https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines
make building of codec_gsm against the system GSM library actually work
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r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines
don't define this type either if LOW_MEMORY is enabled
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r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame header caching in the core if LOW_MEMORY is defined
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines
Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line
added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that.
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r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict.
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r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line
when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults.
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r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line
when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines
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r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line
added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE.
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r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines
* Added check for bridging in misdn_call to avoid setting echocancellation
when 2 mISDN channels are involved and when bridging is set. That lead
to a kernel panic before under different situations, because we switched
about 2 times between hardware bridging and echocancelation
* readded MISDN_URATE variable which got lost before, this should make app_v110
work again
* fixed typo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2 lines
Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines
check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 lines
Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
