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2006-12-06Merged revisions 48322 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option in the sample configuration file. (issue #8526, arkadia) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48323 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Add missing s from another repository. (thanks jcmoore!)oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48268 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Updating sip.conf.sample with information about T38 not workingoej1-0/+2
when chan_local or chan_agent is involved in the call. I don't know how big a fix that would be to solve, but this is the current state of affairs. (Chan_sip currently checks if the other side of the bridge has a SIP tech. We could/should implement another check, possibly for udptl_write or some flag in the ast_channel structure). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48264 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-04Add documentation to voicemail.conf.sample for ODBC storage.qwell1-0/+6
Issue 8499 - patch by blitzrage. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48230 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02- Disable RTP hold timers while T.38 fax transmission happensoej1-8/+18
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send something that video phones support in the RTP stream. I now this is a big architectual change at this stage for 1.4, but decided it was needed to avoid future bug reports. - Document the RTP NAT keepalive option in sip.conf.sample Issue 7679 in the bug tracker. Please test. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48199 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-01Merged revisions 48183 via svnmerge from qwell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines Fix a small typo - issue 8848, reported by pabelanger ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48186 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-01- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues ↵oej1-0/+6
with queues and SIP device states - Remove support for T.38 early media, since it's impossible. (Two patches in one - extra friday evening offer due to being off line from svn today... :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48177 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-30Merged revisions 48142 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48143 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Explain the use device status system implemented in SIP for subscriptions,oej1-0/+9
queues and manager a bit better. Like in 1.2, you will get more detailed information if you set a call limit for a device. When the call limit is reached, the status system will report a device as busy. For queues, setting a call limit per SIP device is propably a requirement. In most cases, it will work much better if you only use type=peer and not type=friend. We might decide to backport the new setting from trunk to apply all call limits to the peer part of a friend only. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48113 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Clarify RTP timers. Sorry, grandma.oej1-3/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48105 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-20Explain properly how videosupport works. oej1-1/+5
Committ from Asterisk Video Task Force meeting in Paris! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47845 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16Make the HOLD notification optional, in order to avoid a lot of extra ↵oej1-1/+4
database lookups for all those realtime users out there. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47755 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16- CANCEL is never authenticated (according to the RFC)oej1-0/+6
- Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47733 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07clean up sample config, and make native file playback the more obvious ↵kpfleming1-27/+28
default choice git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47279 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-31Support ;rport when we're supposed to support ;rport. Issue #7473.oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46628 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-27Merged revisions 46176 via svnmerge from crichter1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46351 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-18update entry to reboot a snom phone (issue #7850, pnlarsson)russell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45439 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Adding information about Marks direct-RTP hack to the docs...oej1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45329 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Now, remove all traces of the option that we did not need :-)oej1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45314 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45265 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45280 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45260 via svnmerge from file1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45262 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06Merged revisions 44334 via svnmerge from crichter1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44561 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-05Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so ↵murf1-1/+1
that's where muted should look for it, right\? git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44465 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04I've been meaning to add some explanation about muted... here it ismurf1-0/+13
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44365 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04CLI reverbification update to this config filemurf1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44353 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-03updated res_jabber for even better component support, soon will be jep-0100 ↵mogorman1-19/+0
compliant. also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44240 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-02Missed part of userconf functionality for chan_h323pcadach1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44186 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-01Merged revisions 44110 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44111 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-27This change to extensions.ael was to fix bug 8031; the install scripts are ↵murf1-45/+45
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. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43739 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-21Twould help if we actually documented how the new features in res_odbc ↵tilghman1-1/+11
actually work. (Oops) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@43464 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Add documentation on rtp packetization.qwell3-2/+3
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. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43344 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Document member name logging functionality.qwell1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43327 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-19Add the h323 config file. Arrr!!! for international talk like a pirate's day.mattf1-0/+192
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43288 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18seperate jingle and gtalk so it will be easier to trackmogorman1-0/+19
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. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43185 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18Clarified the meaning of the callwaiting variable in the zapata.conf file.murf1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43165 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18Merged revisions 43159 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43161 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18Remove the suggestion of realtime hints, since that functionality will not ↵tilghman1-10/+0
be available until post-1.4 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43147 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18Improve documentation of users.conf items.markster1-1/+24
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43135 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-17Skinny hold support.qwell1-2/+5
Original patch by wedhorn, with modifications by me. Issue #7588 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43111 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-16merge markster's usersconf branch with some slight changeskpfleming3-1/+64
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43052 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11Merged revisions 42716 via svnmerge from tilghman20-59/+59
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines Spelling/grammar fixes (Issue 7929) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42717 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11Merged revisions 42697 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42697 | tilghman | 2006-09-11 09:40:13 -0500 (Mon, 11 Sep 2006) | 2 lines Two grammar issues (bug 7927) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42698 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11Board numbers and channel numbers are now 0 based, since vpb drivervoicetronix1-6/+6
version 3.0 (released December 2005) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42671 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-07Make sure we give a little warning about the echotraining optionmattf1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42324 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-07Use lower case 'x' instead of a UTF-8 character (issue #7888 reported by ↵file1-1/+1
flefoll) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42284 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-07Comment out default from extensions.aelmarkster1-5/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42244 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-05Merged revisions 42014 via svnmerge from qwell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ 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 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42015 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Use GLOBAL() in dialplan examplesoej2-15/+20
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41560 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵file1-1/+8
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! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-30change default setting for autofallthroughkpfleming1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41475 f38db490-d61c-443f-a65b-d21fe96a405b
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