https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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/trunk@65769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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/trunk@65077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines
Merged revisions 64513 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines
Merged revisions 62945,63402,63519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines
Merged revisions 61357,61770,62885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines
Merged following patch with a lot of changes for 1.4
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Merged revisions 64514 via svnmerge from
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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines
Issue #9726 - rlister - Better logging for ACL denials
While at it, also added better logging and handling of peers that are not supposed to register.
My patch, stole the issue report from Russell. My apologies, Russell :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
