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)
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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 ...
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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
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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.
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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!)
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Reported by: dimas
Patches:
chan_sip.patch uploaded by dimas (license 88)
Read in subscribecontext option in general to be the default.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@83070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also changed load_module function to use defines rather than numbers for return values.
Issue 10651, patch by rbraun_proformatique, with additions by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
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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)
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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).
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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.
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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.
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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)
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ast_read to return NULL in the case that the channel has been hung up.
(crash reported by anonymouz666 on IRC in #asterisk-dev)
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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).
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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)
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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)
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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!
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
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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.
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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.
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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
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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)
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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)
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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.
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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)
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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)
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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.
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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)
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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. :)
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
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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)
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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)
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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)
........
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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)
........
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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.
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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)
........
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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.
........
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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)
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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)
........
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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.
........
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
........
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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)
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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.
........
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | 1 line
isdn_lib.c didn't compile
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69944 65c4cc65-6c06-0410-ace0-fbb531ad65f3