If a SS7 call comes in requesting a CIC that is in-alarm, the call is
accepted and connects if the extension exists in the dialplan. The call
does not have any audio.
* Made release the call immediately with circuit congestion cause.
(closes issue ASTERISK-20204)
Reported by: Tuan Le
Patches:
jira_asterisk_20204_v1.8.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r375519 | rmudgett | 2012-10-30 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines
chan_misdn: Timer primitives must be handled first.
The frm->addr is a different "address space" than the stack/instance
address of other Lx primitives. The test for B channel instance address
could fail.
Patches:
patch01_timers.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2888
........
r375520 | rmudgett | 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines
chan_misdn: Free memory in error paths and other memory leaks.
The one line commented with BUG is not easily fixable because there is no
de-init function one can call.
Patches:
patch02_memory.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2888
........
r375521 | rmudgett | 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines
chan_misdn: ISDN NT L2 de-establish/establish
* An NT-PTMP cannot de/establish L2 since it doesn't know the TEIs.
* On NT-PTP L2 is started when L1 is finally active in handle_l1.
* L2 deactivation logging cleanup.
* L2 aggregate link status is unknown for NT-PTMP, show as "UNKN".
* Removed unused functions and code for L2 handling.
Patches:
patch03_L2estab.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2888
........
r375522 | rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22 lines
chan_misdn: Fix broken upper_id/lower_id usage.
Sending PH prim via lower_id layer (3 or 1) simply does not work. For TE
(3) it returns an error (len=-6) which is not evaluated by handle_l1(), so
the L1 layer status ends up wrong. Instead PH must be sent via L4, only
then does it reach L1 without an error message.
And NT PH prims only reach L1 when they are sent to layer 2 id.
--> use upper_id to send PH primitives.
* Check for errors in PH_(DE)ACTIVATE | CONFIRM.
* Debug messages are improved.
* The lower_id is now not used for anything, except: Why is lower_id layer
deleted when it wasn't created? I removed this code since it looks very
wrong.
Patches:
patch04_l1activation.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2888
........
r375523 | rmudgett | 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines
chan_misdn: Fix loss of B channels if L1 is down.
If you make 2 calls out an NT PTMP port which is not connected to any
phone, the B channel associated with that call becomes unusable until
Asterisk is restarted.
The problem is the EVENT_SETUP is queued when L1 is not up in
misdn_lib_send_event(). If L1 cannot be activated the event won't be
dequeued. It gets even worse when the call is hung up. The queued
EVENT_SETUP will be overwritten by an EVENT_DISCONNECT. The reserved B
channel then will never be freed. If later someone connects a phone to
the port, L1 will eventually activate and the queued EVENT_DISCONNECT is
sent down the stack. However, it is ignored because it is the wrong call
state.
The real fix would be that activation and queueing for a new SETUP is done
by the NT stack. But since it doesn't, the workaround must be removed
because it doesn't always work.
Fix: The event is no longer queued but immediately sent to the stack. If
L1 cannot be activated, the L3 state machine that was started by the
EVENT_SETUP will do its work, i.e. a timeout will release the B channel
properly. The SETUP possibly cannot be sent the first time but is resent
by T303 in case L1 could be activated.
Patches:
patch05_bchan-loss.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2888
........
r375524 | rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13 lines
chan_misdn: Remove some calls to exit().
Try proper cleanup when something goes wrong in misdn_lib_init().
Especially do not call exit()!
* Fix memory leak because stack_destroy() does not free the stack struct.
Patches:
patch06_cleanup-init.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2888
........
Merged revisions 375519-375524 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While looking at some debug logs, I noticed that it was being reported that the
SDP origin line was unsupported or failed. Upon looking into this on my local
machine, I found that I too was getting this debug message yet everything seemed
to be getting processed properly. What was discovered is, that, the variable to
determine what is displayed in the debug message for the SDP line that was
processed, was not being set for the origin line when the result was successful.
This patch fixes this and was tested on local machine.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a "sip reload" is issued for a SIP peer, then his
IP address will be cleared, thus resulting in forgetting the
public IP address. Asterisk will then attempt to route SIP
traffic to the private IP address.
The fix here is to make "sip reload" ignore realtime peers
when "host = dynamic" is spotted. Realtime peers can now only
have their IP address reset if they have gone from being not
dynamic to being dynamic.
(closes issue ASTERISK-18203)
reported by daren ferreira
(closes issue ASTERISK-20572)
reported by JoshE
Patches:
fix_nat_realtime.diff uploaded by JoshE (license #6075)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The tech support customer was using the AMI Redirect action shortly after
a call was placed. While the channel tried to do an ast_read(), the
masquerade resulting from the channel redirect took place. The masquerade
in the middle of the ast_read() resulted in the segfault.
(closes issue AST-1025)
Reported by: Trey Blancher
Patches:
jira_ast_1025_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.
This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.
Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.
Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.
(closes issue ASTERISK-20212)
reported by Phil Ciccone
Review: https://reviewboard.asterisk.org/r/2123
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
................
r374515 | rmudgett | 2012-10-04 17:52:36 -0500 (Thu, 04 Oct 2012) | 10 lines
chan_misdn: Remove some deadcode
* Made setup_bc() static.
Patches:
patch1_unused-code.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2882
................
r374516 | rmudgett | 2012-10-04 18:01:01 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Remove unused bchan states
Patches:
patch2_unused-states.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) | 16 lines
chan_misdn: Remove unnecessary null pointer checks and checks for stack->nt
* cleanup_bc() is always called with valid bc (or it would've crashed
before).
* Value of stack->nt is known in advance at some places.
* Rename handle_event() to handle_event_te(), handle_frm() to
handle_frm_te().
Patches:
patch3_checks.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
JIRA ABE-2882
................
r374518 | rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Fix spelling in log messages
Patches:
patch4_spelling.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374519 | rmudgett | 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
chan_misdn: Don't cleanup a bc twice.
In handle_frm_te() after calling misdn_lib_send_event(bc,
EVENT_RELEASE_COMPLETE) bc is emptied, cleaned and set not in use,
although misdn_lib_send_event() already did the same. This is bad. When
it's not in use we are not allowed to touch it.
* Moved log message in front of the resulting actions and fixed it to
match the case.
Patches:
patch5_bccleanup.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) | 12 lines
chan_misdn: Fix memory leaks, bc, chan not cleaned up etc., really bad stuff.
* Fix return codes of cb_events() for EVENT_SETUP to use caller's cleanup
mechanisms.
* Move cl_queue_chan() call after bearer check.
Patches:
patch6_leaks.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374521 | rmudgett | 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
chan_misdn: We must initialize cause on sending a DISCONNECT.
We must initialize cause on sending a DISCONNECT, so it is later correctly
indicated to ast_channel in case the answer (RELEASE/RELEASE_COMPLETE)
does not include one.
Patches:
patch7_hangupcause.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374522 | rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Remove unused code for upqueue
Patches:
patch8_unused-upqueue.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374523 | rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
chan_misdn: Improve debugging (port number, messages fixed, dups removed)
Patches:
patch9_debug.diff (license #6372) patch uploaded by Guenther Kelleter
JIRA ABE-2882
................
r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) | 8 lines
chan_misdn: Better debug: we can print_bc_info even if there's no ast leg.
Patches:
patch10_debug-bc-2.diff (license #6372) patch uploaded by Guenther Kelleter
Modified.
JIRA ABE-2882
................
r374534 | rmudgett | 2012-10-05 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines
chan_misdn: setup_bc() is called too early for an incoming SETUP on TE.
This prevents the B channel from being setup for HDLC mode when requested
by the bearer capability and config option hdlc=yes. It violates
ETS300102 Ch.5.2.3.2: "The user, in any case, must not connect to the
channel until a CONNECT ACKNOWLEDGE message has been received."
* Call setup_bc() on receipt of CONNECT_ACKNOWLEGDE for PTMP, and on first
response to SETUP for PTP.
Patches:
abe-2881-2.diff (license #6372) patch uploaded by Guenther Kelleter
Modified.
JIRA ABE-2881
................
r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) | 2 lines
chan_misdn: Remove some more deadcode.
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A check was added for direct media ACLs that immediately forbid remote bridging if there
was no bridged channel. This caused directrtpsetup to no longer function as it needs this
information before bridging actually occurs.
Logic has now been adjusted so if there is no bridged channel a remote bridge will still
be attempted.
(closes issue ASTERISK-20511)
Reported by: kristoff
Review: https://reviewboard.asterisk.org/r/2146/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The chan_local channel driver returns a device state of in use even if a created Local
channel has not yet been dialed. This fix changes the logic to return a state of not
in use until the channel itself has been dialed.
(closes issue ASTERISK-20390)
Reported by: tim_ringenbach
Review: https://reviewboard.asterisk.org/r/2116/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
multiplelogin was removed from chan_agent back in 1.6.0 when
AgentCallbackLogin() was removed.
(closes issue AST-948)
reported by Steve Pitts
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Users of the T.38 API can indicate AST_T38_REQUEST_PARMS on a channel to request that the
channel indicate a T.38 negotiation with the parameters present on the channel. The return
value of this indication is expected to be AST_T38_REQUEST_PARMS upon success but with
chan_local involved this could never occur.
This fix changes chan_local to always return AST_T38_REQUEST_PARMS for this situation. If
the underlying channel technology on the other side does not support T.38 this would have
been determined ahead of time using ast_channel_get_t38_state and an indication would
not occur.
(closes issue ASTERISK-20229)
Reported by: wdoekes
Patches:
ASTERISK-20229.patch uploaded by wdoekes (license 5674)
Review: https://reviewboard.asterisk.org/r/2070/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SIP session timer mechanism contains a mandatory 'refresher' parameter
(included in the Session-Expires header) which is used in the session timer
offer/answer signaling within a SIP Invite dialog. It looks like asterisk is
interpreting the uac resp. uas role only as the initial role of client and
server (caller is uac, callee is uas). The standard rfc 4028 however assigns
the client role to the ((RE)-Invite) requester, the server role to the
((RE)-Invite) responder.
This patch has Asterisk track the actual refresher as "us" or "them" as opposed
to relying on just the configured "uas" or "uac" properties.
(closes issue AST-922)
Reported by: Thomas Airmont
Review: https://reviewboard.asterisk.org/r/2118/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When setting CALLERID(pres)=unavailable in the dialplan, the From header
in the SIP message contains "Anonymous" <sip:Anonymous@anonymous.invalid>.
For consistency, Asterisk should use a lowercase a in the userpart of the
URI.
* Make the From header use a lowercase A in the userpart of the anonymous
URI.
(closes issue ASTERISK-19838)
Reported by: Antti Yrjola
Patches:
chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If conditions were right it was possible for both the PBX core and chan_sip to deadlock by both having a lock that the other
wants. In the case of the PBX core it had the contexts lock and wanted a SIP dialog lock, while in the case of chan_sip it
had the SIP dialog lock and wanted the contexts lock.
This fix unlocks the SIP dialog before getting the extension state so that the other thread will not block on trying to lock
it. Once the extension state is retrieved the SIP dialog is locked again and life carries on.
As the SIP dialog is reference counted it is not possible for it to go away after unlocking.
(closes issue ASTERISK-20437)
Reported by: jhutchins
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk v1.8 and later was not as vulnerable to this issue.
* Made find_call() lock each private as it processes the found dialogs.
(Primary cause of ABE-2876)
* Made the other functions that traverse the dialogs container lock each
private as it examines them.
* Fix race condition in sip_call() if the thread that sent the INVITE is
held up long enough for a response to be processed. The p->initid for the
INVITE retransmission could be added after it was canceled by the response
processing.
* Made __sip_destroy() clean up resource pointers after freeing. This is
primarily defensive in case someone has a stale private pointer.
* Removed redundant memset() in reqprep(). The call to init_req() already
does the memset() and is the first reference to req in reqprep().
* Removed useless set of req.method in transmit_invite(). The calls to
initreqprep() and reqprep() have to do this because they memset() the req.
JIRA ABE-2876
..........
Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A change was committed to fix direct media ACL support. This change wrongly assumed that
only a single channel technology structure exists for chan_sip. This is in fact false as
a second exists for calls using SIP INFO DTMF. The code which performs direct media ACL
checking now checks for both the non-INFO DTMF and INFO DTMF channel technology structures.
(closes issue ASTERISK-20409)
Reported by: michele cicciotti privatewave
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For SS7, the companding law for a call was chosen inconsistently depending
upon ss7type (ITU vs ANSI) and the DAHDI companding default (T1 vs E1).
For incoming calls, the companding law was determined by ss7type. For
outgoing calls, the companding law was determined by the DAHDI default.
With the wrong combination you would get A-law/u-law conflicts. An
A-law/u-law conflict sounds like bad static on the line.
SS7 ITU signaling with E1 line: ok
SS7 ITU signaling with T1 line: noise
SS7 ANSI signaling with E1 line: noise
SS7 ANSI signaling with T1 line: ok
* Fix the companding law used to be determined by the SS7 signaling type
only.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch resolves two sources of memory leaks when using TLS in Asterisk:
1) It removes improper initialization (and multiple re-initializations) of
portions of the SSL library. Asterisk calls SSL_library_init and
SSL_load_error_strings during SSL initialization; collectively this
obviates the need for calling any of the following during initialization
or client connection handling:
* ERR_load_crypto_strings (handled by SSL_load_error_strings)
* OpenSSL_add_all_algorithms (synonym for SSL_library_init)
* SSLeay_add_ssl_algorithms (synonym for SSL_library_init)
2) Failure to completely clean up all memory allocated by Asterisk and by
the SSL library for TLS clients. This included not freeing the SSL_CTX
object in the SIP channel driver, as well as not clearing the error
stack when the TLS client exited.
Note that these memory leaks were found by Thomas Arimont, and this patch
was essentially written by him with some minor tweaks.
(closes issue AST-889)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
patches:
(bugAST-889.patch) by Thomas Arimont (license 5525)
Review: https://reviewboard.asterisk.org/r/2105
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "autodestruct with owner in place" message is typically
indicative of a channel reference leak. Printing out the name
of the channel in the message may be helpful when trying to
debug the issue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes chan_local channels to use an 8 digit hex identifier generated
atomically and sequentially in order to eliminate the chance of having
multiple channels with the same name during high call volume situations.
(issue ASTERISK-20318)
Reported by: Dan Cropp
Review: https://reviewboard.asterisk.org/r/2104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When IAX2 debug was changed from iax_showframe to iax_outputframe,
some instances were missed (or added afterward). This was causing
debug output to not be displayed when expected.
(closes issue ASTERISK-20338)
Reported-by: John Covert
Patch-by: John Covert
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a CLI warning when a SDP offer is rejected due to UDPTL
initialization failure. Previously, there was no indication of the
reason for offer rejection in this case.
(closes issue ASTERISK-20357)
Reported-by: Francesco Usseglio Gaudi
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an
SDP offer and the ability to re-create an SRTP session when the crypto keys
changed. In certain circumstances - most notably when a phone is put on
hold after having been bridged for a significant amount of time - the act
of re-creating the SRTP session causes problems for certain models of phones.
The patch committed in r356604 always re-created the SRTP session regardless
of whether or not the cryptographic keys changed. Since this is technically
not necessary, this patch modifies the behavior to only re-create the SRTP
session if Asterisk detects that the remote key has changed. This allows
models of phones that do not handle the SRTP session changing to continue
to work, while also providing the behavior needed for those phones that do
re-negotiate cryptographic keys.
In addition, in Asterisk 1.8 only, it was found that phones that offer
AES_CM_128_HMAC_SHA1_32 will end up with no audio if the phone is the
initiator of the call. The phone will send an INVITE request specifying
that AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy; Asterisk
will set its policy to that value. Unfortunately, when the call is Answered
and a 200 OK is sent back to the UA, the policy sent in the response's SDP
will be the hard coded value AES_CM_128_HMAC_SHA1_80. This potentially
results in Asterisk using the INVITE request's policy of
AES_CM_128_HMAC_SHA1_32, while the phone uses Asterisk's response of
AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think the other
is crazy.
This patch fixes that by caching the policy from the request and responding
with it. Note that this is not a problem in Asterisk 10 and later, as the
ability to configure the policy was added in that version.
(issue ASTERISK-20194)
Reported by: Nicolo Mazzon
Tested by: Nicolo Mazzon
Review: https://reviewboard.asterisk.org/r/2099
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to 1.8, it was not necessary for an explicit "type" to be set for an
asterisk LDAP realtime peer. Now the routine find_peer actually checks the
type field during registration and fails to find the peer if it is not set.
The attached patches make the realtime type equal whatever type is being
searched for if the type is 0 upon return from routine build_peer.
(closes issue ASTERISK-17222)
Reported by: John Covert
Patch by: David Vossel
Tested by: Darren Sessions
Review: https://reviewboard.asterisk.org/r/2095/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.
This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.
(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes three main issues
* Change asprintf() uses to ast_asprintf() so that it
pairs properly with ast_free() and no longer causes
MALLOC_DEBUG to freak out.
* When ast_asprintf() fails, set the pointer NULL if
it will be referenced later.
* Fix some memory leaks that were spotted while taking
care of the first two points.
(Closes issue ASTERISK-20135)
reported by Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2071
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously the pvt SIP_OUTGOING flag was used instead, which will frequently
flip during reinvites.
(closes issue AST-897)
Reported by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Under certain conditions, a SIP transaction involving directmedia wouldn't
trigger a re-invite because the SDP answer was included in an ACK instead
of in a message that we would have triggered the invite with. This patch
just queues a source change control frame if the dialog is using
directmedia when we find sdp for an ACK.
(closes issue AST-913)
Reported by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The other instance of this bug was fixed by jcolp/file in r121496. If
we are destroying a dialog only set the MWI dialog pointer on the
related peer to NULL if it is the dialog currently being destroyed.
(closes issue ASTERISK-20119)
Patch-by: Misha Vodsedalek
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The flash-hook the bridged peer feature now correctly determines if the
bridged peer is another chan_dahdi channel, that it is an analog channel,
and that it has the correct signaling for an FXO port. It now also
flash-hooks the correct channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on the review request posted by Walter Doekes
(referenced lower in the commit message)
The main fix here is to treat the IPorHost portion of the dial
string as a temporary outbound proxy. This ensures requests
get sent to the proper location.
Due to the age of the request, some parts were no longer relevant.
For instance, the request moved outbound proxy parsing code into
a single method. This is done in a previous commit, so it was not
necessary to do again.
Also, the review request fixed some errors with regards to request
routing for CANCEL and ACK requests. This has also been fixed in
more recent commits.
(closes issue ASTERISK-19677)
reported by Walter Doekes
Review https://reviewboard.asterisk.org/r/1859
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With a large number of SIP peers registered, performing a SIP reload causes a
flood of SIP OPTIONS request packets. These are immediately sent out, and, as
responses come back, can cause peers to be flagged as 'lagged' due to handling
of the many response messages.
This fix prevents this "packet storm" and schedules the pokes for a random
time. That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting.
The committed patch has some very small modifications to the patch schmidts
wrote for the review.
(closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon
patches:
issue19154.patch license #6034 uploaded by schmidts
Review: https://reviewboard.asterisk.org/r/1652
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
