Reported by: casper
Patches:
chan_sip.c.82076.diff uploaded by casper (license 55)
Remove double check for zombie flag and optimize things a bit.
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r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) | 16 lines
(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.
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(closes issue #10651)
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r81523 | qwell | 2007-09-05 10:14:30 -0500 (Wed, 05 Sep 2007) | 5 lines
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.
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r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) | 4 lines
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.
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r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007) | 2 lines
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
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r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) | 5 lines
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)
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r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6 lines
(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.
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Reported by: tootai
Make sure the external IP address has the standard SIP port set for when the user does not specify the port in the externip setting.
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r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) | 11 lines
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.
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r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27 Aug 2007) | 7 lines
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)
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r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6 lines
(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.
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r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) | 7 lines
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.
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r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) | 5 lines
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)
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r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 Aug 2007) | 9 lines
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).
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r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | 34 lines
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.
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r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) | 7 lines
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)
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r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) | 4 lines
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)
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(closes issue #10430)
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r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines
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).
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in place of a very common construct. I also used it in a number of places
in chan_sip.
if (id > -1)
ast_sched_del(sched, id);
id = ast_sched_add(sched, ...);
changes to:
ast_sched_replace(id, sched, ...);
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Joshua Colp
Russell Bryant
Philippe Sultan
Matthew Fredrickson
Jason Parker
Kevin P. Fleming
Tilghman Lesher
Christian Richter
Steve Murphy
Mark Michelson