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3733 Commits

Author SHA1 Message Date
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

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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

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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
Christian Richter
60ad44b8e9 we update the name on any first reply of our setup 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 13:17:18 +00:00
Paul Cadach
6fd3418e40 chan_h323 is very stable, so let it built by default 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 11:07:41 +00:00
Russell Bryant
9aab046002 Merged revisions 53045 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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.

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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
362abb9deb Drop the deprecated show commands since the original ones were changed back. (issue #8937 reported by PCadach) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-30 15:29:50 +00:00
Paul Cadach
a175571e39 Revert reprecation of h.323 gk cycle command from pre-1.4 version instead of duplicated h323 cycle gk 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-30 08:46:31 +00:00
Russell Bryant
a46e6d74df Merged revisions 52762 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines

Fix the extraction of the timestamp from video frames.  It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795, mihai)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-30 00:15:50 +00:00
Joshua Colp
6b462c95c9 Merged revisions 52360 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines

Make the last context entry read in the dominant one. (issue #8918 reported by pj)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-27 00:08:18 +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
Christian Richter
9352eda642 added fix from #8899 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-24 12:57:22 +00:00
Christian Richter
6d2afb98b4 Merged revisions 51966 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 Jan 2007) | 1 line

fixed the busy problem (dialstatus was not busy when we called a busy extension)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-24 11:24:53 +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
Christian Richter
e65aeb27b5 Merged revisions 50495,50506 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | 6 lines

* more additions to make the RESTART message work
* added fix for misdn_call to allow SETUPs with empty 
  extensions, replaced the strtok_r functions with strsep for that 
  (inspired by Sandro Cappellazzo, thanks)


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r50506 | crichter | 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line

when we get L2 UP, the L1 is UP definitely too, so we set the L1 state up as well.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-23 10:28:54 +00:00
Christian Richter
19fdc9b14e manually merged r49922 and r50335, because of conflicts. this commint includes addition of the ISDN RESTART Message 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-23 10:05:54 +00:00
Paul Cadach
964d2d2e42 Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-23 06:51:51 +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
0e37d936da Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are 
actually the same thing.  So, a digit would have been interpreted incorrectly
here.  Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-19 19:08:25 +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
Dwayne M. Hubbard
b62be97f7a chan_zap compiles without libpri after committing 7877 patch 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-19 00:17:32 +00:00
Dwayne M. Hubbard
a4317ce870 Merged revisions 51271 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines

issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads.  Reset configuration variables to default values prior to parsing configuration file.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-18 23:56:49 +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
2b2f8d22f7 Instead of dividing the offset by 2 directly, make it more clear that the 
offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-17 22:09:52 +00:00
Russell Bryant
4244459e31 Merged revisions 51197 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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.

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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
f986c4f2c3 Move rescheduling of lagrq/pings into the scheduler callback. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-17 00:46:29 +00:00
Joshua Colp
531d008066 Merged revisions 51085 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines

Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-16 05:55:23 +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
Joshua Colp
0a51642588 Merged revisions 49889 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 lines

Ensure we use the default refresh value of 60 if the remote server does not send one. (issue #8746 reported by maethor)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-08 05:11:54 +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
Jason Parker
9803b9a357 Make codec_zap and chan_zap also depend on zaptel. 
This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-05 23:16:16 +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

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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
Kevin P. Fleming
ab4da71374 revert the dynamic_list insertion change... that was not the right thing to do 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-05 00:01:40 +00:00
Kevin P. Fleming
aed1467281 create the IAX2 processing threads as background threads so they will use smaller stacks 
when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-04 23:50:15 +00:00
Joshua Colp
62b8b03294 It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-04 23:00:50 +00:00