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1908 Commits

Author SHA1 Message Date
Tilghman Lesher
0b289ed1b6 Back out changes from revision 97077, since it wasn't perfect 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-11 15:52:31 +00:00
Tilghman Lesher
7f61492a42 1) When we get a translated frame out, clone it, because if the 
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running.  Don't just ignore it.
(Closes issue #11698)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@97973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-10 23:08:36 +00:00
Tilghman Lesher
1b289c2998 Apply multiple crash fixes, found in issue #11386, but not completely 
closing that issue.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@97077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-08 18:02:13 +00:00
Tilghman Lesher
58cc575e52 If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie. 
Reported and patched by: one47
(Closes issue #11535)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@96525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-04 19:27:25 +00:00
Joshua Colp
b5087d7743 Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001) 
(closes issue #11637)
Reported by: greyvoip


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@95946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-01-02 20:24:09 +00:00
Russell Bryant
ce3f27c5d5 Remove duplicate increment of the header count in the add_header() function. 
(closes issue #11648)
Reported by: makoto
Patch provided by sergee, committed patch by me, inspired by comments from putnopvut


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@95191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-28 18:24:59 +00:00
Joshua Colp
856bc2ef76 Use ast_strlen_zero to see if our_contact is set or not on the dialog. It is possible for it to be a pointer to NULL. 
(closes issue #11557)
Reported by: FuriousGeorge


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-27 17:27:11 +00:00
Joshua Colp
25aef5fc44 file says... build on the builders. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-25 02:27:08 +00:00
Tilghman Lesher
9dbd385de7 More deadlock avoidance code (this time between sip_monitor and sip_hangup) 
Reported by: apsaras
Patch by: tilghman
(Closes issue #11413)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-24 16:17:01 +00:00
Tilghman Lesher
f26935218f Another bit of bad logic in realtime_peer 
Reported by: dimas
Patch by: dimas
(Closes issue #11631)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-24 15:39:56 +00:00
Tilghman Lesher
1e5463be20 Argh... I suppose third time's the charm. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-23 01:21:03 +00:00
Tilghman Lesher
be17d99712 Merged revisions 93667 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) | 2 lines

Fixing AST-2007-027 (Closes issue #11119)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@93668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-18 18:29:39 +00:00
Olle Johansson
517dacce70 Issue 11574: Add dependencies on res_monitor and res_features. 
I wonder if Asterisk can run at all without res_features. My guess is that 
there's propably a lot of more modules and the core that depends on it.

Reported by: caio1982
(closes issue #11574)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@93182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-17 07:15:13 +00:00
Joshua Colp
6a2c388e18 Up the length of the format on the SIP channel since it can now be rather long. 
(closes issue #11552)
Reported by: francesco_r


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-14 15:16:15 +00:00
Jason Parker
8771f23fff If a typo is found in a config file, we previous continued on with what was already loaded. 
We do not want to do this (see bug below for details).

This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded.

Isue #10690.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-13 00:11:09 +00:00
Joshua Colp
6fcaf2bb4b It is possible for nativeformats to contain more then one codec, so print out multiple ones. 
(closes issue #11366)
Reported by: ovi


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-10 16:13:43 +00:00
Olle Johansson
7efa6b1fe0 Avoid reinvite race situations with two Asterisks trying 
to reinvite each other in 1.4 and trunk. 

This patch implements support for the 491 error code that
Asterisk 1.4 generates on situations where we get an 
incoming INVITE and already has one in progress.

Thanks to mavetju for reporting and to Raj Jain for an
excellent explanation of the problem.

Patch by myself. Tested with 8 Asterisk servers connected
to each other in a training network.

Closes issue #10481



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-10 14:04:44 +00:00
Joshua Colp
4edfc25a1f Add support for accepting and sending T.38 in the initial INVITE. 
(closes issue #9402)
Reported by: thdei


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-12-06 16:14:26 +00:00
Joshua Colp
daf94226d5 Fix locking issues under one legged replaces scenarios. 
(closes issue #11420)
Reported by: irroot
Patches:
      chan_sip_oneleg.patch uploaded by irroot (license 52)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-30 14:43:15 +00:00
Olle Johansson
0c3ec937ce If we get a codec offer using a well-known payload type, but using it for another 
codec that we don't know, Asterisk did not remove that codec from the list.

With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.

(closes issue #11376)
Reported by: lasse
Patches: 
      bug11376.txt uploaded by oej (license 306)
Tested by: lasse


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-27 15:23:17 +00:00
Olle Johansson
9d3be20a0d Don't send re-invites during pending INVITE transactions. 
Patch by one47 - thanks!

Closes issue #9305


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-15 11:26:22 +00:00
Olle Johansson
fc49ddab3a Improve support for multipart messages. Code by gasparz, changes 
by me (mostly formatting). Thanks, gasparz!

Closes issue #10947


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-15 11:15:09 +00:00
Tilghman Lesher
4332b72082 If we set a value for qualify, we should actually pay attention to it, instead of overriding the value 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-13 17:34:11 +00:00
Tilghman Lesher
1a052e0498 Fix two cases of memory corruption caused by background threads. 
Reported by: atis
Patch by: tilghman
Fixes issue #10923


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-12 17:29:17 +00:00
Mark Michelson
3b75ff9010 Rework of the commit I made yesterday to use the already built-in 
ast_uri_decode function as opposed to my home-rolled one. Also added
comments.

Thanks to oej for pointing me in the right direction



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-08 21:00:08 +00:00
Joshua Colp
3aea241b63 Do not add a sip: to the beginning of the To URI unless needed. 
(closes issue #10756)
Reported by: goestelecom


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-08 02:26:48 +00:00
Joshua Colp
0f1ef85f9a Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. 
(closes issue #10164)
Reported by: nic_bellamy
Patches:
      sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-08 01:28:56 +00:00
Joshua Colp
53fd91490e Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. 
(closes issue #10946)
Reported by: flefoll
(closes issue #10915)
Reported by: ramonpeek
(closes issue #9567)
Reported by: atca_pres


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-08 01:11:25 +00:00
Joshua Colp
f1309f2c3f If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. 
(closes issue #11185)
Reported by: spditner


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-07 23:53:25 +00:00
Mark Michelson
0d76379f54 This patch makes it possible for SIP phones to dial extensions defined with '#' characters 
in extensions.conf AND maintain their escaped characters when forming URI's

(closes issue #10681, reported by cahen, patched by me, code review by file)



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-07 22:40:35 +00:00
Joshua Colp
cac21aa19b Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. 
(closes issue #11006)
Reported by: pj


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-06 17:08:05 +00:00
Russell Bryant
ea00780d49 When traversing the list of channel variables here in transmit_invite(), the 
asterisk channel must be locked, as this data may change at any time.

(I have seen numerous reports of crashes related to the handling of channel
variables.  There are a couple of issues on the bug tracker related to it,
but it has also been noted on IRC and mailing lists.  So, I am finding and
fixing some places where channel variables are handled improperly.) 


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-05 21:33:56 +00:00
Russell Bryant
69e42e6096 Fix up some indentation. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-05 21:21:39 +00:00
Joshua Colp
10c172eb00 If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. 
(closes issue #11085)
Reported by: francesco_r
Tested by: blitzrage
(closes issue #10474)
Reported by: acennami


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-05 18:47:13 +00:00
Jason Parker
7fa3a53ec2 Make sure we destroy the config structure on configuration failure. 
Issue 11163, patch by eliel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-05 17:19:41 +00:00
Luigi Rizzo
a0edff2477 Rename ast_string_field_free_pool to ast_string_field_free_memory, 
and ast_string_field_free_all to ast_string_field_reset_all
to avoid misuse (due to too similar names and an error in
documentation). Fix two related memory leaks in app_meetme.

No need to merge to trunk, different fix already applied there.
Not applicable to 1.2



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-04 22:38:13 +00:00
Joshua Colp
986e8a3ae8 Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context). 
(closes issue #10702)
Reported by: slavon


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-02 20:49:45 +00:00
Joshua Colp
e49faa404a If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. 
(closes issue #5747)
Reported by: chandi
Patches:
      infofix-81430-1.patch uploaded by IgorG (license 20)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-02 20:20:21 +00:00
Joshua Colp
143536912e Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason. 
(closes issue #9566)
Reported by: atca_pres
Patches:
      bug9566.patch uploaded by oej


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@87342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-29 17:20:28 +00:00
Joshua Colp
d6d7384355 After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well. 
(closes issue #10113)
Reported by: makoto


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-22 16:35:22 +00:00
Joshua Colp
a5b69b3583 Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote. 
(closes issue #11027)
Reported by: ramonpeek
Patches:
      11027-1.diff uploaded by ramonpeek (license 266)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-19 15:33:49 +00:00
Joshua Colp
918ee9ae5b Set port number in received as information for registrations as well. 
(closes issue #11028)
Reported by: brad-x


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-19 15:08:12 +00:00
Russell Bryant
ed426a7165 If Asterisk is in the middle of shutting down, respond to OPTIONS 
with 503 Unavailable.

(closes issue #10994)
Reported by: eserra
Patches:
      sip-options-503.patch uploaded by eserra (license 45)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-17 17:57:45 +00:00
Joshua Colp
faf4b212eb Whoops, forgot to remove the original sip_scheddestroy. 
(closes issue #11010)
Reported by: vadim


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-17 16:58:03 +00:00
Joshua Colp
c9638888aa Don't schedule dialog destruction if a MESSAGE is received using an existing dialog. 
(closes issue #11010)
Reported by: vadim


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-17 15:06:36 +00:00
Russell Bryant
3ba62c30b9 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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-15 16:54:57 +00:00
Joshua Colp
437a6ffcd3 When creating a new packet don't try to stop retransmission of it. It was just allocated/created so it's impossible for it to have already been scheduled. 
(closes issue #10945)
Reported by: flefoll
Patches:
      chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll (license 244)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-11 15:26:20 +00:00
Joshua Colp
3b9fafc1e0 If devicestate is passed a port number strip it out. 
(closes issue #10930)
Reported by: ibc


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-10 14:42:00 +00:00
Joshua Colp
e568fc66d9 Add support for handling a 182 Queued response. 
(closes issue #10924)
Reported by: ramonpeek
Patches:
      queued-182.diff uploaded by ramonpeek (license 266)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-10 14:28:18 +00:00
Joshua Colp
474e0e8d12 Don't perform a reinvite if a transfer is in progress. 
(issue #10915)
Reported by: ramonpeek


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-09 14:30:16 +00:00