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1674 Commits

Author SHA1 Message Date
Joshua Colp
7a4bed883e Merged revisions 55073 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines

Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-17 01:16:59 +00:00
Olle Johansson
3ca445e34c Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-16 12:06:23 +00:00
Russell Bryant
7bcf1d913a Remove a couple of leftover debug messages 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-13 21:31:22 +00:00
Russell Bryant
b100b69703 If we fail to create the SIP socket, then return -1 from reload_config() so 
that load_module() will return AST_MODULE_LOAD_DECLINE.  Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-13 19:42:00 +00:00
Russell Bryant
93fcd4a354 Change some text to properly state "On Hold", which was already done in trunk. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-10 00:41:57 +00:00
Russell Bryant
7ee02f585d Merge team/russell/sla_rewrite 
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4.  It is now functional and ready for testing.  However, I will be
adding some additional features over the next week, as well.

For information on how to set this up, see configs/sla.conf.sample 
and doc/sla.txt.

In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:

chan_sip:
 - Add the ability to indicate HOLD state in NOTIFY messages.
 - Queue HOLD and UNHOLD control frames even if the channel is not bridged to
   another channel.

linkedlists.h:
 - Add support for rwlock based linked lists.

dial.c:
 - Add the ability to run ast_dial_start() without a reference channel to
   inherit information from.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-10 00:35:09 +00:00
Olle Johansson
e7a0e86756 Formatting 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-05 23:43:59 +00:00
Olle Johansson
8e07358edf Add some comments on queue system behaviour and how it affects the 
SIP channel


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-05 00:18:34 +00:00
Joshua Colp
910898b7be Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-03 21:05:02 +00:00
Olle Johansson
90a4b844a9 Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now 
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-02 00:24:03 +00:00
Joshua Colp
57fe6882ac Merged revisions 53103 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines

Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 22:24:32 +00:00
Joshua Colp
09844a7f1a Merged revisions 53095 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines

Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) 

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 21:54:28 +00:00
Olle Johansson
97efd0be22 - Clean INC_COUNT flag when we decrement call counter 
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
  before we destroy the dialog


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 21:05:34 +00:00
Olle Johansson
6bb6bba6a3 Cleaning up the devicestate callback function 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 20:28:54 +00:00
Joshua Colp
e86275c11c Fix silly logic. We really want to write UDPTL frames out when the call is up. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 17:37:44 +00:00
Russell Bryant
9aab046002 Merged revisions 53045 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines

Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-31 21:32:08 +00:00
Russell Bryant
29b7393d84 Only set the DTMF flag on the rtp structure if the DTMF mode is actually 
RFC2833, not just that it is not INFO.  This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-30 19:33:12 +00:00
Joshua Colp
ed48c69f06 Drop out variables I accidentally put in. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-25 17:49:39 +00:00
Joshua Colp
6b08afd05d Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-25 17:14:53 +00:00
Joshua Colp
5ebd1ecf63 Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-24 17:59:55 +00:00
Olle Johansson
a207a31a97 Show capabilities *and* preference in general settings in "sip show settings" 
(reported by Clona/Telio - Thanks!)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-24 09:30:21 +00:00
Joshua Colp
8f7ddbef0d Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-23 22:46:31 +00:00
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

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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