If a user does not provide a port in the fromdomain setting, chan_sip will set
the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will
then get used unilaterally in certain places. This causes issues with TLS,
where the default port is expected to be 5061.
This patch modifies chan_sip such that fromdomainport is only used if it is
not the standard SIP port; otherwise, the port from the SIP pvt's recorded
self IP address is used.
Review: https://reviewboard.asterisk.org/r/3893/
ASTERISK-24178 #close
Reported by: Elazar Broad
patches:
fromdomainport_fix.diff uploaded by Elazar Broad (License 5835)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@421717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
sip_subscribe_mwi_destroy calls sip_destroy on the reference counted
mwi->call. This results in the fields of mwi->call being freed, but
mwi->call itself it leaked. If other code is still using mwi->call
it can cause problems. This change uses dialog_unref instead, to
balance the ref provided by sip_alloc().
ASTERISK-24087 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3834/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.
Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP. It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed. This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.
NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.
The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.
This commit is a merge of the two patches indicated below.
ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
pri-4.diff (license #6302) patch uploaded by Pavel Troller
jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/3633/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is essentially a backport of a small portion of r397526 from
ASTERISK-21981. In that patch, pass through support and format attribute
negotiation was added for Opus. Part of that included being more tolerant to
whitespace in the fmtp line of an SDP; that part of the patch is being
applied here.
As the author of the backport pointed out, in SDP, the fmtp line is allowed to
include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001)
includes an example for this. This was not removed in the updated RFC 4867 in
2007.
Note that this patch only applies to audio in Asterisk 1.8, which is a bit more
limited in its support for format attributes. It does have limited support for
some codecs, so this patch is still useful in this version.
Review: https://reviewboard.asterisk.org/r/3658
ASTERISK-23916
Reported by: Alexander Traud
patches:
sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From headers were processed using a 256 character buffer on the stack.
This change replaces that with a heap allocation by ast_strdup.
ASTERISK-23790 #close
Reported by: uniken1
Tested by: uniken1
Review: https://reviewboard.asterisk.org/r/3669/
Patches:
chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change debug level for messages in sdp_crypto.c from zero to one. This
ensures the messages are not displayed when debugging is disabled. Change
does not apply to 12+ as it was already fixed in those versions.
ASTERISK-23246 #close
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3605/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Simply establishing a TCP connection and never sending anything to the
configured HTTP port in http.conf will tie up a HTTP connection. Since
there is a maximum number of open HTTP sessions allowed at a time you can
block legitimate connections.
A similar problem exists if a HTTP request is started but never finished.
* Added http.conf session_inactivity timer option to close HTTP
connections that aren't doing anything. Defaults to 30000 ms.
* Removed the undocumented manager.conf block-sockets option. It
interferes with TCP/TLS inactivity timeouts.
* AMI and SIP TLS connections now have better authentication timeout
protection. Though I didn't remove the bizzare TLS timeout polling code
from chan_sip.
* chan_sip can now handle SSL certificate renegotiations in the middle of
a session. It couldn't do that before because the socket was non-blocking
and the SSL calls were not restarted as documented by the OpenSSL
documentation.
* Fixed an off nominal leak of the ssl struct in
handle_tcptls_connection() if the FILE stream failed to open and the SSL
certificate negotiations failed.
The patch creates a custom FILE stream handler to give the created FILE
streams inactivity timeout and timeout after a specific moment in time
capability. This approach eliminates the need for code using the FILE
stream to be redesigned to deal with the timeouts.
This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of
the SSL_read/SSL_write operations.
ASTERISK-23673 #close
Reported by: Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk started counting the session timer at INVITE while the other
end correctly started at 200. This meant that for short session-expiries
(90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk
would wrongly assume that the timer was hit before the other end thought
it was time to send a session refresh. This resulted in prematurely
ended calls.
This changes the session timer to start counting first at 200 like RFC
says it should.
(Also removed a few excess NULL checks that would never hit, because if
they did, asterisk would have crashed already.)
ASTERISK-22551 #close
Reported by: i2045
Review: https://reviewboard.asterisk.org/r/3562/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The fix for ASTERISK-12292 was a bit too aggressive. You could have
generators pointed at each other on local channels but need to get other
kinds of frames such as DTMF or CONNECTED_LINE frames accross.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Check if waitingfordt (waitfordialtone) is enabled in dahdi_read() to
allow the DSP to operate early enough to detect dialtone.
* Made use the correct variable in my_check_waitingfordt().
ASTERISK-23709 #close
Reported by: Steve Davies
Patches:
dialtone_detect_fix (license #5012) patch uploaded by Steve Davies
Review: https://reviewboard.asterisk.org/r/3534/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AST_FLAG_ANSWERED_ELSEWHERE was not propagated back from local channels.
It is now. That means that when a call is picked up from a callgroup of
local channels, the other channels will now properly see it as "picked up".
This occurs when you use a construct like Dial(Local/a@context&Local/b@context)
where a@context and b@context dial two chan_sip devices respectively. If one
device picks up, the other will not see "1 missed call" anymore. In this
respect, it now behaves the same as when doing Dial(SIP/a&SIP/b).
Review: https://reviewboard.asterisk.org/r/3540/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP. sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.
* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.
* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.
* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected. The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.
* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN. This helps interoperability with SIP.
* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available. This helps interoperability with SIP.
This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.
AST-1338 #close
Reported by: Tyler Stewart
Review: https://reviewboard.asterisk.org/r/3521/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.
(closes issue AST-1301)
(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
Review: https://reviewboard.asterisk.org/r/3447/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
THis patch fixes an issue in chan_oss when building on certain platforms. It
ensures that soundcard.h is found.
Review: https://reviewboard.asterisk.org/r/3426
Note that this patch is a part of the patch on ASTERISK-23576; the Makefile
portion only applies to Asterisk 11+.
(issue ASTERISK-23576)
Reported by: Sebastian Wiedenroth
patches:
fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior to this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
---
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
Every run will now blow away the previous run (as large ref files
sometimes caused issues). We now also no longer open/close the file
on each write, instead relying on fflush to make sure data gets written
to the file (in case the ao2 call being performed is about to cause a
crash)
(3) It goes with a comma delineated format for the ref debug file. This
makes parsing much easier. This also now includes the thread ID of the
thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
contrib/scripts folder.
Review: https://reviewboard.asterisk.org/r/3377/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross. Local channel optimization requires frames
flowing to trigger when optimization can happen. When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing. If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received. With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.
* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed. Asterisk now always uses internal
timing when needed if any timing module is loaded. The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used. The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.
* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().
ASTERISK-22846 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3414/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The allowed methods advertised by chan_sip did not previously note the MESSAGE
request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we
should advertise that we support MESSAGE requests.
ASTERISK-23504 #close
ASTERISK-23504 #comment Reported by: Martin Kontsek
ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
Review: https://reviewboard.asterisk.org/r/3396/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior too this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.
(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.
(closes issue ASTERISK-23373)
Reported by: Corey Farrell
Patches:
chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909)
chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a static realtime peer with qualify=yes is loaded, Asterisk will fail to
send an OPTIONS request due to the lastms being equal to 0. This results in
the peer being unable to receive calls from Asterisk because the status is
permanently UNKNOWN.
This patch allows an OPTIONS request to be sent during module load by
ignoring the lastms value on startup only.
Review: https://reviewboard.asterisk.org/r/3294/
(closes issue ASTERISK-17523)
Reported by: Maciej Krajewski
Tested by: wushumasters
patches:
realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix crash in ast_channel_hangupcause_set() because p->owner not checked
before calling. Regression introduced by the fix for ASTERISK-22621.
(closes issue ASTERISK-23135)
Reported by: OK
(issue ASTERISK-23323)
Reported by: Walter Doekes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.
Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.
The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.
Credit to Olle Johansson for pointing me in the right direction on this issue.
(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Backported from trunk/12.
Review: https://reviewboard.asterisk.org/r/726/
........
Merged revisions 394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
(closes issue ASTERISK-22988)
Reported by: adomjan
Original patch by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This restricts direct usage of global oseq so that all accesses are
locked and threads are not racing to get oseq values that they did not
claim.
This also fixes a build error in res_pktccops under dev mode.
(closes issue ASTERISK-23100)
Reported by: adomjan
Patch by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ASTERISK-12117, an improvement to insure consistant local from tags
on outbound registrations resulted in an undesirable behavior - caused
by leftover unexpired sip_pvt dialogs (with the previous cseq number),
resulting in many uncessary REGISTER requests. Instead of significant
rework of transmit_register(), this change deletes the dialogs after a
200 OK response indiciating a successful registration, keeping the old
dialogs from interfering with normal operation.
(closes issue ASTERISK-22946)
Reported by: Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/3109/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When performing a SIP transfer to a Park extension, if the Park fails, chan_sip
will currently not hang up either the transferer or the transfer target. This
results in the channels being orphaned with no thread to service frames,
resulting in stuck channels.
This patch immediately hangs up the two channels if a Park fails.
(closes issue ASTERISK-22834)
Reported by: rsw686
(closes issue ASTERISK-23047)
Reported by: Tommy Thompson
Review: https://reviewboard.asterisk.org/r/3107
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
