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r65853 | russell | 2007-05-24 10:04:14 -0500 (Thu, 24 May 2007) | 4 lines
Ensure that frames are fully initialized. This will probably fix getting
weird timestamp log messages in logs when using the Festival app.
(issue #9781, patch by me)
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r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines
Issue #8536 - Caller ID not set in CDR for jingle
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r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines
Fix the calculation of the RTT for RTCP. The previous code would result in
oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines
Merged revisions 65837 via svnmerge from
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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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but is still ok... Sorry. Lack of Tea, really.
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structure...
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r65768 | crichter | 2007-05-24 11:37:32 +0200 (Do, 24 Mai 2007) | 9 lines
Merged revisions 65767 via svnmerge from
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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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- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
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r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007) | 2 lines
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines
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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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r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) | 2 lines
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)
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r65680 | kpfleming | 2007-05-23 16:35:50 -0400 (Wed, 23 May 2007) | 2 lines
clear the 'delay PBX' flag when we are ready to start the PBX
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r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines
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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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voicemail.
MWI events get cached, so go ahead and always generate them so the cache gets
populated.
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r65589 | russell | 2007-05-23 08:07:13 -0500 (Wed, 23 May 2007) | 11 lines
Merged revisions 65588 via svnmerge from
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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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r65541 | kpfleming | 2007-05-22 16:25:41 -0400 (Tue, 22 May 2007) | 2 lines
when building a version string for a developer branch, include the base branch in the version string
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class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
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r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) | 3 lines
List res_smdi as a dependency for app_voicemail and chan_zap
(Thanks to mnicholson for pointing it out)
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r65408 | bweschke | 2007-05-22 10:02:56 -0400 (Tue, 22 May 2007) | 3 lines
Fix a problem with flag recognition.
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r65452 | file | 2007-05-22 11:04:46 -0400 (Tue, 22 May 2007) | 2 lines
Remove a double const.
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r65394 | russell | 2007-05-22 08:09:34 -0500 (Tue, 22 May 2007) | 12 lines
Merged revisions 65389 via svnmerge from
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r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | 4 lines
Fix a memory leak that I just noticed in the device state handling in app_queue.
On most device state changes, it would leak roughly 8 to 64 bytes (the length of
the name of the device).
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reported by tfbu)
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double free. (issue #9772, reported by Mike Anikienko, fix by me)
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r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines
Merged revisions 65328 via svnmerge from
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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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saw this, I couldn't help myself from changing it. Previously, for *every*
device state change, app_queue would spawn a thread to handle it. Now, the
device state callback just puts the state change in a queue and it gets
handled by a single state change processing thread.
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r65250 | file | 2007-05-20 13:59:58 -0400 (Sun, 20 May 2007) | 2 lines
res_agi needs to export two symbols (ast_agi_register and ast_agi_unregister) for usage by others. (issue #9755 reported by mnicholson)
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They register the function pointers upon loading with their respective stubs.
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to use this build_stub macro strategy.
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no longer make the symbols from res_adsi global since they don't need to be.
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r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1 line
Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call
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r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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Merged revisions 65123 via svnmerge from
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
Merged revisions 65122 via svnmerge from
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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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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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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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r65039 | crichter | 2007-05-18 14:40:46 +0200 (Fr, 18 Mai 2007) | 9 lines
Merged revisions 65007 via svnmerge from
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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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r64904 | crichter | 2007-05-18 10:58:51 +0200 (Fr, 18 Mai 2007) | 9 lines
Merged revisions 64902 via svnmerge from
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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line
we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines
Issue 9487 - stop media flows at hangup of call
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r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines
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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line
in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines
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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line
some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines
Merged revisions 59623-59624,59639 via svnmerge from
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r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line
we can now make 30 channels on a PRI (before we forgot chan 31..)
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r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line
don't be verbose if no need
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r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line
added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour
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