closes issue
#10606, casper)
2007-08-30 14:53 +0000 [r81375] Joshua Colp <jcolp@digium.com>
* main/pbx.c: (closes issue #10603) Reported by: jmls Patches:
pbx.diff uploaded by jmls (license 141) Backport changes from
81372. Add REASON dialplan variable for when an originated call
fails and the failed extension is executed.
2007-08-30 14:43 +0000 [r81373] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: Fixed some warnings.
2007-08-30 14:23 +0000 [r81369] Joshua Colp <jcolp@digium.com>
* res/res_features.c: (issue #10599) Reported by: dimas Handle the
-1 control subclass during feature dialing (it indicates to stop
sounds).
2007-08-30 08:31 +0000 [r81367] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: Fixed a severe
issue where a misdn_read would lock the channel, but read would
not return because it blocks. later chan_misdn would try to queue
a frame like a AST_CONTROL_ANSWER which could result in a
deadlock situation. misdn_read will now not block forever
anymore, and we don't queue the ANSWER frame at all when we
already was called with misdn_answer -> answer would be called
twice. Also we don't explicitly send a RELEASE_COMPLETE on
receiption of a RELEASE anymore, because mISDN does that for us,
this resulted in a problem on some switches, which would block
our port after some calls for a short while.
2007-08-29 16:35 +0000 [r81346-81349] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This patch, in essence, will correctly pause a
realtime queue member and reflect those changes in the realtime
engine. (issue #10424, reported by irroot, patch by me) This
patch creates a new function called update_realtime_member_field,
which is a generic function which will allow any one field of a
realtime queue member to be updated. This patch only uses this
function to update the paused status of a queue member, but it
lays the foundation for persisting the state of a realtime member
the same way that static members' state is maintained when using
the persistentmembers setting
* apps/app_queue.c: Changed some tabs to spaces
2007-08-29 15:57 +0000 [r81342] Russell Bryant <russell@digium.com>
* main/Makefile: If chan_h323 is not being built, don't use g++ to
do the final link of Asterisk. (in response to a question on the
asterisk-dev list)
2007-08-29 15:52 +0000 [r81340] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: This fix creates a more accurate way of
detecting whether realtime members were deleted. (closes issue
10541, reported by Alric, patched by me) The REALLY nice things
about this patch is that queue members now have a "realtime"
field which will be true if the member is a realtime member. This
means we can check this value prior to certain processing if it
should ONLY be done for realtime members.
2007-08-29 14:13 +0000 [r81331] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: (closes issue #9690) Reported by: mattv Make
rtp timeouts work even if two RTP streams are directly bridged in
the RTP stack.
2007-08-28 21:38 +0000 [r81226-81291] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Change the message about receiving a
mini-frame before the first full voice frame to a DEBUG message.
* pbx/pbx_dundi.c: revert unintentional changes in rev 81226
* configs/indications.conf.sample, pbx/pbx_dundi.c: Add Russian
tones. (closes issue #7953, hanabana)
2007-08-28 14:12 +0000 [r81120-81189] Mark Michelson <mmichelson@digium.com>
* contrib/scripts/vmail.cgi: Fixes a forwarding problem when using
res_config_mysql (closes issue #10573, reported by chrisvaughan,
patch suggested by chrisvaughan as well)
* apps/app_queue.c: Resolve a potential deadlock. In this case, a
single queue is locked, then the queue list. In changethread(),
the queue list is locked, and then each individual queue is
locked. Under the right circumstances, this could deadlock. As
such, I have unlocked the individual queue before locking the
queue list, and then locked the queue back after the queue list
is unlocked.
* channels/chan_agent.c: DTMF begin frames should be ignored so
that when an agent acks a call with the '#' key, he doesn't cause
a queue's announce file to be interrupted. Also went ahead and
did the same for the '*' key and for ending a call. (closes issue
#10528, reported by deskhack, patched by me)
2007-08-27 17:27 +0000 [r81042-81074] Russell Bryant <russell@digium.com>
* pbx/pbx_dundi.c: Add a \todo to note that this module leaks most
of the memory it allocates on unload and should be fixed (when
I'm not in the middle of something else ...).
* pbx/pbx_dundi.c: explicity define a variable as a boolean
* res/res_musiconhold.c: (closes issue #10419) Reported by:
mustardman Patches: asterisk-mohposition.diff.txt uploaded by
jamesgolovich (license 176) This patch fixes a few problems with
music on hold. * Fix issues with starting at the beginning of a
file when it shouldn't. * Fix the inuse counter to be decremented
even if the class had not been set to be deleted when not in use
anymore * Don't arbitrarily limit the number of MOH files to 255
2007-08-27 15:01 +0000 [r81012] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: (closes issue #10561) Reported by: jesselang
Patches: chan_sip-ChannelReload-20080825.patch uploaded by
jesselang (license 202) Remove an extra \r\n to make the
ChannelReload event conform with every other event.
2007-08-27 14:55 +0000 [r81010] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Found a case where the queue's membercount is
off. It does not take into account dynamic members on a reload.
2007-08-27 13:20 +0000 [r80974] Joshua Colp <jcolp@digium.com>
* main/rtp.c: (closes issue #10562) Reported by: idkpmiller Correct
jitter value output in the CLI to be as expected.
2007-08-26 18:11 +0000 [r80932] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Remove an extra signal_condition() for the
scheduler thread. (closes issue #10564, patch from casper)
2007-08-25 17:37 +0000 [r80895] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix some issues with the handling of the
scheduler in chan_iax2. Most of the places that scheduled items
to be executed by the scheduler thread did not signal the
scheduler thread to wake up so that it could recalculate the time
until the next action. These changes will make the scheduler
thread more responsive and ensure that actions get executed as
close to when intended as possible instead of it being possible
for very long delays.
2007-08-24 22:59 +0000 [r80878] Dwayne M. Hubbard <dhubbard@digium.com>
* apps/app_zapateller.c: An empty string is an empty callerid ...
so zap it. This closes issue #10502, which was pointed out by
dswartz. Thank you, and may the swartz be with you
2007-08-24 21:22 +0000 [r80820-80849] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: If dnsmgr is in use, and no DNS servers are
available when Asterisk first starts, then don't give up on
poking peers. Allow the poke to get rescheduled so that it will
work once the dnsmgr is able to resolve the host. (closes issue
#10521, patch by jamesgolovich)
* main/dsp.c: Improve the debouncing logic in the DTMF detector to
fix some reliability issues. Previously, this code used a shift
register of hits and non-hits. However, if the start of the digit
isn't clean, it is possible for the leading edge detector to miss
the digit. These changes replace the flawed shift register logic
and also does the debouncing on the trailing edge as well.
(closes issue #10535, many thanks to softins for the patch)
2007-08-24 19:52 +0000 [r80818] BJ Weschke <bweschke@btwtech.com>
* apps/app_queue.c: A minor correction to the available logic of
autofill. If a queue member is paused, they're not really
"available" so don't count them as such. Somewhat related to
issue #10155
2007-08-24 18:52 +0000 [r80789] Steve Murphy <murf@digium.com>
* main/cdr.c: From a complaint by jmls, I realize that the message
in cdr_disposition is unnecessary. To get failure disposition,
just return -1; no use having more than one case do that.
2007-08-24 15:51 +0000 [r80750] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix a possible crash in IMAP voicemail.
2007-08-24 15:41 +0000 [r80747] Tilghman Lesher <tlesher@digium.com>
* main/pbx.c, UPGRADE.txt: Make the deprecation warning inline with
the code, instead of only in documentation (closes issue #10549)
2007-08-24 15:28 +0000 [r80722] Russell Bryant <russell@digium.com>
* utils/ael_main.c: Tweak the formatting of this MODULEINFO block.
I think this would have caused a "*" to get in the
menuselect-tree file.
2007-08-24 14:48 +0000 [r80689-80717] Steve Murphy <murf@digium.com>
* utils/ael_main.c: This change addresses JerJer's complaint that
aelparse builds and installs even if pbx_ael is unchecked in the
menuselect stuff.
* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test6:
backport of 80649, a fix to an unreported problem in the ael
parser, that results in a crash on a 64bit machine
2007-08-24 11:42 +0000 [r80661] Philippe Sultan <philippe.sultan@gmail.com>
* channels/chan_gtalk.c: Closes issue #10509 Googletalk calls are
answered too early, which results in CDRs wrongly stating that a
call was ANSWERED when the calling party cancelled a call before
before being established. We must not answer the call upon
reception of a 'transport-accept' iq packet, but this packet
still needs to be acknowledged, otherwise the remote peer would
close the call (like in #8970).
2007-08-23 21:34 +0000 [r80601-80617] Dwayne M. Hubbard <dhubbard@digium.com>
* channels/misdn/isdn_lib.c: make misdn/isdn_lib compile without
warnings
* channels/chan_misdn.c: make chan_misdn compile without warnings
2007-08-23 20:16 +0000 [r80539-80573] Russell Bryant <russell@digium.com>
* include/asterisk/features.h, res/res_features.c: When executing a
dynamic feature, don't look it up a second time by digit pattern
after we already looked it up by name. This causes broken
behavior if there is more than one feature defined with the same
digit pattern. (closes issue #10539, reported by bungalow, patch
by me)
* funcs/func_timeout.c: Revert very broken fix for issue #10540 ...
none of these values take ms so I don't know what I was thinking
* funcs/func_timeout.c: Fix func_timeout to take values in floating
point so 1.5 actually means 1.5 seconds instead of being rounded.
(closes issue #10540, reported by spendergrass, patch by me)
2007-08-23 17:14 +0000 [r80505-80507] Jason Parker <jparker@digium.com>
* /: *sigh*
* /: use autotagged externals
2007-08-23 17:08 +0000 [r80501] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: report the actual channel number that was
unregistered, instead of assuming that the interface list
consists of channels 1 through <x> with no gaps in the sequence
2007-08-23 17:02 +0000 [r80360-80499] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix some code where it was possible for a
reference to a peer to not get released when it should. Thank you
to Marta Carbone for pointing this out!
* main/astobj2.c, include/asterisk/astobj2.h, channels/chan_iax2.c:
This is a hack to maintain old behavior of chan_iax2. This
ensures that if the peers and users are being stored in a linked
list, that they go in the list in the same order that the older
code used. This is necessary to maintain the behavior of which
peers and users get matched when traversing the container.
* res/res_agi.c: Revert res_agi fix that didn't quite work until we
get it right ...
* include/asterisk/astobj2.h: Add some more documentation on
iterating ao2 containers. The documentation implies that is
possible to miss an object or see an object twice while
iterating. After looking through the code and talking with
mmichelson, I have documented the exact conditions under which
this can happen (which are rare and harmless in most cases).
* main/astobj2.c: When converting this code to use the list macros,
I changed it so objects are added to the head of a bucket instead
of the tail. However, while looking over code with mmichelson, we
noticed that the algorithm used in ao2_iterator_next requires
that items are added to the tail. This wouldn't have caused any
huge problem, but it wasn't correct. It meant that if an object
was added to a container while you were iterating it, and it was
added to the same bucket that the current element is in, then the
new object would be returned by ao2_iterator_next, and any other
objects in the bucket would be bypassed in the traversal.
* channels/chan_sip.c: Don't crash when using realtime in chan_sip
without an insecure setting in the database. (closes issue
#10348, reported by link55, fixed by me)
* main/astobj2.c (added), main/Makefile, include/asterisk/astobj2.h
(added), doc/iax.txt, UPGRADE.txt, include/asterisk/strings.h,
channels/chan_iax2.c: Merge changes from
team/russell/iax_refcount. This set of changes fixes problems
with the handling of iax2_user and iax2_peer objects. It was very
possible for a thread to still hold a reference to one of these
objects while a reload operation tries to delete them. The fix
here is to ensure that all references to these objects are
tracked so that they can't go away while still in use. To
accomplish this, I used the astobj2 reference counted object
model. This code has been in one of Luigi Rizzo's branches for a
long time and was primarily developed by one of his students,
Marta Carbone. I wanted to go ahead and bring this in to 1.4
because there are other problems similar to the ones fixed by
these changes, so we might as well go ahead and use the new
astobj if we're going to go through all of the work necessary to
fix the problems. As a nice side benefit of these changes, peer
and user handling got more efficient. Using astobj2 lets us not
hold the container lock for peers or users nearly as long while
iterating. Also, by changing a define at the top of chan_iax2.c,
the objects will be distributed in a hash table, drastically
increasing lookup speed in these containers, which will have a
very big impact on systems that have a large number of users or
peers. The use of the hash table will be made the default in
trunk. It is not the default in 1.4 because it changes the
behavior slightly. Previously, since peers and users were stored
in memory in the same order they were specified in the
configuration file, you could influence peer and user matching
order based on the order they are specified in the configuration.
The hash table does not guarantee any order in the container, so
this behavior will be going away. It just means that you have to
be a little more careful ensuring that peers and users are
matched explicitly and not forcing chan_iax2 to have to guess
which user is the right one based on secret, host, and access
list settings, instead of simply using the username. If you have
any questions, feel free to ask on the asterisk-dev list.
* res/res_agi.c: Juggie in #asterisk-dev was reporting problems
where fgets would return without reading the whole line when
using fastagi. When this happens, errno was set to EINTR or
EAGAIN. This patch accounts for the possibility and lets fgets
continue in that case.
2007-08-22 18:53 +0000 [r80302-80330] Jason Parker <jparker@digium.com>
* Makefile, build_tools/mkpkgconfig, build_tools/make_build_h,
build_tools/strip_nonapi, build_tools/prep_moduledeps,
build_tools/make_buildopts_h: Fix a few build issues in Solaris
(and likely others). Use GREP and ID variables from autoconf.
Reported to me in #asterisk-dev I forgot who reported this -
sorry. :(
* Makefile: Change a syntax that the GNU make in Solaris dislikes.
* build_tools/make_version: Fix a bashism (we explicitly request
/bin/sh). Remove some oddly placed quotes I found in passing.
2007-08-22 16:21 +0000 [r80257] Russell Bryant <russell@digium.com>
* Makefile: Honor the contents of the COPTS variable as custom
target CFLAGS. Apparently this is what openwrt does. (reported by
Brian Capouch on the asterisk-dev list, patch by me)
2007-08-22 16:14 +0000 [r80255] Joshua Colp <jcolp@digium.com>
* main/rtp.c: (closes issue #10526) Reported by: sinistermidget
Revert commit from issue #10355 and return timestamp skew to 640.
2007-08-21 Russell Bryant <russell@digium.com>
* Asterisk 1.4.11 released.
2007-08-21 18:42 +0000 [r80183] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Don't record SIP dialog history if it's not
turned on. Also, put an upper limit on how many history entires
will be stored for each SIP dialog. It is currently set to 50,
but can be increased if deemed necessary. (closes issue #10421,
closes issue #10418, patches suggested by jmoldenhauer, patches
updated by me) (Security implications documented in AST-2007-020)
2007-08-21 16:39 +0000 [r80166-80167] Steve Murphy <murf@digium.com>
* include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
the diffs from ulaw.h and alaw.h for now; accidentally added them
in 80166
* main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
This patch solves problem 1 in 8126; it should not slow down the
alaw codec, but should prevent signal degradation via multiple
trips thru the codec. Fossil estimates the twice thru this codec
will prevent fax from working. 4-6 times thru would result
hearable, noticeable, voice degradation.
2007-08-21 15:22 +0000 [r80132] Russell Bryant <russell@digium.com>
* channels/chan_mgcp.c: Don't try to dereference the owner channel
when it may not exist (issue #10507, maxper)
2007-08-21 15:03 +0000 [r80130] Jason Parker <jparker@digium.com>
* configs/cdr.conf.sample: (issue #10510) Reported by: casper
Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
errors in sample cdr config file.
2007-08-20 21:57 +0000 [r80088] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Fix the build of app_queue
2007-08-20 21:39 +0000 [r80049-80086] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: After a discussion on #asterisk-dev, it was
decided that this should be in 1.4 as well. (issue #10424,
reported and patched by irroot)
* apps/app_queue.c: Found a pointless ternary if. member->dynamic
was set to 1 and has no opportunity to change between then and
this line, so "dynamic" will ALWAYS be output.
2007-08-20 16:08 +0000 [r80047] Jason Parker <jparker@digium.com>
* configs/extensions.conf.sample: (issue #10499) Reported by:
casper Patches: extensions.conf.sample.diff uploaded by casper
(license 55) Update CLI examples in extensions.conf.sample to
reflect command changes.
2007-08-20 15:34 +0000 [r80044] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Ukrainian language voicemail support.
(closes issue #10458, reported and patched by Oleh)
2007-08-20 02:42 +0000 [r79998] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
snuffy via IRC)
2007-08-18 14:30 +0000 [r79947] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Don't allocate vmu for messagecount when we
could just use the stack instead (closes issue #10490) Also,
remove a useless (and leaky) SQLAllocHandle (closes issue #10480)
2007-08-17 21:01 +0000 [r79912] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: Avoid a crash in the handling of DTMF based
Caller ID. It is valid for ast_read to return NULL in the case
that the channel has been hung up. (crash reported by
anonymouz666 on IRC in #asterisk-dev)
2007-08-17 19:14 +0000 [r79906] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Patch allows for more seamless transition
from file storage voicemail to ODBC storage voicemail. If a
retrieval of a greeting from the database fails, but the file is
found on the file system, then we go ahead an insert the greeting
into the database. The result of this is that people who switch
from file storage to ODBC storage do not need to rerecord their
voicemail greetings.
2007-08-17 19:12 +0000 [r79902-79904] Jason Parker <jparker@digium.com>
* channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
Don't send a semicolon over the wire in sip notify messages.
Caused by fix for issue 9938. I basically took the code that
existed before 9938 was fixed, and copied it into a new function
- ast_unescape_semicolon There should be very few places this
will be needed (pbx_config does NOT need this (see issue 9938 for
details)) Issue 10430, patch by me, with help/ideas from murf
(thanks murf).
* channels/chan_local.c: Re-add the setting of callerid name and
number. Issue 10485, reported by and fix explained by paradise.
2007-08-17 13:37 +0000 [r79857] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix some crashes in chan_sip. This patch
changes various places that add items to the scheduler to ensure
that they don't overwrite the ID of a previously scheduled item.
If there is one, it should be removed. (closes issue #10391,
closes issue #10256, probably others, patch by me)
2007-08-17 08:22 +0000 [r79833] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: sometimes we don't need to signal dtmf
tones to asterisk, we just want them to go through as inband.
Otherwise they might be generated by the other channel partner
and then there is a double tone.
2007-08-16 22:32 +0000 [r79756-79792] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: Fix a little race condition that could
cause a crash if two channels had MOH stopped at the same time
that were using a class that had been marked for deletion when
its use count hits zero.
* res/res_musiconhold.c: This patch fixes a bug where reloading the
module with "module reload" did not delete classes from memory
that were no longer in the config. This patch fixes that problem
as well as another one. Previously, if you reloaded MOH using the
"moh reload" CLI command, which behaved differently than "module
reload ...", MOH had to be stopped on every channel and started
again immediately. However, there was no way to tell what class
was being used, so they would all fall back to the default class.
(closes issue #10139) Reported by: blitzrage Patches:
asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
(license 176) Tested by: jamesgolovich
* channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
introduced by making frame handling and scheduling
multi-threaded. Unfortunately, we have to do some expensive
deadlock avoidance when queueing frames on to the ast_channel
owner of the IAX2 pvt struct. This was already handled for
regular frames, but ast_queue_hangup and ast_queue_control were
still used directly. Making these changes introduced even more
places where the IAX2 pvt struct can disappear in the context of
a function holding its lock due to calling a function that has to
unlock/lock it to avoid deadlocks. I went through and fixed all
of these places to account for this possibility. (issue #10362,
patch by me)
2007-08-16 21:16 +0000 [r79690-79748] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Fixes a problem where agents would get
stuck busy due to their wrapuptime being longer than the queue's
wrapuptime and ringinuse=no for the queue. (closes issue #10215,
reported by Doug, repaired by me) Special thanks to fkasumovic
for pointing out the source of the problem and to bweschke for
helping to come up with a solution!
* apps/app_voicemail.c: base_encode is not trying to open a log
file, so we should not call it a log file in the warning.
(related to issue #10452, reported by bcnit)
2007-08-16 09:37 +0000 [r79665] Philippe Sultan <philippe.sultan@gmail.com>
* res/res_jabber.c: A fix for two critical problems detected while
working with Daniel McKeehan in issue #10184. Upon priority
change, the resource list is not NULL terminated when moving an
item to the end of the list. This makes Asterisk endlessy loop
whenever it needs to read the list. Jids with different resource
and priority values, like in Gmail's and GoogleTalk's jabber
clients put that problem in evidence. Upon reception of a 'from'
attribute with an empty resource string, Asterisk crashes when
trying to access the found->cap pointer if the resource list for
the given buddy is not empty. This situation is perfectly valid
and must be handled. The Gizmoproject's jabber client put that
problem in evidence. Also added a few comments in the code as
well as a handle for the capabilities from Gmail's jabber client,
which are stored in a caps:c tag rather than the usual c tag.
Closes issue #10184.
2007-08-16 08:21 +0000 [r79642] Christian Richter <christian.richter@beronet.com>
* channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
we want to send IA5 Chars.
2007-08-15 14:40 +0000 [r79553] Joshua Colp <jcolp@digium.com>
* main/rtp.c: (closes issue #10440) Reported by: irroot (closes
issue #10454) Reported by: flo_turc Increase maximum timestamp
skew to 120. 20 was apparently far too low.
2007-08-15 14:26 +0000 [r79527] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fixed an error in the Russian language
voicemail intro. (issue #10458, reported and patched by Oleh)
2007-08-15 14:18 +0000 [r79523] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: (closes issue #10456) Reported by: irroot
Patches: sip_timeout.patch uploaded by irroot (license 52) Change
hardcoded timer value to defined value. I'm doing this in 1.4 as
well so if it needs to be changed in the future this place would
not have been forgotten.
2007-08-14 18:49 +0000 [r79436-79470] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix another spot where an iax2_peer would
be leaked if realtime was in use.
* channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
related to the use of realtime. I found these while working on
iax2_peer object reference tracking.
2007-08-14 15:27 +0000 [r79397] Joshua Colp <jcolp@digium.com>
* res/res_features.c: (closes issue #10415) Reported by: atis
Revert fix for #10327 as it causes more issues then it solves.
2007-08-13 22:40 +0000 [r79363] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: memset really, really needs to be used here.
2007-08-13 21:57 +0000 [r79334] Joshua Colp <jcolp@digium.com>
* res/res_speech.c, apps/app_speech_utils.c,
include/asterisk/speech.h: Instead of accepting a single DTMF
character accept a full string.
2007-08-13 20:37 +0000 [r79272-79301] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Don't call find_peer in
registry_authrequest with the pvt lock held to avoid a deadlock.
* channels/chan_iax2.c: Release the pvt lock before calling
find_peer in register_verify to avoid a deadlock. Also, remove
some unnecessary locking in auth_fail that was only done
recursively.
* channels/chan_iax2.c: Don't call find_peer within update_registry
with a pvt lock held. This can cause a deadlock as the code will
eventually call find_callno.
* channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
have tracked them down to a single core issue. You can not call
find_callno() while holding a pvt lock as this function has to
lock another (every) other pvt lock. Doing so can lead to a
classic deadlock. So, I am tracking down all of the code paths
where this can happen and fixing them. The fix I committed
earlier today was along the same theme. This patch fixes some
code down the path of authenticate_reply.
2007-08-13 17:49 +0000 [r79255] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-vtest21 (added),
pbx/ael/ael-test/ref.ael-test19,
pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
pbx/ael/ael-test/ael-vtest21 (added),
pbx/ael/ael-test/ref.ael-vtest17,
pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
fixes bug 10411. I added a new regression test, some regression
test cleanups
2007-08-13 15:28 +0000 [r79214] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a potential deadlock in socket_process.
check_provisioning can eventually call find_callno. You can't
hold a pvt lock while calling find_callno because it goes through
and locks every single one looking for a match.
2007-08-13 14:51 +0000 [r79174-79207] Joshua Colp <jcolp@digium.com>
* res/res_speech.c, apps/app_speech_utils.c,
include/asterisk/speech.h: Add an API call to allow the engine to
know that DTMF was received.
* channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
channels/chan_local.c, channels/chan_misdn.c,
channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
channels/chan_h323.c, channels/chan_gtalk.c,
channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
Don't set the callerid name and number a second time on a newly
created channel. ast_channel_alloc itself already sets it and
setting it twice would cause a memory leak.
2007-08-11 05:23 +0000 [r79142] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c: Ensure the connection gets marked as used at
allocation time (closes issue #10429, report and fix by
mnicholson)
2007-08-10 20:53 +0000 [r79044-79099] Steve Murphy <murf@digium.com>
* main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
a user complaint on #asterisk, I have forced pbx_spool to explain
what reason codes mean, when they are logged
* main/cdr.c: Re bug behavior mentioned in #asterisk, made this
tweak to code, to prevent hundreds of log messages from being
generated
* main/cdr.c: This will help debug; from a question asked on
#asterisk
2007-08-10 Russell Bryant <russell@digium.com>
* Asterisk 1.4.10.1 released.
2007-08-10 15:20 +0000 [r78995] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: The last set of changes that I made to
"core show locks" made it not able to track mutexes unless they
were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
with ast_mutex_init() were not tracked. It should work now.
2007-08-10 14:15 +0000 [r78951-78955] Joshua Colp <jcolp@digium.com>
* main/file.c: Don't bother having the core pass through or emulate
begin DTMF frames when in an ast_waitstream. It only cares about
the end of DTMF.
* configs/queues.conf.sample: (closes issue #10422) Reported by:
bhowell Add note to sample configuration about module load order
and how it can cause perfectly good queue members to be marked as
invalid.
2007-08-10 13:24 +0000 [r78936] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, channels/misdn/ie.c,
channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
information element. We send them now also in the disconnect
message.
2007-08-09 23:47 +0000 [r78907] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Improved a bit of logic regarding
comma-separated mailboxes in has_voicemail. Also added some
braces to some compound if statements since unbraced if
statements scare me in general.
2007-08-09 23:10 +0000 [r78891] Steve Murphy <murf@digium.com>
* Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
output
2007-08-09 22:03 +0000 [r78826-78860] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Removing some extra debug code I left in my
last commit
* apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
1. instead of using inboxcount as the core message counting
function, we use messagecount instead. This makes it possible to
count messages in folders besides just INBOX and Old. 2.
inboxcount and hasvoicemail now use messagecount as their means
of determining return values. 3. Added a copy_message function
for IMAP storage. Unfortunately I don't have the means to test
it, but it seems like a pretty straightforward function. 4.
Removed a #ifndef IMAP_STORAGE and matching #endif from
leave_voicemail for a couple of reasons. One, we want to support
copying mail to multiple IMAP boxes, and two, IMAP was broken
because a STORE macro had been moved into this section of code.
* channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
put some new code in the wrong place and so I've reverted the
canreinvite section to how it was and put my new code where it
should be.
2007-08-09 17:58 +0000 [r78717-78778] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: add a comment to indicate that inboxcount
for ODBC_STORAGE needs to be fixed to support multiple mailboxes
* apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
ODBC_STORAGE. Also, leave a comment for this to be fixed for
IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
working on this code right now for another reason. This is broken
even worse in trunk, but for a different reason. The fact that
the mailbox option supported multiple mailboxes is completely not
obvious from the code in the channel drivers. Anyway, I will fix
that in another commit ...
* apps/app_meetme.c: Fix a problem with the combination of the 'F'
option to pass DTMF through a conference and options that use
DTMF to activate various features. The problem was that the BEGIN
frame would be passed through, but the END frame would get
intercepted to activate a feature. Then, the other conference
members would hear DTMF for forever, which they didn't seem to
like very much. (closes issue #10400, reported by stevefeinstein,
fixed by me)
2007-08-08 19:29 +0000 [r78646] Jason Parker <jparker@digium.com>
* doc/jabber.txt: Fix mogs email address.
2007-08-08 18:16 +0000 [r78575-78620] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fixed some compiler warnings so that
compiling with dev-mode and IMAP storage would not have any
errors. This section of code may get changed again shortly since
my change uncovers a rather silly bit of logic.
* apps/app_queue.c: Changing a bit of logic so that someone will
NEVER exit the queue on timeout unless they have enabled the 'n'
option. This commit relates to issue #10320. Thanks to
jfitzgibbon for detailing the idea behind this code change.
2007-08-08 13:51 +0000 [r78569] Joshua Colp <jcolp@digium.com>
* configs/sip.conf.sample: (closes issue #10335) Reported by:
adamgundy Update sip.conf to include another scenario where
directrtpsetup will fail.
2007-08-07 Russell Bryant <russell@digium.com>
* Asterisk 1.4.10 released.
2007-08-07 20:57 +0000 [r78488] Russell Bryant <russell@digium.com>
* res/res_config_odbc.c: Fix the build of this module on 64-bit
platforms
2007-08-07 19:43 +0000 [r78450] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: The logic behind inboxcount's return value
was reversed in has_voicemail and message_count. (closes issue
#10401, reported by st1710, patched by me)
2007-08-07 19:34 +0000 [r78437] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c: Don't free the environment handle when the
connection fails, because other connections might be depending
upon it
2007-08-07 19:11 +0000 [r78416] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Allow chan_sip to build in devmode
2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, res/res_config_odbc.c,
apps/app_directory.c: Reconnection doesn't happen automatically
when a DB goes down (fixes issue #9389)
2007-08-07 18:25 +0000 [r78375] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Properly check the capabilities count to
avoid a segfault. (ASA-2007-019)
2007-08-07 17:45 +0000 [r78371] Russell Bryant <russell@digium.com>
* channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
4 lines Revert patch committed for issue #9660. It broke E&M
trunks. (closes issue #10360) (closes issue #10364) ........
2007-08-06 21:41 +0000 [r78275] Joshua Colp <jcolp@digium.com>
* main/channel.c: Add additional DTMF log messages to help when
debugging issues.
2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix an issue where dynamic threads can get
free'd, but still exist in the dynamic thread list. (closes issue
#10392, patch from Mihai, with credit to his colleague, Pete)
* include/asterisk/linkedlists.h: Fix the return value of
AST_LIST_REMOVE(). This shouldn't be causing any problems,
though, because the only code that uses the return value only
checks to see if it is NULL. (closes issue #10390, pointed out by
mihai)
2007-08-06 16:32 +0000 [r78182] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: It is possible for a transfer to occur
before the remote device has our tag in which case they send none
in the transfer. In this case we need to not fail the transfer
dialog lookup.
2007-08-06 16:30 +0000 [r78180] Jason Parker <jparker@digium.com>
* main/config.c: Fix an issue with using UpdateConfig (manager
action) where escaped semicolons in a config would be converted
to just semicolons (\; to ;) Issue 9938
2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp <jcolp@digium.com>
* main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
we pass through RTP timestamp information we need to make the
allowed timestamp skew considerably less. There are situations
where a source may change and due to the timestamp difference the
receiver will experience an audio gap since we did not indicate
by setting the marker bit that the source changed.
* configure, configure.ac: (closes issue #10383) Reported by: rizzo
Include stdlib.h so NULL gets defined for gethostbyname_r checks.
2007-08-06 13:33 +0000 [r78164] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fixed a mistake I made in realtime_peer
which caused it to return NULL every time. Thanks to Jon Fealy
for emailing me the correction.
2007-08-05 14:18 +0000 [r78146] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
bug #10382)
2007-08-05 04:15 +0000 [r78143] Russell Bryant <russell@digium.com>
* include/asterisk/lock.h: Fix compilation failure when
MALLOC_DEBUG is enabled, but DEBUG_THREADS is not
2007-08-05 03:29 +0000 [r78139] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* channels/chan_sip.c: If peer is not found, the error message is
misleading (should be peer not found, not ACL failure)
2007-08-03 20:25 +0000 [r78103] Mark Michelson <mmichelson@digium.com>
* main/config.c, channels/chan_sip.c, include/asterisk/config.h:
Changed the behavior of sip's realtime_peer function to match the
corresponding way of matching for non-realtime peers. Now matches
are made on both the IP address and port number, or if the
insecure setting is set to "port" then just match on the IP
address. In order to accomplish this, I also added a new API
call, ast_category_root, which returns the first variable of an
ast_category struct
2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: (closes issue #10194) Reported by:
blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
blitzrage Fix a problem when you call Voicemail() with multiple
mailboxes specified and ODBC_STORAGE is in use. The audio part of
the message was only given to the first mailbox specified.
* main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
improvements to lock debugging. These changes take effect with
DEBUG_THREADS enabled and provide the following: * This will keep
track of which locks are held by which thread as well as which
lock a thread is waiting for in a thread-local data structure. A
reference to this structure is available on the stack in the
dummy_start() function, which is the common entry point for all
threads. This information can be easily retrieved using gdb if
you switch to the dummy_start() stack frame of any thread and
print the contents of the lock_info variable. * All of the
thread-local structures for keeping track of this lock
information are also stored in a list so that the information can
be dumped to the CLI using the "core show locks" CLI command.
This introduces a little bit of a performance hit as it requires
additional underlying locking operations inside of every
lock/unlock on an ast_mutex. However, the benefits of having this
information available at the CLI is huge, especially considering
this is only done in DEBUG_THREADS mode. It means that in most
cases where we debug deadlocks, we no longer have to request
access to the machine to analyze the contents of ast_mutex_t
structures. We can now just ask them to get the output of "core
show locks", which gives us all of the information we needed in
most cases. I also had to make some additional changes to astmm.c
to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
enabled. I disabled tracking of one of the locks in astmm.c
because it gets used inside the replacement memory allocation
routines, and the lock tracking code allocates memory. This
caused infinite recursion.
* channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
frames when the mohinterpret option is set to "passthrough". This
was pointed out by Kevin in the middle of a training session.
* channels/chan_iax2.c: Don't reuse the timespec that was set to 0
in the previous timedwait as it will just return immediately.
Also, fix some logic so the thread's lock isn't unlocked twice in
the weird case of dynamic threads getting acquired right after a
timeout. (pointed out by SteveK)
2007-08-02 21:53 +0000 [r77993-77996] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
actually allow 6 chars to be sent. Also make note of the "A"
option of date format. Issue 9779, modifications by DEA, wedhorn,
and myself.
* channels/chan_skinny.c: If a device disconnects, the session will
go away. If this happens during call setup, we need to give up.
Issue 10325.
2007-08-02 19:25 +0000 [r77949] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix the case where a dynamic thread times
out waiting for something to do during the first time it runs.
This shouldn't ever happen, but we should account for it anyway.
(pointed out by pete, who works with mihai)
2007-08-02 18:42 +0000 [r77947] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Make sure we clear the prompt status
message on a hangup. Also rearrange messages to better fit with
what a wireshark trace shows it should be. Issue 10299, initial
patch and solution by sbisker, modified by me to fit with
wireshark trace.
2007-08-02 18:21 +0000 [r77945] Steve Murphy <murf@digium.com>
* main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
line This patch hopefully solves 10141; The user is running with
it, and it doesn't appear to harm asterisk's operation, and may
prevent a crash. I'll store it in 1.2, as we have shut down
support on 1.2, but since I developed the patch before support
finished, and it might affect 1.4 and trunk, I'm going ahead with
it. ........
2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix another race condition in the handling
of dynamic threads. If the dynamic thread timed out waiting for
something to do, but was acquired to perform an action
immediately afterwords, then wait on the condition again to give
the other thread a chance to finish setting up the data for what
action this thread should perform. Otherwise, if it immediately
continues, it will perform the wrong action. (reported on IRC by
mihai, patch by me) (related to issue #10289)
* channels/chan_iax2.c: Add another sanity check to
vnak_retransmit(). This check ensures that frames that have
already been marked for deletion don't get retransmitted. (closes
issue #10361, patch from mihai)
2007-08-02 15:15 +0000 [r77890-77894] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Make sure that we show the correct
extension if dialed from a macro "From: 5555" rather than "From:
s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
tested by sbisker
* channels/chan_skinny.c: Put in some additional debug information
for softkey/stimulus messages. Issue 10291, patch by DEA.
2007-08-01 22:16 +0000 [r77887] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix some race conditions which have been
causing weird problems in chan_iax2. The most notable problem is
that people have been seeing storms of VNAK frames being sent due
to really old frames mysteriously being in the retransmission
queue and never getting removed. It was possible that a dynamic
thread got created, but did not acquire its lock before the
thread that created it signals it to perform an action. When this
happens, the thread will sleep until it hits a timeout, and then
get destroyed. So, the action never gets performed and in some
cases, means a frame doesn't get transmitted and never gets freed
since the scheduler never gets a chance to reschedule
transmission. Another less severe race condition is in the
handling of a timeout for a dynamic thread. It was possible for
it to be acquired to perform at action at the same time that it
hit a timeout. When this occurs, whatever action it was acquired
for would never get performed. (patch contributed by Mihai and
SteveK) (closes issue #10289) (closes issue #10248) (closes issue
#10232) (possibly related to issue #10359)
2007-08-01 22:14 +0000 [r77886] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
not compile cleanly (missing def)
2007-08-01 21:08 +0000 [r77883] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix an issue that caused one-way audio on
some newer devices (specifically the 7921), due to sending
packets in the wrong order during hangup. Also make sure we clear
tones/messages on the correct line/instance. Issue 10291, patch
by DEA, tested by sbisker and myself.
2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp <jcolp@digium.com>
* main/cli.c: (closes issue #10351) Reported by: ftarz Some
platforms don't like it when you pass NULL to vsnprintf so pass
"" instead.
* include/asterisk/threadstorage.h, channels/chan_mgcp.c,
apps/app_voicemail.c, main/acl.c, utils/smsq.c,
channels/chan_iax2.c: Add some fixes for building on Solaris.
* main/utils.c: Whoops, I meant R_5 not R5.
* configure, configure.ac: And for my last trick... make sure that
if gethostbyname_r is exported by a library that it is used.
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/utils.c: Extend autoconf logic to determine which version of
gethostbyname_r is on the system.
2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Fixes an issue I introduced to queues wherein a
queue with joinempty=yes would kick people out of the queue
because of erroneously thinking the 'n' option was in use.
(closes issue #10320, reported by jfitzgibbon, patched by me,
tested by blitzrage and me) Thank you blitzrage for all the
testing you've done lately with queues! It's much appreciated!
* apps/app_queue.c: If a queue uses dynamic realtime members, then
the member list should be updated after each attempt to call the
queue. This fixes an issue where if a caller calls into a queue
where no one is logged in, they would wait forever even if a
member logged in at some point. (closes issue #10346, reported by
and tested by blitzrage, patched by me)
2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon <telesistant@hotmail.com>
* apps/app_rpt.c: Much newer version, 0.70 with much additions
* main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
logic in TONE_VERIFY/RELAX mode in chan_zap.
2007-07-31 20:59 +0000 [r77844] Steve Murphy <murf@digium.com>
* /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
line This probably isn't super-general, but it's a first stab at
using kill -11 to generate a core file instead of gcore. ........
2007-07-31 16:17 +0000 [r77831] Joshua Colp <jcolp@digium.com>
* res/res_speech.c, include/asterisk/speech.h: Add a flag to the
speech API that allows an engine to set whether it received
results or not.
2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
DEBUG_THREADS or it does nothing
2007-07-31 15:21 +0000 [r77824] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: This patch makes Asterisk send 100 Trying
provisional responses upon receipt of re-invites. This makes it
so that if there are two or more Asterisk servers between
endpoints, the Asterisk servers will not keep retransmitting the
re-invites. (closes issue #10274, reported by cstadlmann, patched
by me with approval from file)
2007-07-30 20:17 +0000 [r77795] Jason Parker <jparker@digium.com>
* main/say.c: Applications like SayAlpha() should not hang up the
channel if you request an "unknown" character such as a comma.
Instead, skip the character and move on. Issue 10083, initial
patch by jsmith, modified by me.
2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix an issue that could potentially cause
corruption of the global iax frame queue. In the network_thread()
loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
macro. However, to remove an element of the list within this
loop, it used AST_LIST_REMOVE, instead of
AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
internal variables of the SAFE macro invalid. Mihai says that he
already made this change in his local copy and it didn't help his
VNAK storm issues, but I still think it's wrong. :)
* res/res_agi.c: (closes issue #10279) Reported by: seanbright
Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
uploaded by seanbright (license 71) Allow the "agi_network: yes"
line to be printed out in the AGI debug output. Also, allow
partial writes to be handled when writing out this line just like
it is for all of the others.
* main/channel.c: file and I both committed changes for issue
#10301. Remove a duplicated assignment to restore the original
value of the previous channel.
2007-07-30 18:43 +0000 [r77783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, res/res_agi.c: Merged revisions 77782 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
| 2 lines Revert change in revision 71656, even though it fixed a
bug, because many people were depending upon the (broken)
behavior. ........
2007-07-30 17:29 +0000 [r77780] Russell Bryant <russell@digium.com>
* main/channel.c: (closes issue #10301) Reported by: fnordian
Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
(license 110) Additional changes by me Fix some problems in
channel_find_locked() which can cause an infinite loop. The
reference to the previous channel is set to NULL in some cases.
These changes ensure that the reference to the previous channel
gets restored before needing it again. I'm not convinced that the
code that is setting it to NULL is really the right thing to do.
However, I am making these changes to fix the obvious problem and
just leaving an XXX comment that it needs a better explanation
that what is there now.
2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp@digium.com>
* res/res_features.c: (closes issue #10327) Reported by: kkiely
Instead of directly mucking with the extension/context/priority
of the channel we are transferring when it has a PBX simply call
ast_async_goto on it. This will ensure that the channel gets
handled properly and sent to the right place.
* main/channel.c: (closes issue #10301) Reported by: fnordian
Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
(license 110) Restore previous behavior where if we failed to
lock the channel we wanted we would return to exactly the same
point as if we had just reentered the function.
* /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
lines (closes issue #10334) Reported by: ramonpeek Pass through
the return value from macro_exec through the MacroIf application.
........
2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c: Missing newline
2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp@digium.com>
* cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
Patches: cdr_pgsql.patch uploaded by prashant (license 114)
Finish the Postgresql connection after the log messages are
printed so we don't access invalid memory.
* channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
julianjm (license 99) Clear ONHOLD flag when decrementing the
onHold peer count. If we did not do this the count may keep
decreasing.
2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: "re-invite" was misspelled
2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp@digium.com>
* main/channel.c: (closes issue #10302) Reported by: litnialex If a
DTMF end frame comes from a channel without a begin and it is
going to a technology that only accepts end frames (aka INFO)
then use the minimum DTMF duration if one is not in the frame
already.
2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming@digium.com>
* doc/mp3.txt: change protocol for downloads as well
* doc/mp3.txt, sounds/Makefile: use new canonical name for download
server
2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell@digium.com>
* Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
in trunk, but not in 1.4. When Asterisk is compiled under dev
mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
define it in the same way that trunk does. Also, revert the
change that added this define in the Makefile The advantage to
doing it this way is that buildopts.h gets installed when you
install Asterisk. Then, when building any out of tree modules, or
building asterisk-addons, these modules know which options the
rest of Asterisk was built with.
2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson@digium.com>
* Makefile, main/logger.c: Fixes to get ast_backtrace working
properly. The AST_DEVMODE macro was never defined so the majority
of ast_backtrace never attempted compilation. The makefile now
defines AST_DEVMODE if configure was run with --enable-dev-mode.
Also, changes were made to acccomodate 64 bit systems in
ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
their roles in allowing me to get this committed
2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/logger.c: Missed one
* main/logger.c: Oops, that builtin define should be all-lowercase.
2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson@digium.com>
* cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
meaning free gets called on the same variable twice. This patch
sets the connection to NULL after calls to PQfinish so that the
problem does not occur. Also in this patch, prashant_jois
informed me that it is safe to pass a null pointer to PQfinish,
so I have removed the check for conn's existence from
my_unload_module. (closes issue 10295, reported by junky, patched
by me with input from prashant_jois)
2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf@digium.com>
* apps/app_meetme.c: This fix solves problem with intense squelch
noise when someone joins conf in bug 9430; We repro'd the problem
with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
I'm applying it. It looks like playing the recorded username will
louse up the next thing played into the channel. Josh rearranged
the code so as to start things over before playing data directly
into the conference.
2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
digit to terminate a DTMF string with. If none is specified then
no terminator will be used.
2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson@digium.com>
* main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
not a signed int, so use %u instead of %d in the format string
2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker@digium.com>
* /: so are my fingers...
* /: autotagexternals script is still obviously misbehaving...
* /: use autotagged externals
2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp@digium.com>
* configure, acinclude.m4: Fix autoconf logic for finding OpenH323
when it is not in the first place searched (/usr/share/openh323).
2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo@icir.org>
* main/rtp.c: set the sequence number in a frame for all frame
types
2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf@digium.com>
* channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
line this fixes bug 10293, where the error message because
defaultzone or loadzone was not defined was confusing ........
2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
Jul 2007) | 2 lines Oops, res contains the error code, not errno.
I was wondering why a mutex was reporting "No such file or
directory"... ........
* main/app.c: Found another place where we should be using the
umask (thanks jcmoore)
2007-07-24 Jason Parker <jparker@digium.com>
* Asterisk 1.4.9 released.
2007-07-24 16:42 +0000 [r76803-76805] Jason Parker <jparker@digium.com>
* channels/chan_iax2.c: Don't create the Asterisk channel until we
are starting the PBX on it. (ASA-2007-018)
2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Added a membercount variable to call_queue
struct which keeps track of the number of logged in members in a
particular queue. This makes it so that the 'n' option for
Queue() can act properly depending on which strategy is used. If
the strategy is roundrobin, rrmemory, or ringall, we want to ring
each phone once before moving on in the dialplan. However, if any
other strategy is used, we will only ring one phone since it
cannot be guaranteed that a different phone will ring on
subsequent attempts to ring a phone. As a side effect of this,
the QUEUE_MEMBER_COUNT dialplan function now just reads the
membercount variable instead of traversing through the member
list to figure out how many members there are. Special thanks to
blitzrage for helping to test this out. (closes issue #10127,
reported by bcnit, patched by me, tested by blitzrage)
2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: It was our stated intention for 1.4 that
files created in app_voicemail should depend upon the umask.
Unfortunately, mkstemp() creates files with mode 0600, regardless
of the umask. This corrects that deficiency.
2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix some incorrect softkey labels in
messages. Don't try to play dialtone in some unimplemented
features.
2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp@digium.com>
* /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
force channel format changes when a generator is present. The
generator may have changed the formats itself and changing them
back would cause issues. ........
2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Don't try to queue up hold/unhold frames
on a non-existent channel. Issue 10276.
2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp@digium.com>
* apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
by putting the value of the digit char in an int.
* /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
lines (closes issue #10236) Reported by: homesick Patches:
rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
Remote Party ID on guest calls. ........
* channels/chan_skinny.c: (closes issue #10268) Reported by:
mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
(license 7) Add another OS that has to use the Macros for byte
ordering.
2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Use a signed integer for storing the number
of bytes in the packet read from the network. Using an unsigned
value here made it impossible to handle an error returned from
recvfrom(). Furthermore, in the case that recvfrom() did return
an error, this would cause a crash due to a heap overflow.
(closes issue #10265, reported by and fix suggested by
timrobbins)
2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
4 lines Backport a fix for a memory leak that was fixed in trunk
in reivision 76221 by rizzo. The memory used for the localaddr
list was not freed during a configuration reload. ........
2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf@digium.com>
* sounds/Makefile: This patch from 10249 is worth applying! It
prevents downloading sound files if they are already downloaded.
Darn Practical, if you ask me
2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Allow getting a call from an existing
"sub" channel. Cancel ringing if endpoint hangs up before
answering. Fixes were backported from trunk (there was apparently
a bit of confusion during merge of a previous patch). (closes
issue #10241)
* main/manager.c: Eliminate a compiler warning with gcc 4.2 by
constifying a char *
* channels/chan_skinny.c: It's possible for sub->owner to be NULL
here if you cancel the call immediately after/during sending a
digit.
2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson@digium.com>
* apps/app_directory.c: When using users.conf for the entries in
the directory, if multiple users had the same last name, only the
first user listed would be available in the directory. (closes
issue #10200, reported by mrskippy, patched by me)
2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell@digium.com>
* main/channel.c: Use the define that specifies the default length
of an artificially created DTMF digit in the ast_senddigit()
function. The define is set to 100ms by default, which is the
same thing that this function was using. But, using the define
lets changes take effect in this case, as well as the others
where it was already used.
2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
lines (closes issue #10247) Reported by: fkasumovic Patches:
chan_sip.patch uploaded by fkasumovic (license #101) Drop any
peer realm authentication entries when reloading so multiple
entries do not get added to the peer. ........
* res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
Patches: res_conver.patch uploaded by fkasumovic (license #101)
Use the last occurance of . to find the extension, not the first
occurance.
* apps/app_queue.c: Move makeannouncement variable declaration to
proper place.
2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Remove some duplicate code.
2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: The diff on this looks pretty big but all I did
was remove a pointless if statement (always evaluates true).
* apps/app_queue.c: Changes in handling return values of several
functions in app_queue. This all started as a fix for issue
#10008 but now includes all of the following changes: 1.
Simplifying the code to handle positive return values from ast
API calls. 2. Removing the background_file function. 3. The fix
for issue #10008 (closes issue #10008, reported and patched by
dimas)
2007-07-19 15:53 +0000 [r75928] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) |
6 lines When processing full frames, take sequence number
wraparound into account when deciding whether or not we need to
request retransmissions by sending a VNAK. This code could cause
VNAKs to be sent erroneously in some cases, and to not be sent in
other cases when it should have been. (closes issue #10237,
reported and patched by mihai) ........
2007-07-18 22:59 +0000 [r75807] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Need to make sure we set milliseconds and
timestamp - pointed out by the recent ast_ time stuff from
Tilghman
2007-07-18 21:09 +0000 [r75759] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) |
5 lines When traversing the queue of frames for possible
retransmission after receiving a VNAK, handle sequence number
wraparound so that all frames that should be retransmitted
actually do get retransmitted. (issue #10227, reported and
patched by mihai) ........
2007-07-18 20:40 +0000 [r75749] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007)
| 2 lines Store prior to copy (closes issue #10193) ........
2007-07-18 20:17 +0000 [r75732] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Umm, why are we transmitting dialtone on
cfwdall?
2007-07-18 20:00 +0000 [r75712] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c,
pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these
Asterisk won't build under devmode using GCC 4.2.
2007-07-18 19:54 +0000 [r75707-75711] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fixes for 7935/7936 conference phones.
Issue 9245, patch by slimey.
* channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue
9887, patches by DEA
2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard <dhubbard@digium.com>
* /, apps/app_queue.c: Merged revisions 75657 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007)
| 1 line removed the word 'pissed' from ast_log(...) function
call for BE-90 ........
2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Few more places that needs to check for
onhold state.
* channels/chan_sip.c: (closes issue #10165) Reported by: elandivar
It is possible for hold status to exist without call limits set,
so we need to ensure update_call_counter is executed regardless.
* channels/chan_h323.c: Don't bother reloading chan_h323 if it did
not load successfully in the first place. This would otherwise
cause a crash.
* pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record
the threadid of each running thread before shutting them down as
the thread themselves may change the value.
2007-07-18 12:29 +0000 [r75529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_meetme.c: Using a freed frame causes crashes (closes
issue #9317)
2007-07-17 Russell Bryant <russell@digium.com>
* Asterisk 1.4.8 released.
2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant <russell@digium.com>
* /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17
Jul 2007) | 3 lines Properly check for the length in the skinny
packet to prevent an invalid memcpy. (ASA-2007-016) ........
* main/rtp.c: cast arguments to ast_log so that it builds without
warnings for me
* channels/iax2-parser.c, channels/iax2-parser.h, /,
channels/chan_iax2.c: Merged revisions 75444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) |
5 lines Ensure that when encoding the contents of an ast_frame
into an iax_frame, that the size of the destination buffer is
known in the iax_frame so that code won't write past the end of
the allocated buffer when sending outgoing frames. (ASA-2007-014)
........
* /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) |
4 lines After parsing information elements in IAX frames, set the
data length to zero, so that code later on does not think it has
data to copy. (ASA-2007-015) ........
2007-07-17 20:40 +0000 [r75439] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Ensure that the pointer to STUN data does not go to
unaccessible memory. (ASA-2007-017)
2007-07-17 20:33 +0000 [r75437] Russell Bryant <russell@digium.com>
* res/res_agi.c: (issue #10210) Reported by: juggie Patches:
10210-1.4-grr.patch uploaded by juggie (license #24) Tested by:
juggie, blitzrage Log a warning if someone uses DeadAGI on a live
channel.
2007-07-17 20:03 +0000 [r75405] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Fixing an error I made earlier. ast_fileexists
can return -1 on failure, so I need to be sure that we only enter
the if statement if it is successful. Related to my fix to issue
#10186
2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant <russell@digium.com>
* main/pbx.c: (closes issue #10209) Reported by: juggie Patches:
10209-trunk-2.patch uploaded by juggie Tested by: juggie,
blitzrage In ast_pbx_run(), mark a channel as hung up after an
application returned -1, or when it runs out of extensions to
execute. This is so that code can detect that this channel has
been hung up for things like making sure DeadAGI is used on
actual dead channels, and is beneficial for other things, like
making sure someone doesn't try to start spying on a channel that
is about to go away.
* res/res_agi.c: Remove a duplicated newline character in AGI debug
output. (closes issue #10207, patch by seanbright)
2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming <kpfleming@digium.com>
* main/dns.c, /: Merged revisions 75304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007)
| 3 lines provide proper copyright/license attribution for this
structure that was copied from a BSD-licensed header file long,
long ago... ........
* /: another fix that is not needed here (finishing up 75251)
2007-07-16 18:16 +0000 [r75253] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c: Restoring functionality from 1.2 wherein
Retrydial will not exit if there is no announce file specified.
This change makes it so that if there is no announce file
specified, the application will continue until finished (or
caller hangs up). If a bogus announce file is specified, then a
warning message will be printed saying that the file could not be
found, but execution will still continue. (closes issue #10186,
reported by jon, patched by me)
2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming <kpfleming@digium.com>
* /: block change that is not relevant here
2007-07-13 20:36 +0000 [r75108] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13
Jul 2007) | 3 lines Fix a couple potential minor memory leaks.
load_moh_classes() could return without destroying the loaded
configuration. ........
2007-07-13 20:15 +0000 [r75078] Mark Michelson <mmichelson@digium.com>
* apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul
2007) | 5 lines Fixed an issue where chanspy flags were
uninitialized if no options were passed. What triggered this
investigation was an IRC chat where some people's quiet flags
were set while others' weren't even though none of them had
specified the q option. ........
2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13
Jul 2007) | 6 lines Ensure that adding a user to the list of
users of a specific music on hold class is not done at the same
time as any of the other operations on this list to prevent list
corruption. Using the global moh_data lock for this is not ideal,
but it is what is used to protect these lists everywhere else in
the module, and I am only changing what is necessary to fix the
bug. ........
* channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) |
12 lines (closes issue #9660) Reported by: mmacvicar Patches
submitted by: bbryant, russell Tested by: mmacvicar, marco,
arcivanov, jmhunter, explidous When using a TDM400P (and probably
other analog cards) there was a chance that you could hang up and
pick the phone back up where it has been long enough to be not
considered a flash hook, but too soon such that the device
reports that it is busy and the person on the phone will only
hear silence. This patch makes chan_zap more tolerant of this and
gives the device a couple of seconds to succeed so the person on
the phone happily gets their dialtone. ........
2007-07-12 23:00 +0000 [r74998] Mark Michelson <mmichelson@digium.com>
* channels/chan_agent.c: Change to my previous fix regarding agent
logoff soft. Now uses deferlogoff instead of loginstart since
loginstart is used after logoff. Thanks to makoto for pointing
this out and suggesting the fix. (closes issue #10178, reported
and patched by makoto, with modification by me)
2007-07-12 20:42 +0000 [r74955] Steve Murphy <murf@digium.com>
* channels/chan_sip.c: This patch resolves 10143; thanks to irroot
for the patch; looked acceptable. Let the community decide if it
messes things up
2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp <jcolp@digium.com>
* main/channel.c: Whoops... didn't want this to be returned to 0
each iteration.
* main/channel.c: When waiting for a digit ensure that a begin
frame was received with it, not just an end frame. (issue #10084
reported by rushowr)
2007-07-12 16:53 +0000 [r74839-74866] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: It helps if I actually add this stuff for
the 7921 too - otherwise it won't actually do much of anything.
* channels/chan_skinny.c: Add device ID for 7921 wireless skinny
phone
* channels/chan_skinny.c: Fix dialing in skinny that was broken in
some cases. Issue 10136, fix provided by DEA.
2007-07-12 15:53 +0000 [r74815] Joshua Colp <jcolp@digium.com>
* /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul
2007) | 2 lines Only print out a warning for situations where it
is actually helpful. (issue #10187 reported by denke) ........
2007-07-11 22:57 +0000 [r74767] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) |
5 lines The function make_trunk() can fail and return -1 instead
of a valid new call number. Fix the uses of this function to
handle this instead of treating it as the new call number. This
would cause a deadlock and memory corruption. (possible cause of
issue #9614 and others, patch by me) ........
2007-07-11 21:14 +0000 [r74722] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_agent.c: Merged revisions 74719 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11
Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft"
did not work...at all. Now it does. (closes issue #10178,
reported and patched by makoto, with slight modification for 1.4
and trunk by me) ........
2007-07-11 18:34 +0000 [r74657] Russell Bryant <russell@digium.com>
* res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11
Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows
the condition that uses LIKE. This fixes realtime extensions with
ODBC. (closes issue #10175, reported by stuarth, patch by me)
........
2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy <murf@digium.com>
* Makefile: This fixes 10172, where the entire man8 dir gets
removed during an uninstall of asterisk
* utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt:
further reversion of previously applied floating point stuff for
expr2
2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp <jcolp@digium.com>
* channels/chan_phone.c, configure,
include/asterisk/autoconfig.h.in, configure.ac: Instead of
figuring out kernel versions that have compiler.h and not...
let's just use autoconf to check for it's presence. (issue #10174
reported by francesco_r)
* channels/chan_phone.c: Only check if we need to do a SIGMA based
tone generation if we have a card. (issue #10179 reported by
mikowhy)
2007-07-10 23:32 +0000 [r74476] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Forwarding a message with IMAP storage was
storing the message in the sender's box instead of the forwarded
mailbox. (closes issue #10138, reported and patched by jaroth)
2007-07-10 19:58 +0000 [r74374-74428] Jason Parker <jparker@digium.com>
* /, apps/app_queue.c: Merged revisions 74427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6
lines Fix an issue where it was possible to have a service level
of over 100% Between the time recalc_holdtime and update_queue
was called, it was possible that the call could have been hungup.
Move both additions to the same place, so this won't happen.
Issue 10158, initial patch by makoto, modified by me. ........
* main/dns.c: Don't use #if to check if something is defined - use
#ifdef instead. Pointed out by kpfleming
* /, channels/chan_agent.c: Merged revisions 74376 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul
2007) | 4 lines Fix an issue with wrapuptime not working when
using AgentLogin. Issue 10169, patch by makoto, with a minor mod
by me to not re-break issue 9618 ........
* main/dns.c, /, configure, include/asterisk/autoconfig.h.in,
configure.ac: Merged revisions 74373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5
lines Use res_ndestroy on systems that have it. Otherwise, use
res_nclose. This prevents a memleak on NetBSD - and possibly
others. Issue 10133, patch by me, reported and tested by scw
........
2007-07-10 Russell Bryant <russell@digium.com>
* Asterisk 1.4.7.1 released.
2007-07-10 16:00 +0000 [r74323] Russell Bryant <russell@digium.com>
* res/res_musiconhold.c: fix an uninitialized variable
2007-07-10 15:38 +0000 [r74317] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4
lines Fix a small typo in description in of Voicemail()
application. Issue 10170, patch by casper. ........
2007-07-10 15:31 +0000 [r74314] Russell Bryant <russell@digium.com>
* res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10
Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue
#10075, this part reported by jmls on IRC, patch by me) ........
2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp <jcolp@digium.com>
* /, main/app.c: Merged revisions 74264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2
lines Ensure the group information category exists before trying
to do a string comparison with it. (issue #10171 reported by
mlegas) ........
* channels/chan_sip.c: Only spit out an inringing warning message
when it is applicable. Since call limits are already toast in
realtime let's not scare the user if they are using it. (issue
#10166 reported by bcnit)
2007-07-09 Russell Bryant <russell@digium.com>
* Asterisk 1.4.7 released.
2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant <russell@digium.com>
* configure, configure.ac: Update the configure script to check for
a required function that is not present in the 1.2 version of
libpri. This will prevent the configure script from thinking that
it has compatible libpri support for Asterisk 1.4, when it
actually does not because the installed version is from 1.2.
* res/res_musiconhold.c: (closes issue #10123) Reported by:
blitzrage Patches submitted by: juggie, qwell, me Tested by:
blitzrage When trying to find a music on hold class to use, try
all of the options, instead of only the first one that is set.
Also, change the MusicOnHold applications to not hang up on the
channel when a class can not be found.
2007-07-09 20:19 +0000 [r74159] Jason Parker <jparker@digium.com>
* channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8
lines Several chan_zap options were not working on reload because
they were arbitrarily disallowed when reloading some/most PRI
options (such as signalling) was disallowed. Options such as
polarityonanswerdelay and answeronpolarityswitch can safely be
changed on a reload. This corrects that behavior. Issue 9186,
patch by tzafrir. ........
2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Forgot to get rid of an extraneous debug
message.
* apps/app_queue.c: The n option for Queue should make the queue
exit immediately after failure to reach any members and should
not be dependent on the timeout value passed to Queue (closes
issue #10127, reported by bcnit, repaired by me)
2007-07-09 15:32 +0000 [r74082] Joshua Colp <jcolp@digium.com>
* channels/chan_skinny.c: Only destroy the scheduler context if it
was allocated. (issue #10124 reported by gzero)
2007-07-09 14:57 +0000 [r74047] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fixed a logic error in leave_voicemail.
Pass the mailbox instead of the context to inbox_count when the
context is "default." (closes issue #10135, reported by yannj,
repaired by me)
2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp <jcolp@digium.com>
* channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread
synchronization tweaks. (issue #10124 reported by gzero)
* configure, acinclude.m4: Use AC_CHECK_HEADER to check for
ptlib/openh323 to allow for cross compiling. (issue #9675
reported by zandbelt)
2007-07-09 04:03 +0000 [r73985] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while
'make progdocs'. (Closes issue #10104)
2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp <jcolp@digium.com>
* main/cdr.c: Give Agent channel names priority when doing CDR
merging. (issue #10011 reported by krtorio)
* pbx/pbx_config.c: Add a few sanity checks when writing out the
dialplan. (issue #10157 reported by dome)
2007-07-08 09:47 +0000 [r73849] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: While tracking down a bug, I need some more
history. Dumphistory is very useful, indeed.
2007-07-06 23:02 +0000 [r73769] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) |
4 lines If a sip_pvt struct has already registered an extension
state callback, remove the old one before adding a new one. If
this isn't done, Asterisk will crash. (issue #10120) ........
2007-07-06 16:36 +0000 [r73727] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fixing a rare case which causes voicemail
to crash when compiled with IMAP storage. inboxcount has the
possibility of finding an "interactive" vm_state when no
persistent "non-interactive" vm_state exists for that mailbox. If
this should happen when someone attempts to leave a message, it
results in a crash. This patch, along with my commit in revision
72670 fix issue 10053, reported by jaroth. closes issue #10053
2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant <russell@digium.com>
* res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06
Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras
Patches submitted by: Corydon76 Tested by: apsaras Fix a problem
with MSSQL 2005 by explicitly stating that '\' is being used as
an escape character. ........
* /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) |
7 lines (closes issue #10125) Reported by: makoto Patches
submitted by: makoto This fixes a crash in chan_sip that happens
when the bindaddr setting is not valid on Asterisk startup, gets
fixed, and then a reload gets issued. ........
2007-07-06 15:27 +0000 [r73675] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_agent.c: Merged revisions 73674 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06
Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy.
(issue 9618, reported by jiddings, patched by moi) closes issue
#9618 ........
2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant <russell@digium.com>
* BUGS: fix a little spelling error
* channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop
the monitor thread if it was never started. (closes issue #10124,
reported by gzero, fixed by me)
* channels/chan_iax2.c: copy from the correct buffer when deferring
a full frame (related to issue #9937)
* channels/chan_iax2.c: * Store the call number that a thread is
processing without the full frame bit set to ease debugging *
When deferring a full frame for processing, stick it into the
queue for the thread that is processing frames for that call, not
the one that read the current frame and is about to go back into
the idle list (related to issue #9937)
2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007)
| 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just
like we don't support it for G.729 ........
2007-07-05 20:50 +0000 [r73512] Russell Bryant <russell@digium.com>
* res/res_features.c: Pass HOLD and UNHOLD frames to the other
channel when they are returned from a native bridge function.
This fixes a problem where when two zap channels are natively
bridged and one does a flash hook, the other channel did not
receive music on hold. (Reported to me directly by Doug Bailey at
Digium)
2007-07-05 19:18 +0000 [r73467] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2
lines Copy language information to the dialog structure when
calling a peer for situations where a PBX may be started on the
dialed channel. (issue #10121 reported by clegall_proformatique)
........
2007-07-05 15:59 +0000 [r73400] Mark Michelson <mmichelson@digium.com>
* apps/app_queue.c: Correcting a minor CLI bug I found. When
issuing the queue show command, if you type queue show and then
press tab, you can continue pressing tab and it will keep
auto-completing queue names even though only 1 queue can be used
as an argument.
2007-07-05 15:28 +0000 [r73398] Russell Bryant <russell@digium.com>
* channels/chan_vpb.cc, channels/Makefile: Make this module build
for me in dev-mode
2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp <jcolp@digium.com>
* apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2
lines Tweak spy locking. (issue #9951 reported by welles)
........
* channels/chan_local.c, /: Merged revisions 73318 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul
2007) | 2 lines Actually check to make sure a PBX was started on
one of the Local channels instead of blindly assuming it was.
(issue #10112 reported by makoto) ........
* /, apps/app_queue.c: Merged revisions 73315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2
lines Reset ServicelevelPerf variable back to 0 if we are unable
to calculate it each time... otherwise we will get previous
values. (issue #10117 reported by noriyuki) ........
2007-07-04 14:53 +0000 [r73208-73253] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04
Jul 2007) | 1 line bchannel configurations like echocancel and
volume control, need to be setuped on inbound calls too. ........
* channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04
Jul 2007) | 1 line bad bug in overlapdial case, we called
start_pbx multiple times, because the state wasn't changed..
........
2007-07-03 20:17 +0000 [r73143] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c, main/Makefile,
main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing
expr floating patch from 1.4; too much of a behavior change. If
you want this fix, try trunk instead. bug 9508.
2007-07-03 15:42 +0000 [r73104-73106] Jason Parker <jparker@digium.com>
* /: What the heck. This should not have happened.
* /: use autotagged externals
2007-07-03 12:38 +0000 [r73053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_dial.c, /: Merged revisions 73052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007)
| 2 lines RetryDial should accept a 0 argument, but it does not,
because atoi does not distinguish between 0 and error (closes
issue #10106) ........
2007-07-03 08:17 +0000 [r73005] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03
Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only
be called from mISDN Source channels.. #9449 ........
2007-07-02 20:16 +0000 [r72933] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput,
main/Makefile, main/ast_expr2.h, main/ast_expr2.y,
main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support
for floating point numbers added to ast_expr2 $\[...\] exprs.
Fixes bug 9508, where the expr code fails with fp numbers. The
MATH function returns fp numbers by default, so this fix is
considered necessary.
2007-07-02 18:18 +0000 [r72926] Russell Bryant <russell@digium.com>
* main/manager.c: Remove a bogus comment and add proper locking to
the handler function for the CLI command to show information on
manager actions.
2007-07-02 14:32 +0000 [r72888] Joshua Colp <jcolp@digium.com>
* main/channel.c: Added additional DTMF debug messages for when
emulation occurs.
2007-07-02 08:41 +0000 [r72850-72852] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
revisions 72585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) |
1 line check if the bchannel stack id is already used, if so
don't use it a second time. Also added a release_chan lock, so
that the same chan_list object cannot be freed twice. chan_misdn
does not crash anymore on heavy load with these changes. ........
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
Merged revisions 72099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) |
1 line simplified generation for dummy bchannels, also we mark
them as dummies, so they are not used later as real-bchannels,
optimized the RESTART mechanisms, we block a channel now on
cause:44, and send out a RESTART automatically, then on reception
of RESTART_ACKNOWLEDGE we unblock the channel again. ........
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged
revisions 72087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) |
1 line simplified channel finding and locking a lot. removed
unnecessary #ifdefed areas. ........
2007-07-01 23:52 +0000 [r72806] Russell Bryant <russell@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) |
5 lines When appending lines to call files to keep track of
retries, write a leading newline just in case the original call
file did not have a newline at the end. This fix is in response
to a problem I saw reported on the asterisk-users mailing list.
........
2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant <russell@digium.com>
* configure, configure.ac: Tweak the configure script so that error
output isn't spewed to the console when searching for GTK2 libs,
and they aren't found.
* formats/format_pcm.c: give format_pcm a more concise destription
2007-06-29 19:07 +0000 [r72665] Luigi Rizzo <rizzo@icir.org>
* main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for
absence of the function. This was already done in trunk.
2007-06-29 Russell Bryant <russell@digium.com>
* Asterisk 1.4.6 released.
2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp <jcolp@digium.com>
* main/cdr.c: Minor change for older GCC versions.
* Makefile, configure, configure.ac, makeopts.in: Backport fix for
GCC versions without support for declaration-after-statement.
2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/manager.c: Issue 10055 - Change memory allocation to use the
heap for a command, since the output has the potential to
overflow the stack (as it did here)
* res/res_jabber.c: Fix 1.4 breakage
2007-06-28 19:44 +0000 [r72493] Russell Bryant <russell@digium.com>
* configure, include/asterisk/autoconfig.h.in: regenerate the
configure script for rizzo
2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo <rizzo@icir.org>
* configure.ac: add a check for gethostbyname_r so we can simplify
the handling e.g. in utils.c Also add comments on a couple of
features which are not working on FreeBSD. All the above has been
already done in trunk so the merge must be blocked. Can someone
please regenerate ./configure ?
* Makefile, channels/chan_zap.c, main/say.c: Add
-Wdeclaration-after-statement to AST_DEVMODE flags to catch
variable declarations in the middle of a block. Fix the few
instances of the above spotted out by the compiler. All of this
has been already done or is not applicable in trunk, so the merge
of this change will be blocked.
* apps/app_meetme.c: cast a time_t so that it does not conflict
with the print format. This change was already done on trunk so
this change needs to be blocked from merging.
2007-06-27 23:29 +0000 [r72383] Brett Bryant <bbryant@digium.com>
* main/asterisk.c, /: Merged revisions 72373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) |
3 lines Reinstating patch. This actually fixes the problem,
however I was running a development branch without it and
mistakenly thought it wasn't fixed. Fixes issue #10010, and
#9654: 100% CPU usage caused by an asterisk console losing it's
controlling terminal. ........
2007-06-27 23:25 +0000 [r72381] Joshua Colp <jcolp@digium.com>
* apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun
2007) | 2 lines Update documentation to clarify variable usage
with MixMonitor. (issue #9494 reported by netoguy) ........
2007-06-27 23:03 +0000 [r72335] Brett Bryant <bbryant@digium.com>
* main/asterisk.c, /: Merged revisions 72333 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) |
2 lines Reverted changes for earlier revisions 72259 to 72261.
Issue #9654, #10010 ........
2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp <jcolp@digium.com>
* channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to
our list. Since these are dynamic payloads the other side
shouldn't care. (issue #9426 reported by irroot)
* /, apps/app_queue.c: Merged revisions 72327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2
lines Fix issue where queue log events might be missing. (issue
#7765 reported by mtryfoss) ........
2007-06-27 21:08 +0000 [r72272] Russell Bryant <russell@digium.com>
* /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) |
5 lines Fix a minor issue with parsing the priority number. You
could have as much whitespace as you want around a numeric
priority, but you couldn't have any whitespace around a special
priority like "n" or "hint". (issue #10039, reported by mitheloc,
fixed by me) ........
2007-06-27 20:46 +0000 [r72260] Brett Bryant <bbryant@digium.com>
* main/asterisk.c, /: Merged revisions 72259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) |
4 lines Fixes 100% load when controlling terminal disappears.
Issue #9654, #10010 ........
2007-06-27 20:25 +0000 [r72257] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 72256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2
lines I may possibly get shot for doing this... but... defer CDR
processing until after the channel has been dealt with. This
should eliminate all of the issues with channels going funky
(SIP/PRI) when you are posting CDRs to a database that is either
slow or unavailable and do not want to enable batching. ........
2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: use the proper type for storing group number
bits so that if someone specifies 'group=42' it will actually
work instead of being silently ignored
2007-06-27 18:40 +0000 [r72182-72185] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Fix another problem in voicemail with
missing symbols. Issue 10074, patch by kryptolus, extended to
include #if 0'd blocks (just in case)
2007-06-27 17:31 +0000 [r72148] Joshua Colp <jcolp@digium.com>
* main/channel.c: Make the ast_read_noaudio API call behave better
under circumstances where DTMF emulation was happening and a
generator was setup. (issue #10065 reported by stevefeinstein)
2007-06-27 17:10 +0000 [r72125] Jason Parker <jparker@digium.com>
* channels/chan_gtalk.c: Don't modify a variable that we don't want
modified. Make a copy of it instead. Issue 10029, patch by
phsultan with slight modifications by me (to remove needless
casts).
2007-06-27 16:34 +0000 [r72112] Russell Bryant <russell@digium.com>
* main/rtp.c: Only output debug information related to RTCP
timestamps when RTCP debug is turned on (issue #10066, patch by
me)
2007-06-27 07:58 +0000 [r72042] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) |
1 line for inbound TE calls, we setup the bchannel when we get
the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready.
removed some #if 0 areas which weren't used anymore. ........
r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) |
1 line isdn_lib.c didn't compile ........
2007-06-27 00:58 +0000 [r72006] Joshua Colp <jcolp@digium.com>
* pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work.
2007-06-26 23:02 +0000 [r71953] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Removing a pointless line. This variable
was already set earlier and between then and this line, there is
no way that the values on the right side of the assignment could
have changed.
2007-06-26 20:36 +0000 [r71915] Jason Parker <jparker@digium.com>
* main/rtp.c: Don't dereference a pointer that may be NULL here.
Issue 10017.
2007-06-26 19:00 +0000 [r71877] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: A few changes, the ultimate goal of which
is to keep better track of the number of messages that a mailbox
currently has. A description of the changes: 1. Changed the
"updated" field of the vm_state struct to act more as a binary
semaphore than a counting semaphore, since its current
implementation made the inboxcount function not work properly.
This change falls in line with a change made by UPenn with their
IMAP setup and helps to sync our changes with theirs. 2.
Eliminated some redundant calls to get_vm_state_by_mailbox inside
leave_voicemail 3. Use the play_folder variable to keep track of
the number of old and new messages in a mailbox as the messages
are deleted 4. Added an increment to the number of new messages
that was not there previously in the leave_voicemail function
2007-06-26 15:47 +0000 [r71796] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fixing bug where the authuser was
mistakenly pulled from the mailbox string instead of the IMAP
user. (closes issue 10054, reported and patched by jaroth)
2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007)
| 2 lines Issue 10062 - Trying to move a message without
selecting one first results in memory corruption ........
* /, res/res_agi.c: Merged revisions 71656 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007)
| 2 lines Issue 10035 - handle_exec returns a result inconsistent
with all of the other AGI commands ........
2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp <jcolp@digium.com>
* channels/chan_h323.c: Build a peer as well when hash323 is
enabled in users.conf (issue #9599 reported by asagage)
* channels/chan_agent.c: Minor tweak for queueing up the unhold
frame... this will teach me to do bugs while half asleep. (issue
#10046 reported by dimas)
2007-06-25 12:40 +0000 [r71519] Russell Bryant <russell@digium.com>
* doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue
#10048, Matti)
2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2
lines Ignore other URIs after the first in a 300 Multiple Choice
response. (issue #10041 reported by homesick) ........
* main/cdr.c: Fix it so 1.4 actually compiles on my box.
* channels/chan_agent.c: Check to make sure the channel pointer is
present before queueing up an unhold frame on it. (issue #10046
reported by dimas)
2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant <russell@digium.com>
* build_tools/prep_tarball: Include the menuselect-tree file in
tarballs to make builds from tarballs a little bit faster
* main/asterisk.c, /: Merged revisions 71358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) |
2 lines Revert the patch from issue 9654 due to an unexpected
side effect ........
2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_features.c: Issue 10044 - chan->cdr is NULL here, so
peer->cdr is what we really wanted to use
* main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24
Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to
be able to set variables to the empty string. ........
2007-06-23 03:29 +0000 [r71230] Steve Murphy <murf@digium.com>
* main/cdr.c, res/res_features.c: This patch is meant to fix 8433;
where clid and src are lost via bridging.
2007-06-22 22:44 +0000 [r71214] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20
Jun 2007) | 1 line fixed a bug that was introduced by copy and
paste in the last commit ..bchannels weren't cleaned properly.
........
2007-06-22 15:38 +0000 [r71096-71123] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
revisions 70672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) |
1 line we activate the bchannels in TE mode on incoming calls
only when we want to connect the call. ........
* channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20
Jun 2007) | 1 line forgot one place .. ........
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20
Jun 2007) | 1 line on receiption of cause:44 we mark the channel
as in use and inform the user about the situation, we need to
test the RESTART stuff then. Also shuffled the
empty_chan_in_stack function after the bchannel cleaning
functions, to avoid race conditions. ........
* channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19
Jun 2007) | 1 line when we send out a SETUP, but get no response,
we should cleanup everything after reception of a hangup.
........
* /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) |
1 line restart indicator 0x80 is correct, at least that's what
libpri does. ........
* channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12
Jun 2007) | 1 line if the bridged partner is mISDN too we should
not send dtmf tones, they are transmitted inband always ........
* channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12
Jun 2007) | 1 line if we have already some digits, we just stop
the tones. ........
2007-06-22 15:00 +0000 [r71068] Jason Parker <jparker@digium.com>
* apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged
revisions 71065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4
lines Fix a few silly usages of ast_playstream() - it only ever
returns 0... Issue 10035 ........
2007-06-22 14:53 +0000 [r71066] Brett Bryant <bbryant@digium.com>
* main/asterisk.c, /: Merged revisions 71064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) |
10 lines Fixed infinite loop when controlling terminal was lost
and return value of input function wasn't checked for errors.
This would cause 100% cpu to be taken up. (closes issue #9654,
issue #10010) Reported by: mnicholson, and eserra Idea for the
patch from mnicholson, patched by me ........
2007-06-22 14:10 +0000 [r71063] Steve Murphy <murf@digium.com>
* main/cdr.c: My conditions for merging amaflags info was naive;
DOCUMENTATION is the default, although null is possible; theft of
user-settable fields is not good. Just copy them, leave them
alone.
2007-06-22 03:14 +0000 [r71003] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a small typo which ... well ...
completely broke chan_iax2. oops! (issue #9937, patch by me)
2007-06-21 22:34 +0000 [r70949] Steve Murphy <murf@digium.com>
* main/cdr.c, /: Merged revisions 70948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1
line This little fix is in response to bug 10016, but may not
cure it. The code is wrong, clearly. In a situation where you set
the CDR's amaflags, and then ForkCDR, and then set the new CDR's
amaflags to some other value, you will see that all CDRs have had
their amaflags changed. This is not good. So I fixed it. ........
2007-06-21 21:40 +0000 [r70899] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2
lines Don't explode if the gain option is specified without a
value. (issue #9274 reported by mfarver) ........
2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Put the thread reading from the socket back
in the idle list if it deferred the processing of a full frame to
another thread
* channels/chan_iax2.c: If a full frame is received while one of
the iax2 threads is in the middle of handling a full frame for
the same call, queue it up for processing by that same thread
later instead of dropping it. (issue #9937, patch by me)
2007-06-21 20:19 +0000 [r70841] Steve Murphy <murf@digium.com>
* cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1
line it was pointed out that the cdr_custom config load could get
a lock, and under certain circumstances, would never release it.
I also noted that the situation where more than one mapping spec
was warned about, but did not ignore further mappings as it had
promised. I think I have fixed both situations. ........
2007-06-21 19:49 +0000 [r70808] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: When volgain is used don't leave a
temporary file behind. (Closes Issue 8514, Reported and patched
by ulogic, code reviewed by Jason Parker)
2007-06-21 15:22 +0000 [r70727] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Do not Packet2Packet bridge if packetization settings
do not allow it. (issue #9117 reported by phsultan)
2007-06-21 15:21 +0000 [r70726] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Remove a couple of duplicate unlocks
2007-06-21 13:58 +0000 [r70677] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Fix building with ODBC storage enabled.
(issue #10025 reported by denisgalvao)
2007-06-21 13:00 +0000 [r70656] Steve Murphy <murf@digium.com>
* main/cdr.c: Via complaints aired in asterisk-users, I submit
these changes, which allow cdr updates to see macro
context/exten, whether hung up or not
2007-06-20 23:32 +0000 [r70554-70612] Jason Parker <jparker@digium.com>
* cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql.
Issue 10020, patch by my, with credit to prashant_jois for
pointing out the problem.
* cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race
condition fix
* cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur
when reloading the module. Issue 10022, patch by me, with credit
to prashant_jois for finding the bug.
2007-06-20 22:22 +0000 [r70552] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2
lines Don't overwrite the configured username setting upon a
REGISTER. (issue #8565 reported by jsmith) ........
2007-06-20 20:53 +0000 [r70494] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Make sure we clear the previously dialed
number if it did not exist. Issue 9958.
2007-06-20 19:29 +0000 [r70445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_dial.c, /: Merged revisions 70444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007)
| 2 lines Issue 9997 - Timelimit times out the wrong channel
........
2007-06-20 18:46 +0000 [r70397] Russell Bryant <russell@digium.com>
* channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) |
5 lines Fix a problem where an established call would not be
properly disconnected when a PRI disconnect is received depending
on which cause code was received. (issue #9588, original patch by
softins, updated patch from jtexter3, and some additional
feedback from mhardeman) ........
2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp <jcolp@digium.com>
* main/rtp.c, main/frame.c: Put the speex packetization values back
in but disable it when setting up the smoother.
* main/frame.c: Don't do packetization/smoother stuff with speex,
it doesn't work.
2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant <russell@digium.com>
* contrib/scripts/ast_grab_core: don't delete the backtrace in
ast_grab_core
* channels/chan_gtalk.c: Only attempt to queue a hangup on the
owner channel if it actually exists. (issue #9795, patch from
zandbelt)
2007-06-19 18:23 +0000 [r70062] Steve Murphy <murf@digium.com>
* main/channel.c, /: Merged revisions 70053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1
line This fixes 9246, where channel variables are not available
in the 'h' exten, on a 'ZOMBIE' channel. The fix is to
consolidate the channel variables during a masquerade, and then
copy the merged variables back onto the clone, so the zombie has
the same vars that the 'original' has. ........
2007-06-19 17:07 +0000 [r70003] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 69992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2
lines Handle the CC field in the RTP header. (issue #9384
reported by DoodleHu) ........
2007-06-19 16:24 +0000 [r69987] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 69986 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2
lines Update BRIDGEPEER variable if set to the new channel name
when a masquerade happens. (issue #9699 reported by dimas)
........
2007-06-19 15:22 +0000 [r69944] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Fix a crash that could occur when handing
device state changes. When the state of a device changes, the
device state thread tells the extension state handling code that
it changed. Then, the extension state code calls the callback in
chan_sip so that it can update subscriptions to that extension. A
pointer to a sip_pvt structure is passed to this function as the
call which needs a NOTIFY sent. However, there was no locking
done to ensure that the pvt struct didn't disappear during this
process. (issue #9946, reported by tdonahue, patch by me, patch
updated to trunk to use the sip_pvt lock wrappers by eliel)
2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2
lines Perform an extra hangup check just in case. (issue #9589
reported by bcnit) ........
* /, res/res_features.c: Merged revisions 69846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2
lines Add parked call extension AFTER the parking slot has been
announced, otherwise two threads will try to handle the same
channel and it will go kaboom. (issue #9191 reported by japple)
........
* main/callerid.c: Fix for building on PowerPC under Linux.
2007-06-18 19:48 +0000 [r69796] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* channels/chan_sip.c: Issue 10005 - Segfault with missing
arguments, plus fix a missing define for SIP INFO channels
2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't count RTP timeout when involved in a
T38 fax session. (issue #9222 reported by ivoc)
* /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2
lines Set the peer name on the dialog to the one configured in
sip.conf and NOT the username to be used for authentication
attempts. (issue #9967 reported by achauvin) ........
2007-06-18 17:46 +0000 [r69744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* contrib/scripts/safe_asterisk, /: Merged revisions 69743 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007)
| 2 lines Issue 9998 - Remove SIG prefix, since it's not
supported by ksh ........
2007-06-18 16:51 +0000 [r69708] Joshua Colp <jcolp@digium.com>
* main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called
for the first time so that it does not needlessly spit out
changed messages when the host really didn't change.
2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant <russell@digium.com>
* res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c,
build_tools/menuselect-deps.in, configure, funcs/func_odbc.c,
include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c:
To prevent 92138749238754 more reports of "I have unixodbc
installed, but still can't build *_odbc.so!", check for ltdl
directly, instead of just listing it as another library to
include in the unixodbc check in the configure script. This also
makes ltdl show up as a dependency in menuselect so people know
what to go install. (related to issue #9989, patch by me)
* build_tools/prep_moduledeps: Change the use of "echo -e" to
"printf". On systems where /bin/sh is not bash, most of the lines
in menuselect-tree were getting a "-e" at the beginning of every
line. I'm surprised nobody noticed this, but I think the XML
parser was being very nice and ignoring them.
2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't defer the BYE till later on a transfer
when the transfer itself goes kaboom and has no hope of working.
* channels/chan_sip.c: Few minor transfer tweaks. We can't unlock
something we never locked, and better handle a specific scenario
with doing an attended transfer between two non-bridged calls.
2007-06-18 15:46 +0000 [r69660] Russell Bryant <russell@digium.com>
* Makefile: Tweak paths for BSD systems (issue #10001, stuarth)
2007-06-18 13:55 +0000 [r69625] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix issue where it would be possible for the
negotiated codecs to get set back to nothing. (issue #9992
reported by yehavi)
2007-06-15 Russell Bryant <russell@digium.com>
* Asterisk 1.4.5 released.
2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell@digium.com>
* res/res_features.c: Fix a silly deadlock in res_features that I
found while debugging on one of blitzrage's test machines. It was
one of the situations where he was seeing hung channels, and may
be the cause of some of the reports from other people. (related
to issue #9235)
2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Add support for setting the maximum
length of acceptable DTMF in SpeechBackground.
2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: The SLATRUNK_STATUS variable indicated
"SUCCESS" for both an answer of the incoming call on the trunk,
or if the trunk reached its ring timeout. This patch changes the
variable to say "RINGTIMEOUT" in that case. (issue #9973,
reported by n00dle, patch by me)
2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker@digium.com>
* main/config.c, /: Merged revisions 69469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4
lines Fix an issue where the line number in an unterminated
comment block error message would show the wrong line number.
"Reported" to me on #asterisk (somebody posted an error message,
and I happened to catch it) ........
* sounds/Makefile: Update to latest versions of sound files.
2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming@digium.com>
* cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c,
cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c,
main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c,
apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c,
main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c,
channels/chan_iax2.c: use ast_localtime() in every place
localtime_r() was being used
2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell@digium.com>
* main/say.c: Fix some problems with saying dates and times for the
"tw" langauge (issue #9964, ljmid)
2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker@digium.com>
* funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun
2007) | 4 lines Change a quite broken while loop to a for loop,
so "continue;" works as expected instead of eating 99% CPU...
Issue 9966, patch by me. ........
2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Whoops...
* channels/chan_iax2.c: Let's make chan_iax2 media only native
transfers actually work. (issue #9376 reported by simone
cittadini)
* channels/iax2-parser.c: Add TXMEDIA to list so that it is
properly displayed during iax2 packet output.
2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Move the logic for destroying a call when no
response is received to a BYE outside of the block that checks
for FLAG_FATAL to be set. This flag is only set when the packet
is transmitted with the reliability set to XMIT_CRITICAL when the
original packet is transmitted. A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered.
This resulted in seeing some SIP channels that would never go
away with the last packet sent being a BYE. (part of issue #9235,
patch from jcmoore)
2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Contains a patch for fixing an encoding
problem when using Outlook to view voicemail emails and
attachments. This fix has also been tested on Thunderbird,
Evolution, Pine, and Mutt. (Issue 9336, reported by marwick,
patched by mutterc)
2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Really ignore NULL frames and check whether
the channel hungup or not. (issue #9912 reported by junky)
* /, main/app.c: Merged revisions 69127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2
lines Return group counting to previous behavior where you could
only have one group per category. (issue #9711 reported by
irroot) ........
2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Clarify a bit of logic. This doesn't change
behavior in any way, but it is helpful when following the logic
to debug problems like 9235.
* channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct
was accessed without the lock held. This issue was reported to me
via email by Dmitry Mishchenko. Thanks!
* cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois
in #asterisk-bugs. PQclear() was not called on the result
structure after doing a PQexec(). Also, fix up some formatting in
passing.
2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Change the full frame dropping log message
to debug to avoid future bug reports.
* channels/chan_iax2.c: Schedule the sending of a PING packet a
second later than previously so that it does not collide with the
LAGRQ.
2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell@digium.com>
* main/channel.c: In ast_channel_make_compatible(), just return if
the channels' read and write formats already match up. There are
code paths that call this function on a pair of channels multiple
times. This made calls fail that were using g729 in some cases.
The reason is that codec_g729a will unregister itself from the
list of available translators will all licenses are in use. So,
the first time the function got called, the right translation
path was allocated. However, the second time it got called, the
code would not find a translation path to/from g729 and make the
call fail, even if the channel actually already had a g729
translation path allocated. (SPD-32)
2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 68921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2
lines Bring RTP back to Asterisk at the end of a native bridge no
matter what. ........
2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker@digium.com>
* include/asterisk/time.h: Solaris 10 sometimes (?) needs this
include in order to have NULL defined.
2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly
used
2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
Merged revisions 68732 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) |
1 line added check for NULL Pointer when calling misdn_new.
Asterisk does not allow us to create channels anymore when stop
gracefully is used :). also modified the restart_indicator to 0
........
2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp@digium.com>
* main/channel.c, /: Merged revisions 68682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2
lines Improve deadlock handling of the channel list. (issue #8376
reported by one47) ........
2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /, channels/misdn/ie.c,
channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) |
1 line fixed problem that the dummybc chanels had no lock,
checking for the lock now. Also fixed the channel restart stuff,
we can now specify and restart particular channels too. ........
2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* pbx/pbx_config.c: "dialplan save" produced garbage in the config
file
2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell@digium.com>
* /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) |
4 lines Don't automatically hang up after running Dictate so that
callers can exit cleanly using '#' (closes issue #9577, patch
from Thomas Andrews) ........
2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: actually remember the type/subclass of full
frames that are in process
2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp@digium.com>
* /, main/say.c: Merged revisions 68397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2
lines Don't call ast_waitstream_full when the control file
descriptor and audio file descriptor are not set, simply call
ast_waitstream! (issue #8530 reported by rickead2000) ........
* main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2
lines Do a DNS lookup immediately upon calling the dnsmgr
function, don't wait until a refresh happens. (issue #9097
reported by plack) ........
2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell@digium.com>
* /, main/say.c: Merged revisions 68351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) |
3 lines Fix a problem where saying a character wouldn't properly
break out when the caller pressed '#' (issue #8113, reported by
patbaker82, patch from jamesgolovich (hey, long time no see!) and
patbaker82) ........
2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Fix incorrect French syntax of "old
messages". Request for feedback was sent to asterisk-dev mailing
list, with little response. Issue 9118, patch by junky.
2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: some improvements to the IAX2 full frame
dropping logic recently added: - use inaddrcmp(), since we have
it - output the type of frame and subclass being dropped, and the
type/subclass that is already being processed (which caused the
drop)
2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell@digium.com>
* channels/chan_agent.c, apps/app_queue.c: Fix loading persistent
queue members when using realtime configuration for queues. Also,
remove an unneeded leading slash for the astdb family. (issue
#9911, patch by atis)
2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix an issue with newer phones which
require packets be padded out to the correct length. Issue 9887,
patch by DEA.
* apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4
lines Don't try to save voicemail greetings unless the user
presses '1' to accept/save. Issue 9904, patch by me. ........
2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a
check to make sure that greetings get stored properly. (Issue
8016, reported by edhorton, patched by alamantia with
modification by me. Thanks to Jason Parker for the advice on
this).
2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej@edvina.net>
* channels/chan_features.c: Disable chan_features by default in
menuselect
2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell@digium.com>
* main/strcompat.c: Include stdarg.h for build issues on Solaris
(issue #9381)
2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp@digium.com>
* main/channel.c: Fix logic when doing a name based channel search
for a structure when you want to start from a specific point in
the channel list. (issue #9324 reported by slavon)
* apps/app_dial.c, /: Merged revisions 68070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2
lines Allow the 'g' option to work if used with the 'S' option.
(issue #9888 reported by gasparz) ........
2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej@edvina.net>
* res/res_jabber.c: Adding a few Todo's to res_jabber so we don't
forget.
* res/res_jabber.c: Ok, we found out that this is not about if you
have any *active* clients using TLS, but if you have initialized
TLS at all during the lifetime of the module. So if you reload to
disable TLS, it won't help.
* res/res_jabber.c: If you have a jabber client that uses TLS,
refuse unload. Bad fix, but will prevent crashes while we are
trying to find a workaround. Iksemel development seems to have
stalled and we might have to stop using the TCP/TLS connections
in that library and use our own, which would scale better from a
poll/select perspective I guess. It would also make it easier to
migrate to OpenSSL and stop Asterisk from depending on both
OpenSSL and GnuTLS.
* include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make
sure we can unload res_jabber. Patch by phsultan - thanks! Due to
a bug in the iksemel library, this will not work if you are using
GTLS in the connection. That's being investigated. If you figure
out a way to handle that without us having to patch iksemel, let
us know in the bug report. Thanks.
2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2
lines Only notify the devicestate system of a peer state change
when the peer is built from the config file. (issue #9900
reported by arkadia) ........
* main/file.c: Properly handle cases where a stream can't be
written to. (issue #9757 reported by junky)
2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell@digium.com>
* res/res_snmp.c: Disable reload functionality in res_snmp. It is
not possible to initialize the snmp library more than once
without completely unloading the module and loading it again.
(issue #9571, reported by hristo, additional helpful debug
information from festr, patch from me)
* channels/chan_sip.c: Fix a crash when doing call pickups with SIP
phones. The code unlocked the channel when it should not have.
(issue #9652, reported by corruptor, fixed by me)
2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix for Issue 9810. There was a segfault
under a specific set of circumstances: 1. VoiceMailMain was
configured in the dialplan with an extension as its argument 2. A
message was left for this mailbox 3. Tried to call VoiceMailMain
but hung up before entering password. This was fixed by checking
that a pointer was non-null prior to trying to dereference it.
(Issue 9810, reported by xmarksthespot, patched by Corydon76 with
modifications by me).
2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell@digium.com>
* main/channel.c, /, include/asterisk/linkedlists.h: Merged
revisions 67715 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) |
5 lines We have some bug reports showing crashes due to a double
free of a channel. Add a sanity check to ast_channel_free() to
make sure we don't go on trying to free a channel that wasn't
found in the channel list. (issue #8850, and others...) ........
2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 67649 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2
lines Reinvite the RTP back to the Asterisk machine when the
timeout happens. (issue #9888 reported by gasparz) ........
* main/translate.c: Fix plc_samples warning when registering a
translator. (issue #9897 reported by xylome)
* apps/app_directed_pickup.c: Include macroexten while searching
for a channel to pick up in case they are in a macro. (issue
#9491 reported by jamesb63)
* res/res_agi.c: Make the new "agi debug off" CLI command work.
(issue #9890 reported by eliel)
* /, main/devicestate.c: Merged revisions 67593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2
lines Revert channel name splitting fix for Zap. The moral of the
story is don't use - in your user/peer names. (issue #9668
reported by stevedavies) ........
2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Fix some crashes related to the use of the
"meetme" CLI command. The code for this command was not locking
the conference list at all. (issue #9351, reported by and patch
submitted by Junk-Y, committed patch is different and by me)
2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf@digium.com>
* pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug
9883, wherein macros were not allowing the includes construct.
fixed and tested, looks OK. Now includes can serve as an adjunct
to catch.
2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell@digium.com>
* include/asterisk/linkedlists.h: This bug has been hanging over my
head ever since I wrote this SLA code. Every time I tried to go
debug it by adding some debug output, the behavior would change.
It turns out I wasn't crazy. I had the following piece of code:
if (remove) AST_LIST_REMOVE_CURRENT(...); Well,
AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my
conditional statement didn't do much good at all. It always ran
at least all of the macro minus the first statement, so I was
seeing list entries magically disappear when they weren't
supposed to. After many hours of debugging, I have come to this
extremely irritating fix. :) (issues #9581, #9497)
* channels/chan_zap.c: Suppress a bunch of debug output unless
option_debug is on
2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails
saved to IMAP storage using extensions other than gsm were unable
to be played over the phone. (Issue 9786, reporter:
xmarksthespot, Patched by xmarksthe spot with revisions by me,
reviewed by Russell Bryant).
2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Correctly update date/time on devices
throughout the life of the device, instead of just at
registration. Issue 9152, yet another patch by DEA.
2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: Added code to automatically add a default case to
switches that don't have one. In some cases, rather than fall
thru, it results in a goto with -1 result, which terminates the
extension; a sort of dialplan seqfault, sort of. This was
required to fix bug reported in 9881
2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell@digium.com>
* main/channel.c: Handle a failure in malloc() in
ast_safe_string_alloc()
* main/channel.c: Fix a problem that showed itself by causing Zap
channel names to be completely bogus on my machine.
ast_safe_string_alloc() was broken. It called vsnprintf() on a
va_args list twice without re-initializing it. After the first
usage, va_end() and va_start() must be called again.
2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter@beronet.com>
* /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) |
1 line briding is a bool, fixed copy and paste issue. ........
* channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05
Jun 2007) | 1 line simplified the EVENT_SETUP handling in the
cb_events function a lot. Commented the different possibilities a
bit and made functions of shared code. When the dialed extension
does not exist in the extensions.conf we'll jump into the 'i'
extension if this does exist, else we disconnect the call with
the cause:1 = No Route to Destination. ........
2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell@digium.com>
* main/asterisk.c, main/loader.c, include/asterisk/module.h: When
shutting down "gracefully", go through and run the unload()
callbacks for all of the modules. "stop now" is considered a
non-graceful shutdown and will not go through this process.
(issue #9804, reported by chrisost, patch by me)
2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Only muck with the thread structure if an
idle one was found/created.
2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: ensure that a burst of full frames
(AST_FRAME_DTMF being the prime example) will not be processed
out of order... this is a brute force fix, but seems to be the
safest fix for now (thanks to the Digium PQ department for
finding this bug)
2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter@beronet.com>
* channels/misdn_config.c, channels/chan_misdn.c, /,
channels/misdn/chan_misdn_config.h: Merged revisions 67209 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) |
1 line added possibility to deactivate bridging per port ........
2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, funcs/func_math.c: Merged revisions 67161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007)
| 2 lines According to MATH, 0+1181000386 = 1181000448. Oops.
........
2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt
structure can disappear and the code did not account for it and
crashes. (issues #9642, #9569, #9666, probably others ... based
on the work by stevedavies and mihai, with additional changes
from me)
2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix for skinny keepalives. If there is no
traffic from the phone for (keep_alive * 1100) ms (arbitrarily
adding 10% for network issues, etc), unregister the device. Issue
8394, patch by DEA.
* channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar
to the recent fix for chan_skinny) Issue 9855, patch by DEA.
2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Add comments for two functions that get
called with the appropriate call locked, but perform operations
that could result in the pvt structure getting destroyed before
returning again, causing numerous seg faults all over the module.
(inspired by issues #9642, #9569, and #9666, and the work done by
stevedavies and mihai)
2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf@digium.com>
* main/cdr.c: This typo has been here since 1.4 forked. It has been
the source of heartburn to many a dialplan/CDR programmer.
2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell@digium.com>
* main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC)
2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Better handle SIP devices that say they have
SDP content... but really don't. (issue #9398 reported by
mthomasslo)
* apps/app_dial.c: Initialize cidname variable to nothing since it
may be used without having been touched. (issue #9661 reported by
dimas)
* res/res_features.c: Returning a value that indicates the parking
of a call was a success when it really wasn't (because the
parking slot selected was in use) is the wrong thing to do.
(issue #9723 reported by mdu113)
2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* contrib/init.d/rc.debian.asterisk,
contrib/init.d/rc.mandrake.asterisk, /,
contrib/init.d/rc.redhat.asterisk,
contrib/init.d/rc.gentoo.asterisk,
contrib/init.d/rc.mandrake.zaptel,
contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007)
| 2 lines Add revision Id tags (by request of tzafrir) ........
2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell@digium.com>
* configure, configure.ac: Change the configure script to build a
test program against libcurl to make sure the results from
curl-config can be used to compile successfully. This is intended
to help prevent a situation where you are cross compiling, and
the configure script finds the curl library installed on the
host. (issue #9865, reported and patched by zandbelt)
2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_jabber.c: Issue 9739 - Malformed jid causes a crash
2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When
handling an implicit ACK to a frame that was marked as the final
transmission for a call, don't call iax2_destroy() for that call
while the global frame queue is still locked. There is a very
nice explanation of the deadlock in the report. (issue #9663,
thorough report and patch from stevedavies, additional positive
test reports from mihai and joff_oconnell)
* include/asterisk/stringfields.h: Fix some compiler warnings in
C++ modules. (issue #9866, reported by osk, patch by Corydon76)
2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_odbc.c: On some drivers, deallocating the statement
handle isn't enough. We also have to clear the cursor (nice,
Oracle)
2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Removing extraneous debugging lines from
revision 66897. Sorry :)
* apps/app_voicemail.c: Submitting a fix for voicemail with IMAP
storage. Attachments with format specified as gsm were duplicated
(i.e. two attachments) were left. Thank you very much to
xmarksthespot for submitting the patch that fixed this. (Issues
9787 and 8873, Reported by xmarksthespot and jerjer, patched by
xmarksthespot)
2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell@digium.com>
* channels/chan_skinny.c: Changes to the way DTMF is handled in the
core broke dialing in chan_skinny. This patch makes chan_skinny
usable again. I did not end up testing this, but there are
multiple positive test reports listed in the bug report. (issue
#9596, reported by pj, testing by pj and mvanbaak, and the fix
was written by DEA)
* apps/app_page.c: List app_meetme as a module that app_page
depends on.
2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* doc/asterisk.8: Issue 9850 - update preferred command line syntax
2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell@digium.com>
* res/res_speech.c, include/asterisk/app.h,
include/asterisk/speech.h: Change a couple of header files to not
use "new", which is a reserved keyword in C++. (issue #9830,
reported by osk)
2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, apps/app_macro.c: Merged revisions 66744 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007)
| 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime.
Issue 8329 will remain unfixed for pbx_realtime, but only because
we lack core API to do it. ........
2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2
lines It is now possible for this path of execution to have the
frame pointer be NULL, therefore we need to check for it before
trying to access it. (issue #9836 reported by barthpbx) ........
2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson@digium.com>
* apps/app_voicemail.c: Fixed seg-faults when recording greetings
in voicemail with IMAP enabled. (Issue No. 9735, reported by
xmarksthespot, patched by me)
2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Silly me for having out of date source! Oh
well... I'm still leaving my comment.
* channels/chan_sip.c: When calling some peer/host that may not
exist/reply back... don't keep the dialog in memory for all of
eternity.
* channels/chan_zap.c, channels/chan_features.c: Change how channel
names are generated a bit. (issue #9825 reported by eldadran)
2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007)
| 2 lines If the value of a variable passed to FIELDQTY is blank,
then FIELDQTY should return 0, not 1. ........
2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Properly handle 408 request timeout -
according to the RFC, the dialog dies if a request in a dialog
gets this response.
* channels/chan_sip.c: Don't issue hangup on hangup on hangup on
hangup (for jcmoore)
2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Handle cases where a frame may have no data. (issue
#9519 reported by dmb)
2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't reset hangupcause if we already have
one
* channels/chan_sip.c: Tracking down hanging channels, killing them
one by one. Issue #9235 and related
2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp@digium.com>
* doc/datastores.txt: Update datastores documentation. (issue #9801
reported by mnicholson)
2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2
lines Issue #9802 - Change inuse counter on CANCEL ........
2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp@digium.com>
* channels/chan_zap.c: Make the usedistinctiveringdetection option
work again. (issue #9823 reported by premeau)
2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker@digium.com>
* channels/chan_zap.c: I don't know what this was trying to do, but
it's clearly incorrect. Issues 9808 and 9809.
2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac: have to check for OSP toolkit _after_
checking for OpenSSL
2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, main/say.c: Merged revisions 66127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007)
| 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch
........
2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac, channels/chan_gtalk.c, makeopts.in,
res/res_jabber.c: handle the GNUTLS library properly in the
configure script and build system don't build in OSP support
unless we have found and are allowed to use SSL support
2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell@digium.com>
* main/channel.c: if the string field init fails, clean up the
stuff that was allocated already
2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp@digium.com>
* main/slinfactory.c: Fix slinfactory logic when dealing with
frames coming in that may already be in the signed linear format.
2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell@digium.com>
* main/channel.c: Check the result of ast_string_field_init() in
ast_channel_alloc()
* main/rtp.c: Make 1.4 build on my machine, too..
2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker@digium.com>
* configure: Rebuild configure script for previous ar fix.
* configure.ac: Following moving strip to AC_PATH_TOOL, we need to
do something similar for ar.
2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac:
Checking for the strip application needs to be done with
AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross
compilation environments.
* Makefile: Clear CFLAGS before running make for menuselect. (issue
#9784, reported by ovi, patch by me)
2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_gtalk.c: oops, use #ifdef instead of #if
* channels/chan_gtalk.c: don't reference GnuTLS headers and
functions unless the configure script found it
* main/rtp.c: don't use uninitialized variables
2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp@digium.com>
* main/manager.c: Add the ability to blacklist certain commands
from being executed using the Command AMI action. (issue #9240
reported by junky)
2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej@edvina.net>
* channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk
core dump since the GnuTLS interface did not support
multithreading correctly.
* channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN.
Patch by phsultan. Thanks!
2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker@digium.com>
* .cleancount: Update cleancount for that last commit - just for
good measure.
* include/asterisk/translate.h, codecs/codec_speex.c,
main/translate.c, codecs/codec_ilbc.c: Fix handling of
zero-length frames when a codec is capable of native PLC. Issue
9183, patch by Mihai.
2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard@digium.com>
* funcs/func_math.c: merged qwell's func_math patch for issue 9507
2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp@digium.com>
* main/rtp.c: I like it when the RTP stack compiles myself...
2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej@edvina.net>
* channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues
when calling from gtalk to SIP over nat.
2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell@digium.com>
* apps/app_festival.c: Ensure that frames are fully initialized.
This will probably fix getting weird timestamp log messages in
logs when using the Festival app. (issue #9781, patch by me)
* main/rtp.c: Fix the calculation of the RTT for RTCP. The previous
code would result in oscillating and incorrect data.
Additionally, the RTT would sometimes report negative values due
to incorrect calculations. (issue #9601, patch from davetroy)
2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej@edvina.net>
* channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for
jingle
2007-05-24 14:42 +0000 [r65839] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2
lines Allow RFC2833 to be negotiated when an INVITE comes in
without SDP and is not matched to a user or peer. (issue #9546
reported by mcrawford) ........
2007-05-24 14:38 +0000 [r65836] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan -
Fix "login" as component to jabber server. ...and, by accident,
fix a bug in chan_sip for stopping a loop on retransmits of BYE
requests.
2007-05-24 09:37 +0000 [r65768] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24
Mai 2007) | 1 line we should only activate the generator in
chan_misdn, when asterisk hask not yet taken the call
(WAITING4DIGS state). Alerting audio will be generated fomr
asterisk for example. ........
2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: start the delayed PBX when receive voice or
video full frames as well, and comment this delayed-PBX activity
* /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007)
| 2 lines ensure that variables are set on a newly created
channel before we start a PBX on it ........
* channels/chan_iax2.c: clear the 'delay PBX' flag when we are
ready to start the PBX
* channels/chan_iax2.c: don't start a PBX on a new incoming IAX2
channel until we have some sort of response to our ACCEPT (ACK or
anything else)
* /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007)
| 2 lines if we are going to set variables on a newly created
channel, it should be done *before* we start the PBX on it
........
2007-05-23 13:07 +0000 [r65589] Russell Bryant <russell@digium.com>
* channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) |
3 lines Revert revision 62417 as someone reported problems with
it to Mark. This was related to issue #9588. ........
2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/make_version: when building a version string for a
developer branch, include the base branch in the version string
2007-05-22 18:40 +0000 [r65501] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a
dependency for app_voicemail and chan_zap (Thanks to mnicholson
for pointing it out)
2007-05-22 15:04 +0000 [r65452] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Remove a double const.
2007-05-22 14:02 +0000 [r65408] BJ Weschke <bweschke@btwtech.com>
* apps/app_followme.c: Fix a problem with flag recognition.
2007-05-22 13:09 +0000 [r65394] Russell Bryant <russell@digium.com>
* /, apps/app_queue.c: Merged revisions 65389 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) |
4 lines Fix a memory leak that I just noticed in the device state
handling in app_queue. On most device state changes, it would
leak roughly 8 to 64 bytes (the length of the name of the
device). ........
2007-05-22 08:12 +0000 [r65342] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22
Mai 2007) | 1 line we stop the tones only when we're in the
pre-call phase, otherwise e.g. when in CONNECTED state we should
not stop tones when we receive an Information Message ........
2007-05-20 17:59 +0000 [r65250] Joshua Colp <jcolp@digium.com>
* res/res_agi.c: res_agi needs to export two symbols
(ast_agi_register and ast_agi_unregister) for usage by others.
(issue #9755 reported by mnicholson)
2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy <murf@digium.com>
* main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c
to main/cdr.c, and neither did I. This is the remainder of the
9717 patch, the fix for the run-away FAIL status for a call
* apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions
65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1
line This update will fix the situation that occurs as described
by 9717, where when several targets are specified for a dial, if
any one them reports FAIL, the whole call gets FAIL, even though
others were ringing OK. I rearranged the priorities, so that a
new disposition, NULL, is at the lowest level, and the
disposition get init'd to NULL. Then, next up is FAIL, and next
up is BUSY, then NOANSWER, then ANSWERED. All the related set
routines will only do so if the disposition value to be set to is
greater than what's already there. This gives the intended
effect. So, if all the targets are busy, you'd get BUSY for the
call disposition. If all get BUSY, but one, and that one rings is
not answered, you get NOANSWER. If by some freak of nature, the
NULL value doesn't get overridden, then the disp2str routine will
report NOANSWER as before. ........
2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2
lines Not getting an ACK to a 200 OK in the initial invite is
critical to the call. ........
* /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5
lines Issue 9235 - part of the problem, maybe not all. Please
retry with this patch (and no other patch) if you have problems
with hanging SIP channels. Thank you. A special Thank You to
WeBRainstorm that gave me access to his system. ........
* channels/chan_sip.c: - Adding support for putting calls OFF hold
with a re-invite with blank SDP. This was a bug found while doing
tests at SIPit in Antwerp. - In order to not duplicate code, I
restructured some of the code for putting calls on/off hold.
Thanks DEA for reminding me. This fix has been asleep in the
videocaps branch until now.
2007-05-18 12:40 +0000 [r65039] Christian Richter <christian.richter@beronet.com>
* /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
revisions 65007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) |
1 line fixed a warning regarding Keypad encoding. encode the IE
sending_complete at the right position. ........
2007-05-18 10:37 +0000 [r64974] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue 9487 - stop media flows at hangup of
call
2007-05-18 08:58 +0000 [r64904] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18
Mai 2007) | 1 line we *need* to send a PROCEEDING when
sending_complete is set, even if need_more_infos is requested.
........
2007-05-18 02:48 +0000 [r64868] Russell Bryant <russell@digium.com>
* apps/app_queue.c: Fix a small bug I noticed while working on
something else. app_queue did not unregister its device state
monitoring callback in unload_module(). So, this would make
Asterisk crash on the first device state change after you unload
the module.
2007-05-17 21:19 +0000 [r64820] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, include/asterisk/linkedlists.h: Merged revisions 64819 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007)
| 2 lines How is it that we never caught that this is returning
the opposite of our documentation, until now? ........
2007-05-17 16:53 +0000 [r64761] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4
lines If we have a negative current message, we shouldn't go back
even further... Issue 9727. ........
2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant <russell@digium.com>
* contrib/scripts/astxs (removed): Remove script that is no longer
functional since the build system was redone. (issue #9340,
reported by junky)
* apps/app_dial.c: Increase the size of a buffer to support longer
dial strings for channels. (issue #9291, reported and fix
suggested by meni)
2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Even more direct RTP setup fixes! Don't
allow a codec that isn't supported to creep into the SDP of
either side. (issue #9446 reported by marcelbarbulescu)
* apps/app_voicemail.c: Fix authuser support. (issue #9740 reported
by xmarksthespot)
2007-05-17 06:13 +0000 [r64686] Russell Bryant <russell@digium.com>
* README: Update the main README to reflect the new build process
for 1.4 and above. (issue #9725, patch by eliel)
2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson <oej@edvina.net>
* /: Blocking patch already in this code
* channels/chan_sip.c: Fix auth on BYE. (Different patch than for
1.2)
* channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE
* channels/chan_sip.c: Final part of issue #9483 - fixing
transfer() of sip calls in the dial plan (twilson)
* channels/chan_sip.c: Issue #9439 - properly handle username
parameters in SIP uri.
* /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2
lines Support SIP uri's starting with SIP: and sip: (reported by
Tony Mountfield on the mailing list. Thanks!) ........
* /, channels/chan_sip.c: Merged following patch with a lot of
changes for 1.4 ------ Merged revisions 64514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6
lines Issue #9726 - rlister - Better logging for ACL denials
While at it, also added better logging and handling of peers that
are not supposed to register. My patch, stole the issue report
from Russell. My apologies, Russell :-) ........
2007-05-16 08:44 +0000 [r64515] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16
Mai 2007) | 1 line in the case immediate=yes, we directly jump
into the dialplan, where people can use PlayTones to indicate a
Dialtone, so we don't need to to that by ourself. also we should
not do a dialtone_indicate for incoming calls on a TE port in
overlapdialmode. ........
2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant <russell@digium.com>
* res/res_features.c: Properly fix a problem that occurs when you
set PARKINGEXTEN to an exten where a call is already parked.
(issue #9723, patch by me)
* res/res_features.c: When someone requests a specific parking
space using the PARKINGEXTEN variable, ensure that no other
caller is already there. (issue #9723, reported by mdu113, patch
by me)
2007-05-14 19:26 +0000 [r64324] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit
more
2007-05-14 19:13 +0000 [r64306] Russell Bryant <russell@digium.com>
* channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by
just ignoring it. An unknown indication will trigger an error and
cause sounds to stop, which in this case, is ringing.
2007-05-14 18:52 +0000 [r64280] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Handle network errors, like host or network
unreachable, in a better way. This means that calls to hosts or
qualify (OPTION) messages will fail quicker if the TCP/IP stack
tells us that there is an issue. Since this is an unconnected UDP
socket, we will not get error messages directly in most cases,
but maybe on the second and third try. This is already
implemented in trunk.
2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp <jcolp@digium.com>
* codecs/codec_speex.c: Properly set datalen field when doing PLC
in codec_speex. (issue #9722 reported by mihai)
* /, main/devicestate.c: Merged revisions 64275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2
lines Only perform stripping of - strings from the channel name
for Zap channels. Anywhere else we might remove a legitimate part
of a device name. (issue #9668 reported by stevedavies) ........
* main/channel.c: Fix scenario where if a phone that simply called
Echo() put itself on hold it could never get off hold.
2007-05-14 13:58 +0000 [r64193] Steve Murphy <murf@digium.com>
* main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570,
worrisome CDR warnings have been removed, that are either not
helpful, or not relevant.
2007-05-14 10:39 +0000 [r64157] Olle Johansson <oej@edvina.net>
* main/channel.c: Add hangupcause when we lack codecs for
transcoding
2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: This concludes my final adventure with
bitmasks and the onhold flag. Would anyone care for some peanuts?
* channels/chan_sip.c: Tweak hold flags some more. They can be of
three states when active: active, inactive, one direction.
* channels/chan_sip.c: Ensure the onhold flag is set no matter what
when being put on hold.
2007-05-11 20:16 +0000 [r63982] Jason Parker <jparker@digium.com>
* main/manager.c: Hide manager password from "manager show user
foo". I realize that there are other ways to get this, but we
really don't need to just show it in plain text so easily. Issue
9273, patch by junky
2007-05-11 16:35 +0000 [r63905] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* contrib/scripts/safe_asterisk, Makefile, /: Merged revisions
63903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007)
| 2 lines Issue 9121 - fixups for safe_asterisk script ........
2007-05-11 16:05 +0000 [r63886] Russell Bryant <russell@digium.com>
* main/manager.c: When MD5 authentication is not possible because
there is no challenge present, either because the Challenge
action was never issued, or some other reason, give a proper
error message and return an error instead of claiming that the
user wasn't found. (reported by jsmith on IRC)
2007-05-11 15:43 +0000 [r63872] Joshua Colp <jcolp@digium.com>
* res/res_features.c: Make the PARKINGEXTEN feature of parking
actually work. (issue #9708 reported by mdu113)
2007-05-10 23:15 +0000 [r63830] Jason Parker <jparker@digium.com>
* /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4
lines Fix an issue with trying to kill a thread before it gets
created. Issue 9709, patch by nic_bellamy. ........
2007-05-10 22:23 +0000 [r63804] Russell Bryant <russell@digium.com>
* main/manager.c: Strip terminal escape sequences from CLI command
output that is going to be sent out over the manager interface.
(issue #9659, reported by pari, fixed by me)
2007-05-10 20:48 +0000 [r63750] Doug Bailey <dbailey@digium.com>
* main/callerid.c: Add test for negative offsets in cid data to
prevent infinite loops.
2007-05-10 20:46 +0000 [r63749] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4
lines Do not allocate SIP pvt's for PEERs we can not reach. This
was seen as a lot of dialogs being created then immediately
destroyed at reload/restart of the SIP channel. ........
2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp <jcolp@digium.com>
* main/channel.c: Use the DTMF frame on the channel when returning
a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.
* channels/chan_sip.c: Do not prematurely go on hold if sendonly
was not actually set.
2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson <creslin@digium.com>
* channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2
lines Make sure we only create a DSP if it's requested on
SUB_REAL ........
2007-05-09 16:55 +0000 [r63612] Russell Bryant <russell@digium.com>
* main/channel.c: Modify ast_senddigit_begin() to use the same
assumptions used elsewhere in the code in that if a channel does
not have a send_digit_begin() callback, it only cares about DTMF
END events. (pointed out by Michael Neuhauser on the asterisk-dev
list)
2007-05-09 16:54 +0000 [r63611] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2
lines Properly handle hints that point to multiple devices in
chan_sip. Why chan_sip is even doing this I have no idea but I
would rather not go into a rant. (issue #9536 reported by
rlister) ........
2007-05-09 16:43 +0000 [r63608] Russell Bryant <russell@digium.com>
* main/channel.c: Only call ast_senddigit_begin() in
ast_senddigit() if the channel has a send_digit_begin() callback.
Checking the END_DTMF_ONLY flag was the wrong thing to do,
because that flag indicates that a *bridged* channel only wants
DTMF END events coming from this channel.
2007-05-09 14:50 +0000 [r63566] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, apps/app_directory.c: Merged revisions 63565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007)
| 2 lines Replicate fix from 51158 (app_voicemail) to
app_directory (Issue 9224) ........
2007-05-09 13:24 +0000 [r63535] Russell Bryant <russell@digium.com>
* Makefile: I have seen multiple people post questions trying to
figure out what the message "The configure script must be
executed before running 'make'" means. So, add another like that
says to specifically run ./configure. If this isn't obvious
enough, then they should be using something like AsteriskNOW and
not installing from source.
2007-05-09 13:17 +0000 [r63534] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
channels/misdn/isdn_msg_parser.c: Merged revisions
62945,63402,63519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) |
1 line when we're in state WAITING4DIGS, we use the asterisk
tone-generator which prods us, so we can't just return -1 in
misdn_write in this case. Added a MISDN_KEYPAD channel variable,
and fixed the sending of keypad. this enables us to modify the
call forward parameters in the switch. ........ r63402 | crichter
| 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added
application misdn_check_l2l1 which tries to pull up the L1/L2 on
all ports that have the layers down in a group. It waits then for
a timeout. This helps for scenarios where multiple PMP BRIs are
grouped together, or where a provider has a faulty PTP
Implementation, that looses the L2 after a while. ........ r63519
| crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after
release_chan, this may cause segfaults. ........
2007-05-09 13:04 +0000 [r63532] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't retransmit 200 OK's on ignore status.
(Reported on asterisk-users)
2007-05-08 22:38 +0000 [r63478] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, apps/app_macro.c: Merged revisions 63477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007)
| 2 lines Issue 9602 - segfault in app_macro ........
2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant <russell@digium.com>
* res/res_features.c: I mixed up the use of the find_feature()
function, so I renamed it find_dynamic_feature, and changed the
code to use the correct lock when using it.
* res/res_features.c: Use a read/write lock when accessing the
built-in features.
* contrib/scripts/realtime_pgsql.sql (added),
contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to
contrib/scripts to be with the rest of the sql examples. (issue
#9676, suretec)
2007-05-08 06:22 +0000 [r63360] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007)
| 2 lines Issue 9527 - upon entering a folder, no message is
selected (curmsg == -1), so deleting causes memory corruption
(beyond bounds) ........
2007-05-07 22:28 +0000 [r63329] Russell Bryant <russell@digium.com>
* configs/res_pgsql.conf.sample (added),
configs/extconfig.conf.sample, contrib/realtime_pgsql.sql
(added): Add a sample configuration file and example tables for
use with res_config_pgsql. (issue #9676, suretec)
2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp <jcolp@digium.com>
* main/channel.c, include/asterisk/app.h, /, main/app.c: Merged
revisions 63285 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2
lines Properly handle what happens during a masquerade in
relation to group counting. (issue #9657 reported by ramonpeek)
........
* channels/chan_sip.c: Minor backport of revision 59083 in trunk.
Don't queue an unhold frame up if the call was never on hold to
begin with.
2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson <oej@edvina.net>
* main/config.c: Don't remove configuration from memory just
because one section failed.
* /: Guess svnmerge doesn't handle files that move around. Blocking
patch to ./config.c
2007-05-06 12:28 +0000 [r63152] Olle Johansson <oej@edvina.net>
* main/file.c: Stop the video stream when you stop playback of all
streams for a call
2007-05-04 20:03 +0000 [r63099] Jason Parker <jparker@digium.com>
* res/res_jabber.c: Fix a crash when checking version attribute in
an incoming XML caps element. Issue 9667, patch by phsultan.
2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni <paripurnachand@digium.com>
* configs/manager.conf.sample: explanation for httptimeout in
manager.conf
2007-05-03 16:44 +0000 [r62989] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2
lines When a peer is seeded or built tell the devicestate core to
update it's status. This is easier then having chan_sip load
before pbx_config. (issue #9658 reported by dlynes) ........
2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming <kpfleming@digium.com>
* main/loader.c: improve loader a bit, by avoiding trying to
initialize embedded modules twice and avoiding trying to load
modules from disk when they have been loaded already during the
'preload' pass (reported by blitzrage on IRC, patch by me)
2007-05-03 15:23 +0000 [r62942] Russell Bryant <russell@digium.com>
* main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell
Bryant, 2007, TM, Patent Pending). This set of changes came from
a debugging session I had with Dwayne Hubbard. When he called
into his home FXO, ran the Echo application, and pressed a digit,
the digit would be echoed back and would never end. This is
fixed, along with a couple other little improvements. * When
chan_zap is in the middle of playing a digit to a channel, it
feeds back null frames, not voice frames. So, I have modified
ast_read to check the timing on emulated DTMF when it receives
null frames, in addition to where it was doing this on voice
frames. * Make a tweak to setting the duration on emulated DTMF
digits. If there was no duration specified, it set it to be the
minimum, instead of the default. * Instead of timing the emulated
digits off of the number of samples in audio frames that pass
through, just use time values. Now there is no code in this
section that assumes 8kHz audio.
2007-05-03 14:41 +0000 [r62913] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19
(added), pbx/ael/ael-test/ael-test18/extensions.ael,
pbx/ael/ael-test/ael-test19/extensions.ael (added),
pbx/ael/ael-test/ael-test19 (added),
pbx/ael/ael-test/ref.ael-test20 (added),
pbx/ael/ael-test/ael-test20/extensions.ael (added),
pbx/ael/ael-test/ael-test20 (added): updated the ael regressions
to match what's in trunk
2007-05-03 14:36 +0000 [r62912] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h,
channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged
revisions 61357,61770,62885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) |
1 line some fixes for PMP Hold/Retrieve, it should work now, when
briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200
(Di, 24 Apr 2007) | 1 line added lock for sending messages to
avoid double sending. shuffled some empty_chans after the
cb_event calls, this avoids that a release_complete from a quite
different call releases a fresh created setup by accident.
........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03
Mai 2007) | 1 line fixed the problem that misdn_write did not
return -1 when called with 0 samples in a frame this resultet in
a deadlock in some circumstances, when the call ended because of
a busy extension. added encoding of keypad. ........
2007-05-03 13:54 +0000 [r62883] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test18 (added),
pbx/ael/ael-test/ref.ael-vtest13,
pbx/ael/ael-test/ael-test18/extensions.ael (added),
pbx/ael/ael-test/ael-test18 (added),
pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c,
pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7:
These mods fix bug 9623, where an '@' in the eswitch contents
causes a syntax error. I also updated the regressions.
2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming <kpfleming@digium.com>
* res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02
May 2007) | 2 lines doh... initializing the pointer variable will
work just a bit better ........
* res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02
May 2007) | 7 lines increase reliability and efficiency of static
Realtime config loading via ODBC: don't request fields we aren't
going to use don't request sorting on fields that are pointless
to sort on explicitly request the fields we want, because we
can't expect the database to always return them in the order they
were created (reported by blitzrage in person (!), patch by me)
........
* res/res_config_pgsql.c: improve static Realtime config loading
from PostgreSQL: don't request sorting on fields that are
pointless to sort on use ast_build_string() instead of snprintf()
don't request the list of fieldnames that resulted from the query
when we both knew what they were before we ran the query _AND_ we
aren't going to do anything with them anyway (patch by me,
inspired by blitzrage's bug report about res_config_odbc)
2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant <russell@digium.com>
* main/channel.c: Merge changes from team/russell/inband_dtmf ...
Fix some issues related to generating inband DTMF. There are two
changes here: 1) The list of DTMF tones in the senddigit_begin()
function explicitly specified 100ms of the tone followed by 100ms
of silence. This really broke things with the way that Asterisk
now wants complete control over when the digit begins and ends.
So, regardless of what Asterisk really wanted to do, this was
going to play out the tone at the length it wanted to. This
caused various problems like DTMF translation to inband to be
extremely unreliable. The list of tones has been changed so that
the correct DTMF tone is played indefinitely until Asterisk tells
it to stop. 2) ast_write() had to be modified to let a DTMF_END
frame get processed even when a generator is present. This is how
the tone will finally get stopped. (issues #8944, #9250, #9348,
maybe others. Thanks to mdu113 from #8944 for the testing and
feedback!)
* main/manager.c: Backport the change that only went in to trunk
that fixes the command manager action over http. (reported
internally by pari and bkruse)
2007-05-02 20:46 +0000 [r62738] Steve Murphy <murf@digium.com>
* main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May
2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being
in 'h' extension louses up the dst field ........
2007-05-02 17:43 +0000 [r62692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007)
| 4 lines Issue 9638 - if a text frame is sent with no
terminating NULL through a bridged IAX connection, the remote end
will receive garbage characters tacked onto the end. ........
2007-05-02 17:10 +0000 [r62689] Steve Murphy <murf@digium.com>
* configs/extensions.conf.sample, main/channel.c, main/pbx.c,
channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the
clid, src fields in channel_alloc call. b)in the channel_alloc
func, set the cid_num and name fields from the arglist[blush]. c)
don't update the channel app & app data fields if you are in the
'h' extension. d)the load_module func in cdr_radius needs to
return DECLINE, SUCCESS.
2007-05-02 06:15 +0000 [r62624] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Don't unlock a channel that we already know
does not exist (propably isue 8228)
2007-05-01 21:57 +0000 [r62548] Russell Bryant <russell@digium.com>
* /, res/res_features.c: Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) |
4 lines Remove an unnecessary check that makes it so if you hang
up after doing an attended transfer before the target extension
answers the channel, the transfer is not successful. (issue
#9338, patch by svanlund) ........
2007-05-01 21:34 +0000 [r62545] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user()
(found by rayjay, different fixes by me)
2007-05-01 16:26 +0000 [r62497] Russell Bryant <russell@digium.com>
* /, configs/indications.conf.sample: Merged revisions 62496 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) |
3 lines Add indications.conf information for the Philippines.
(issue #9525, reported and patched by loloski) ........
2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant <russell@digium.com>
* channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) |
4 lines This patch fixes an issue where depending on the cause
code, when the network sends a PRI disconnect, the call may not
be properly hung up. (issue #9588, reported and patched by
softins) ........
* include/asterisk/http.h, main/http.c: When serving dynamic
content, include a Cache-Control header to instruct the browsers
to not store the resulting content. (issue #9621, reported by
Pari, patch by me)
2007-04-30 14:52 +0000 [r62371] Jason Parker <jparker@digium.com>
* configs/iax.conf.sample: Remove unused (and potentially
confusing) jitterbuffer options from sample config.
2007-04-30 14:36 +0000 [r62369] Joshua Colp <jcolp@digium.com>
* main/asterisk.c, /: Merged revisions 62368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2
lines Update copyright notice. It's now the year 2007! ........
2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: Fix a bug that made the "language" setting
in zapata.conf not functional. (issue #9626, reported and fixed
by sergee)
* apps/app_meetme.c: Note that the "talker optimization" option
will be enabled by default in 1.6
2007-04-27 Russell Bryant <russell@digium.com>
* Asterisk 1.4.4 released.
2007-04-27 21:10 +0000 [r62218] Russell Bryant <russell@digium.com>
* channels/chan_agent.c: Fix a weird problem where when a caller
talking to someone sitting behind an agent channel sent a digit,
the digit would be played to the agent for forever. This is
because chan_agent always returned -1 from its send_digit_begin
and _end callbacks. This non-zero return value indicates to the
Asterisk core that it would like an inband DTMF generator put on
the channel. However, this is the wrong thing to do. It should
*always* return 0, instead. When the digit begin and end
functions are called on the proxied channel, the underlying
channel will indicate whether inband DTMF is needed or not, and
the generator will be put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed
by me)
2007-04-27 16:17 +0000 [r62174] Jason Parker <jparker@digium.com>
* /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3
lines This transcoder message needn't be a NOTICE. I've seen it
cause confusion more than a few times. ........
2007-04-27 16:14 +0000 [r62171] Russell Bryant <russell@digium.com>
* main/pbx.c: If no variables were passed into
pbx_substitute_variables_helper_full(), then don't even bother
creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they
were on a channel. Most importantly, this fixes a crash. (issue
#9613, reported by callguy, fixed by me)
2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4
lines Issue #7351 - SIP Cancel fails due to the wrong contact
uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka
- THANKS!!!! THis was a hard one to catch. ........
* channels/chan_zap.c, main/manager.c: Issue #9608 - fix some
annoying DEBUG messages not controlled by option_debug (DEA).
Thanks!
2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp <jcolp@digium.com>
* /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2
lines Revert previous fix for when the IAX2 channel goes funky
(that's the technical term). This is causing legit calls to be
prematurely hung up. (issue #9600 reported by justdave) ........
* main/channel.c: Missed an ast_app_group_discard during merge.
Thanks blitzrage!
* res/res_monitor.c: Don't always say that the channel is being
paused if it is actually being unpaused in the Manager ack
message. (reported by jsmith in #asterisk-bugs)
* main/config.c, /: Merged revisions 61958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2
lines Don't count failed include attempts against the
configuration include level. (issue #9593 reported by mostyn)
........
2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007)
| 2 lines handle a very bizarre race condition with channels
being redirected before a simple switch can be started on them
(issue #9286) ........
2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) |
2 lines If the callerid= option is specified, but empty, clear
any previous data. ........
* /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) |
2 lines Ensure that callerid settings are reset on a reload.
........
2007-04-25 19:21 +0000 [r61805] Joshua Colp <jcolp@digium.com>
* main/cli.c, main/channel.c, include/asterisk/app.h,
funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2
lines Merge rewritten group counting support. No more storing
data on the variable list of the channels. That was bad, mmmk?
(issue #7497 reported by sabbathbh) ........
2007-04-25 16:22 +0000 [r61799] Russell Bryant <russell@digium.com>
* channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) |
3 lines Fix a typo where cid_num got copied instead of cid_ani.
(issue #9587, reported and patched by xrg) ........
2007-04-24 Russell Bryant <russell@digium.com>
* Asterisk 1.4.3 released.
2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 61786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) |
4 lines Don't crash if a manager connection provides a username
that exists in manager.conf but does not have a password, and
also requests MD5 authentication. (ASA-2007-012) ........
* main/channel.c, include/asterisk/channel.h: Improve DTMF handling
in ast_read() even more in response to a discussion on the
asterisk-dev mailing list. I changed the enforced minimum length
of a digit from 100ms to 80ms. Furthermore, I made it now enforce
a gap of 45ms in between digits. These values are not
configurable in a configuration file right now, but they can be
easily changed near the top of main/channel.c.
2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard@digium.com>
* channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007)
| 1 line removed #if 0 block from chan_phone, chan_zap, and
chan_modem restart_monitor() ........
2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell@digium.com>
* main/dial.c: Add a few more state changes in
handle_frame_ownerless() so that the SLA code will get notified
of these changes even when an owner channel is not provided. This
isn't from a specific bug report, it's just something I noticed
while poking around.
2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2
lines Allow RFC2833 to be sent in the response SDP when an INVITE
comes in without SDP. (issue #9546 reported by mcrawford)
........
2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell@digium.com>
* main/pbx.c: Some dialplan functions, such as CUT(), expect to
operate on variables on a channel. So, this little hack lets them
work in places where a channel doesn't exist, such as within
DUNDi configuration. (issue #9465, reported and patched by
Corydon76, testing by blitzrage)
* main/channel.c: Ensure that digits passing through Asterisk have
a reasonable minimum length. It is currently 100 ms. If someone
thinks this should be different, feel free to speak up. (related
to issues #8944, #9250, and #9348)
2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker@digium.com>
* main/rtp.c: Avoid invalid seqno cycling detection. Per comment
from Dave Troy: This adds back in some simple typecasting I had
in an earlier version which I realize now may be breaking things.
Issue #9554.
* main/loader.c, /: Merged revisions 61704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4
lines Fix an issue that I noticed while looking over issue 9571.
The reload timestamp was getting set after reloading the built-in
stuff, and before the modules. ........
2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell@digium.com>
* main/rtp.c: Remove a stray debug message introduced by a recent
commit.
2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker@digium.com>
* /, apps/app_queue.c: Merged revisions 61692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5
lines If the '* to hangup' option is not enabled, we don't need
to disable * as a valid exit key. If it was enabled, this
statement would've never been checked in the first place. Issue
#9552 ........
2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell@digium.com>
* main/config.c, apps/app_voicemail.c, main/manager.c,
include/asterisk/config.h: Fix the UpdateConfig manager action to
properly treat "variables" and "objects" differently (a=b versus
a=>b). (issue #9568, reported by pari, patch by me)
2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3
lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by
Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........
2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/manager.c: Bug 9557 - simple reason why reading a function
always returned NULL
* funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c,
funcs/func_groupcount.c, /, funcs/func_timeout.c,
funcs/func_cdr.c: Merged revisions 61680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007)
| 5 lines Bug 9557 - Specifying the GetVar AMI action without a
Channel parameter can cause Asterisk to crash. The reason this
needs to be fixed in the functions instead of in AMI is because
Channel can legitimately be NULL, such as when retrieving global
variables. ........
2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: allow external build systems to extract the
required sound file versions
2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej@edvina.net>
* main/rtp.c: Clean upp formatting, add some doxygen stuff while
we're in cleaning mode... Thanks Kevin!
* main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy)
2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: #9483, half of patch by twilson to solve 302
redirect issues
* /: Blocking AstHoloPatch from 1.2
2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf@digium.com>
* main/cdr.c: This is a fix to the way CDR merge handles the data
that results from ForkCDR.
2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 61655 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2
lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves
the same as OUTBOUND_GROUP except it will get unset after use so
it won't get accidentally inherited. (issue #BE-140) ........
* apps/app_speech_utils.c: Do not bother looking for a result if
none are present.
* channels/chan_sip.c: For those very verbose SIP implementations
that attach tons of info to the Contact header... let's increase
our variable sizes. (issue #9535 reported by jeffg)
2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Eliminate a compiler warning with
ODBC_STORAGE enabled so that it will build under dev-mode.
2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf@digium.com>
* channels/chan_oss.c: A fix for chan_oss that resulted from the
CDR changes; it helps to use the right info.
2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't assume the callid of a dialog will be
set, as in some circumstances it may not. (issue #9534 reported
by tecnoxarxa)
2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) |
5 lines If someone sets the "useragent" option in sip.conf to be
empty, then don't add the User-Agent header at all. It is an
optional header, anyway. Also, the bug report says that some of
Japan's SIP providers don't allow it for some weird reason.
(issue #9488, reported by makoto, fixed by me) ........
2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns@beronet.com>
* channels/chan_misdn.c: Don't export AOCD variables on
misdn_hangup anymore, this was mainly a fix for trunk..
2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) |
6 lines Fix a bug with switching between host=dynamic and using
specific hosts for peers. The code would only reset the peer's
address when it is dynamic if it was a new peer structure. Now,
it will also reset the address if it was already in the peer
list, but before the reload, it was not dynamic. (issue #9515,
reported by caio1982, fixed by me) ........
* main/http.c: Add "svgz" to the mimetypes table. (issue #9510,
bkruse) In passing, constify the elements of the mimetypes table.
* /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) |
5 lines Remove the attempt at reporting configuration errors in
sip.conf. This can cause a bunch of improper messages when using
realtime. I give up. As oej tried to convince me when I put this
in, there is just no easy way to do it. (inspired by a message on
the -dev list) ........
2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns@beronet.com>
* channels/chan_misdn.c: Export AOCD variables on misdn_hangup.
* channels/chan_misdn.c: Ignore facility messages in case we don't
have a corresponding channel object.
* channels/chan_misdn.c: AOCD's are now exported to asterisk
channel variables.
2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell@digium.com>
* main/Makefile, main/http.c, main/minimime (removed): File upload
support was added to solve some needs for the Asterisk GUI.
However, after much discussion, it has been decided that adding
this to 1.4 is not in the best interests of the project. It has
been removed here, but will remain in trunk.
2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns@beronet.com>
* channels/misdn_config.c, /: Merged revisions 61170 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr
2007) | 2 lines msns config parameter defaults to '*' ........
2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf@digium.com>
* apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a
previous fix to overcome a compiler warning; the app NoCDR() has
been updated to mark the channel CDR as POST_DISABLED instead of
destroying the CDR; this way its flags are propagated thru a
bridge and the CDR is actually dropped. The cases where only one
channel in a bridge has a CDR was cleaned up.
2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3
lines - Don't send ActionID before Response: header. - Don't use
a blank in an AMI header ........
2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming@digium.com>
* main/minimime/mm_envelope.c, res/res_features.c: fix up some
warnings found using --enable-dev-mode
* main/minimime/Doxyfile (removed),
main/minimime/tests/messages/CVS (removed),
main/minimime/tests/CVS (removed): remove some more stuff we
don't need
2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell@digium.com>
* main/minimime/test (removed): Remove another directory that
should no longer be there
* main/minimime/Make.conf (removed), main/minimime/mytest_files
(removed), main/minimime/.cvsignore (removed), main/minimime/sys
(removed), main/minimime/mm-docs (removed): Remove various files
that I thought I already removed.
2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker@digium.com>
* apps/app_queue.c: Use the appropriate interface name with
COMPLETECALLER. Issue 9395.
2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf@digium.com>
* channels/chan_oss.c, main/channel.c, main/cdr.c,
channels/chan_phone.c, channels/chan_misdn.c,
channels/chan_skinny.c, channels/chan_features.c,
channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c,
channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c,
channels/chan_sip.c, res/res_features.c, channels/chan_agent.c,
include/asterisk/channel.h, channels/chan_gtalk.c,
channels/chan_iax2.c: This is a big improvement over the current
CDR fixes. It may still need refinement, but this won't have as
many folks bothered.
2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej@edvina.net>
* res/res_jabber.c: Add final new line after JabberEvent
2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker@digium.com>
* /, apps/app_directory.c: Merged revisions 60935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5
lines Allow matching on names shorter than 3 chars. This also
fixes the case where somebody wants to match on less then 3
chars. Issue 9071 ........
2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/asterisk.c, include/asterisk.h, /: Merged revisions 60849
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007)
| 2 lines Don't check for error when lowering priority (according
to the manpage, it should never happen anyway). It might could
happen, though, if another thread messed with the priority, so
safeguard against that (reported via -dev list). ........
* channels/chan_local.c, /: Merged revisions 60846 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08
Apr 2007) | 2 lines Bug 9505 - If the return value for
local_queue_frame is set, then p->lock is no longer valid.
........
2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 60797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2
lines When calling a device that then forwards us elsewhere... we
have to make our channels compatible if it is the only channel
being dialed. (issue #9445 reported by marcelbarbulescu) ........
* apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if
MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy)
2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, apps/app_macro.c: Merged revisions 60711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007)
| 2 lines Gosub called within a Macro resets the arguments
improperly and causes general weirdness. (Issue 8329) ........
* main/http.c: Fix --enable-dev-mode
* channels/chan_oss.c: Off by one error, resulting in a crash
(Issue 9500)
* /, main/file.c: Merged revisions 60660 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007)
| 2 lines Bug 9486 - memory leak when opening a filestream
........
2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell@digium.com>
* main/minimime/sys/mm_queue.h, main/minimime/Doxyfile,
main/minimime/mimeparser.yy.c, main/minimime/minimime.c,
main/manager.c, main/minimime/mm_mimepart.c,
main/minimime/test.sh, configure, include/asterisk/compat.h,
main/strcompat.c, main/minimime/mm_internal.h, main/http.c,
main/minimime/tests/parse.c, main/minimime/mm_base64.c,
main/minimime/mm_mimeutil.c, main/minimime/mm.h,
main/minimime/tests, main/minimime/mm_header.c,
main/minimime/mm_error.c, main/Makefile,
main/minimime/mm_codecs.c, main/minimime/mm_param.c,
configure.ac, main/minimime/Makefile, main/minimime/mm_init.c,
include/asterisk/manager.h, main/minimime/strlcpy.c,
configs/http.conf.sample, main/minimime/mm_parse.c,
main/minimime/tests/create.c, main/minimime/mm_contenttype.c,
main/minimime/mm_util.c, main/minimime/mm_envelope.c,
main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c,
main/minimime/tests/messages/test2.txt,
main/minimime/tests/messages/test3.txt,
main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c,
main/minimime/tests/messages/test4.txt,
main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h,
main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c,
main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt,
main/minimime/mimeparser.l, main/minimime/mm_context.c,
main/minimime/mimeparser.tab.h, main/minimime (added),
main/minimime/mm_warnings.c, main/minimime/mm_queue.h,
main/minimime/tests/messages, include/asterisk/autoconfig.h.in,
main/minimime/mimeparser.y, Makefile.moddir_rules,
main/minimime/sys, main/minimime/tests/Makefile: To be able to
achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface
with access to the Asterisk manager interface. One of the things
that was intended to be a part of this system, but was never
actually implemented, was the ability for the GUI to be able to
upload files to Asterisk. So, this commit adds this in the most
minimally invasive way that we could come up with. A lot of work
on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to
check permissions of active manager sessions was added by Dwayne
Hubbard. Then, hacking this all together and do doing the
modifications necessary to the HTTP interface was done by me.
2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard@digium.com>
* UPGRADE.txt: clarified a sentence in the format_wav section
* UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and
plan to remove GAIN code from trunk
2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: When a station picks up a trunk that was on
hold, make the hints reflect that nobody has the trunk on hold
anymore.
* apps/app_meetme.c: Fix a few problems with SLA. (issue #9459,
reported by francesco_r, fixed by me) * The original behavior was
that if one station put a call on hold, another one picked it up,
and then hung up, the code would still consider the call on hold
by the first station, so the trunk would not be hung up. However,
to better comply with what most people seem to expect it to
behave, it will now hang up the trunk. * Fix a problem with
"barge=no". This was only intended to prevent people from joining
calls that are in progress. However, it also prevented other
people from picking up a call that was on hold. This has been
fixed. * When there are no active stations on a trunk and it is
on hold, the code now indicates the HOLD and UNHOLD conditions to
the trunk channel. This allows music on hold to be played to the
trunk when it is on hold.
2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin@digium.com>
* channels/chan_zap.c: Make sure we check the faxdetect option
before doing fax processing
* channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2
lines There should only be one code path for doing DTMF
conditionals on channels. This fixes it. ........
2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming@digium.com>
* /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007)
| 2 lines remove undocumented 'cardsmode' parameter and stop
searching for transcoders during reload() ........
2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp@digium.com>
* res/res_speech.c, apps/app_speech_utils.c,
include/asterisk/speech.h: Add support for returning different
types of results (ie: NBest).
2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard@digium.com>
* formats/format_wav.c: modified default GAIN for issue 5823,
thanks jrwalliker
2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf@digium.com>
* configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added
some clarification to the example configs for CDRs, on how to
select a backend. Also, made cdr-csv the default if you 'make
samples', and no other changes.
2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5
lines Just because we can't find the voicemail configuration
file, doesn't mean that the module failed to load. The user could
be using realtime. Issue #9473 ........
2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell@digium.com>
* main/http.c: Add the MIME type for gif by request from Pari
2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2
lines Only unlock our pvt and net locks if we are actually going
to try to lock the owner again. (issue #9472 reported by zoa)
........
2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 60134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) |
6 lines It is valid to redirect channels via the manager
interface that are not in the UP state. Instead of checking for
that to prevent to ensure a dead channel doesn't get redirected,
just use the ast_check_hangup() API call. (issue #9457, reported
by Callmewind, patch by me) (related to issue #8977) ........
* channels/chan_sip.c: Add a Content-Length of 0 to the response
built by transmit_response_with_unsupported(). (issue #9454,
reported by makoto, fixed by me)
* /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) |
4 lines Fix the return value of handle_common_options() so that
it always properly indicates whether it handled the option or
not. (issue #9455, reported by Netview, fixed by me) ........
* apps/app_meetme.c: Fix a problem where if a trunk was hung up
while it was on hold, all of the hints would reflect the line
still on hold, even though it should reflect that it is back to
not in use. (issue #9459, reported by francesco_r, fixed by me)
2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Don't clash when a person both speaks
and uses DTMF.
2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell@digium.com>
* /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) |
4 lines Don't attempt to report configuration errors in
build_user(). oej pointed out that for a "friend" entry, this
won't work, because all user options are valid for peers, but not
the other way around. ........
* /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) |
3 lines Make chan_sip report when it encounters an unknown
option. (issue #9440, reported by nightcrawler) ........
* /, main/app.c: Merged revisions 59886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) |
5 lines When doing a built-in blind or attended transfer, restore
the ability to use '#' to terminate the number and immediately do
the transfer instead of having to dial the number and just wait
for the feature digit timeout. (issue #8366, xueliangliang)
........
* Makefile: Ensure that menuselect gets executed in dependency
check mode every time you run make.
2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns@beronet.com>
* channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h:
Merged revisions 59788,59803 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2
lines Use the new sysfs way of mISDN 1.2 to check if a port is NT
or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di,
03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........
2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h:
Merged revisions 59623-59624,59639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) |
1 line we can now make 30 channels on a PRI (before we forgot
chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200
(Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........
r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) |
1 line added option which allows us to accept incoming SETUP
Messages without automatically sending Proceeding or Setup
Acknowledge, this is useful with some broken switches and if you
want to Release incoming calls without previously having
acknowledged them. The new option is
noautorespond_on_setup=yes|no default is no, so we don't break
the existing behaviour ........
2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2
lines Increase the maximum size for a string of mailboxes to
1024. (issue #9270 reported by rtucker) ........
2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: continue in for-loop should go to the incrementer,
not the test. As per 9435, thanks to marcelbarbulescu
2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell@digium.com>
* main/netsock.c, /: Merged revisions 59608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) |
6 lines Add the SO_REUSEADDR flag to sockets handled by netsock.
This is needed by the patch that went in for issue 7874.
chan_iax2 needs to be able to create socket that is lisetning on
INADDR_ANY, but also be able to bind sockets to specific
addresses. (Thanks to Stevenson on the asterisk-dev mailing list
for explaining why this flag was needed.) ........
2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker@digium.com>
* configure, main/Makefile, acinclude.m4: Add linux-uclibc host
arch..."thingy". Sorry, I don't know what it's called...
2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf@digium.com>
* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
include/asterisk/cdr.h: several changes via kpflemings review
* main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c,
include/asterisk/cdr.h: These mods fix CDR issues from 8221,
8593, 8680, 8743, and perhaps others. Mainly with CDRs generated
from transfer situations.
* configs/extensions.conf.sample: A small clarification to keep
bugs from being filed, and confusion from rising, if
clearglobalvars is set, and globals are set in the AEL file.
(9419)
2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell@digium.com>
* res/res_jabber.c: When building a response to a subscription, the
"from" must be the full Jabber ID. This fixes some problems where
jabber users are not able to add their Asterisk account to their
user list, since they are unable to get Asterisk to approve their
subscription. (issue #8210, reported by caspy, and verified by
bradtem)
2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2
lines Keep a global array of variables indicating whether certain
conference rooms are in use. This ensures that two people going
into a new dynamic conference when the 'e' option is set don't go
into the same conference room. (issue #8835 reported by eliel)
........
2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell@digium.com>
* main/rtp.c, /: Merged revisions 59357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) |
5 lines If an error occurs when reading from an RTP socket, and
the error code does not indicate that we should try again, then
return NULL instead of a "null frame". This will prevent Asterisk
from trying over and over again, and eventually causing the
system to crash. (issue #8285, john) ........
* channels/chan_iax2.c: When the IAX2 read callback gets called,
return NULL instead of a "null frame". This will cause Asterisk
to hangup the call instead of keep trying whatever it was doing.
Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that
will cause it to get called every 100 thousand calls or so. When
this does happen, it puts the channel in a loop that eventually
brings down the system. So, hangup up the call is certainly a
better alternative. (issue #8286, john)
* Makefile: Export the GTK2 library and include information to sub
Makefiles.
2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007)
| 3 lines Issue 9415 - No point to getting a diagnostic field if
we aren't doing anything with the information. (Plus, it tends to
crash the Postgres ODBC driver.) ........
2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c: Another crash that I thought we had fixed already
- Issue 9396
* apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007)
| 2 lines Oops ........
* apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007)
| 2 lines Fix a few remaining bad mmap(2) return values ........
2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell@digium.com>
* /, apps/app_directory.c: Merged revisions 59277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) |
3 lines Fix the check of the return value from mmap(). Thanks to
Corydon for catching this one. ........
* apps/app_directory.c: Fix app_directory to actually compile with
ODBC_STORAGE, and update the code to the latest res_odbc API.
* apps/Makefile: Fix app_directory when ODBC_STORAGE is being used.
The Makefile did not properly ensure that this information got
copied from what was selected for app_voicemail. (issue #9224)
* channels/chan_sip.c: Fix the check that ensures that the CHANNEL
function's first argument is "rtpqos". Thanks, Corydon. :)
2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes
asterisk), kpfleming pointed on asterisk-dev, that DECLINE in
this case the proper thing to do. This change now has it doing
the proper thing.
2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) |
4 lines Fix the use of the "sourceaddress" option when "bindaddr"
is set to 0.0.0.0 instead of having each interface explicitly
listed. (issue #7874, patch by stevens) ........
* channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS
function to just be additional parameter of the CHANNEL function.
This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27
Mär 2007) | 1 line fixed #9355 ........
2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* channels/chan_sip.c: Oops, this should be case insensitive
2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes
asterisk). I turned a duplicate context from a WARNING to an
ERROR. Now you get a module load failure, and asterisk just
exits. That's better than a crash, right\?
2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* channels/chan_sip.c: Change this to a single dp function to make
oej happy.
2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf@digium.com>
* main/config.c: Fix for 9257; by eliminating the globals in
main/config.c, we make it thread-safe, which is a minimum
requirement.
2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Add ability to specify no timeout. This
means as soon as the prompt is done playing it moves on to the
next priority.
2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Somehow the code for building the email for
voicemail got out of sync. This change makes a few tweaks to get
1.4 in sync with trunk. (issue #9301)
* apps/app_meetme.c: Fix some codec negotiation problems when
CallerID support is not enabled in SLA. (issue #9308, reported by
twilson)
2007-03-26 18:13 +0000 [r59213] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Make SpeechBackground obey the digit
timeout value.
2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Rename the new dialplan functions to match
the variable name
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The
AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in
some because they get set in sip_hangup. So, there are common
situations where the variables will not be available in the
dialplan at all. So, this patch provides an alternate method for
getting to this information by introducing AUDIORTPQOS and
VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76,
with some testing by blitzrage)
2007-03-26 17:38 +0000 [r59206] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE,
and STANDALONE_AEL
2007-03-26 15:25 +0000 [r59202] Nadi Sarrar <ns@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure,
include/asterisk/autoconfig.h.in, channels/misdn/Makefile,
channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2
provides a dsp pipeline for i.e. echo cancellation modules, make
chan_misdn use it. * add a check for linux/mISDNdsp.h to
configure.ac and update the autogenerated files: 'configure',
'autoconfig.h.in' (the 'configure' script was not in sync with
the latest configure.ac, so the diff is a bit bigger than
expected).
2007-03-26 15:16 +0000 [r59200] Joshua Colp <jcolp@digium.com>
* pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the
aelparse binary! DONT_OPTIMIZE should now work once again.
2007-03-24 01:39 +0000 [r59195] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2
lines Only try to handle a response if it has a response code.
(ASA-2007-011) ........
2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy <murf@digium.com>
* /: blocking out the fix in 59187... already incorporated here
* /, apps/app_macro.c: Merged revisions 59186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1
line Added a few words in the Macro doc strings about the
behavior of macros with hangups (et al.), as per 9337 ........
2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: don't allow string input to overrun the
buffer to hold it (ASA-2007-010)
* channels/chan_misdn.c: remove variables that are no longer used
(--enable-dev-mode is good, developers should be using it)
2007-03-22 14:40 +0000 [r59145] Steve Murphy <murf@digium.com>
* utils/Makefile: The stuff in utils was compiling with -O6 even if
DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp <jcolp@digium.com>
* main/http.c: Add svg mimetype for pari.
* res/res_monitor.c, /: Merged revisions 59086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2
lines Indicate the filename changed when it is changed. (issue
#9311 reported by jsmith) ........
* channels/chan_sip.c: Until we can do media level parsing for
sendrecv/etc just use the first value found. This crept up when a
phone was offered audio+video and returned an inactive video
stream. chan_sip thought the phone said to put the person on hold
but that was totally wrong. (issue #9319 reported by benbrown)
2007-03-20 21:04 +0000 [r59078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/logger.c: Fix defines for inline stack backtraces (only used
by developers anyway)
2007-03-20 20:42 +0000 [r59076] Joshua Colp <jcolp@digium.com>
* channels/iax2-parser.c: Copy len variable as well, should fix
remaining IAX2 DTMF issues.
2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy <murf@digium.com>
* apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should
return it to its previous, untouched, state.
* apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h:
The fix for the AEL <<security hole>> (bug 9316) is here...
2007-03-20 13:16 +0000 [r59064] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
channels/misdn/chan_misdn_config.h: Merged revisions
58849-58850,59062-59063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) |
1 line added method standard_dec for dialing out on groups, to
avoid conflicts, which caused issues with some ISDN providers
........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13
Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 |
crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one. ........
r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) |
1 line modified a loglevel ........
2007-03-19 Jason Parker <jparker@digium.com>
* Asterisk 1.4.2 released.
2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_strings.c: Oops, this should have been a %d all along
2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp@digium.com>
* funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295
reported by ajohnson)
2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* configs/sip_notify.conf.sample: Fix unescaped semicolon (reported
via -dev list)
2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return
code 0 (reported by qwerty1979)
2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke@btwtech.com>
* apps/app_followme.c: Don't return a non-zero return code if the
profile doesn't exist, to match what the documentation says it
already does. (#9307 Reported by kkiely)
2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp@digium.com>
* apps/app_page.c: Wait for the async thread to exit when hanging
up all of the paged phones under all circumstances. (issue #9181
reported by PhilSmith)
2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell@digium.com>
* configs/sla.conf.sample: fix a couple SLA documentation
references
* doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex
(removed), doc/freetds.txt (added), doc/odbcstorage.txt (added),
doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added),
doc/channelvariables.txt (added), doc/ael.txt (added),
doc/billing.tex (removed), build_tools/prep_tarball,
doc/callingpres.txt (added), doc/enum.txt (added),
doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added),
doc/cdrdriver.tex (removed), build_tools/make_buildopts_h,
doc/security.txt (added), doc/imapstorage.txt (added),
doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed),
doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac,
doc/iax.txt (added), doc/ael.tex (removed),
doc/channelvariables.tex (removed), doc/enum.tex (removed),
doc/security.tex (removed), doc/math.txt (added), Makefile,
doc/imapstorage.tex (removed), doc/privacy.tex (removed),
doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt
(added), apps/app_voicemail.c, doc/cliprompt.txt (added),
doc/chaniax.txt (added), doc/app-sms.txt (added),
doc/ast_appdocs.tex (removed), doc/realtime.tex (removed),
doc/ices.txt (added), doc/dundi.tex (removed),
doc/linkedlists.txt (added), doc/queuelog.txt (added),
doc/extconfig.txt (added), doc/radius.txt (added),
doc/cliprompt.tex (removed), doc/chaniax.tex (removed),
doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex
(removed), doc/ices.tex (removed), doc/asterisk.tex (removed),
doc/queuelog.tex (removed), doc/configuration.txt (added),
doc/asterisk-conf.txt (added), doc/sla.pdf (added),
doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt
(added), doc/mp3.tex (removed), doc/configuration.tex (removed),
doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added),
doc/channels.txt (added), doc/ip-tos.tex (removed),
doc/extensions.txt (added), doc/queues-with-callback-members.txt
(added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added),
doc/misdn.txt (added), doc/manager.txt (added),
doc/jitterbuffer.tex (removed), doc/extensions.tex (removed),
doc/billing.txt (added), doc/localchannel.txt (added),
doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt
(added), doc/00README.1st (added): Making these documentation
changes in the 1.4 branch upset various people, so these chanes
will only be done in the trunk.
* build_tools/prep_tarball: Add the --pdf option to the usage of
rubber in prep_tarball
* Makefile, build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add
configure script checking for GTK2 and some additional Makefile
targets to support gmenuselect
2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match
common syntax and update the resulting appdocs TeX file
2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell@digium.com>
* doc/asterisk.tex: add a link to the rubber homepage
2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_setcdruserfield.c, main/pbx.c,
apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c:
Expand deprecation warnings from simply warning on use to the
builtin documentation.
2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell@digium.com>
* doc/asterisk.tex, Makefile: Add Asterisk version information to
the generated PDF
* build_tools/prep_tarball: have prep_tarball attempt to build
asterisk.pdf
2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_realtime.c: Function works fine, but the documentation
is backwards.
2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell@digium.com>
* doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex
(added), doc/freetds.txt (removed), doc/odbcstorage.txt
(removed), configure, doc/sla.tex, doc/cygwin.txt (removed),
doc/model.txt (removed), doc/channelvariables.txt (removed),
doc/ael.txt (removed), doc/billing.tex (added),
doc/callingpres.txt (removed), doc/enum.txt (removed),
doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed),
doc/cdrdriver.tex (added), build_tools/make_buildopts_h,
doc/security.txt (removed), doc/imapstorage.txt (removed),
doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added),
doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac,
doc/iax.txt (removed), doc/ael.tex (added),
doc/channelvariables.tex (added), doc/enum.tex (added),
doc/security.tex (added), doc/math.txt (removed), Makefile,
doc/imapstorage.tex (added), doc/privacy.tex (added),
doc/realtime.txt (removed), doc/dundi.txt (removed),
doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt
(removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed),
doc/ast_appdocs.tex (added), doc/realtime.tex (added),
doc/ices.txt (removed), doc/dundi.tex (added),
doc/linkedlists.txt (removed), doc/queuelog.txt (removed),
doc/extconfig.txt (removed), doc/radius.txt (removed),
doc/cliprompt.tex (added), doc/chaniax.tex (added),
doc/hardware.txt (removed), doc/mp3.txt (removed),
doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex
(added), doc/queuelog.tex (added), doc/configuration.txt
(removed), doc/asterisk-conf.txt (removed), doc/sla.pdf
(removed), doc/ip-tos.txt (removed), doc/hardware.tex (added),
doc/h323.txt (removed), doc/mp3.tex (added),
doc/configuration.tex (added), doc/asterisk-conf.tex (added),
doc/jitterbuffer.txt (removed), doc/channels.txt (removed),
doc/ip-tos.tex (added), doc/extensions.txt (removed),
doc/queues-with-callback-members.txt (removed), doc/apps.txt
(removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt
(removed), doc/manager.txt (removed), doc/jitterbuffer.tex
(added), doc/extensions.tex (added), doc/billing.txt (removed),
doc/localchannel.txt (removed),
doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt
(removed), doc/00README.1st (removed): Merge changes from
svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc
directory into a single LaTeX formatted document so that we can
generate a PDF, HTML, or other formats from this information. *
Add a CLI command to dump the application documentation into
LaTeX format which will only be include if the configure script
is run with --enable-dev-mode. * The PDF turned out to be close
to 1 MB, so it is not included. However, you can simply run "make
asterisk.pdf" to generate it yourself. We may include it in
release tarballs or have automatically generated ones on the web
site, but that has yet to be decided.
2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Don't assume that the pvt structure will
still exist after calling schedule_delivery as it may not. (issue
#9278 reported by fmachado)
2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Some people like to put "limitonpeer"
instead of "limitonpeers" in their configuration. While we're at
it, support "limitonpeerz" and "limitonpeerssssss". (inspired by
issue #9172)
* doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the
examples section
* doc/security.txt, /: Merged revisions 58896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) |
3 lines Add a note to the security file that the Asterisk CLI and
log files may contain sensitive information, and that people
should keep this in mind. ........
* configs/sla.conf.sample, apps/app_meetme.c: By default, don't
attempt to do any CallerID handling at all with SLA because it is
known to not work properly in some situations. However, add an
option to enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID
with SLA, we need the ability to change the CallerID on an
existing call, and we are not ready to handle that.
2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_strings.c: Issue 9162 -
pbx_substitute_variables_helper assumes the buffer is initialized
to all zeroes. This fixes a case where it wasn't.
2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Ensure that the blinky lights show that the
trunk stopped ringing when the trunk hangs up before a station
has answered it. (issue #9234, reported by francesco_r)
* configs/sla.conf.sample: fix the reference to the SLA
documentation
2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2
lines Issue #9229 - No port in request URI on register to non
default SIP ports (neelakantan) ........
* channels/chan_sip.c: Don't hangup the call on OK or errors on
MESSAGE and INFO inside of a dialog (like video update requests).
* channels/chan_sip.c: Issue #9251 - Clear From URI from user
attributes (tgrman)
2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
revisions 57034,57523,57753,58558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) |
1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com
bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02
19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........
r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) |
1 line fixed another place where the out_cause was hardcoded to
16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09
Mar 2007) | 1 line we can free channel 31 as well, since we can
occupy it ........
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, channels/misdn/ie.c,
channels/misdn/isdn_msg_parser.c: added UU transceiving and
corect handling for rdnis
2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Allow RFC2833 compensation to compensate for even
stupider implementations by queueing up the end frame at the
start, not the actual end. (issue #8963 reported by AndrewZ)
* channels/chan_sip.c, configs/sip.conf.sample: Add
matchexterniplocally setting which only substitutes your
externip/externhost setting if it matches the localnet setting. I
know of at least two people who need opposite settings, so I made
it an option! (issue #8821 reported by kokoskarokoska)
2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a few more places in chan_iax2 where
the ast_frame used for receiving a frame was not properly
initialized. - Interpolating a frame when the jitterbuffer is in
use - decrypting a frame when IAX2 encryption is on - frames in
an IAX2 trunk
* apps/app_meetme.c: Make the compiler happy and initialize a
variable.
* doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added):
Merge some updates to the SLA documentation. I plan to keep
working on this to explain all of the expected behavior with call
handling, configuration details for specific phones, and other
things. However, I got tired of doing it in plain text, so I
switched to using LaTeX. I have included the PDF version. I
haven't been able to get a nice looking plain text version out of
it yet, but I'm not terribly concerned since this is supposed to
be more of the manual, while the plain text sample configuration
file is the reference.
2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Fix spelling of unavailable in voicemail
documentation. (issue #9248 reported by tensai)
* /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2
lines If we are unable to lookup the host in a c line we have to
abort, otherwise the previous data is gone and we will
(potentially) have no data when all is said and done. ........
2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Hang up the channel that put the call on hold
in the event processing thread to avoid a race condition. Also,
if the station originated the call that it is putting on hold,
don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
* channels/chan_zap.c: Add a missing break statement so that
handling the above event does not incorrectly destroy the
channel. (issue #9242, andrew)
2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c: Fix segfault (Issue 9236)
2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Refactor hold handling a bit so that it does
not require keeping the call up when a call is put on hold.
2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Make early SDP seeding even smarter! We have to check
codecs in the make_compatible function too. (issue #9221 reported
by marcelbarbulescu)
* main/dsp.c, /: Merged revisions 58388 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2
lines Only print out debug message if the definition that makes
the variables shows up was actually defined. (issue #9233
reported by serginuez) ........
2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming@digium.com>
* main/http.c: this change was not needed; fclose() handles closing
the file descriptor already
* apps/app_meetme.c: fix a compiler warning, and overwriting 'res'
value
* main/http.c: fix two cases where HTTP session file descriptors
would not be closed
2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell@digium.com>
* channels/chan_zap.c, configure, configure.ac: If we receive
ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256,
tzafrir) Also, update the configure script to make sure that we
don't try to build chan_zap if the installed version of zaptel
does not include ZT_EVENT_REMOVED.
* /, channels/chan_iax2.c: (This bug was reported to me by Kinsey
Moore) Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) |
7 lines Fix a problem where the Asterisk channel name could be
that of the wrong IAX2 user for a call. This is because the first
step of choosing this name is to look for an IAX2 peer that
happens to have the same IP/port number that this call is coming
from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication
happens since at that point, we have an exact match. ........
2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp@digium.com>
* main/rtp.c, channels/chan_sip.c: Ensure we have (or should have)
at least one matching codec before attempting early bridge SDP
seeding. (issue #9221 reported by marcelbarbulescu)
2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell@digium.com>
* main/manager.c, /: Merged revisions 58164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) |
4 lines If the channels acquired using the manager Redirect
action are not up, then don't attempt to do anything with them.
It could lead to weird behavior, including crashes. (issue #8977)
........
2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf@digium.com>
* /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1
line Fix for 9220: Eyebeam cannot renew subscriptions for
presence info. Reason: re-SUBSCRIBE requests don't include Accept
headers, which the rfc says are optional (to put it tersely), (it
uses MAY), and luckily, the sip_pvt struct has the format info
stored, so we simply leave it if the format is set, and the
accept header null. ........
2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell@digium.com>
* configs/voicemail.conf.sample: Clarify the documentation of the
dialout and sendvoicemail options. (issue #9000, caio1982 and
serge-v)
2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2
lines Change error message to proper message ........
2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell@digium.com>
* channels/chan_skinny.c: Return an error of transmit_response is
called without a session. (issue #9002)
2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Since chan_iax2 does not support reception
of DTMF with duration ensure that it is set to 0 on the frame.
(issue #8521 reported by gdhgdh)
* apps/app_meetme.c: Don't create a listen channel and record the
conference unless the option is turned on. (issue #9204 reported
by francesco_r)
* apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2
lines Make create_dirpath use our standard for return values. -1
is failure, 0 is success. (issue #9205 reported by ballares)
........
2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 57825 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1
line Fixed a typo introduced via 9156 (either the gotos or their
doc strings are wrong) ........
2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp@digium.com>
* main/slinfactory.c: Don't allow a NULL pointer to reach
ast_frdup. (issue #9155 reported by cmaj)
* res/res_jabber.c: Don't reference a potentially NULL pointer.
(issue #9199 reported by klolik)
* main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198
reported by edgreenberg)
2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2,
pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7:
Updated the regression tests
2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007)
| 2 lines Memory leak of a list, if call recording was abandoned
........
2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard@digium.com>
* main/say.c: submitted patch for Georgian language, issue 9010,
submitted by Alexander Shaduri
2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell@digium.com>
* configs/sla.conf.sample: add missing configuration template.
Thanks to Lacy Moore on asterisk-users for pointing this out\!
2007-03-02 Russell Bryant <russell@digium.com>
* Asterisk 1.4.1 released.
2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com>
* configure, configure.ac: Update the check that is used to
determine whether zaptel transcoder support is present. The
interface has changed.
2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2
lines If a SIP message comes in and goes to a method handler that
requires additional values that may not be present then send back
an error. ........
2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 57458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1
line further refinement in wording of goto documentation, as per
9156, goto not proceeding to next instruction ........
* pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes
right, but 9184 points out the problem-- the escape is removed by
pbx_config, and pbx_ael should also, before sending it down into
the pbx engine. Also, you have to insert it back in, if you are
generating extensions.conf code from the AEL.
2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com>
* main/file.c: Return the correct digit that interrupted the
stream. This fixes exiting the Background application when using
the m option. (issue #9176, mjagdis)
* configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt,
include/asterisk/channel.h: Merge changes from
svn/asterisk/team/russell/sla_updates * Originally, I put in the
documentation that only Zap interfaces would be supported on the
trunk side. However, after a discussion with Qwell, we came up
with a way to make IP trunks work as well, using some things
already in Asterisk. So, here it is, this now officially supports
IP trunks. * Update the SLA documentation to reflect how to setup
IP trunks. * Add a section in sla.txt that describes how to set
up an SLA system with voicemail. * Simplify the way DTMF
passthrough is handled in MeetMe. * Fix a bug that exposed itself
when using a Local channel on the trunk side in SLA. The
station's channel needs to be passed to the dial API when dialing
the trunk. * Change a WARNING message to DEBUG in channel.h. This
message is of no use to users.
2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 57317 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar
2007) | 2 lines Don't even attempt to optimize things when a
proxy channel is involved. It will just explode in weird and
unexplaineable ways. (issue #9175 reported by
clegall_proformatique) ........
2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com>
* doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick.
2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com>
* configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla
docs
* configs/sla.conf.sample, apps/app_meetme.c: Merge more changes
from svn/asterisk/team/russell/sla_updates * Add support for
private hold. By setting "hold=private" for a trunk, only the
station that put the call on hold will be able to retrieve it
from hold. Also, by setting "hold=private" for a station, any
call that station puts on hold can only be retrieved by that
station.
* apps/app_meetme.c: Minor formatting change
* configs/sla.conf.sample, apps/app_meetme.c: Merge changes from
svn/asterisk/team/russell/sla_updates * Add support for the
"barge=no" option for trunks. If this option is set, then
stations will not be able to join in on a call that is on
progress on this trunk.
2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com>
* main/pbx.c, /: Merged revisions 57118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1
line a small documentation update, to reflect reality in the goto
doc strings, as per 9156, Goto does not proceed to next prio if
jump fails ........
2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com>
* /, channels/chan_agent.c: Merged revisions 57092 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb
2007) | 2 lines Fix a few more issues with the agent logoff CLI
command. (issue #9123 reported by arbrandes) ........
2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com>
* configs/sla.conf.sample, apps/app_meetme.c: Merge current set of
changes from svn/asterisk/team/russell/sla_updates * Add support
for station ring delays. Ring delays can be set globally for a
station or for specific trunks on the station. * Fix a few bugs
in existing code. * Restructure and Reorganize code to improve
readability and maintainability. * Improve formatting of the "sla
show (trunks|stations)" CLI commands.
2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Picky compiler...
* apps/app_speech_utils.c: Better handle timeouts when the
individual speaks after everything has been played but before the
timeout ends.
2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: I was surprised that I had not yet downgraded
missing goto targets and macro call defs to a warning, in case
they are in extensions.conf; I rectified this problem. Also, A
goto in a macro to a target in a catch block was not being found;
I fixed this too; the cause was that I needed to treat catch
statements like an extension in the find_match code.
2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Fix voicemail email attachments. I missed
the conversion of one of the line endings and there was an extra
one where it should not have been. (issue #9128)
2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky,
picky... show deprecation warning in application help, too
(reported via list)
2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com>
* channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where
if a device was not specified in alsa.conf, then we just use the
system default, instead of creating our own default of hw:0,0.
(issue #9139)
2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com>
* /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2
lines Obey the clearglobalvars option in extensions reload (or
dialplan reload depending on your version). (issue #9146 reported
by ramonpeek) ........
2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Fix a crash in my last change to
iax2_indicate(). (issue #9150)
2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com>
* apps/app_record.c: Update app_record documentation to use new CLI
command, core show file formats. (issue #9151 reported by junky)
* main/pbx.c: Use ast_strlen_zero to see if the language and/or
context argument is not present for Background instead of just
checking if it is NULL. (issue #9141 reported by mjagdis)
2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Do more complete locking of the
chan_iax2_pvt struct in the indicate callback. (Problem brought
up by Ben Smithurst on the asterisk-dev list)
2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com>
* main/asterisk.c: Allow both of the show version files and core
show file versions CLI commands to work. (issue #9135 reported by
mvanbaak)
2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Move a comment to be in the correct struct.
2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/channel.c, /: Merged revisions 56684 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007)
| 3 lines Issue 9130 - If prev is the last item on the channel
list, then evaluating additional conditions (e.g. name prefix)
will cause a NULL dereference. ........
2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Make sure to set a speeddials parent on
creation. Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com>
* codecs/codec_zap.c: update to match zaptel 1.4 API change that
was committed a few minutes ago
2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com>
* main/asterisk.c, /: Merged revisions 56504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) |
8 lines Fix up a couple more signal handlers to not do bad things
that could cause various undesirable results. The other day, I
made Asterisk deadlock by hitting Control-C because of a bad
signal handler. Now, signal handlers just set a flag and write to
an alert pipe for the flag to be handled. Then, there is another
thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the
background, a thread is created to do it. ........
2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com>
* main/sched.c: Change log notice to debug. It is possible for a
scheduled item to execute and be deleted at close to the same
time and unavoidable. If this happens this message creeps up.
2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) |
4 lines Don't destroy mutexes before unregistering all of the
entry points from the core. Also, fix a potential memory leak
from not destroying the locks for all of the possible call
numbers (about 32k of them). ........
2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com>
* build_tools/make_version_h: build special version strings for
AADK/S800i builds
2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: The IMAP storage code uses the same code to
build the email that is used when voicemail is sent via email
using something like sendmail. In the patch from bug 8033 to fix
various IMAP storage problems, the line endings in the email file
were changed in the code from "\n" to "\r\n". However, this
breaks sending regular voicemail to email. So, this change
conditionally sets line endings to "\r\n" only if IMAP_STORAGE is
enabled. (issue #9128, patch by jarjarbinks, modified by me to
not break IMAP storage)
2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com>
* configs/sla.conf.sample, main/dial.c, apps/app_meetme.c,
doc/sla.txt: Merge changes from team/russell/sla_updates. This
batch of changes to the SLA code does a few different things. * I
made the SLA code event driven instead of having to act in a lot
of busy loops while dialing things to wait for state changes.
This makes the code more efficient and readable at the same time.
* I have implemented a couple of new features. The first is
inbound trunk ringing timeouts. This is an option that defines
how long to let an incoming call on a trunk to ring. * I have
also implemented ring timeouts for stations. They may be
specified for the entire station, meaning it is how long to let
the station ring before giving up. You can also specify a ring
timeout for a specific trunk on a station. So, you can say that
you only want a specific station to ring 5 seconds if it is line1
ringing, but otherwise, there is no timeout.
2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com>
* main/channel.c, /, channels/chan_sip.c: Merged revisions 56230
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2
lines Only change the original or clone channel if it's the
channel behind the proxy channel, not if it's just a regular
bridged channel. ........
2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com>
* doc/osp.txt: Update OSP documentation for v1.4.
2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Move message from verbose to debug
2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com>
* sounds/Makefile: updated the sound tarball versions in Makefile
2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Restructure a little bit of code to reduce
nesting. There is no functionality change here.
* /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) |
3 lines If we receive a frame that is not in any of the
negotiated formats, then drop it. (potentially issue #8781 and
SPD-12) ........
2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com>
* main/cli.c: Print out deprecation notice on usage output of CLI
commands. (issue #8925 reported by blitzrage)
2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com>
* main/loader.c: disable unloading of embedded modules... there is
a fundamental problem with doing so that will not be fixed in
this version of Asterisk due to its invasiveness
2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com>
* /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2
lines Change naughty warning message to provide useful
information. If a write now fails on a channel in meetme it will
tell you the channel name instead of spitting out the wrong error
message. ........
2007-02-21 20:27 +0000 [r55954] Jason Parker <jparker@digium.com>
* channels/chan_gtalk.c: Fix locking issue, and accept
"transport-accept" as a valid accept message. This should solve
issues 8970 and 8503.
2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Simplify the last change to app_meetme, and
move the call to dispose_conf() up into the block where we know a
conf exists.
2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Only dispose of the conference if one was
created.
* apps/app_speech_utils.c: Only start playing the next file if we
have not been quieted.
* channels/chan_sip.c: Add a flag that indicates whether a SIP
dialog is an outgoing call or not. SIP_OUTGOING originally did it
but it was repurposed to the direction of the last transaction,
which can cause update_call_counter to falsely decrease the wrong
counters. (please don't hurt me oej) (issue #8943 reported by
mdu113)
2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com>
* /, build_tools/make_version: Merged revisions 55868 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21
Feb 2007) | 2 lines use new tag version script ........
2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly
after transfer (decrement inuse early on transferer's call leg)
2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com>
* channels/chan_gtalk.c: Fix segfault when buddy couldn't be found.
Issue 7764, patch by sailer
2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Improve the reference counting to fix bugs
where people report seeing conferences listed that have no
members. (issue #9073)
2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Better handle dropped IMAP connections.
(issue #9054 reported by bsmithurst)
* channels/chan_sip.c: Return behavior I removed. I did not
remember that you could just add a localnet entry to make it
work.
* channels/chan_sip.c: Don't test our own address against the
localnet settings. At least one person has had issues as a result
of this from #7051 so I'm reversing it. (issue #8821 reported by
kokoskarokoska)
* /, channels/chan_agent.c: Merged revisions 55669 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb
2007) | 2 lines Defer clearing callback information if channels
are up until they are hung up. This ensures the hangup process
goes smoothly and no channels get hung in limbo. (issue #8088
reported by kebl0155) ........
2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com>
* main/http.c: Add the Asterisk version information to the Server
header in HTTP responses. (requested by Pari)
* include/asterisk/manager.h: Increase the maximum number of
manager headers to 128, at the request of Pari.
2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com>
* channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free
with strdupa (thanks file) 55555!
2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com>
* configs/sla.conf.sample: Change the formatting of sla.conf.sample
to make it more readable. (issue #9112, blitzrage)
2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net>
* res/res_jabber.c: - Not sending arguments to an application is
not "out of memory" - Making error messages a bit more clear
2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007)
| 2 lines forcename and forcegreetings options should check to
see if the recording already exists ........
2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com>
* channels/chan_iax2.c: Changed iax2 process thread to detached to
correct memory leak due to left over thread context on thread
exit. Modified module unload process to avoid deadlocks on
pthread cancels
2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net>
* /, apps/app_record.c: Merged revisions 55277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2
lines Documentation update (#9053, jsmith) ........
* /: Block patch that was made only for 1.2 (already implemented in
1.4 and trunk)
2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com>
* apps/app_queue.c: Add missing membername option to AddQueueMember
documentation. (issue #9088 reported by seanbright)
2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix an issue where callerid would not be
displayed on some phones. Issue 8995, initial patch and research
done by wedhorn
2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 55153 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2
lines Answer the channel before recording privacy information.
(issue #8926 reported by lmamane) ........
* apps/app_queue.c: Make the 'i' option of Queue actually work.
(issue #8986 reported by utis)
* /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2
lines Allow chan_sip to handle attended transfers from a SIP
phone that is sitting behind chan_agent. Yes folks, all it took
was one line of code. (issue #8784 reported by pzieba) ........
2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac: If the
pg_config application is found, but there is probably executing
it, then consider postgres unavailable. (issue #8637)
* codecs/gsm/Makefile: Filter out yet another architecture that
does not work with the optimizations in the built-in libgsm.
(issue 8637, ovi)
* /, apps/app_meetme.c, configs/meetme.conf.sample: Merged
revisions 55005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) |
9 lines Revert the change I did in revisions 54955, 54969, and
54970, in 1.2, 1.4, and trunk. I decided that once a conference
is created from meetme.conf, it is acceptable behavior that the
pin can not be changed until the conference goes away. I also
added a note in meetme.conf to describe this behavior. We still
have another issue in 1.4 and trunk where some conferences with
no users don't go away. That is the real bug that needs to be
addressed here. ........
2007-02-16 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com>
* /, channels/chan_agent.c: Merged revisions 54999 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb
2007) | 2 lines Do not send indications through ast_indicate in
chan_agent but instead go directly to the technology. This way
when indications are emulated they happen on the Agent channel
and do not screw up formats on the channels. (issue #8439
reported by punkgode) ........
2007-02-16 21:12 +0000 [r54969] Russell Bryant <russell@digium.com>
* /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) |
5 lines For conferences that are configured in meetme.conf, check
the configuration file every time someone joins the conference
instead of only when the conference is first created. This is to
ensure that changes to the pin numbers in the config file are
always honored. (issue #9073) ........
2007-02-16 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c: Need to check macro extension as well as macro
context for directed pickup.
2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com>
* pbx/pbx_config.c: Fix setting "autofallthrough" to yes by
default. It was set to enabled in pbx.c. However, if the option
was not present in extensions.conf, then pbx_config.c would set
it back to disabled.
* res/res_features.c: Clean up a few coding guidelines issues -
spaces to tabs, use sizeof() to pass the size of a static buffer,
add spaces ...
2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com>
* main/asterisk.c: Clarify a restart message. It's silly, but the
reporter had a very valid point. Issue 9079
2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c: Allow directed pickup to pick up the real
context instead of the macro context if a Macro is used. (issue
#8984 reported by jamesb63)
2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue #7541 - Handle multipart attachments
to SIP messages - even if boundary is quoted.
* /, res/res_agi.c: Merged revisions 54771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2
lines Issue #9069 - If we open with TH we should not close with
/TD. (seanbright) ........
2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Don't let dtmf leak over into the engine
and let it skew the results... also give DTMF results priority.
(issue #9014 reported by surftek)
* apps/app_dial.c, /: Merged revisions 54622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2
lines Use a separate variable to indicate execution should
continue instead of the return value. (issue #8842 reported by
pluto70) ........
* apps/app_dial.c: Forward begin DTMF frames as well as end. (issue
#9068 reported by mhardeman)
2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net>
* /: Block patch only needed in 1.2
2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com>
* channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2
lines When handling glare on a PRI, move the requested channel
rather than hang up the old one. Fix for 8957 and 9011. ........
2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com>
* main/channel.c: Add G722 to ast_best_codec. If anyone disagrees
with it's placement, feel free to change it. (issue #9045
reported by gork)
2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Remove a couple of leftover debug messages
* include/asterisk/devicestate.h: Fix the documentation on the
return values from device state provider registration and
deletion.
* channels/chan_sip.c: If we fail to create the SIP socket, then
return -1 from reload_config() so that load_module() will return
AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get
spammed with error messages every time chan_sip tries to send a
message.
2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net>
* /: Blocking patch for 1.2 only
2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com>
* main/dial.c, include/asterisk/dial.h: Change
ast_set_state_callback() to ast_dial_set_state_callback()
* main/dial.c, apps/app_meetme.c, apps/app_page.c,
include/asterisk/dial.h: - Add the ability to register a callback
to monitor state changes in an asynchronous dial operation. -
Rename the various references to "status" to "state" in the dial
API
2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com>
* configure, configure.ac: Make the --without-oss argument work.
(issue #9026 reported by puzzled)
2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com>
* configs/users.conf.sample: Fix a typo where "vmpassword" should
be "vmsecret"
2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_h323.c: Fix VLDTMF reception
* apps/app_echo.c: Much simpler than previous one ;-)
* main/channel.c: Provide correct DTMF duration
* main/cli.c: Bring deprecated 'debug channel <x|all>' command back
2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac, acinclude.m4: don't display the
--with-imap message unless --with-imap was specified without a
path use '-n' instead of '! -z' for tests
2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Add some output for "show application
SLAStation/SLATrunk"
* channels/chan_sip.c: Change some text to properly state "On
Hold", which was already done in trunk.
* configs/sla.conf.sample, include/asterisk/app.h,
include/asterisk/utils.h, main/dial.c, apps/app_meetme.c,
channels/chan_sip.c, doc/sla.txt (added),
include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge
team/russell/sla_rewrite This is a completely new implementation
of the SLA functionality introduced in Asterisk 1.4. It is now
functional and ready for testing. However, I will be adding some
additional features over the next week, as well. For information
on how to set this up, see configs/sla.conf.sample and
doc/sla.txt. In addition to the changes in app_meetme.c for the
SLA implementation itself, this merge brings in various other
changes: chan_sip: - Add the ability to indicate HOLD state in
NOTIFY messages. - Queue HOLD and UNHOLD control frames even if
the channel is not bridged to another channel. linkedlists.h: -
Add support for rwlock based linked lists. dial.c: - Add the
ability to run ast_dial_start() without a reference channel to
inherit information from.
* apps/app_echo.c: When the Echo() application receives the digit
'#', echo that back as well. Since we already sent the BEGIN
frame for that digit, it makes sense to send the END as well.
2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_gtalk.c: another dependency
* apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c,
funcs/func_odbc.c, res/res_adsi.c: add some inter-module
dependencies
* build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk
scripts to work when both MODULEINFO and MAKEOPTS are present in
a source file
2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c: Temporarily change musicclass on channel to one
specified in Dial so that the 'm' option functions properly.
(issue #8969 reported by christianbee)
2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com>
* doc/imapstorage.txt, configure, configure.ac: clarify the fact
that voicemail IMAP storage cannot be built against a distro's
binary c-client library package (at least not at this time)
2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net>
* main/acl.c: Don't output debug unless we asked for it
2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Fix timeout issue when utterance is
longer then timeout itself.
2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/loader.c: Issue 9007 - Mutex not released on early return
* apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007)
| 2 lines Issue 9003 - If fullname is empty, quote() passes back
"\"" ........
2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com>
* main/db1-ast/Makefile: When building libdb1.a, put the additional
flags needed at the beginning of ASTCFLAGS, instead of at the
end. This way, we ensure that we find the local headers first
before accidentally trying to use headers that exist in locations
specified in the ASTCFLAGS passed from the main Makefile. (issue
#8637, ovi)
* main/Makefile: The clean target actually needs to run "distclean"
on editline. This is because we need to make sure that its
configure script gets executed again, because the CFLAGS we want
to pass to editline may have changed.
2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com>
* main/rtp.c: We can not reliably do P2P bridging with DTMF passing
back with compensation if we need to listen for DTMF frames.
(issue #8962 reported by caio1982)
2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com>
* main/rtp.c: When parsing the NTP timestamp in a sender report
message, you are supposed to take the low 16 bits of the integer
part, and the high 16 bits of the fractional part. However, the
code here was erroneously taking the low 16 bits of the
fractional part. It then shifted the result 16 bits down, so the
result was always zero. This fix makes it grab the appropriate
high 16 bits, instead. (issue #8991, pointed out by
andre_abrantes)
2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com>
* apps/app_playback.c: Directly load say.conf in load_module
instead of calling the reload function. (issue #8946 reported by
junky)
* /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2
lines Fix a few potential memory leaks with realtime users and
peers. (issue #8999 reported by bsmithurst) ........
2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, apps/app_macro.c: Merged revisions 53354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007)
| 2 lines Issue 7440 - Macro called from Macro from the h
extension exits prematurely ........
2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged
revisions 52843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) |
1 line fixed some possible segfaults. also fixed an very
important bug which occurs on high load (when calls are very fast
generated) ........
2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_jabber.c: Text fix for jabber reload command (reported by
bkruse via IRC)
* main/manager.c, /: Merged revisions 53245 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007)
| 2 lines Issue 8987 - Status could return two responses
(mnicholson) ........
2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Formatting
2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com>
* apps/app_playback.c: Ensure say_cfg is NULL when the module is
loaded. (issue #8946 reported by junky)
* apps/app_playback.c: Unregister Playback CLI commands as well as
dialplan application. (issue #8946 reported by junky)
2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Add some comments on queue system behaviour
and how it affects the SIP channel
2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make SIPDtmfMode application work with
recent capability changes, and also fix an RTP stack issue when
the auto option was used. (issue #8972 reported by mdu113)
2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com>
* apps/app_dial.c, /: Merged revisions 53133 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) |
4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when
the dial application exits early because of invalid arguments
instead of just leaving it empty. (issue #8975) ........
2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string
because due to compatibilities with CS1000 reported at
www.voip-info.org
2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com>
* UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a
warning to the console that things might possibly be
misconfigured when queue member's states are still 'Not in Use'
when we're about to bridge them with a caller from queue. Also,
put some documentation quoted from oej's queues.txt efforts
started in /trunk today. This commit puts #7433 into feedback
state for 1.4, and pending no further negative feedback, it will
finally be closed.
2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Correct a copy/pasted error message line for RTCP.
* main/config.c, /: Merged revisions 53117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2
lines Pass the glob expanded filename to process_text_line so
that error messages contain the actual filename, not the original
include one. (issue #8959 reported by tzafrir) ........
* Makefile: Add systemname to asterisk.conf generation per recent
discussions about it. (issue #8968 reported by blitzrage)
2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: Disable the direct
p2p RTP call setup in SIP. You can enable it in sip.conf, but it
is now considered experimental until we solve the
AST_CONTROL_ANSWER with payload and videocaps stuff.
2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2
lines Copy noncodeccapability over to the joint variable so that
telephone-event will get transmitted in the sent INVITE. ........
* main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile
here as well, but it apparently required both dev mode and no
optimizations to creep up.
* /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2
lines Don't negotiate RFC2833 when not configured to do so.
(issue #8799 reported by mdu113) ........
2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com>
* funcs/func_strings.c: Fix the FIELDQTY function to not crash.
(reported by blitzrage and Corydon on IRC)
2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net>
* /: Going backwards, blame file.
2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com>
* /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb
2007) | 2 lines Return previous behavior of having MOH pick up
where it was left off. (issue #8672 reported by sinistermidget)
........
* funcs/func_strings.c: Make func_strings build under dev mode.
Didn't I do this today already in the berkeley DB?
2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Clean INC_COUNT flag when we decrement
call counter - If it's still set at time of dialog destruction,
make sure we decrement the device call counter properly before we
destroy the dialog
* apps/app_queue.c: Change debug level for state change message
that is not really informative when debugging app_queue
* channels/chan_sip.c: Cleaning up the devicestate callback
function
2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_strings.c: Oops.
* /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007)
| 2 lines Bug 8965 ........
2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com>
* main/asterisk.c: Add missing 'F' letter to getopt so it magically
becomes a valid option. (issue #8960 reported by tzafrir)
2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007)
| 2 lines No wonder FIELDQTY doesn't work with functions... the
documentation in pbx.c was wrong ........
2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Fix silly logic. We really want to write
UDPTL frames out when the call is up.
2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Add explanation of port= in combination
with defaultip= (thanks jsmith)
2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: we update the name on any first reply of
our setup
2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_h323.c: chan_h323 is very stable, so let it built
by default
2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com>
* main/rtp.c: When going on hold have the side that was put on hold
reinvite back to Asterisk. When going off hold have the side that
was taken off hold reinvited back to the other party.
* main/rtp.c: Add more frame types to forward in the RTP bridge
loops.
2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com>
* main/cdr.c, main/manager.c, pbx/pbx_spool.c,
channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c,
main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c,
channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c:
Merged revisions 53045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) |
3 lines Fix a bunch of places where pthread_attr_init() was
called, but pthread_attr_destroy() was not. ........
* apps/app_userevent.c: Remove an extra \r\n from manager user
events. (issue #8955, mnicholson)
* main/rtp.c, /: Merged revisions 53039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) |
3 lines Use the proper format string to print unsigned values in
the rtp debug output. (issue #8954, wmis) ........
* apps/app_queue.c: Only changed the paused status in an existing
queue member if the paused column exists.
* apps/app_queue.c: Instead of always creating a realtime queue
member as unpaused, read the "paused" column and use that value
for the paused status of the member. (issue #8949, jmls)
* contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10.
(issue #8363, johnlange)
* doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue
#8942, lters)
* configure, include/asterisk/autoconfig.h.in, configure.ac,
codecs/codec_gsm.c: When we are checking for a system installed
version of libgsm, we need to check for gsm.h as well.
Furthermore, when checking for this header, it may be located in
a gsm/ sub directory, so check for that, as well. (issue #8773)
* channels/chan_sip.c: Only set the DTMF flag on the rtp structure
if the DTMF mode is actually RFC2833, not just that it is not
INFO. This makes it get set for inband DTMF as well, which is not
valid. (issue #8936)
* main/asterisk.c, /: Merged revisions 52903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) |
9 lines The SIGHUP handler was implemented to allow admins to
send SIGHUP to a running Asterisk process to reload the
configuration. However, doing the actual reload in the signal
handler itself is a very bad thing to do, because the reload
process includes calling non-reentrant functions such as
malloc/calloc/etc. If Asterisk is running in the background, then
the reload will happen immediately. However, if running in
console mode, the reload doesn't work until something is typed at
the console. That sort of defeats the purpose, but I don't see an
easy way to get around it at this point. ........
2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Drop the deprecated show commands since the
original ones were changed back. (issue #8937 reported by
PCadach)
2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_h323.c: Revert reprecation of h.323 gk cycle
command from pre-1.4 version instead of duplicated h323 cycle gk
* res/res_odbc.c: Don't play with free()'d pointers
* configure, acinclude.m4: Handle non-standard OpenH323/PWLib
library names
2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) |
5 lines Fix the extraction of the timestamp from video frames. It
was using the mapping for a mini-frame instead of a video-frame,
which caused it to get invalid data. (issue #8795, mihai)
........
2007-01-29 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com>
* apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan
2007) | 2 lines Now that filename is part of the structure and
since it comes before postprocess... we have to add it to our
postprocess line. (reported on asterisk-dev by Boris Bakchiev)
........
2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com>
* main/Makefile: Add a missing quotation mark. This was pointed out
by jcmoore on #asterisk-dev.
* main/manager.c: Remove a recursive lock of the manager session.
This was pointed out by zandbelt in issue #8711.
2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* pbx/pbx_config.c: Argument number correction
2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com>
* main/Makefile: ASTLDFLAGS needs to be passed to the editline
configure script as LDFLAGS. (issue #8928, zandbelt)
* main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF
mode translation. P2P bridging can only be used when the DTMF
modes don't match if the core is monitoring DTMF in both
directions. Then, the core will handle the translation.
Otherwise, this bridging method can not be used. (issue #8936)
* main/manager.c: The session lock can not be held while calling
action callbacks. If so, then when the WaitEvent callback gets
called, then no event can happen because the session can't be
locked by another thread. Also, the session needs to be locked in
the HTTP callback when it reads out the output string. This fixes
the deadlock reported in both 8711 and 8934. Regarding issue
8711, there still may be an issue. If there is a second action
requested before the processing of the first action is finished,
there could still be some corruption of the output string buffer
used to build the result. (issue #8711, #8934)
2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Use ast_calloc instead of malloc.
2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com>
* apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR
backport to 1.4). It was committed to trunk via 7663. But it
wasn't so much an enhancement as a fix for the bad language
output for portuguese in Brazil, so, after a lot of prodding from
patient Brazilians, here is the same fix for 1.4
2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Set quota information to 0 when creating a
vm_state. (issue #8924 reported by neutrino88)
2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com>
* main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in
the last commit to the adaptive jitterbuffer code. - Specifically
indicate to the compiler that the "dropem" variable only needs
one but. - Change formatting to conform to coding guidelines.
2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com>
* main/jitterbuf.c, include/jitterbuf.h: Fixed problem with
jitterbuf, whereas it would not complain about, and would allow
itself to be overfilled (per the max_jitterbuf parameter). Now it
rejects any data over and above that size, and complains about
it.
2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* configure, configure.ac: Suggested change to fix normal usage of
--with-tds=/usr/local (Sean Bright, via asterisk-dev mailing
list)
2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com>
* /, apps/app_queue.c: Merged revisions 52415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2
lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log
follow documentation. (issue #7677 reported by amilcar) ........
* main/manager.c: Have the manager interface send back an "Already
logged in" message instead of "Invalid/Unknown Command" when the
client authenticates for a second time. (issue #8509 reported by
pari)
* /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2
lines Make the last context entry read in the dominant one.
(issue #8918 reported by pj) ........
* main/file.c: Fix core show file formats CLI command.
2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com>
* /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2
lines Allow dequeueing of frames with negative timestamp by
moving jitterbuffer frames check to jb_next. (issue #8546
reported by harmen) ........
* channels/chan_sip.c: Drop out variables I accidentally put in.
* channels/chan_sip.c: Decrement onHold count if we are hung up on
and still on hold. (issue #8909 reported by alexh42)
* apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan
2007) | 2 lines Add another note about audio files being played
back to each bridged party. (issue #8718 reported by ppyy)
........
2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c, configs/users.conf.sample: By suggestion
from kpfleming last week, change "vmpassword" to "vmsecret".
* configure, configure.ac: Remove libnsl as a required lib for
libiksemel to work. This change was already made in the trunk.
(issue #8762)
* include/asterisk/dial.h: Fix the formatting of doxygen comments
to properly indicate that the comment documents the previous
entity, as opposed to the next one.
2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com>
* utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1
line updated check_expr via 8322 (refactoring of expression
checking impl); elfring contributed a nice code reorg, I
contributed some time to get it working again, better messages
........
2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com>
* main/dial.c (added), apps/app_page.c, main/Makefile,
include/asterisk/dial.h (added): Merge in dialing API and the
app_page that uses it. (issue #BE-118)
* channels/chan_sip.c: Fix changing channel formats when joint
capability changes and there are no audio formats... I didn't
break it originally! (issue #8535 reported by ivoc)
2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com>
* configure: rebuild configure script to reflect last chan_h323
related changes.
2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: added fix from #8899
* channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24
Jan 2007) | 1 line fixed the busy problem (dialstatus was not
busy when we called a busy extension) ........
2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Show capabilities *and* preference in
general settings in "sip show settings" (reported by Clona/Telio
- Thanks!)
2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz>
* acinclude.m4: Allow x64 builds of H.323 (please, rebuild
configure)
2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com>
* main/channel.c, /: Merged revisions 51843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) |
6 lines Fix an issue related to synchronization of recordings
when using Monitor(). The bug is a miscalculation of the amount
to seek the stream for writing to disk when the number of samples
coming in and out of a channel do not match up. (issue #8298,
#8887, report and patch by guillecabeza, patch files created and
testing done by whoiswes) ........
* apps/app_while.c, /: Merged revisions 51828 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) |
4 lines Don't set a new value for the END_ variable on the
channel before using the old value. If you do, it will lead to
accessing a memory address that has been free()'d. (issue #8895,
arkadia) ........
2007-01-23 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com>
* channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c,
channels/chan_sip.c, channels/chan_skinny.c,
channels/chan_features.c, channels/chan_alsa.c,
channels/chan_gtalk.c, channels/chan_iax2.c: Update channel
drivers to use module referencing so that unloading them while in
use will not result in crashes. (issue #8897 reported by junky)
2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com>
* main/manager.c: Fix some bugs in process_message(). The manager
session lock needs to be held when sending some sort of response,
or calling one of the manager action callbacks. This resolves an
issue where people using the GUI would get random crashes when
they start clicking around a lot. (issue #8711, reported and
debugged by zandbelt)
* main/http.c: Fix setting the default port of 8088 on 64-bit or
big-endian machines.
* main/manager.c: When traversing the list of manager actions, the
iterator needs to be initialized to the list head *after* locking
the list. Also, lock the actions list in one place it is being
accessed where it was not being done.
2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com>
* res/res_features.c: this mod from 8593 (dstchannel in cdr is
empty when transfer call).
* main/callerid.c: via 8748 (callerid.c loses name when returning
PRIVATE_NUMBER flag), the user suggested this mod, saying it
would allow 'WITHHELD' to appear in the name field, which would
be useful
2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c, /,
channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) |
6 lines * more additions to make the RESTART message work * added
fix for misdn_call to allow SETUPs with empty extensions,
replaced the strtok_r functions with strsep for that (inspired by
Sandro Cappellazzo, thanks) ........ r50506 | crichter |
2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get
L2 UP, the L1 is UP definitely too, so we set the L1 state up as
well. ........
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c: manually merged r49922 and r50335, because
of conflicts. this commint includes addition of the ISDN RESTART
Message
2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_h323.c, channels/Makefile: Do not abort Asterisk
startup if h323 configuration file not found (reported by
mithraen)
2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only change audio formats on the channel if
we have an audio format to change to. (issue #8535 reported by
ivoc)
* /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan
2007) | 2 lines Yield before reading from zaptel timing source
under Solaris so that other threads get a chance to do things.
(issue #7875 reported by bob) ........
2007-01-22 19:28 +0000 [r51409] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This fixes 8836, according to dnatural
2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com>
* apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan
2007) | 2 lines Move filestream creation to Mixmonitor loop. This
will prevent a blank file from being created if no frames ever
pass through to be recorded. (issue #7589 reported by
steve_mcneil) ........
2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com>
* configs/say.conf.sample: Fix Italian numeral support in say.conf
for "_[2-9]00" case. "2131" would've translated to something
along the lines of (pardon my..Italian {or lack thereof})
"duecentocentotrentuno", which makes no sense at all.
* configs/say.conf.sample: Fix German language support in say.conf
Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
einundzwanzig has the same format as zweiundzwanzig (as do all
other "_ZX" spoken numerals) Fix support for numbers in the
10,000,000 to 99,999,999 range. Add support for numbers in the
100,000,000 to 999,999,999 range.
2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com>
* apps/app_meetme.c: Remove an unused instance of an unnamed enum.
* apps/app_meetme.c: Remove another duplicated definition
* apps/app_meetme.c: Remove a variable that was declared twice.
* codecs/gsm/Makefile: Add a couple more processors that need
optimizations excluded. (issue #8637)
* channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk.
AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same
thing. So, a digit would have been interpreted incorrectly here.
Since the channel driver will always have the begin and end
callbacks called for a digit, only support the button-down and
button-up messages.
* .cleancount: Bump the cleancount since my last commit changed the
channel structure.
* channels/chan_oss.c, main/rtp.c, main/channel.c,
channels/chan_phone.c, channels/chan_misdn.c,
channels/chan_skinny.c, channels/chan_features.c,
channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c,
channels/chan_zap.c, channels/chan_local.c, main/frame.c,
channels/chan_sip.c, channels/chan_agent.c,
include/asterisk/channel.h, channels/chan_gtalk.c,
channels/chan_iax2.c: Merge the changes from the
/team/group/vldtmf_fixup branch. The main bug being addressed
here is a problem introduced when two SIP channels using SIP INFO
dtmf have their media directly bridged. So, when a DTMF END frame
comes into Asterisk from an incoming INFO message, Asterisk would
try to emulate a digit of some length by first sending a DTMF
BEGIN frame and sending a DTMF END later timed off of incoming
audio. However, since there was no audio coming in, the DTMF_END
was never generated. This caused DTMF based features to no longer
work. To fix this, the core now knows when a channel doesn't care
about DTMF BEGIN frames (such as a SIP channel sending INFO
dtmf). If this is the case, then Asterisk will not emulate a
digit of some length, and will instead just pass through the
single DTMF END event. Channel drivers also now get passed the
length of the digit to their digit_end callback. This improves
SIP INFO support even further by enabling us to put the real
digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the
frame and passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
* main/asterisk.c: Merged revisions 51300 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) |
4 lines Fix a memory leak on command line tab completion. The
container for the matches was freed, but the individual matches
themselves were not. (issue #8851, arkadia) ........
2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com>
* channels/chan_zap.c: chan_zap compiles without libpri after
committing 7877 patch
* channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007)
| 3 lines issue 7877: chan_zap module reload does not use
default/initialized values on subsequent loads. Reset
configuration variables to default values prior to parsing
configuration file. ........
2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com>
* /: block this patch since it is already here
2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c, main/channel.c, main/pbx.c,
funcs/func_strings.c, main/app.c: Add some more checks for
option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832,
patch(es) by tgrman
2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com>
* Makefile, configure, main/Makefile, acinclude.m4, makeopts.in:
Ensure that the locations given to the Asterisk configure script
for ncurses, curses, termcap, or tinfo are further passed along
to the editline configure script. This fixes some
cross-compilation environments. (issue #8637, reported by ovi,
patch by me)
2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18
Jan 2007) | 2 lines If a timezone is not specified, assume
localtime (instead of gmtime) (Issue #7748) ........
2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com>
* apps/app_speech_utils.c: Only start timeout once we reach the end
of the files to play back.
2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com>
* main/cli.c: Fix an issue with file name completion in "module
load" and "load". Issue 8846
2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Copy MOH settings when calling a peer so
that if they put someone on hold or get put on hold themselves
they get the right music class. (issue #8840 reported by mdu113)
2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com>
* main/channel.c: Fix an issue with deprecated commands
2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18
Jan 2007) | 2 lines Document all the fields, including the
indication that "uniqueid" should not be renamed. ........
2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com>
* main/manager.c: Make the "hasmanager" option in users.conf
actually have an effect. (issue #8740, LnxPrgr3)
2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Build the IMAP remote directory string
better and properly. Fix an issue with encoding the GSM voicemail
when attaching to the voicemail. (issue #8808 reported by
akohlsmith)
* main/rtp.c: Pass data as well for hold/unhold/vidupdate frames.
(issue #8840 reported by mdu113)
2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com>
* funcs/func_odbc.c: Fix some instances where when loading
func_odbc, a double-free could occur. Also, remove an unneeded
error message. If the failure condition is actually a memory
allocation failure, a log message will already be generated
automatically.
* channels/chan_zap.c: Instead of dividing the offset by 2
directly, make it more clear that the offset is being scaled by
the size of the elements in the buffer. (Inspired by a discussing
on the asterisk-dev list about this code)
* /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) |
3 lines Move the check for a failure of ast_channel_alloc() to
before locking the pvt structure again. Otherwise, on a failure,
this will cause a deadlock. ........
2007-01-17 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, main/utils.c: Merged revisions 51194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007)
| 4 lines When ast_strip_quoted was called with a zero-length
string, it would treat a NULL as if it were the quoting character
(and would thus return the string in memory immediately following
the passed-in string). ........
2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: re-add "password" for realtime voicemail
2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Return the correct result when directly writing out a
packet so that the core doesn't then decide to handle it the
regular way again. (issue #8833 reported by rcourtna)
2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_voicemail.c: a few more coding style cleanups and one
bug fix (from AnthonyL)
2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Move rescheduling of lagrq/pings into the
scheduler callback.
2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com>
* main/rtp.c: Fix issue with dtmf continuation packets when the
dtmf digit is 0... Issue 8831
* apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with
IMAP storage and realtime voicemail. Also update the vmdb sql
script for IMAP specific options. Issue 8819, initial patches by
bsmithurst (slightly modified by me)
* doc/voicemail_odbc_postgresql.txt: change documentation to
reflect new procedure in 1.4/trunk
2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions
51161 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007)
| 2 lines Add documentation walkthrough on getting Postgres to
work with voicemail (from Issue 8513) ........
* apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007)
| 2 lines Postgres driver doesn't like a NULL pointer when
retrieving the length (Bug 8513) ........
2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c: minor things i missed before i get jumped
on
2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com>
* /, res/res_features.c: Merged revisions 51145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2
lines Return previous behavior. ParkedCalls will be able to do
DTMF based transfers again. trunk however will get an option to
allow this to be set on/off. (issue #8804 reported by nortex)
........
2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com>
* main/file.c: Display more useful output when streaming files.
Include the channel name to which the file is being played. Issue
8828, patch by junky.
2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com>
* channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2
lines Add none as a valid callgroup/pickupgroup option. I
consider it a bug that it would inherit it all the way down and
not have any way to reset it to nothing - so that's why it is in
1.2. (issue #8296 reported by gkloepfer) ........
2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com>
* main/config.c: It is possible for the config pointer to be NULL
here, so it needs to be checked before dereferencing it.
2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c, configs/users.conf.sample: Patch allows for
changing voicemail password in users.conf from voicemail main,
written by AnthonyL bug #8436
2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com>
* Makefile.rules: Filter out a few CFLAGS that are not valid
CXXFLAGS.
2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946
| mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4
lines Solves issue with forwarding voicemails from folders other
than inbox. patch by anthonyl. ........
2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com>
* main/asterisk.c: re-add deprecated "show version" CLI command.
2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com>
* main/manager.c: Move event processing into do_message so that it
gets executed again when events are tripped.
2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com>
* configure, include/asterisk/autoconfig.h.in, main/Makefile,
configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the
ACX_PTHREAD macro from the Autoconf macro archive for setting up
compiler pthreads support... should improve portability to
platforms with unusual pthreads requirements
2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com>
* main/astmm.c: Add missing newlines for two memory CLI commands.
2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c,
main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c,
main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c,
main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c,
main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c,
main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c,
main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c,
main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c,
main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c,
main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h,
main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c,
main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c,
main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c,
main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c,
main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c,
main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13
Jan 2007) | 2 lines Bug 8814 - db should look for its header
using a relative path, instead of the system path (Fixes FreeWRT)
........
2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, build_tools/make_sample_voicemail (added): when
building the sample greetings for maibox 1234@default during
'make samples', build a greeting for each language and file
format the user selected to install with menuselect (reported by
Brian Capouch on asterisk-dev)
2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com>
* main/channel.c: Only write a frame out to the channel if one
exists. There are cases where one may not and would therefore
cause the channel driver to segfault. (issue #8434 reported by
slimey)
* res/res_snmp.c: Only join the snmp thread on an unload if the
thread is actually running. (issue #8810 reported by junky)
2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com>
* configs/voicemail.conf.sample: Update documentation to state that
you shouldn't use realtime static with voicemail.conf
2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com>
* main/manager.c: We need to check for res being 0 in do_message
itself, otherwise our headers will get lost.
2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com>
* main/pbx.c, /: Merged revisions 50561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007)
| 2 lines minor documentation clarification ........
2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Remove check for channel state as it can
definitely be something other then ring, and also clean up the
code a bit. This should solve the parking issues and maybe some
attended transfer issues people have been seeing.
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add
support to see whether NAT was detected (yay symmetric RTP) and
also add a check in chan_sip so that if NAT has been detected and
the reinvite behind nat option has been turned off, then just do
partial bridge. (issue #8655 reported by mnicholson)
* apps/app_speech_utils.c: Merge speech-multi branch which adds
support for joining multiple sound files together to be played
one after another in SpeechBackground.
* main/config.c: Fix parsing when using something like ldap
settings. (done by anthonyl)
* channels/chan_sip.c: Fix chan_sip not working issue. Let's not
prematurely return 0. (issue #8783 reported by st41ker)
2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com>
* cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made
it fail to load if the config file existed. Issue 8777
2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 50295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2
lines Add another return value to dial_exec_full that indicates
execution is going to continuing at a new
extension/context/priority and to just let it slide. (issue #8598
reported by jon) ........
* main/pbx.c: Ensure data's existence before trying to access it.
(issue #8774 reported by rcourtna)
2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com>
* Makefile, /: Merged revisions 50227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) |
6 lines Make the number that represents the major version number
a single digit instead of 2. Using two digits makes it an octal
number when put into version.h, which breaks the compilation of
any out of tree module that checks the version for any version
after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev
mailing list, who gave credit to vihai for pointing it out)
........
2007-01-09 17:11 +0000 [r50186] Jason Parker <jparker@digium.com>
* main/cli.c: Re-add CLI command that should have only been
deprecated in 1.4. Thanks kshumard! (reported in person, so no
associated issue #)
2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007)
| 4 lines The advent of realtime has enabled people to use commas
in the fullname field. This could cause an issue with sending
voicemails, when the field is unquoted. (Issue 8595) ........
2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - handle re-invites properly in sip_hangup()
- Add some invitestate status changes just to be sure
2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Fix an issue with voicemail and users.conf,
where it wouldn't ever parse a password, since it was using
"secret" instead of "password" Issue 8761, reported by and patch
suggestion from ssokol.
2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com>
* apps/app_senddtmf.c: we can't unlock a channel if we cant find
it. - AnthonyL bug #8741
2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Disable the more intense packet2packet bridging until
the bugs can be worked out.
2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue #8677 - Handle failure of T.38
re-invite This is not a fix, but adding an error message to tell
the admin that we have a bad configuration. We should not send
T.38 re-invites to devices that can't handle it (with the current
architecture where you have to hard-code t.38 support per
device). To really fix this, we need to figure out a way to tell
the incoming call that the re-invite failed, so we can signal
failure on that end and go back to the original call.
* channels/chan_sip.c: Issue #8524, support multiple via header
values (tardieu) Thanks!
* channels/chan_sip.c: We only need one forward declaration
* channels/chan_sip.c: Issue 8735: Terminate state when extension
is unavailable for subscription
2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com>
* /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2
lines Ensure we use the default refresh value of 60 if the remote
server does not send one. (issue #8746 reported by maethor)
........
2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac: since we use AC_PATH_TOOL to find tools,
we should use the results it provides for us (reported by Brian
Capouch on the asterisk-dev list)
2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007)
| 2 lines If openstream fails, then we crash (Issue 8564)
........
* channels/chan_sip.c: Second condition was a subset of the first,
so hold was never decremented, thus hint stayed stuck (Issue
8747)
2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com>
* main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping
byte of allocated memory! This looks like it may have been a
chicken/egg scenario.. You had to call a cleanup func, because
everything was allocated. Then since you had to call a cleanup
func, you were forced to allocate - ie; strdup("").
2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com>
* configure, acinclude.m4: one more time...
* configure, acinclude.m4: proper fix for r49712
* configure, acinclude.m4: if --with-foo=<path> is specific for a
configure option, ensure that it is used for header file checking
as well
* main/manager.c: ast_func_read() needs a writable copy of the
function name to be passed
2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com>
* channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and
chan_zap also depend on zaptel. This fixes an issue (8727) with
zaptel being in a different directory, using --with-zaptel.
2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com>
* main/manager.c: don't 'consume' the params list before we try to
use it again
* res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c,
main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c,
main/db.c, channels/chan_zap.c, channels/chan_sip.c,
apps/app_meetme.c, res/res_features.c, channels/chan_agent.c,
utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c,
apps/app_queue.c, res/res_jabber.c: reduce stack consumption for
AMI and AMI/HTTP requests by nearly 20K in most cases
2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com>
* main/channel.c: Don't keep repeating the warning over and over
when the end of the call is reached. (issue #8724 reported by
xrg)
2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com>
* /, channels/chan_sip.c, channels/chan_skinny.c,
channels/chan_iax2.c: Merged revisions 49635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007)
| 2 lines ensure that threads which are supposed to be detached
(because we aren't going to wait on them) are created properly
........
* channels/chan_iax2.c: revert the dynamic_list insertion change...
that was not the right thing to do
* channels/chan_iax2.c: create the IAX2 processing threads as
background threads so they will use smaller stacks when we create
a dynamic thread, put it on the dynamic_list right away so we
don't lose track of it
2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: It's possible for the iax2 pvt to
disappear, so if it has... don't bother looking for dpentries.
2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/threadstorage.h, main/asterisk.c,
build_tools/cflags.xml, include/asterisk.h, main/Makefile,
main/threadstorage.c (added), main/utils.c: add support for
tracking thread-local-storage objects that exist via
'threadstorage' CLI commands
2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com>
* main/config.c: Only free comments and line buffer once we reach
the first level. (issue #8678 reported by ssokol, fixed by
anthonyl)
2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com>
* channels/iax2-parser.c, main/frame.c: don't mark these
allocations as 'cache' allocations when caching has been disabled
* channels/iax2-parser.c: if we're going to decrement the frame
count when we free a frame, we should inrement it when we create
one :-)
* channels/iax2-parser.c, channels/iax2-parser.h,
channels/chan_iax2.c: only do IAX2 frame caching for voice and
video frames
* main/frame.c: don't do frame header caching in the core if
LOW_MEMORY is defined
* channels/iax2-parser.c: don't define this type either if
LOW_MEMORY is enabled
2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447
| mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2
lines converted a lot of 256 to PATH_MAX and some white space
fixes. ........
2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com>
* channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode
* codecs/Makefile: make building of codec_gsm against the system
GSM library actually work
2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412
| mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3
lines good catch russell sorry i missed that. fix magic number
with proper sizeof ........
2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com>
* funcs/func_realtime.c: Fix the REALTIME() dialplan function.
ast_build_string() advances the string pointer to the position to
begin the next write into the buffer. So, this pointer can not be
used to copy the contents of the string later. The beginning of
the buffer must be saved. Interestingly enough, this code could
not have ever worked. (Pointed out by Sebb on IRC, thanks!)
2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354
| mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6
lines When using ODBC_STORAGE VoicemailMain doesn't create the
subdirectories for a mailbox such as the INBOX directory. this
patch solves that problem, was written by anthony be-125 ........
2007-01-03 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c,
/, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c,
configs/misdn.conf.sample: Merged revisions
48319,48321,48467,48552,48576,49135,49303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) |
1 line changed a few debugs to higher debug levels ........
r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) |
1 line added the export and import of the MISDN_ADDRESS_COMPLETE
Variable to inidcate wether the extension is already completely
dialed or if there might come additional digits by information
elements. also added some docs for that. ........ r48467 |
crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict
when we create a setup while the other site creates a setup on
the same channel, besides the check we resolve this conflict.
........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18
Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a
preselected channel we just accept it, even when we're NT. added
some checks for segfaults. ........ r48576 | crichter |
2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we
reject a channel, because it's in use already, we shouldn't
process the setup anymore. made the channel allocation a bit
easier and more understandable, removed a few unused lines
........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02
Jan 2007) | 1 line added check for channel ranges in the
set/empty channel functions. set pmp_l1_check default to no.
added misdn restart pid cli command. added cleaning of channel
when we send a RELEASE_COMPLETE. ........ r49303 | crichter |
2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added
check for bridging in misdn_call to avoid setting
echocancellation when 2 mISDN channels are involved and when
bridging is set. That lead to a kernel panic before under
different situations, because we switched about 2 times between
hardware bridging and echocancelation * readded MISDN_URATE
variable which got lost before, this should make app_v110 work
again * fixed typo ........
2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, Makefile.rules: various Makefile improvements to get
chan_vpb (and any other C++ modules) to build properly
2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Check pvt structure presence before passing
to send_command. This gets rid of the irritating message about a
packet without pvt structure. This happens because the scheduled
item is getting cancelled at almost the exact moment it is
getting executed.
2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c,
pbx/ael/ael.flex: This is a slight modification to Josh's edits
for #8579; both files edited were the produced by flex; so the
source files need to be changed instead, and the generated files
regenerated.
2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Small cleanup of add_t38sdp - it's always
enabled at that point in the code
2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com>
* main/pbx.c: Allow fractions of a second in the Wait()
application, like it says it allows.
2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c: remove comment that is unrelated to this
function
2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net>
* configs/features.conf.sample: Adding note on effect of
applicationmap features on re-invites
2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_zap.c, build_tools/menuselect-deps.in, configure,
configure.ac, codecs/codec_zap.c: check specifically for VLDTMF
and transcoding support in the system's Zaptel installation, and
make only the modules that need those features dependent on them
(this will allow building the other Zaptel-using parts of
Asterisk against older versions of Zaptel or those on other
platforms that haven't caught up yet to the Linux version)
* Makefile: use a simpler (and portable) method to ensure that
menuselect is built as a host binary
* Makefile: revert this change until a better solution can be
found... 'env -i' was not being used properly, but even when
changed to do so, this process fails during cross-compilation
because the menuselect build still sees 'CC' as set to the
cross-compiler
2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: remove incomplete implementation of dnsmgr.
Let's fix this in trunk.
2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com>
* pbx/pbx_config.c: IAX has been deprecated for quite some time so
we had better use IAX2 when creating the dial string for users.
(issue #8697 reported by ssokol)
* channels/chan_zap.c: Use asprintf to build the channel names
instead of custom function. I believe the custom function is
doing some things that are not portable across all
implementations. (issue #8570 reported by hterag & issue #8692
reported by nicolasg)
* main/rtp.c: If the Packet2Packet bridge is being broken because
of a masquerade then attempt to read a frame in so the masquerade
actually happens. Otherwise weirdness will occur. (issue #8696
reported by kjotte)
* channels/chan_iax2.c: Initialize the packet queue in load_module
instead of just declaring the list with the default value. (issue
#8695 reported by ssokol)
2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have
comma args converted to vertical bars. I hope this change does
little harm.
2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com>
* /: put this value into the correct property
* /, BUGS: Merged revisions 49045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006)
| 2 lines location of the bug posting guidelines has changed
........
* sample.call: simple commit to test CIA integration
2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com>
* main/cli.c: Fix some deprecated commands. Issue 8682, patch by me
* main/http.c: saw this in passing... fix a small typo
2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: new versions of sounds
2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com>
* main/http.c: make the uris_lock a rwlock instead of a mutex lock
- needs to be forward ported to trunk
2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
include/asterisk/lock.h: Backport support for read/write locks.
2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com>
* main/ast_expr2.fl, main/ast_expr2.c, main/frame.c,
pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c,
pbx/ael/ael_lex.c, include/asterisk/ael_structs.h,
pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk
from the ael stuff. Also, threw in a minor fix to frame.c to
avoid build-killing compiler warnings.
2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com>
* main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not
available when LOW_MEMORY is used and things are being built in
the utils directory, so we need to resort to the old method of
strncpy. (issue #8579 reported by mottano)
2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com>
* main/enum.c, main/asterisk.c, main/rtp.c, main/term.c,
main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c,
main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c,
main/http.c, main/logger.c: since these variables all have static
duration, none of them need initializers (they default to zero
anyway)
* include/asterisk/options.h, main/asterisk.c, main/file.c: move
extern declaration for this option to a header file where it
belongs provide an initial value for 'languageprefix' option,
instead of relying on randomness to provide a useful value
2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Only include acl.h and lock.h once
* channels/chan_sip.c: Only set rfc2833compensate flag once
(handle_common_options)
* channels/chan_sip.c: - Remove checking for T38 options twice.
Keeping them in handle_common_options
2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: make the option actually match the
documentation
* channels/iax2-parser.c, include/asterisk/utils.h,
include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show
memory' and 'show memory summary' to distinguish memory
allocations that were done for caching purposes, so they don't
look like memory leaks
2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: Be a bit more
politically correct
* channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy
cisco MWI support. Normally we try not to change our software for
bugs in other devices. But in this case, the Cisco phones are so
widespread so we try to implement a fix while waiting for a
bugfix from Cisco.
* channels/chan_sip.c: - Make sure handle_common_options return 1
when we found a common option - Move uncommon (only global)
option away from handle_common_options Reported by rizzo. Thanks!
* channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before
re-sending invite with auth.
* /, channels/chan_sip.c: Fix bogus content-length in t38 sdp.
(rizzo, #8600)
2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Get rid of a needless memory allocation and
only create a conference structure in find_conf_realtime if data
was read from realtime. (issue #8669 reported by robl)
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an
API call that initializes an RTP structure. We need this because
chan_sip is cheeky and uses a temporary RTP structure for codec
purposes, and the API calls that are used rely on the lock.
(Pointed out on asterisk-dev by Andy Wang)
* configure, configure.ac: Clean up autoconf file (gets rid of
warnings seen when rebuilding configure) and rebuild configure.
2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com>
* /, funcs/func_math.c: Merged revisions 48955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) |
6 lines Fix an error introduced by copying and pasting the
handling of the >= operator for the MATH function. If a single
equal sign was used as an operator, the function would treat it
is as if it were the >= operator. Now, it properly handles it as
an invalid operator. (issue #8665, patch by tempest1) ........
* channels/chan_oss.c: Fix a typo in an error message that
indicated that the MGCP channel type could not be registered,
instead of the correct type, OSS.
* /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) |
3 lines Check for the proper return value on an error in a call
to mmap(). This was reported by Andy Wang on the asterisk-dev
list. Thanks! ........
* /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) |
3 lines Remove a couple of misplaced dots in log messages. This
was reported by Andrea Spadaccini on the asterisk-dev mailing
list. ........
* main/http.c: Implement locking for the list of URI handlers to
make it thread-safe.
2006-12-23 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0 released.
2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com>
* Makefile, main/stdtime/localtime.c: Minor fixes for Solaris.
* channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia
2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com>
* /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2
lines Add new silence sound files to the spec for Redhat. (issue
#8652 reported by alvaro_palma_aste) ........
2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: vms doesn't exist on non-IMAP storage
builds.
* apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so
it is then passed to the IMAP store file function. (issue #8614
reported by punknow)
* doc/snmp.txt: find is not the same as bind when it comes to
documentation. (issue #8626 reported by johann8384)
2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com>
* channels/Makefile: suppress compiler warnings in this module
until it can be improved
2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, /: Merged revisions 48584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2
lines Free localuser structure when we fail to dial (issue #8612
reported by rizzo) ........
2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org>
* apps/app_sms.c: fix a bogus datalen in the frames generated by
app_sms (causing noisy output if you listen to the output!) This
affects trunk as well, whereas 1.2 is ok.
2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com>
* res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable
type for these unixODBC API calls, eliminating warnings on 64-bit
platforms that use the 'new' 64-bit types for ODBC API calls
2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com>
* Makefile: Use env -i to start a fresh environment when going to
build menuselect. This is more portable then using unset. (issue
#8543 reported by jtodd)
2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org>
* include/asterisk/channel.h: unbreak the macro used for
incrementing the frame counters. I don't know when the bug was
introduced, but with the typical usage c->fin =
FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects
trunk as well (fix coming).
2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Put thread into proper list if we abort
handling due to an error, and also hold the lock while putting it
back into the proper idle list so we don't prematurely get a
signal. (issue #8604 reported by arkadia)
2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com>
* codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile,
utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile,
utils/ael_main.c: remove some now-unnecessary explicit includes
of autoconfig.h clean up per-file dependencies during 'make
clean'
* build_tools/prep_tarball: need an additional argument here to
make the downloads actually occur
* configure, include/asterisk/autoconfig.h.in, configure.ac,
acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep
these calls from thinking they have multiple arguments
* codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile,
funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast,
main, codecs/gsm, pbx, res, channels, codecs, utils, agi,
main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr:
simplify dependency tracking system, using the compiler's
built-in method for generating them, and only doing dependency
tracking if developer mode is enabled via the configure script
* Makefile, include/asterisk.h, main/stdtime/localtime.c: since we
really, really have to have autoconfig.h included before all
other headers (especially system headers), the Makefile will now
force it to happen (this will fix build problems with files like
ast_expr2f.c, where we can't control the inclusion order in the
file itself)
* funcs/func_curl.c: instead of initializing the curl library every
time the CURL() function is invoked, do it only once per thread
(this allows multiple calls to CURL() in the dialplan for a
channel to run much more quickly, and also to re-use connections
to the server) (thanks to JerJer for frequently complaining about
this performance problem)
2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Turn payload_lock into bridge_lock and make it
encompass all RTP structure contents that may relate to bridge
information, including who we are bridged to.
* channels/chan_iax2.c: Hold call structure lock in places where a
qualify or peer action can destroy it.
* channels/chan_iax2.c: Lock network retransmission queue in all
places that it is used.
2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported
from 1.2)
* channels/chan_sip.c: Update to latest IANA spec
2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Use a wakeup variable so that we don't wait
on IO indefinitely if packets need to be retransmitted.
* main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP
structure can change AFTER a bridge has started. This comes from
the packet handling of the SIP response when indication that it
was answered has been sent. Therefore we need to protect this
data with a lock when we read/write. (issue #8232 reported by
tgrman)
* main/rtp.c: Remove direct RTCP bridging. I've come to the
conclusion that we should handle this through the core and not
just forward it on. Should solve a few bugs.
2006-12-12 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta4 released.
2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This
is the way it should have been done.
2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com>
* sounds/Makefile: new sounds package with 100% more silence
* /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge
from https://svn.digium.com/svn/asterisk/branches/1.2 ........
r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006)
| 4 lines app_externalivr needs a real silence file, and
additional changes to add silence files into core instead of
extra patch provided by bug 8177 with minor additions. ........
2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Return non-existant callerid handling to
that which it was before. In 1.4 and trunk callerid can be
allocated but not have any contents so we have to use
ast_strlen_zero before passing it to the relevant functions.
(issue #8567 reported by pabelanger)
2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_strings.c: STRFTIME() does not actually require an
argument (issue 8540)
2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Merge in my latest RTP changes. Break out RTP and
RTCP callback functions so they no longer share a common one.
* apps/app_meetme.c: Use the correct API call to say a device state
changed. (Yes, I'm a nub.)
* apps/app_meetme.c: Don't access the conference structure after it
has been freed.
2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c,
res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c,
apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006)
| 5 lines When doing a fork() and exec(), two problems existed
(Issue 8086): 1) Ignored signals stayed ignored after the exec().
2) Signals could possibly fire between the fork() and exec(),
causing Asterisk signal handlers within the child to execute,
which caused nasty race conditions. ........
2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com>
* channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1
line This version applies the patch suggested by stevens in bug
7836 (make inbound channel RINGING state consistent with other
channels). ........
2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com>
* channels/chan_iax2.c: Use locking when accessing the
registrations list. This list is not actually used very often, so
the likelihood of there being a problem is pretty small, but
still possible. For example, if the CLI command to list the
registrations was called at the same time that a reload was
occurring and the registrations list was getting destroyed and
rebuilt, a crash could occur. In passing, go ahead and convert
this list to use the linked list macros.
2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07
Dec 2006) | 3 lines Ensure that the file position is not
incremented beyond the total number of files available for
playback. (issue #8539, ulogic) ........
2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com>
* main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that
killed bug 8423 -- OriginateSuccess and OriginateError incomplete
channel name. May it rest in peace.
2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being
retransmitted to Asterisk
2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com>
* configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06
Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option
in the sample configuration file. (issue #8526, arkadia) ........
2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Don't send Contact on MESSAGE
2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com>
* configure.ac: Fix curl version number testing to be much more
friendly to non-bash shells. Issue 8508, patch by me. This
*SHOULD* be POSIX compliant now..
2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Merging the invitestate-1.4 branch after
successful testing. Will check if I can solve this with less
changes in 1.2.
* configs/sip.conf.sample: Add missing s from another repository.
(thanks jcmoore!)
* configs/sip.conf.sample: Updating sip.conf.sample with
information about T38 not working when chan_local or chan_agent
is involved in the call. I don't know how big a fix that would be
to solve, but this is the current state of affairs. (Chan_sip
currently checks if the other side of the bridge has a SIP tech.
We could/should implement another check, possibly for udptl_write
or some flag in the ast_channel structure).
2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Oops, forgot to release the odbc handle
* apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006)
| 6 lines If the recording in the database is too large, it will
fail to retrieve with an mmap error. Not too sure why this
doesn't happen when we put it in the database, also, but since
that doesn't seem to be broken, I'm not going to fix it (at least
until someone reports it). Solution is to ask for the file in
smaller chunks. (Bug 8385) ........
2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com>
* apps/app_voicemail.c: Fix an issue which didn't allow
unavail/greet/busy/etc messages from being saved into ODBC (and
probably IMAP).
2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com>
* configs/voicemail.conf.sample: Add documentation to
voicemail.conf.sample for ODBC storage. Issue 8499 - patch by
blitzrage.
* doc/snmp.txt: Attempt to document some of the dependencies that
are needed for net-snmp Issue 8499 - initial patch by blitzrage.
2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com>
* sounds/Makefile: When "fetch" is in use, instead of "wget",
--continue is not a valid option. (issue #8451)
2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Removing one of two pieces of code to
handle 481 response on INVITE - Move handling of REFER response
to handle_response_refer()
* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax
transmission happens - Encapsulate RTP timers in the rtp
structure so we have one for video and one for audio The video
one is not used in 1.4, really. Will be used for RTP keepalives
when we can send something that video phones support in the RTP
stream. I now this is a big architectual change at this stage for
1.4, but decided it was needed to avoid future bug reports. -
Document the RTP NAT keepalive option in sip.conf.sample Issue
7679 in the bug tracker. Please test.
2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com>
* include/asterisk/utils.h: Backport the comment containing the
warning regarding the limitations on the usage of this function.
It is thread safe, but not technically reentrant.
2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_dial.c, /: Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006)
| 2 lines if Dial() is going to send music-on-hold to the calling
party, it has to send PROGRESS first to ensure that the reverse
audio path has been setup first (BE-106) ........
2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com>
* Makefile, configure, configure.ac, makeopts.in, sounds/Makefile:
FreeBSD 6.1 does not include wget by default. However, it has
fetch which will work just fine for our purposes of downloading
the sounds packages. So, check for both wget and fetch and the
configure script and use what was found to download them. If
neither one was found, and sound packages are selected that must
be downloaded, the install process will print out an informative
error message indicating the situation. Also, fix a couple places
where "make" was hard coded into some output messages by
replacing them with the $(MAKE) variable. (issue #8451, initial
patch by pabelanger, with additional modifications by me)
2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com>
* configs/extensions.conf.sample, /: Merged revisions 48183 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2
lines Fix a small typo - issue 8848, reported by pabelanger
........
2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/cli.c: Double-unlock error (reported by blitzrage on IRC)
2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: - Backport of the
"limitonpeers" patch from trunk, to fix a lot of issues with
queues and SIP device states - Remove support for T.38 early
media, since it's impossible. (Two patches in one - extra friday
evening offer due to being off line from svn today... :-)
2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com>
* main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not
do a partial bridge for Google Talk since we need to handle STUN.
(issue #8448 reported by phsultan)
2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue 8319 - change noncecount before
using it.
2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com>
* /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2
lines Only print out debug message if bridged channel is not
NULL. (issue #8412 reported by jubilex) ........
* /, res/res_features.c: Merged revisions 48154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2
lines Do not listen for DTMF on the bridge that comes into
existence when ParkedCall is executed. This means native bridging
can now occur for this. (issue #8406 reported by kebl0155)
........
* main/cdr.c, /: Merged revisions 48151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2
lines Print certain CDR messages out at the NOTICE level versus
WARNING since they can occur when used with the CDR applications
and are perfectly fine. (issue #8367 reported by dartvader)
........
* /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov
2006) | 2 lines Document 'port' for SIP peers, came up because of
the current mailing list thread. (issue #8450 reported by
blitzrage) ........
2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net>
* doc/manager.txt: Explain status reports and make codefreeze more
happy :-)
* /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by
GS 487 adapter without CSEQ on separate line in the REGISTER
request. Imported from 1.2.
2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in
mm_login. (issue #8420 reported by slimey)
2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Explain the use device status system
implemented in SIP for subscriptions, queues and manager a bit
better. Like in 1.2, you will get more detailed information if
you set a call limit for a device. When the call limit is
reached, the status system will report a device as busy. For
queues, setting a call limit per SIP device is propably a
requirement. In most cases, it will work much better if you only
use type=peer and not type=friend. We might decide to backport
the new setting from trunk to apply all call limits to the peer
part of a friend only.
2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com>
* main/rtp.c, /: Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2
lines If the frame was duplicated before writing out then we need
to free it. (issue #8429 reported by edguy3) ........
2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma.
2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c: Don't crash if the mailstream was not
created.
2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com>
* Makefile: Export several more variables in top level Makefile.
Inspired by issue 8438.
2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com>
* channels/chan_phone.c, /: Merged revisions 48087 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov
2006) | 2 lines According to the research I have done we never
needed to include compiler.h in the first place so let's not!
(issue #8430 reported by edguy3) ........
* apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2
lines Use the proper function to get the new message count
instead of always using the filesystem. (issue #8421 reported by
slimey) ........
2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27
Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381)
........
2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com>
* main/manager.c: Remove a couple of unused variables (issue #8380,
casper)
2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com>
* pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2
lines Do not reference the freed outgoing structure in the debug
message. (issue #8425 reported by arkadia) ........
2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Change logging message
2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com>
* funcs/func_cdr.c: might as well also document the raw values of
the flag vars
* /, funcs/func_cdr.c: A little bit of func_cdr documentation
upgrade-- no bug# involved, although 8221 may have inspired it.
2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4
and future releases, you can disable subscription support totally
or per peer in sip.conf with allowsubscribe = yes | no
2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com>
* main/translate.c: bug 8189 posted this fix for main/translate.c
for PLC
2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23
Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE.
beatufied some logs, changed some loglevels. changed the default
value of block_on_alarm ........
2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Don't allocate unused variable.
2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Video will never reach Packet2Packet bridging and can
do more harm then good.
2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com>
* main/rtp.c: If we have the non standard G726-32 setting turned on
we want to return G726-32 to the SDP, not our AAL2 string. (issue
#8330 reported by voipgate)
2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Apparently Exosip sends a 101 after a 100
provisional response. Let's not treat that as early media.
(discovered at the AVTF meeting in Paris).
2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Oops, merge missed release of odbc object
* apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006)
| 2 lines Failing to trap -1 error from mmap causes segfault
(Issue 8385) ........
2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com>
* main/frame.c, /: Merged revisions 47859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2
lines Don't forget to byte swap if we are exiting the smoother
feed early. (issue #8287 reported by arturs) ........
2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com>
* /, doc/billing.txt: update documentation regarding IAX2 transfers
and CDRs Merged revisions 47776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006)
| 2 lines update clearly wrong documentation regarding cdr_custom
........
2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Compare technology using the pointers
instead of a straight comparison based on name. (issue #8228
reported by dean bath)
2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com>
* configure, configure.ac: check for pre-1.4 versions of Zaptel and
abort the configure script if found with an appropriate error
message
2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD
notification optional, in order to avoid a lot of extra database
lookups for all those realtime users out there.
2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 47750 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov
2006) | 2 lines Because of the way chan_local is written we
should be extra careful and make sure our callback functions have
a tech_pvt. (issue #8275 reported by mflorell) ........
* apps/app_meetme.c: Don't unreference the SLA object if there is
no SLA object in the devicestate callback. (issue #8354 reported
by loloski)
2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Don't fixup if there's nothing to fixup
* UPGRADE.txt: Warn users about change in canreinvite
* channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never
authenticated (according to the RFC) - Update docs on
canreinvite. "nonat" is the recommended setting for most users
with phones behind a NAT.
2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c, /: Merged revisions 47711 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov
2006) | 2 lines Make sure that the pvt structure exists before
trying to do fixup on Local channels. (issue #7937 reported by
mada123, fix by alamantia with mods by me) ........
2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL
2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com>
* main/channel.c: We need to ensure timelimit stuff is included as
well so warnings get played. (issue #8050 reported by KNK)
2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com>
* main/file.c: don't try to call fclose() if fopen() failed
2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: - Improve SIP history - Never send reply to
ACK (again...)
2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006)
| 4 lines ensure that message duration is included in email
notifications for forwarded messages (BE-96, fix by me after
corydon used his clue-bat on me) ensure that duration in the
message metadata is updated if prepending is done during
forwarding (related to BE-96) remove prototype for API call that
does not exist ........
* main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15
Nov 2006) | 2 lines clear the category's variable tail pointer as
well when variables are detached from it ........ r47688 |
kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2
lines when appending a list of variable to a category, ensure the
tail pointer points to the last variable in the list ........
r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006)
| 2 lines when re-writing the config file, don't repeat the path
if it hasn't changed ........
* main/config.c, /: Merged revisions 47682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006)
| 2 lines ouch... don't use printf, use ast_log/ast_verbose
........
2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org>
* main/cli.c: fix longest match search in find_cli. Trunk already
fixed. 1.2 not affected (well, i have no idea, the code is
totally different there).
2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Send error message when we can't allocate
SIP dialog, possibly due to limitation of file descriptors.
(imported from 1.2)
2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com>
* main/rtp.c: If NAT detection is turned on or already detected
then say NAT is active when setting the remote RTP peer when
doing early bridging. (issue #8365 reported by marcelbarbulescu)
2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com>
* main/term.c: more formatting cleanup, and avoid running off the
end of the string
2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Turn notice about unknown RTCP packet type into a
debug message instead.
2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com>
* channels/misdn/isdn_lib.c: silence compiler warning on 64-bit
platforms (this variable is an 'int' anyway, comparing it to
'signed long' is not useful)
2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com>
* apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2
lines Update copyright information in the ADSI logo blob.
........
* channels/chan_sip.c: Only keep the video RTP structure around if
1. Video support is enabled and 2. A video codec is enabled on
the dialog
* funcs/func_uri.c: Small documentation clarification for
URIENCODE. (issue #8294 reported by salaud)
2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Conversion of res_odbc API to include ast_
prefix did not completely transition app_voicemail when
ODBC_STORAGE is used (reported on IRC by caio1982, not in
bugtracker)
2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com>
* apps/app_amd.c: Use LOG_DEBUG to print out the indication that
app_amd is using default settings instead of using LOG_NOTICE.
This stops needless logging of this information under normal
circumstances. (issue #8361 reported by Seb7)
2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Update documentation to fit the
implementation...
* /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in
retransmission system if it's an OPTION packet from peerpoke
2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com>
* /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2
lines Initialize global pointers for connection and result to
NULL. (issue #8356 reported by james) ........
2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006)
| 2 lines Having more than 255 old messages caused corruption in
the new/old count ........
2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com>
* main/config.c: This solves bug 8342, whereby a crash occurs under
certain circumstances while reading a config file with comments--
a call to CB_ADD shouldn't happen if withcomments is zero
2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/cli.c, channels/chan_sip.c: Re-enable old deprecated
commands
2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: - Don't reply to INVITE already replied
to when we get BYE - Declare errmsg as int. Oops.
2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing
the messed if, but we all forgot to update the regressions. Until
now.
2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: AEL need not complain about parkedcalls not being
found... just confuses users
2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com>
* /, apps/app_sms.c: Merged revisions 47549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2
lines When sending an SMS with a user data header properly set
the UDH flag in the first byte. (issue #8347 reported by
hoffmeis) ........
* main/cli.c: Free full command string upon unregistering of CLI
command. Backported from revision 47536 from rizzo.
2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Only produce error message about sip history
once
2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com>
* configure, acinclude.m4: AC_PROG_SED is included in autoconf
2.60, but apparently it is not included in 2.59. So, to maintain
compatability with 2.59 since it is a small change, copy this
macro into acinclude.m4 and rename it to AST_PROG_SED. (issue
#8345)
2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c, /: Merged revisions 47525 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006)
| 2 lines If the execute fails a second time, make sure that we
don't pass back a stale handle ........
* channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006)
| 2 lines Don't play dialtone if the seizing the channel fails
(Bug 7754) ........
2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks
DEA!!!)
* channels/chan_sip.c: Part of issue 8078 - parse even if udptl is
UDPTL in sdp...
* channels/chan_sip.c: - Don't destroy SIP dialog because of a
failed T.38 re-invite. Wait for a bye. Final response to a
re-invite does not mean that the session dies, only that the
re-invite fails. - Keep RTP active during processing of T.38
re-invite. If the re-invite fails, RTP needs to remain as before
the re-invite. Issue 8338 - darren1713. Please test.
* channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp
-Add some comments to t.38 code
2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com>
* /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) |
4 lines Only do the check to determine whether the channel
calling this function is an IAX2 channel when getting the IP
address using the special argument, CURRENTCHANNEL. (issue #8341,
jcovert) ........
* Makefile: Add the target "menuconfig" as an alias for the
"menuselect" target. This is just a favor to users so that if you
accidentally type "make menuconfig" instead of "make menuselect",
it still works. (inspired by a comment on IRC from wangster
calling me an "especially devious asterisk developer" for having
it be menuselect instead of menuconfig. :) )
* main/term.c: Tweak the formatting of this new function to better
conform to coding guidelines.
2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com>
* main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo
safe output!
2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com>
* channels/chan_zap.c: Make sure we don't use 32 bits when we only
need one bit.
2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: ...and make sure that the dialog is
destroyed, even if we don't get any answer on the bye... This is
the channel that remains dead after the SIP transfer
* channels/chan_sip.c: Add debug output while trying to trace bug
in bug report
* channels/chan_sip.c: Make sure we destroy dialog...
* /, channels/chan_sip.c: Small cleanup of handle_request_invite()
- imported from 1.2 with changes
2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com>
* channels/chan_zap.c: Fix for #7321. Be able to explicitly hide
callerid name for switches that bork on it.
2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Issue 8010 - Fix support for multipart
SDP (alphaque)
2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org>
* build_tools/prep_moduledeps: grep -m is not available on BSD, so
use head -1 instead
2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com>
* apps/app_chanspy.c: Only split up extension and context if a
value exists. (issue #8332 reported by loloski)
2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c,
channels/chan_skinny.c, channels/chan_h323.c,
channels/chan_iax2.c: Discussion of these CLI changes resulted in
more consistency (Bug 8236)
2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: if adding a queue member is LOG_NOTICE, then
removing them should be LOG_NOTICE, not LOG_DEBUG
* apps/app_queue.c: reflect addition/removal of dynamic queue
members in queue_log, so that people using dialplan replacement
for AgentCallbackLogin can still track login/logout (issue #7736,
reported/patched by whoiswes but this commit was written by me
and covers all three paths for AQM/RQM)
2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Rip out half implementation of 491 response
support, since it wasn't implemented properly and caused memory
leaks in the case of us getting 491's, which Asterisk actually
sends... Since it is a bit too complicated to fix this, I'll rip
it out of 1.4 and put it on the to-do-list for future releases.
Now, we handle this as congestion, which it really is. Issue
#8331
* channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD.
Thanks fenlander!
2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com>
* channels/chan_h323.c: Fix building of chan_h323 by completeing
some structure definitions. (issue #8327 reported by Mithraen)
* apps/app_voicemail.c: Do conversion in a more easier to read and
working way for \r, \n, and \t. (issue #8324 reported by
johnlange)
2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c, channels/chan_zap.c,
build_tools/prep_moduledeps: Work around an issue that caused
menuselect to display a bogus description for app_voicemail and
chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description.
However, the way we extract this information from the source
files for menuselect is not smart enough to figure this out.
(issue #8326, #8328)
2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com>
* channels/chan_phone.c, /: Merged revisions 47379 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov
2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and
higher as, well, it's apparently going to be removed. This should
make all you FC6 fans happy as your Asterisk will now build
without any mods. ........
2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com>
* main/cli.c: fix tab completion for "core debug channel" and "core
no debug channel"
* main/cli.c: Fix "core show channel". Also, fix tab completion for
both "core show channel" and "core show channels".
* main/cli.c: Fix "core debug channel <whatever>". I guess someone
needs to go through and audit every CLI command that changed
number of arguments ...
* main/asterisk.c: revert the previous change, which actually
modified the deprecated command, "show profile". Now, actually
apply the change to "core show profile".
* main/asterisk.c: Fix argument parsing for the "core show profile"
CLI command (fixed by rizzo in his branch, team/rizzo/astobj2)
* main/cli.c: Fix another CLI command, "core show uptime" ...
(issue #8323, reported by johnlange, fixed by myself)
* main/asterisk.c: fix "core show version" to reflect the new
number of arguments for this CLI command (issue #8316, kshumard)
2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com>
* main/channel.c: This update fixes 7531
* channels/chan_skinny.c: Committed in behalf of 8190.
2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c: the battle over CLI command formats has broken
stuff...
* channels/chan_sip.c: add simple fix for SDP to report proper
sample rate for G.722 media sessions
2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com>
* utils/streamplayer.c: I occasionally get email from users that
are trying to figure out what this does, or due to some
misunderstanding as to what it is supposed to do, can't get it to
work. So, I have added some text here to hopefully explain what
this application does and does not do.
* channels/chan_gtalk.c: Make this module build again
* configure, configure.ac, acinclude.m4: Copy the macros from
libtool.m4 to our own acinclude.m4 such that libtool is no longer
required to be installed to be able to generated the configure
script.
2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo)
2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com>
* channels/chan_oss.c, main/channel.c, channels/chan_phone.c,
channels/chan_misdn.c, channels/chan_skinny.c,
channels/chan_features.c, channels/chan_h323.c,
channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c,
include/asterisk/stringfields.h, apps/app_voicemail.c,
main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c,
channels/chan_zap.c, channels/chan_sip.c, res/res_features.c,
channels/chan_agent.c, main/utils.c, include/asterisk/channel.h,
channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to
solve the problem in bug 7506. It's a lot of rework to solve a
fairly small problem... such is life.
2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c: Make MOH work as it did before in
chan_local, without this then it can go funky when transfers and
MOH are involved. (issue #7671 reported by jmls)
2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com>
* configs/musiconhold.conf.sample: clean up sample config, and make
native file playback the more obvious default choice
2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com>
* apps/app_voicemail.c: large overhaul to voicemail imap support.
Allows support for more imap servers, also a better
implementation of several parts of the original work. patch
provided by 8033 with major upgrades.
2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of
continue.
2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Fixing the attack shield so it doesn't
produce attacks... Issue 8265 - never reply to an ACK
2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06
Nov 2006) | 5 lines If random order is enabled for files mode
music on hold, set a random initial position, instead of always
starting at the first file, and doing the random operation only
when switching to the next file. (bug reported by John Lange on
the asterisk-dev mailing list) ........
2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and
transfer from "john" Thank you!
2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com>
* main/cli.c: Fix another bug in "core set debug" ...
* main/asterisk.c, main/cli.c: Really fix the "core set debug" and
"core set verbose" CLI commands.
* main/cli.c: fix the "atleast" option to the "core set verbose"
and "core set debug" CLI commands
2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com>
* channels/chan_sip.c: This fix introduced via bug 8233
2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org>
* bootstrap.sh: align bootstrap.sh with the version in trunk (needs
to be blocked as it is already in trunk)
* configure.ac: add proper environment vars to detect modules on
freebsd. (already applied to trunk so it needs to be blocked
there)
2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c,
channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More
changes making the CLI more consistent with "category verb
arguments" (continuation of issue 8236)
* main/config.c, main/cli.c, main/channel.c, main/manager.c,
channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c,
main/http.c, main/file.c, main/logger.c, main/image.c,
res/res_indications.c, main/asterisk.c, res/res_odbc.c,
channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c,
channels/chan_local.c, main/frame.c, channels/chan_sip.c,
res/res_features.c, channels/chan_agent.c, res/res_crypto.c,
res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c:
Reverse change of "show" to "list" and make several other
commands more consistent with "category verb arguments"
2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Move check for codec translation to
sip_call() instead of in add_sdp. No one bothers with the result
of add_sdp anyway... Yet...
* channels/chan_sip.c: Disable code for T38 over TCP and RTP since
there's no trace of actual functionality for it :-)
2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com>
* /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02
Nov 2006) | 3 lines ignore files in a music on hold directory
that begin with '.' (issue #8249, cboie) ........
2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com>
* channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix
2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: don't send INVITE when we have determined
that we can't offer any audio formats due to lack of transcoding
support (or incorrect configuration)
2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2
lines Repeat after me oej: I will at least make sure my code
compiles before I commit it. ........
2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2)
2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com>
* /, main/callerid.c: Add the missing call to free described in
issue #8268. Also, add a bunch of missing calls to free in
callerid_feed_jp().
* main/say.c: fix saying one hundred and two hundred in hebrew
(issue #7810, eldadran)
* Makefile, configure, codecs/gsm/Makefile, configure.ac,
build_tools/strip_nonapi, makeopts.in: Fixes for
cross-compilation on mips (issue #8058, ywalther, with some
modifications)
* aclocal.m4, build_tools/menuselect-deps.in, configure,
build_tools/embed_modules.xml, configure.ac: Add a check in the
configure script to determine whether ld is GNU ld or not. This
is needed because module embedding only works for gnu ld. GNU ld
is now listed as a dependency for all of the module embedding
options in menuselect. (issue #8143)
2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com>
* channels/chan_gtalk.c: bind address support from bug 8164
2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com>
* res/res_config_odbc.c: a fix for bug 8251; the var_val needs to
accept longer strings or mass confusion and a lot of lost time is
the result
2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com>
* main/Makefile: Force poll() emulation for Darwin to always be on.
It's too broken to consider being used. This resolves the console
issue OSX users have been seeing. I would have liked to autoconf
this but I haven't been able to come up with a test case that
works. Que sera.
2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com>
* res/res_monitor.c, /: Merged revisions 46776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) |
9 lines soxmix and Asterisk expect different file extensions for
certain formats. This was already handled for the wav49 format.
However, it was not handled for ulaw and alaw. I fixed this in
such a way that using the alternate extensions for ulaw and alaw
will only happen if we know we're calling soxmix, and not a
custom script defined using the MONITOR_EXEC variable. The wav49
processing was left alone so that external scripts will see no
behavior change. (issue #7550, reported by mnicholson, proposed
patch by junky, committed fix is a bit different) ........
2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: It's another round of chan_iax2 fixes!
Should hopefully fix the deadlock issues people have been
reporting. IAXtel now has qualify turned on for 800 peers and it
is handling it fine.
2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com>
* main/config.c: Cleanups suggested by Russell.
2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: Prevent an infinite loop when config
processing gets to a jitterbuffer option
2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com>
* main/translate.c: Fix "core show translation" output. Issue
#8243, patch by Damin.
2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/translate.h, main/translate.c: add an API so
that translators can activate/deactivate themselves when needed
* include/asterisk/translate.h, main/translate.c: revert changes
that were the wrong way to address this... proper fix coming
* main/translate.c: let's set the seen flag early enough to
actually make a difference...
* include/asterisk/translate.h, main/translate.c: don't re-do setup
operations for translators that can dynamically register
themselves
2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net>
* main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue
#8089 - Fix the ENUM support (picking one record by number).
Thanks otmar!
* /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport
when we're supposed to support ;rport. Issue #7473.
* /, channels/chan_sip.c: If peer fails ACL check, fail peer at
REGISTER
* channels/chan_sip.c: Fix T38 too. Thanks, tgrman !
2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com>
* contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the
boot process to ensure it starts after stuff like MySQL (issue
#8253, Alric)
* /, main/utils.c: Merged revisions 46560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) |
3 lines When handling the case where the hostname is just an IPV4
numeric address, be sure to set the address type. (issue #8247,
alexr) ........
* /, res/res_agi.c: Merged revisions 46557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) |
3 lines fix some copy/paste bugs in the checking of arguments for
the "control stream file" AGI command (issue #8255, mnicholson)
........
* main/translate.c: Add a small tweak to the code that checks to
see whether destination formats are translatable based on the
source format. If we have already determined that there is no
translation path in one direction, don't bother checking the
other direction.
2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com>
* main/translate.c: when unregistering a translator, don't rebuild
the translation matrix unless needed when filtering formats out
of an offer, ensure we check for translation ability in both
directions
* include/asterisk/linkedlists.h: ensure that items removed from a
list are always unlinked from the list (next pointer set to NULL)
2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com>
* configure, configure.ac: Don't explicitly link in crypt as it is
not used on some platforms.
* channels/chan_iax2.c: We need to lock the pvt structure during
retransmission as another worker thread may be doing something as
well.
2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net>
* main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h,
include/asterisk/doxyref.h, channels/chan_sip.c,
main/ast_expr2f.c, include/asterisk/module.h,
formats/format_ogg_vorbis.c, main/app.c,
include/asterisk/channel.h, include/asterisk/lock.h,
include/asterisk/frame.h: Issue #8246 - Doxygen fixes from
kshumard. An extra big thankyou is given to everyone that
contributes to doxygen! THANK YOU!
* main/rtp.c, /: Bind RTCP to the same IP as RTP
* /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302
redirects (imported from 1.2)
* /, channels/chan_sip.c: Issue #7608 - Notifications sent with
wrong content-type (imported from 1.2, modified)
* channels/chan_sip.c, CHANGES: Backport of patch for #7828 that
was reported for trunk, but obviously exists in 1.4 too.
* channels/chan_sip.c: Restoring the old logic, since working
around it and fixing it seemed too complicated. - The
SIP_OUTGOING flag indicates the direction of the last transaction
in the dialog. - The initreq stores the last request in the
dialog, the request that opened the latest transaction. Please
now retry all the 1.4 bug reports with mixed to/from headers,
tags etc in ACK, BYE, CANCEL. Thanks!
* channels/chan_sip.c: Accepting a message twice may be
misinterpreted...
* channels/chan_sip.c: - 183 is not reliable message... - Error
should not have SDP
2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com>
* utils/Makefile: Don't build muted on OpenBSD, it is not
supported.
2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com>
* channels/chan_zap.c: move the copy of the default settings to the
global settings back out of process_zap, so that they aren't
overwritten when process_zap is called multiple times
2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net>
* contrib/asterisk-ng-doxygen: Put some doxygen pressure on
Christian :-)
2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com>
* main/asterisk.c, res/res_agi.c, apps/app_externalivr.c,
res/res_musiconhold.c: We should always be using _exit() after a
fork() or vfork() instead of exit(). This is because exit() does
some extra cleanup which in some implementations of vfork(), for
example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
* channels/chan_zap.c: Instead of iterating all of the options once
to look for jitterbuffer options, and then again for everything
else, move the processing of jitterbuffer options into the main
loop so that there are no erroneous messages about ignoring
unknown options. (issue #8226)
2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c:
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) |
1 line fixed a bug which caused chan_misdn to try to allocate 2
times the same channel on high load, which then caused
instability of mISDN. removed a useless function from isdn_lib.c
........
* channels/misdn_config.c: fixed not compile issue, which was just
introduced
* channels/misdn_config.c, channels/chan_misdn.c, /,
channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) |
1 line added nttimeout option to configure wether we disconnect
calls on NT timeouts or not during an overlapdial session
........
2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com>
* /, contrib/scripts/astgenkey.8: Merged revisions 46337 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2
lines oops - somebody forgot to change this - long ago, probably.
........
* CHANGES: grammar check
2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net>
* CHANGES: Corrections to changes (Multiparking is not included)
2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com>
* main/translate.c: - If the source has no audio or no video
portion, do not call powerof() to get the format index. - Don't
run through the audio and video loops if there is no audio or
video portion of the source If 0 is passed to powerof, it will
return -1. This value of -1 was then being used as an array index
in these loops, which caused a crash on some systems. Other than
this issue, this code works as we expected it to. If a format is
not in the source, and we have to translation path to it, it is
not offered in the list of acceptable destination formats. (fixes
issue #8231)
2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: update to reflect G.722 addition
2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com>
* doc/backtrace.txt: update backtrace documentation to reflect
changes in 1.4 (issue #8230, kshumard)
2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com>
* main/config.c, main/manager.c: Fix config comment code
preservation code (thanks murf!)
2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Old todo note - Don't add Contact header on
BYE and Cancel
2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com>
* configure.ac: fix error output when checking for openh323 to
refer to openh323 instead of pwlib (issue #8222, misaksen)
2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Somewhat ugly code to try to fix issue
#7608. Since the problem was not very well defined, the fix is a
bit fuzzy too... Thanks to Luigi for accidentally spotting the
possible problem!
2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com>
* apps/app_queue.c: update warning message to include "agi" option
(issue #8225, jmls)
2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: use 1.4.3 extra sounds with corrected silence
files
* sounds/sounds.xml, sounds/Makefile: add support for prebuilt
G.722 prompts and music on hold files
2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: show settings doesn't produce a list of
similar objects, it should stay a "show"
2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com>
* main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c,
channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c,
pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c,
main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c,
cdr/cdr_custom.c, channels/chan_mgcp.c,
apps/app_parkandannounce.c, apps/app_voicemail.c,
channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c,
res/res_adsi.c, main/utils.c, apps/app_ices.c,
pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c,
apps/app_getcpeid.c: apparently developers are still not aware
that they should be use ast_copy_string instead of strncpy... fix
up many more users, and fix some bugs in the process
2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/pbx.c: WaitExten truncates decimals of times to wait,
instead of accepting them (Bug 8208)
2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c,
channels/chan_h323.c, channels/chan_iax2.c,
include/asterisk/frame.h: add passthrough and file format support
for G.722 16KHz audio (issue #5084, original patch by andrew,
updated by mithraen)
* channels/chan_sip.c, main/translate.c: code zone experiment:
don't offer formats in the outbound INVITE that aren't either
passthrough or translatable
* main/translate.c: if multiple translators are registered for the
same source/dest combination, ensure that the lowest-cost one is
always inserted earlier in the list
2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com>
* res/res_agi.c: Fix FastAGI when there is no pid (bug #7628,
#8147)
2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: We need to initialize our scheduler pthread
condition... yes.
2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org>
* main/http.c: merge 45152 don't leak descriptors in http.c
* channels/chan_sip.c: merge 45966 refer_to_domain potentially
containing options
* channels/chan_sip.c: merge 46026 improper checks on get_header()
return values
* channels/chan_sip.c: merge 46045 prevent NULL args to
ast_strdupa() in chan_sip.c
2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com>
* Makefile: Restore the ability to remove the firmware directory
without causing the installation to fail (issue #8111)
2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com>
* main/translate.c: ensure that the translation matrix is properly
lock-protected every place it is used
* include/asterisk/translate.h, main/translate.c: add an API call
to allow channel drivers to determine which media formats are
compatible (passthrough or transcode) with the format an existing
channel is already using
* doc/imapstorage.txt: simplify and correct voicemail IMAP storage
build instructions
2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* main/channel.c: Pass through a frame if we don't know what it is,
rather than trying to pass a NULL, which will segfault a channel
driver (Bug 8149)
2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com>
* utils/muted.c, utils/ael_main.c: In muted.c, check the return
value of strdup. In ael_main.c, check the return value of calloc.
(issue #8157) In passing fix a few minor bugs in ael_main.c. The
last argument to strncpy() was a hard-coded 100, where it should
have been 99. I changed this to use sizeof() - 1.
* apps/app_meetme.c: Fix the descriptions of some of the
MeetMeAdmin options (issue #8098, mflorell)
* res/res_jabber.c: don't crash when an incoming message has no
"from" (issue #8205, jmls)
2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com>
* /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2
lines Don't leak memory mmmk? ........
2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21
Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and
couldn't be initialized it would cause a segfault after 'reload'.
Reported by Drew/Matt thx. ........
2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com>
* res/res_monitor.c: Add a couple missing unregistrations of
manager actions and remove duplicate unregistrations of
applications. (issue #8194, jmls)
2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com>
* main/loader.c: Don't use promotion on Darwin because it doesn't
seem to work quite right in all cases, this should solve the
unresolved symbol issue people have been seeing.
* Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get
installed in the proper location (reported on asterisk-dev
mailing list)
2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Let's understand SIP: - REFER can create
dialog, Asterisk does not support it yet - NOTIFY can create
dialog in Asterisk's implementation (voicemail) even though we
don't support the server side of it. In this case, the standard
is a side issue ;-) - Added extened functionality for unsupported
methods (PING, PUBLISH) so we don't create PVT's for those
either. Russellb needs to judge what to do with this in 1.2, but
I think the current implementation n 1.2 is a bug since we're
sending bad replies to NOTIFY and REFER outside of dialogs
2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com>
* res/res_jabber.c: Let's remember to unregister JabberStatus too
(issue #8184 reported by jmls)
* /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.2
........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct
2006) | 2 lines Respect language selection when seeing if the
file exists (issue #8178 reported by mnicholson) ........
* channels/chan_sip.c: If the jitterbuffer is forced on then we
can't partially bridge (reported by wangster on #asterisk-dev)
2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Don't leak the actual thread-specific
sip_pvt struct
2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: don't leak memory when a chan_sip thread is
destroyed that has a thread-local temp_pvt allocated
2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com>
* main/asterisk.c: Don't modify things if we are using vfork as
this is very bad and may cause unexpected behavior (issue #7970
reported by Nick Gavrikov)
2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: remove duplicate declarations
2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org>
* main/http.c: merge from trunk: move ast_variables_destroy() to a
better place in handle_uri() to avoid leaking memory on non
existing files.
2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Don't segfault if you're using a channel driver that
doesn't turn RTCP on
2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com>
* main/channel.c: Don't attempt to access private data members of
the pthread_mutex_t object, because this does not work on all
linux systems. Instead, just access the reentrancy field in the
ast_mutex_info struct when DEBUG_THREADS is enabled. If
DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
DEBUG_THREADS on as well. (issue #8139, me)
* configs/sip_notify.conf.sample: update entry to reboot a snom
phone (issue #7850, pnlarsson)
2006-10-17 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta3 released.
2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/stringfields.h, main/ast_expr2.c,
main/channel.c, channels/chan_sip.c, channels/chan_iax2.c:
optimize the 'quick response' code a bit more... no more malloc()
or memset() for each response expand stringfields API a bit to
allow reusing the stringfield pool on a structure when needed,
and remove some unnecessary code when the structure was being
freed
2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't create a "real" pvt structure for
requests that shouldn't be able to create one. Instead use a
temporary pvt and fill it with enough information so we can send
a reply.
2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net>
* configs/sip.conf.sample: Adding information about Marks
direct-RTP hack to the docs...
2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com>
* LICENSE: provide licensing language for IAXy firmware file
2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com>
* apps/app_dial.c, apps/app_directed_pickup.c: Backport of new
directed pickup (BE-85).
2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net>
* CREDITS: Adding Inotel to credits for SIP transfers. Thanks for
your support!
* channels/chan_sip.c: Don't destroy dialog for unexpected REFER
response...
2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com>
* funcs/func_rand.c: update the doc string for both AEL and
extensions.conf users.
2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com>
* main/acl.c don't drop the entire permit/deny list when an attempt
is made to add an invalid entry (BE-92)
2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com>
* res/res_speech.c: Clear the quiet flag too since we are
restarting a recognition again (reported on -dev by Stephan
Edelman)
* res/res_speech.c: Check return value from engine in case of
failure (ie: out of licenses) (reported on -dev mailing list)
2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-vtest17 (added),
pbx/ael/ael-test/ael-vtest17/extensions.ael (added),
pbx/ael/ael-test/ael-vtest17 (added),
pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in
this release via these changes
2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: avoiding warning, fixing potential bug
2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com>
* codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c,
codecs/lpc10/decode.c, codecs/lpc10/dcbias.c,
codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c,
codecs/lpc10/difmag.c, codecs/lpc10/hp100.c,
codecs/lpc10/synths.c, codecs/lpc10/preemp.c,
codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c,
codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c,
codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c,
codecs/lpc10/lpcini.c, codecs/lpc10/random.c,
codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c,
codecs/lpc10/placea.c, codecs/lpc10/tbdm.c,
codecs/lpc10/analys.c, codecs/lpc10/onset.c,
codecs/lpc10/energy.c, codecs/lpc10/deemp.c,
codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c,
codecs/lpc10/median.c, codecs/lpc10/encode.c,
codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c,
codecs/lpc10/invert.c: And file said... let the compiler warnings
STOP!
* apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136
reported by mnicholson)
* apps/app_playback.c: Move say.conf existence check to do_say
function since it is called from multiple places (issue #8144
reported by kshumard)
2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_iax2.c: when sending a call to a peer, use the proper socket if
we have multiple bindings (reported on asterisk-dev)
2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Complete merging in RPID screen changes
(issue #8101 reported by hristo, patch by oej in revision 44757)
* main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add
the background refresh item back into the scheduler if enabled
since it is deleted during reload. (issue #8142 reported by
p_lindheimer)
2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com>
* configure, include/asterisk/autoconfig.h.in, configure.ac,
main/utils.c: use a configure script test for PMTU discovery
control instead of just assuming it's available on Linux
2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some
echocandisable issues when bridged. this caused a kernel panic
sometimes.. also some minor formatting fixes
* channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause
got a wrong isdn cause at RELEASE_COMPLETE
2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: merge formatting and minor code
simplifications from trunk
2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com>
* channels/chan_gtalk.c: fix for bug 7764.
2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: we can only send one 'a=ptime' attribute per
media session, not one for each format
* main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c,
main/utils.c: ensure that IAX2 and SIP sockets allow UDP
fragmentation when running on Linux (thanks to Brian Candler on
the asterisk-dev list for the tip)
2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com>
* main/manager.c: fix a silly typo in a comment that I saw while
reading the commit list
2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com>
* Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue
#8135 reported by ssokol)
2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com>
* main/manager.c: append_event must be called while holding the
session lock
2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com>
* res/res_jabber.c: change some debug output to use LOG_DEBUG
instead of verbose output
2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com>
* main/db1-ast/Makefile: These are already set by the parent
Makefile.. There is no need to have this here (it doesn't
actually work anyways).
2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c: removed warning because of missing
prototype declaration
2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Do not set default/global values in the
variable declaration, set it in reload_config()
2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Move some stuff around so that a NOTIFY
dialog won't hang around until the end of the world under certain
circumstances
2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz>
* main/channel.c, funcs/func_channel.c, include/asterisk/channel.h:
CHANNEL() function sometime mix parameter and value
2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* funcs/func_logic.c: Lost of a bit of logic when this was
simplified between 1.2 and 1.4 (Bug 8117)
2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Bail out if we have no refer structure and
we get a refer response
2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: more merge from trunk (comments and change a
static function name)
2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Only set DTMF information if an RTP
structure exists
2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added
support of dynamically enabling hdlc on bchannels
2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: whitespace changes related to previous
commit
* channels/chan_sip.c: merge a few code simplifications that have
gone into trunk during last week, to reduce differences between
the two branches and make porting fixes easier.
2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Fix a problem where phones that go
"missing" never got unregistered. Issue #8067, reported by pj,
patch by Anthony LaMantia (with minor whitespace modifications)
2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid
the deadlock
* channels/chan_iax2.c: Properly avoid a collision with iax2_hangup
(issue #8115 reported by vazir)
2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: do not dereference p if we
know it is NULL
2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate
caller's transfer capability too
2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: put common code in a
function to avoid repetitions.
* channels/chan_sip.c: remove hardwired usage of 5060, use
DEFAULT_SIP_PORT instead
* channels/chan_sip.c: option_debug checking
before printing to debug channel.
* channels/chan_sip.c: backport simplifications on sip_register,
usage of ast_set2_flag(), and fixes to the handling of failed
module loading.
* channels/chan_sip.c: improve and document function
get_in_brackets(), introducing a helper function
find_closing_quote() of more general use.
2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com>
* include/asterisk/linkedlists.h: ensure that mutex locks inside
list heads are initialized properly on platforms that require
constructor initialization (issue #8029, patch from timrobbins)
* CHANGES: remove Jingle as per mog
2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Remove the seqno check for RFC2833, the handler is
smart enough to not need it.
2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com>
* CHANGES: various cleanups
2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com>
* main/rtp.c: When the sequence number rolls over then reset the
recorded sequence number for DTMF (issue #8106 reported by
bungalow)
* main/file.c: Even more frames to treat as though the remote side
disappeared (issue #8097 reported by eldadran)
2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org>
* main/manager.c, main/http.c: make sure sockets are blocking when
they should be blocking.
2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c: fixed segfault which happens during
hold/transfer action
* channels/chan_misdn.c: if INFORMATION Message come with keypad
instead of called party number, we just use the keypad as called
party number.
* channels/misdn/isdn_lib.c, channels/misdn_config.c,
channels/misdn/isdn_lib.h, channels/chan_misdn.c,
channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample:
added the option 'reject_cause' to make it possible to set
the RELEASE_COMPLETE - cause on the 3. incoming PMP channel,
which is automatically rejected because chan_misdn does not
support that kind of callwaiting. Therefore chan_misdn supports
now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc
now gets the info if the requested channel is incoming or
outgoing to make the 3. channel possible
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c: fixed the hold/retrieve/transfer issues,
removed a useless bc field, added setting of frame.delivery fields,
some minor code cleanups
2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com>
* main/file.c: Treat busy control frames as hangup in the file streaming
core (issue #8097 reported by eldadran)
2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang.
Many thanks to Doug!
2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite
hanging by a thread if the other side is already setup with T.38
2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com>
* main/app.c: don't segfault when an argument without a close
parenthesis is found stop parsing as soon as that situation
occurs
2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com>
* CHANGES: I put the accumulated changes from the commit logs and
inspection, into CHANGES. Hope everyone approves!
* configs/muted.conf.sample, utils/muted.c: Hang on a minute, the
install process sticks muted.conf in /etc/asterisk, so that's
where muted should look for it, right?
2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Don't totally bail out if T.38 was
negotiated
2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: fix Polycom presence notification again
2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org>
* utils/Makefile: as far as i can tell astman only uses newt...
* Makefile: put linker flags in ASTLDFLAGS where they belong
2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE
requests add workaround for new Polycom firmware SUBSCRIBE
requests (bug is known to exist in 2.0.1 firmware)
* include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually
work
2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com>
* pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c,
pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12,
pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3,
pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4,
pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6,
pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8,
pbx/ael/ael-test/ael-test16/extensions.ael (added),
pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y,
pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14,
pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9,
pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the
problems reported in bug 8090
2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_oss.c, main/cdr.c, channels/chan_phone.c,
main/manager.c, pbx/pbx_spool.c, res/res_smdi.c,
channels/chan_skinny.c, channels/chan_h323.c, main/http.c,
channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c,
main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c,
include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c,
channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c,
main/devicestate.c, main/utils.c, res/res_musiconhold.c,
channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update
thread creation code a bit reduce standard thread stack size
slightly to allow the pthreads library to allocate the stack+data
and not overflow a power-of-2 allocation in the kernel and waste
memory/address space add a new stack size for 'background'
threads (those that don't handle PBX calls) when LOW_MEMORY is
defined
2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com>
* configs/muted.conf.sample: I've been meaning to add some
explanation about muted... here it is
* configs/manager.conf.sample: CLI reverbification update to this
config file
* apps/app_macro.c: In response to bug 7776, a Warning has been
added to the doc string for Macro().
2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com>
* main/asterisk.c, main/loader.c, main/term.c, Makefile,
include/asterisk.h: ensure that local include files are always
used avoid a duplicate function name (term_init())
2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com>
* channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing
client without resource.
2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: fix a logic error in my previous fix to the queue
reload code
2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Change default presentation indicator
to "user provided not screened" if octet 3a missed in
CallingPartyNumber IE
2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Use VideoSupport instead so it is considered
a valid XML attribute name. (issue #8075 reported by renemendoza)
2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Fix preparation of type and
presentation of calling number
2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com>
* doc/jingle.txt, channels/chan_jingle.c (removed),
include/asterisk/jabber.h, configs/jingle.conf.sample (removed),
res/res_jabber.c: updated res_jabber for even better component
support, soon will be jep-0100 compliant. also removed
chan_jingle and infromed info from jingle.txt, chan_gtalk still
works and should be used in this version.
2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Change the fd on the I/O context in case it
changed during the reload, which is indeed possible. (issue #7943
reported by eclubb)
* contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN
instead of hardcoding the path for the error message (issue #7942
reported by eclubb)
2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz>
* configs/users.conf.sample, pbx/pbx_config.c: Missed part of
userconf functionality for chan_h323
2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com>
* main/io.c: Shrink when current_ioc is unused. It is set to -1 when
unused, not 0. (issue #7941 reported by eclubb)
2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz>
* doc/realtime.txt: Typo fix
* channels/chan_h323.c: Optimization of oh323_indicate(): less
locks - less problems, plus single exit point
2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com>
* channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when
you're not talking about a channel :)
2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_h323.c: Do not simulate any audio tones if we got
PROGRESS message
2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com>
* Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to
be empty. The cause is that since ASTDATADIR is explicitly
exported using "export ASTDATADIR" at the top of the Makefile,
make no longer considers the variable "undefined", so the
Makefile can't use ?= to set ASTDATADIR if not yet set. (issue
#8063, reported by akohlsmith, fixed by me)
* configs/queues.conf.sample: Fix the name of the "eventmemberstatus"
option in the sample queues.conf (issue #8065, adamg)
2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org>
* channels/chan_sip.c: sync with trunk - move variable declarations
to the beginning of a block.
2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz>
* main/rtp.c: Allow one-way RTP streams (device->Asterisk)
2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org>
* codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent
build problems: - with AST_DEVMODE, building codecs/lpc10 fails
because of lots of warnings, and the configure step in editline
fails as well. Fix this by removing the -Werror in these steps. -
on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile (a
possible alternative way is to add "export AST_LIBS" near the
beginning of the file). With this fix, i believe that some of the
platform-specific conditionals in main/Makefile are redundant
(because they should be already dealt with in the top level
Makefile) but i don't have a platform to check. Merging to head
will happen in a moment.
2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz>
* channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment
of previous fix: Issue #7928 - Don't send both 404 and 503. Fix
by phsultan with a small fix by me, myself or I. Thanks,
Philippe! (This was caused by my changes to the transaction
handling)
* channels/chan_sip.c: Found some buggy SIP clients (phones Planet
VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which
sends ACK not on OK message only (when remote party answers) but
on RINGING message too, so when we send 200 OK message, we get
unidentified ACK message (because INVITE acknowledged on RINGING
message already), so 200 OK retransmits within its retransmission
interval then call gets dropped. If someone else knows how to
provide workaround for such cases, please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com>
* agi, utils: ignore temporary files made by the Makefiles during a
build
* codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile,
codecs/Makefile, utils/Makefile, configure,
build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac,
Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile,
pbx/Makefile, res/Makefile, channels/Makefile: fix a few build
system bugs, and convert Makefiles to be compatible with GNU make
3.80
2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com>
* main/asterisk.c, main/cli.c: Fix a bug with the removal of
'atleast' argument to 'core verbose' and 'core debug'. Add that
argument back in.
2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more
carefully when no CallingNumber IE available
* channels/h323/ast_h323.cxx: Fake display name by called number on
incoming calls (until passing connected number/connected name is
not implemented)
* channels/h323/ast_h323.cxx: Ported code refers to H.450 - add
includes
* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly
pass TON/PRESENTATION information - original
H323Connection::SendSignalSetup() destroys Q.931 fields.
2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile: yet another place where we were not using the
correct CFLAGS by default
* main/Makefile: missed one conversion to ASTCFLAGS
2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx, channels/chan_h323.c,
channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass
TON/PRESENTATION information too
2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com>
* main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile,
main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse
CFLAGS and LDFLAGS for build of Asterisk components, because they
are also then used for non-Asterisk components (like menuselect);
use our own variables instead
* configure, configure.ac: support --without-curl in configure
script
* Makefile.rules: another cross-compile fix
* Makefile: a couple more environment settings that can't leak into
the menuselect build
* main/cli.c: proper fix for ast_group_t change
* include/asterisk/lock.h: eliminate compiler warning when
DEBUG_CHANNEL_LOCKS is enabled and users of this header file
don't also include channel.h
2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com>
* apps/app_queue.c: Fix incorrect argument order for member names,
on persisted members. Issue 8047, patch by jmls.
2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com>
* apps/app_playback.c, res/res_monitor.c,
include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c,
channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c,
main/udptl.c, main/frame.c, funcs/func_timeout.c,
channels/chan_sip.c, apps/app_festival.c,
channels/iax2-provision.c, apps/app_alarmreceiver.c,
res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c:
Put in missing \ns on the end of ast_logs (issue #7936 reported
by wojtekka)
2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com>
* apps/app_queue.c: fix buggy (and overly complex) loop used during reload
of app_queue for static member list updating
2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Extend call establishment timeout
2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com>
* channels/chan_iax2.c: Make sure the pvt exists before accessing
it again as it may have gone away (issue #7562 reported by Seb7
and issue #7939 reported by sorg)
* main/cli.c: Warning be gone!
2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com>
* apps/app_queue.c: app_queue is comparing the device names incorrectly
while checking their statuses. It's internal list of interfaces
includes the dial string, while the argument passed to this
function does not have the dial string (/n for a local channel).
This causes it to ignore the device state changes because it
thinks it belongs to none of its members. (#8040 reported and
patch by tim_ringenbach)
2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c: Stop the stream after waitstream returns so that our
formats get restored. (issue #7370 reported by kryptolus)
2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Fix compiler warning
2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com>
* apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 -
tim_ringenbach reported and patched)
* apps/app_queue.c: Autopause not working for queue members. (#8042
- jmls reported and patch)
2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force
remote side to start media on outgoing PROGRESS message
* include/asterisk/compiler.h: Put attribute tag at correct place
2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com>
* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h,
channels/chan_misdn.c: fixed a bug which led to chan_list zombies,
when the call could not be properly established in misdn_call.
also removed the ACK_HDLC stuff which is not really needed.
2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/ast_h323.cxx: Do not open transmit channel until
TCS is received
* main/file.c: Don't warn on HOLD/UNHOLD control frames
* main/file.c: Don't treat unknown control frames as voice
2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Avoid inability to lock directory log message by
creating the directory ahead of time. (Issue 7631)
2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com>
* apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS
not being set under certain circumstances. Fix a minor issue, to
make it use the filenames that were parsed, instead of the entire
argument string. Fix Background() to return -1 like Playback(),
if no args are specified.
2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com>
* main/rtp.c: Compensate for out of order packets better if RFC2833
compensation is turned on.
* channels/chan_iax2.c: Get rid of two functions from a time now
past (we THINK these are from pre-recursive lock time) that may
be contributing to two open issues on the bug tracker (7562/7939)
and that has the potential to just make bad things happen if the
timing is right.
2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com>
* main/channel.c,res/res_features.c: Fix a problem that occurred if
a user entered a digit
that matched a bridge feature that was configured using multiple
digits, and the digit that was pressed timed out in the feature
digit timeout period. For example, if blind transfer is
configured as '##', and a user presses just '#'. In this
situation, the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by
michaels and valuable input provided by mneuhauser and kuj. Fixed
by me, with testing help and peer review from Joshua Colp). There
are a couple of issues involved in this fix: 1) When
ast_generic_bridge determines that there has been a timeout, it
returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets
this result, it calls ast_generic_bridge over again with the same
timestamp for the next event. This results in an endless loop of
nothing until the call is terminated. This is resolved by simply
changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it
sees a timeout. 2) I also changed ast_channel_bridge such that if
in the process of calculating the time until the next event, it
knows a timeout has already occured, to immediately return
AST_BRIDGE_COMPLETE instead of attempting to bridge the channels
anyway. 3) In the process of testing the previous two changes, I
ran into a problem in res_features where ast_channel_bridge would
return because it determined that there was a timeout. However,
ast_bridge_call in res_features would then determine by its own
calculation that there was still 1 ms before the timeout really
occurs. It would then proceed, and since the bridge broke out and
did *not* return a frame, it interpreted this as the call was
over and hung up the channels. The reason for this was because
ast_bridge_call in res_features and ast_channel_bridge in
channel.c were using different times for their calculations.
channel.c uses the start_time on the bridge config, which is the
time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after
start_time in the bridge config, and sometimes enough to round up
to one ms. This is fixed by making ast_bridge_call use the same
time as ast_channel_bridge for the timeout calculation. ........
2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com>
* channels/chan_misdn.c, channels/Makefile: removed the chan_misdn
versioning, since Asterisk has it's own
2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c: Make rfc2833compensate a global option.
2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com>
* apps/app_voicemail.c: Backport revision 43754 from the trunk,
which removes an unused buffer from mm_login to close bug 8038,
as well as addresses some formatting and coding guidelines issues
in passing. Originally, I did not commit this to 1.4 since it is
not necessarily fixing a bug. However, since the IMAP storage
code is brand new, I decided it would be better to make the
change here as well, in case someone has to work on this code to
address issues in the very near future. I don't want to make
unnecessary merge problems going to the trunk.
2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com>
* configs/extensions.ael.sample: This change to extensions.ael was
to fix bug 8031; the install scripts are causing it to be copied
to /etc/asterisk/extensions.ael, and because it is a fairly
direct conversion of the original extensions.conf, the macro and
context names clash with the existing extensions.conf. So, I put
an ael- in front of all macros and contexts, and checked every
goto and macro call. Also, this file compiles under aelparse.
2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com>
* main/asterisk.c: Back in revision 4798, this message was changed from
using ast_cli() to directly calling write(). During this change,
checking if this was a remote console was removed. This caused
this message about using "exit" or "quit" to exit an Asterisk
console to come up in times where it did not make sense. This
change restores the check to see if this is a remote console
before printing the message. (fixes BE-65)
2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com>
* .cleancount, main/cli.c, channels/chan_sip.c,
include/asterisk/channel.h: Use proper type to represent the group variable
(issue #8025 reported by makoto)
2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com>
* channels/chan_sip.c: Add missing newline character in the warning
message about deprecated TOS values in configuration.
* apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain
mailbox definitions, don't introduce a length limit on the
definition by using a 256 byte temporary storage buffer. Instead,
make the temporary buffer just as big as it needs to be to hold
the entire mailbox definition. (fixes BE-68)
2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com>
* channels/chan_local.c: Strip options off the argument passed for
devicestate in chan_local. (issue #8034 reported by pcardozo)
* apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight
overhaul of the whisper support. 1. We need to duplicate the
frame from ast_translate 2. We need to ensure we always have
signed linear coming in for signed linear combining. 3. We need
to ensure we are always feeding signed linear out. 4. Properly
store and restore write format when beeping on the channel we are
whispering on. 5. Properly discontinue the stream on the channel
for the beep. (issue #8019 reported by timkelly1980)
2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: update to use 1.4.3 core sounds, with corrected
beep/beeperr/tt-monkeys files
2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com>
* doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by
Dan Austin. Maximum values were incorrect, which is why this is
being put in 1.4
* channels/chan_skinny.c: Add proper codec support to chan_skinny.
Works with at least ulaw, alaw, and g729a. This is technically a
"new feature", but there are justifications for it. I found a bug
with the recent rtp packetization changes, which caused the media
setup to fail under certain circumstances, particularly when
using allow=all, or having no allow= statements (globally or on
the device). I could have either removed the rtp packetization
features, or I could add proper codec support (which, without, I
think most people would consider to be a bug anyways).
2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_voicemail.c: Should have moved these lines up in the
merge, instead of removing them
* apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1)
delete=yes was ignored 2) maxmessages was ignored
2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h,
channels/h323/cisco-h225.asn: Fix ASN1 description of
non-standard Cisco extensions
* channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport
changes of trunk: 1) r43540: Avoid possible deadlock on channel
destruction 2) r43590: Disable fastStart if requested by remote
side
2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com>
* sounds/Makefile: One more fix for sounds installation - this time
for portability. Reported to asterisk-dev mailing list.
2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com>
* formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from
crashing if trying to play an OGG moh file.
2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz>
* channels/h323/caps_h323.cxx, channels/h323/compat_h323.h,
channels/chan_h323.c: Merged revisions 43472,43495 from trunk
2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com>
* channels/iax2-provision.c: Fix a CLI command registration issue
where an erroneous message claiming that "iax2 show provisioning"
was already registered. This was because this command was
registering itself as both the command, as well as the command it
is deprecating. (issue #8022, reported by bjweeks, fixed by
myself)
* channels/chan_iax2.c:Check to see if the channel that is activating the
IAXPEER function is actually an IAX2 channel before proceeding to
process it to avoid crashing. (issue #8017, reported by admott,
fixed by myself)
2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: don't output the 'build complete' message when the
target being run is already going to do an installation
2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: Allow chan_skinny.so to be unloaded
properly. Remove reload support, since it doesn't
actually...work.
2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com>
* pbx/pbx_ael.c: This commits a change to return
MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all
goes well for bug 8004
* pbx/pbx_ael.c: If the extensions.ael file not found, or
unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004.
2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com>
* main/cli.c: Make sure we explicitly set the CLI command to not be
deprecated, if it isn't.
2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com>
* sounds/Makefile: use rebuilt extra sounds
* main/channel.c: all the Linux systems I have don't use
'__m_count' for this field, so I don't know where this came
from...
2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com>
* include/asterisk/threadstorage.h: backport the compatability fix
to use attribute_malloc instaed of __attribute__ ((malloc))
* channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN
could not be configured (issue #8006, Mithraen)
* main/frame.c: Suppress a compiler warning about the use of a
potentially uninitialized variable. It couldn't actually happen,
though.
2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com>
* channels/chan_skinny.c: First shot at unload_module in
chan_skinny.. More to come.
2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com>
* include/asterisk/jabber.h, channels/chan_gtalk.c,
res/res_jabber.c: updates for better compontent support
2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we
actually documented how the new features in res_odbc actually
work. (Oops)
2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com>
* channels/chan_oss.c: Some more clean up in the load function for
chan_oss (issue #8002 reported by Mithraen with minor mods by
moi)
* channels/chan_mgcp.c: Clean up chan_mgcp's module load function
(issue #8001 reported by Mithraen with mods by moi)
2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile, build_tools/strip_nonapi (added): add another
attempt to strip non-API symbols from the final binary... script
will need to be extended to work on non-Linux systems
2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
* apps/app_url.c: Fix documentation to reflect how Url() really
works
* cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates
2006-09-21 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta2 released.
2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile: remove this change... it requires binutils 2.17
2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com>
* build_tools/make_version: fix minor typo in the way version is
handled
2006-09-20 Kevin P. Fleming <kpfleming@digium.com>
* Asterisk 1.4.0-beta1 released.
Reference in New Issue View Git Blame Copy Permalink