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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines
Use responses rather then replies even though they mean the same thing.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines
Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line
added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible
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that's where muted should look for it, right\?
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compliant.
also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01 Oct 2006) | 3 lines
Fix the name of the "eventmemberstatus" option in the sample queues.conf
(issue #8065, adamg)
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causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse.
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actually work. (Oops)
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Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon.
Issue #7989, patch by DEA, slightly modified.
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43159 | file | 2006-09-18 11:05:39 -0400 (Mon, 18 Sep 2006) | 2 lines
Add number unobtainable tone for New Zealand (issue #7969 reported by nic_bellamy)
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be available until post-1.4
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Original patch by wedhorn, with modifications by me.
Issue #7588
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines
Spelling/grammar fixes (Issue 7929)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42697 | tilghman | 2006-09-11 09:40:13 -0500 (Mon, 11 Sep 2006) | 2 lines
Two grammar issues (bug 7927)
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version 3.0 (released December 2005)
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flefoll)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42014 | qwell | 2006-09-05 09:27:46 -0700 (Tue, 05 Sep 2006) | 4 lines
Small typo in zapata.conf.sample
Reported by ppyy in 7881
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41282 f38db490-d61c-443f-a65b-d21fe96a405b
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(from Matt Nicholson in Digium Express Services)
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longer needed or supported
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40979 | file | 2006-08-24 13:13:04 -0400 (Thu, 24 Aug 2006) | 2 lines
I can type english. Honest. Thanks Kenny.
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r40971 | file | 2006-08-24 12:44:41 -0400 (Thu, 24 Aug 2006) | 2 lines
Minor documentation fix to add the 'dynamic' dialplan option from angler
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r40392 | kpfleming | 2006-08-18 14:10:51 -0500 (Fri, 18 Aug 2006) | 2 lines
make a feeble attempt to avoid the 'how do I enable my hardware echo canceler' questions
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junky)
- rename SLAS to SLAStation, and SLAT to SLATrunk
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dialtone generation to the board. If you're using a quicknet board where this might work, give it a try as well. (issue #6092 reported by ywalther - minor mods by moi)
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amplified after recording
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
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post to the asterisk-dev mailing list:
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022174.html
This implements full control over both which channel(s) can activate a dynamic
feature, as well as which channel to run the application on. I also updated
the documentation on the applicationmap in features.conf.sample in hopes that
the configuration is more clear.
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hints by using func_odbc
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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