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1602 Commits

Author SHA1 Message Date
Joshua Colp
5a3acb0511 Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-23 03:00:12 +00:00
Russell Bryant
33235b40d6 Merge the changes from the /team/group/vldtmf_fixup branch. 
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged.  So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio.  However,
since there was no audio coming in, the DTMF_END was never generated.  This
caused DTMF based features to no longer work.

To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf).  If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.

Channel drivers also now get passed the length of the digit to their digit_end
callback.  This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.

(issue #8597, maybe others...)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-19 17:49:38 +00:00
Joshua Colp
1e3557c636 Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-18 18:36:35 +00:00
Russell Bryant
4244459e31 Merged revisions 51197 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines

Move the check for a failure of ast_channel_alloc() to before locking the
pvt structure again.  Otherwise, on a failure, this will cause a deadlock.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-17 21:18:35 +00:00
Joshua Colp
915f9315e1 Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-11 05:53:09 +00:00
Joshua Colp
240ca25bea Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-11 05:19:39 +00:00
Joshua Colp
9aca2b2a54 Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-10 18:32:29 +00:00
Olle Johansson
1a33c38a15 - handle re-invites properly in sip_hangup() 
- Add some invitestate status changes just to be sure


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-09 11:25:20 +00:00
Olle Johansson
3394598f93 Issue #8677 - Handle failure of T.38 re-invite 
This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).

To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-08 14:26:14 +00:00
Olle Johansson
0f96f73768 Issue #8524, support multiple via header values (tardieu) 
Thanks!


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-08 13:28:18 +00:00
Olle Johansson
be32fad9b8 We only need one forward declaration 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-08 09:08:10 +00:00
Olle Johansson
484add6554 Issue 8735: Terminate state when extension is unavailable for subscription 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-08 08:55:03 +00:00
Tilghman Lesher
dcbf36432e Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-07 21:24:04 +00:00
Kevin P. Fleming
444adcb477 reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-05 22:16:33 +00:00
Kevin P. Fleming
fb010e49aa Merged revisions 49635 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines

ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-05 17:09:00 +00:00
Olle Johansson
5edb7fa173 Small cleanup of add_t38sdp - it's always enabled at that point in the code 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-02 19:58:45 +00:00
Olle Johansson
7db2ca152c remove incomplete implementation of dnsmgr. Let's fix this in trunk. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-01 20:14:33 +00:00
Olle Johansson
1e9c141c2d Only include acl.h and lock.h once 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 21:06:48 +00:00
Olle Johansson
4f157be79e Only set rfc2833compensate flag once (handle_common_options) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 20:27:59 +00:00
Olle Johansson
f1f8bbaabe - Remove checking for T38 options twice. Keeping them in handle_common_options 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 20:24:03 +00:00
Kevin P. Fleming
0f5aae9688 make the option actually match the documentation 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 18:33:22 +00:00
Olle Johansson
d2b7e8b247 Be a bit more politically correct 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 17:59:53 +00:00
Olle Johansson
bfe4bb0f1e Issue #8575 - Buggy cisco MWI support. 
Normally we try not to change our software for bugs in other devices. But in
this case, the Cisco phones are so widespread so we try to implement a fix while
waiting for a bugfix from Cisco.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 16:49:45 +00:00
Olle Johansson
e25756dfac - Make sure handle_common_options return 1 when we found a common option 
- Move uncommon (only global) option away from handle_common_options
Reported by rizzo. Thanks!


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 16:31:55 +00:00
Olle Johansson
4ea530f2dc Issue 8599 (rizzo) Change invitestate before re-sending invite with auth. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 16:12:21 +00:00
Olle Johansson
8a42650605 Fix bogus content-length in t38 sdp. (rizzo, #8600) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 15:58:13 +00:00
Joshua Colp
9cc04e026d Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-26 04:31:58 +00:00
Russell Bryant
1208869c00 Merged revisions 48939 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines

Remove a couple of misplaced dots in log messages.  This was reported by
Andrea Spadaccini on the asterisk-dev mailing list.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-24 06:49:31 +00:00
Olle Johansson
c1f1cf48ac Issue #8592 - treat 504 as 503 (imported from 1.2) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-15 10:55:41 +00:00
Olle Johansson
a8fe77b95b Update to latest IANA spec 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-15 10:18:04 +00:00
Olle Johansson
28c36660c7 Issue #8258 - fix handling of 487 being retransmitted to Asterisk 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-06 16:25:06 +00:00
Olle Johansson
0fe91149ae Don't send Contact on MESSAGE 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-06 12:27:43 +00:00
Olle Johansson
694205de93 Merging the invitestate-1.4 branch after successful testing. 
Will check if I can solve this with less changes in 1.2.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-05 17:29:43 +00:00
Olle Johansson
9636bccb7c - Removing one of two pieces of code to handle 481 response on INVITE 
- Move handling of REFER response to handle_response_refer()


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-02 21:45:58 +00:00
Olle Johansson
f89143bd13 - Disable RTP hold timers while T.38 fax transmission happens 
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
   The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
   something that video phones support in the RTP stream.
   I now this is a big architectual change at this stage for 1.4, but decided it was needed
   to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample

Issue 7679 in the bug tracker. Please test.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-02 11:32:51 +00:00
Olle Johansson
98d3fb64ed - Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states 
- Remove support for T.38 early media, since it's impossible.

(Two patches in one - extra friday evening offer due to being off line from svn today... :-)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-01 17:41:56 +00:00
Olle Johansson
76e35bac11 Issue 8319 - change noncecount before using it. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-30 20:51:04 +00:00
Olle Johansson
073e937757 Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line 
in the REGISTER request. Imported from 1.2.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-30 09:12:30 +00:00
Olle Johansson
9436ecaad7 Change logging message 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-27 06:41:33 +00:00
Olle Johansson
5a3dcaff3c Not having a HINT is not an ERROR. In 1.4 and future releases, you can disable 
subscription support totally or per peer in sip.conf with allowsubscribe = yes | no



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-25 09:28:28 +00:00
Olle Johansson
c3a7e9d7c2 Don't allocate unused variable. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-23 11:01:47 +00:00
Olle Johansson
5c0be47e69 Apparently Exosip sends a 101 after a 100 provisional response. Let's not treat that as 
early media. (discovered at the AVTF meeting in Paris).


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-21 15:20:03 +00:00
Olle Johansson
a7c1ac60b7 Make sure we destroy scheduled items and not use them ever again after destruction (rizzo) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-20 11:29:19 +00:00
Luigi Rizzo
d20e81d829 fix bug 7450 - Parsing fails if From header contains angle brackets 
(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-18 17:59:35 +00:00
Joshua Colp
f373cad528 Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-16 21:11:06 +00:00
Olle Johansson
e1e6a1b2a8 Make the HOLD notification optional, in order to avoid a lot of extra database lookups 
for all those realtime users out there.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-16 19:24:23 +00:00
Olle Johansson
d1ee29fc20 Don't fixup if there's nothing to fixup 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-16 16:51:59 +00:00
Olle Johansson
5bd53e3588 - CANCEL is never authenticated (according to the RFC) 
- Update docs on canreinvite. "nonat" is the recommended setting for most users with
  phones behind a NAT.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-16 15:03:49 +00:00
Olle Johansson
c78840b900 - Improve SIP history 
- Never send reply to ACK (again...)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-15 20:31:11 +00:00
Olle Johansson
63f87d112d Send error message when we can't allocate SIP dialog, possibly due to limitation of file descriptors. 
(imported from 1.2)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-15 15:25:50 +00:00