4fd3b973bf
clean up sample config, and make native file playback the more obvious default choice
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2006-11-07 18:56:21 +00:00
9ab1cc22a4
Support ;rport when we're supposed to support ;rport. Issue #7473 .
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2006-10-31 10:26:16 +00:00
6964f148ba
Merged revisions 46176 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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2006-10-27 09:49:20 +00:00
8273d95be3
update entry to reboot a snom phone (issue #7850 , pnlarsson)
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2006-10-18 02:19:07 +00:00
590698e583
Adding information about Marks direct-RTP hack to the docs...
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2006-10-17 17:39:18 +00:00
45fc0eaba4
Now, remove all traces of the option that we did not need :-)
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2006-10-17 16:23:27 +00:00
d28fd24747
Merged revisions 45265 via svnmerge from
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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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2006-10-16 20:06:18 +00:00
3f24dceeca
Merged revisions 45260 via svnmerge from
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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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2006-10-16 19:37:34 +00:00
13825dab85
Merged revisions 44334 via svnmerge from
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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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2006-10-06 12:50:25 +00:00
743097a6c1
Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so that's where muted should look for it, right\?
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2006-10-05 15:04:22 +00:00
caa0d129f2
I've been meaning to add some explanation about muted... here it is
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2006-10-04 17:04:21 +00:00
7778d017fc
CLI reverbification update to this config file
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2006-10-04 16:49:47 +00:00
5058b9e13f
updated res_jabber for even better component support, soon will be jep-0100 compliant.
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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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2006-10-03 00:01:34 +00:00
6b37705130
Missed part of userconf functionality for chan_h323
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2006-10-02 18:52:56 +00:00
3e2fa16670
Merged revisions 44110 via svnmerge from
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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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2006-10-01 15:20:12 +00:00
740233659f
This change to extensions.ael was to fix bug 8031; the install scripts are 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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2006-09-27 02:32:47 +00:00
5ac0ca6af1
Twould help if we actually documented how the new features in res_odbc actually work. (Oops)
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2006-09-21 23:24:41 +00:00
8bd82ebc0d
Add documentation on rtp packetization.
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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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2006-09-20 17:39:59 +00:00
3c224654c2
Document member name logging functionality.
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2006-09-20 14:47:59 +00:00
2e5118cc49
Add the h323 config file. Arrr!!! for international talk like a pirate's day.
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2006-09-19 19:25:18 +00:00
ec4bf7a849
seperate jingle and gtalk so it will be easier to track
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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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2006-09-18 16:36:14 +00:00
4ed1578104
Clarified the meaning of the callwaiting variable in the zapata.conf file.
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2006-09-18 15:44:18 +00:00
956b837a41
Merged revisions 43159 via svnmerge from
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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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2006-09-18 15:07:59 +00:00
ee27f9efee
Remove the suggestion of realtime hints, since that functionality will not be available until post-1.4
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2006-09-18 14:53:54 +00:00
c2d959c2f9
Improve documentation of users.conf items.
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2006-09-18 05:40:17 +00:00
6d5809297b
Skinny hold support.
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Original patch by wedhorn, with modifications by me.
Issue #7588
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2006-09-17 22:24:27 +00:00
c2c4f86c72
merge markster's usersconf branch with some slight changes
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2006-09-16 23:53:58 +00:00
091e1aed8d
Merged revisions 42716 via svnmerge from
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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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2006-09-11 16:41:49 +00:00
b0666488f3
Merged revisions 42697 via svnmerge from
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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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2006-09-11 14:41:35 +00:00
e02897acd4
Board numbers and channel numbers are now 0 based, since vpb driver
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version 3.0 (released December 2005)
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2006-09-11 03:29:08 +00:00
75822388a4
Make sure we give a little warning about the echotraining option
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2006-09-07 20:05:37 +00:00
34eb4f54ba
Use lower case 'x' instead of a UTF-8 character (issue #7888 reported by flefoll)
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2006-09-07 17:09:55 +00:00
47d8e14871
Comment out default from extensions.ael
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2006-09-07 15:35:52 +00:00
4ba458e352
Merged revisions 42014 via svnmerge from
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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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2006-09-05 16:38:22 +00:00
3f888b84f3
Use GLOBAL() in dialplan examples
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2006-08-31 10:40:24 +00:00
c6977b9983
Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla 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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2006-08-31 01:59:02 +00:00
b281acf0f8
change default setting for autofallthrough
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2006-08-30 21:44:05 +00:00
0850bb72ed
Kevins last commit made me spot a typo.
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2006-08-29 21:52:40 +00:00
ece7018515
add one remaining bit of functionality to the features.conf applicationmap (from Matt Nicholson in Digium Express Services)
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2006-08-29 21:20:43 +00:00
7b65cdc0c7
remove documentation of 'global' section in modules.conf, since it is no longer needed or supported
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2006-08-25 14:14:27 +00:00
236c23269d
Merged revisions 40979 via svnmerge from
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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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2006-08-24 17:13:57 +00:00
e5c0665caf
Merged revisions 40971 via svnmerge from
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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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2006-08-24 16:46:13 +00:00
749cd217c3
Merged revisions 40392 via svnmerge from
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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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2006-08-18 19:11:34 +00:00
87ac16847e
- unregister SLA apps on module unload and add sample config (issue #7701 , junky)
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- rename SLAS to SLAStation, and SLAT to SLATrunk
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2006-08-14 01:32:46 +00:00
a0bd41f79b
Add support for Sigma Designs cards. These basically allow you to offload 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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2006-08-08 17:02:45 +00:00
0f0323dbab
Clarify volgain option a bit, it needs sox to work.
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2006-08-08 15:53:51 +00:00
f37a4e3e12
Bug 6237 - add volgain parameter, such that voicemail messages may be amplified after recording
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2006-08-08 15:37:53 +00:00
9d26f46fc7
remove some extraneous 'followme' in prompt names
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2006-08-07 18:47:33 +00:00
2f69bec40e
Add placeholder for sla.conf sample in configs/. Please update with
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
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2006-08-07 17:01:40 +00:00
4d7c67fc72
Merge my applicationmap_fixup branch to address the issues described in this
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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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2006-08-07 04:15:52 +00:00
Kevin P. Fleming
Olle Johansson
Christian Richter
Russell Bryant
Joshua Colp
Steve Murphy
Matt O'Gorman
Paul Cadach
Tilghman Lesher
Jason Parker
Matthew Fredrickson
Mark Spencer
Voicetronix Support