755 Commits (897e4b5eaaa4dbbae307be9652ebf8815544677d)

Author	SHA1	Message	Date
Matt O'Gorman	cc003179d4	Patch allows for changing voicemail password in users.conf from voicemail main, written by AnthonyL bug #8436 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	6de5768987	Update documentation to state that you shouldn't use realtime static with voicemail.conf ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	fb52698667	Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ 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. ........ 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. ........ 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. ........ 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 ........ 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. ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	ab6ee2376a	Adding note on effect of applicationmap features on re-invites ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	d2b7e8b247	Be a bit more politically correct ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	bfe4bb0f1e	Issue #8575 - Buggy cisco MWI support. ... Normally we try not to change our software for bugs in other devices. But in this case, the Cisco phones are so widespread so we try to implement a fix while waiting for a bugfix from Cisco. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	4ee818eb8f	Merged revisions 48322 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	7945d4ca35	Add missing s from another repository. (thanks jcmoore!) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	027096b3a3	Updating sip.conf.sample with information about T38 not working ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	56c03478ab	Add documentation to voicemail.conf.sample for ODBC storage. ... Issue 8499 - patch by blitzrage. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	f89143bd13	- Disable RTP hold timers while T.38 fax transmission happens ... - 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	8cbe6025b6	Merged revisions 48183 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	98d3fb64ed	- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	802c3c3ecf	Merged revisions 48142 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	a68edf400f	Explain the use device status system implemented in SIP for subscriptions, ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@48113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	3fe8e34039	Clarify RTP timers. Sorry, grandma. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	7da1a54fe6	Explain properly how videosupport works. ... Committ from Asterisk Video Task Force meeting in Paris! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	e1e6a1b2a8	Make the HOLD notification optional, in order to avoid a lot of extra database lookups ... for all those realtime users out there. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	5bd53e3588	- CANCEL is never authenticated (according to the RFC) ... - Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	4fd3b973bf	clean up sample config, and make native file playback the more obvious default choice ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	9ab1cc22a4	Support ;rport when we're supposed to support ;rport. Issue #7473. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	6964f148ba	Merged revisions 46176 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@46351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	8273d95be3	update entry to reboot a snom phone (issue #7850, pnlarsson) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	590698e583	Adding information about Marks direct-RTP hack to the docs... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	45fc0eaba4	Now, remove all traces of the option that we did not need :-) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	d28fd24747	Merged revisions 45265 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@45280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	3f24dceeca	Merged revisions 45260 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@45262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	13825dab85	Merged revisions 44334 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@44561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	743097a6c1	Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so that's where muted should look for it, right\? ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	caa0d129f2	I've been meaning to add some explanation about muted... here it is ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	7778d017fc	CLI reverbification update to this config file ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	5058b9e13f	updated res_jabber for even better component support, soon will be jep-0100 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@44240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Paul Cadach	6b37705130	Missed part of userconf functionality for chan_h323 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3e2fa16670	Merged revisions 44110 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@44111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	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. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43739 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	5ac0ca6af1	Twould help if we actually documented how the new features in res_odbc actually work. (Oops) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	8bd82ebc0d	Add documentation on rtp packetization. ... 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: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	3c224654c2	Document member name logging functionality. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matthew Fredrickson	2e5118cc49	Add the h323 config file. Arrr!!! for international talk like a pirate's day. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	ec4bf7a849	seperate jingle and gtalk so it will be easier to track ... 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: https://origsvn.digium.com/svn/asterisk/trunk@43185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	4ed1578104	Clarified the meaning of the callwaiting variable in the zapata.conf file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	956b837a41	Merged revisions 43159 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@43161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	ee27f9efee	Remove the suggestion of realtime hints, since that functionality will not be available until post-1.4 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Mark Spencer	c2d959c2f9	Improve documentation of users.conf items. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	6d5809297b	Skinny hold support. ... Original patch by wedhorn, with modifications by me. Issue #7588 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	c2c4f86c72	merge markster's usersconf branch with some slight changes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	091e1aed8d	Merged revisions 42716 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@42717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	b0666488f3	Merged revisions 42697 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@42698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Voicetronix Support	e02897acd4	Board numbers and channel numbers are now 0 based, since vpb driver ... version 3.0 (released December 2005) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matthew Fredrickson	75822388a4	Make sure we give a little warning about the echotraining option ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago