Don't improperly memset() over an ast_str. This was leftover from before it
got changed to use ast_str.
(closes issue #11003, reported by pj)
(closes issue #10770, reported by yehavi)
(patched by me)
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SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Modifications :
- set bridge type to partial ;
- process media candidates from the remote peer properly.
Now we have Jingle audio, at least between two Asterisk Jingle
clients.
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r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines
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)
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in the Dial command. The 'j' option _must_ be used in conjunction with the 'n'
option.
This feature will allow you to use the existing jitterbuffer implementation to
put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by
putting a local channel in the middle.
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r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | 6 lines
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 ...
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r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6 lines
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
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a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string
(closes issue #10843)
Reported by: Corydon76
Patches:
2007092900_10843.diff uploaded by mvanbaak (license 7)
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r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | 5 lines
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.
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No real Jingle implementation being available, testing was made using
two Asterisk servers relaying SIP calls over their Jingle channels:
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Thus, it was possible to test the code in both ways, and make the
Jingle channel comply with the latest specifications. No sound available yet.
Main modifications include :
- modified the 'jingle_candidate' structure and the
'jingle_create_candidates' function according to XEP-0176 ;
- modified the 'jingle_action' function in order to properly terminate
a Jingle session, in conformance with XEP-0166 ;
- modified username format used in STUN requests ;
- actually make the bindaddr configuration field useable.
Todo :
- set audio paths up (no native bridging) ;
- make the CLI gtalk functions available to jingle ;
- clean up the storage space used in strings.
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
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)
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r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) | 8 lines
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!)
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r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) | 1 line
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.
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r83024 | crichter | 2007-09-19 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line
removed comment which violates the coding guidelines.
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This is theoretically a potential deadlock, but it's the way it was before so
I'm going to leave it this way for now.
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Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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Joshua Colp
Jason Parker
Matthew Fredrickson
Russell Bryant
Philippe Sultan
Tilghman Lesher
Kevin P. Fleming
Dwayne M. Hubbard
Christian Richter
Steve Murphy
Mark Michelson