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4624 Commits

Author SHA1 Message Date
Olle Johansson
2d491b0eae Disable chan_features by default in menuselect 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 19:46:10 +00:00
Joshua Colp
b58a48d672 Merged revisions 67938 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines

Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-07 00:10:48 +00:00
Russell Bryant
6595debbc5 Fix a crash when doing call pickups with SIP phones. The code unlocked the 
channel when it should not have.
(issue #9652, reported by corruptor, fixed by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 21:14:46 +00:00
Russell Bryant
83744c244a Suppress a bunch of debug output unless option_debug is on 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 19:48:02 +00:00
Jason Parker
4a7ae2ebde Correctly update date/time on devices throughout the life of the device, instead of just at registration. 
Issue 9152, yet another patch by DEA.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 18:18:24 +00:00
Christian Richter
619cdb016a Merged revisions 67307 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | 1 line

briding is a bool, fixed copy and paste issue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 16:14:07 +00:00
Christian Richter
b0dd31cb52 Merged revisions 67306 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line

simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 16:11:57 +00:00
Joshua Colp
24922406f4 Only muck with the thread structure if an idle one was found/created. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 15:22:30 +00:00
Kevin P. Fleming
5a5d549b1d ensure that a burst of full frames (AST_FRAME_DTMF being the prime example) will not be processed out of order... this is a brute force fix, but seems to be the safest fix for now (thanks to the Digium PQ department for finding this bug) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 14:35:52 +00:00
Christian Richter
9f749257ef Merged revisions 67209 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line

added possibility to deactivate bridging per port
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-05 10:25:32 +00:00
Russell Bryant
3b5bd316ed Fix up a bunch of places where the iax2 pvt structure can disappear and the 
code did not account for it and crashes.
(issues #9642, #9569, #9666, probably others ... based on the work by
 stevedavies and mihai, with additional changes from me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 23:31:40 +00:00
Jason Parker
0b7607ad55 Fix for skinny keepalives. 
If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily
 adding 10% for network issues, etc),  unregister the device.

Issue 8394, patch by DEA.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 23:26:28 +00:00
Jason Parker
6b150d7b9c Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny) 
Issue 9855, patch by DEA.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 22:36:57 +00:00
Russell Bryant
885d846739 Add comments for two functions that get called with the appropriate call locked, 
but perform operations that could result in the pvt structure getting destroyed
before returning again, causing numerous seg faults all over the module.
(inspired by issues #9642, #9569, and #9666, and the work done by stevedavies
 and mihai)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 22:28:55 +00:00
Joshua Colp
01456184f9 Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 19:31:09 +00:00
Russell Bryant
0b7719e9a1 Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that 
was marked as the final transmission for a call, don't call iax2_destroy() for
that call while the global frame queue is still locked.  There is a very nice
explanation of the deadlock in the report.
(issue #9663, thorough report and patch from stevedavies, additional positive
 test reports from mihai and joff_oconnell)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 15:47:40 +00:00
Russell Bryant
0ecb2ecf42 Changes to the way DTMF is handled in the core broke dialing in chan_skinny. 
This patch makes chan_skinny usable again.  I did not end up testing this,
but there are multiple positive test reports listed in the bug report.
(issue #9596, reported by pj, testing by pj and mvanbaak, and the fix was
 written by DEA)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-01 19:41:30 +00:00
Joshua Colp
c7112015ba Merged revisions 66764 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines

It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-31 16:14:48 +00:00
Joshua Colp
612f61a9b2 Silly me for having out of date source! Oh well... I'm still leaving my comment. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-30 17:28:12 +00:00
Joshua Colp
3d8d697e34 When calling some peer/host that may not exist/reply back... don't keep the dialog in memory for all of eternity. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-30 17:21:06 +00:00
Joshua Colp
7905fca326 Change how channel names are generated a bit. (issue #9825 reported by eldadran) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-30 16:06:37 +00:00
Olle Johansson
332eabcc07 Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 19:32:57 +00:00
Olle Johansson
8d06f379fe Don't issue hangup on hangup on hangup on hangup (for jcmoore) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 19:02:04 +00:00
Olle Johansson
9f15005143 Don't reset hangupcause if we already have one 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 16:07:44 +00:00
Olle Johansson
ff9e2751c6 Tracking down hanging channels, killing them one by one. Issue #9235 and related 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 16:02:50 +00:00
Olle Johansson
bab6473879 Merged revisions 66349 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines

Issue #9802 - Change inuse counter on CANCEL

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 09:41:40 +00:00
Joshua Colp
3a5a649415 Make the usedistinctiveringdetection option work again. (issue #9823 reported by premeau) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-28 23:16:56 +00:00
Jason Parker
a8ab3e5738 I don't know what this was trying to do, but it's clearly incorrect. 
Issues 9808 and 9809.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-27 04:12:37 +00:00
Kevin P. Fleming
5f9897ba56 handle the GNUTLS library properly in the configure script and build system 
don't build in OSP support unless we have found and are allowed to use SSL support


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-25 14:28:46 +00:00
Kevin P. Fleming
0b14e93d00 oops, use #ifdef instead of #if 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 18:28:48 +00:00
Kevin P. Fleming
e49f14e6cb don't reference GnuTLS headers and functions unless the configure script found it 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 18:25:21 +00:00
Olle Johansson
4eab565b25 Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 15:26:10 +00:00
Olle Johansson
e8addde3f1 Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks! 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 15:20:54 +00:00
Olle Johansson
abc8683ce7 Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 15:05:10 +00:00
Olle Johansson
e4f9cc15e6 Issue #8536 - Caller ID not set in CDR for jingle 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:48:55 +00:00
Joshua Colp
0df2a42f96 Merged revisions 65837 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines

Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:42:12 +00:00
Olle Johansson
6cfe6a550e Issue 8409 - phsultan - Fix "login" as component to jabber server. 
...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
   of BYE requests.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:38:30 +00:00
Christian Richter
9f54cd55af Merged revisions 65767 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line

we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 09:37:32 +00:00
Kevin P. Fleming
e1518f42ae start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:59:19 +00:00
Kevin P. Fleming
ca6b421be4 Merged revisions 65682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines

ensure that variables are set on a newly created channel before we start a PBX on it

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:51:56 +00:00
Kevin P. Fleming
b89faf596b clear the 'delay PBX' flag when we are ready to start the PBX 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:35:50 +00:00
Kevin P. Fleming
09dc4253d8 don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:30:24 +00:00
Kevin P. Fleming
f608d90283 Merged revisions 65676 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines

if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 20:07:59 +00:00
Russell Bryant
7398856cea Merged revisions 65588 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines

Revert revision 62417 as someone reported problems with it to Mark.  This was
related to issue #9588.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-23 13:07:13 +00:00
Russell Bryant
c556cc222c List res_smdi as a dependency for app_voicemail and chan_zap 
(Thanks to mnicholson for pointing it out)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 18:40:38 +00:00
Christian Richter
e7355ec53b Merged revisions 65328 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line

we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-22 08:12:20 +00:00
Olle Johansson
4483fa12e8 Merged revisions 65122 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines

Not getting an ACK to a 200 OK in the initial invite is critical to the call.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 18:16:09 +00:00
Olle Johansson
7fe3608300 Merged revisions 65075 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines

Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.

A special Thank You to WeBRainstorm that gave me access to his system.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 15:18:13 +00:00
Olle Johansson
50f79ba4b2 - Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp. 
- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.

Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-18 12:58:39 +00:00
Christian Richter
012fe116fa Merged revisions 65007 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line

fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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 2007-05-18 12:40:46 +00:00