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2009-01-28Merged revisions 171838 via svnmerge from oej1-0/+19
https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@171857 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-24Merged revisions 170837 via svnmerge from tilghman1-5/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170839 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-23Merged revisions 170720 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines Add notes to the idlecheck explanation in res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170722 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-17Add discriminator for when ring pulse alert signal is used to preface MWI spillsdbailey1-6/+14
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@169154 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Merged revisions 168722 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r168722 | oej | 2009-01-15 19:47:14 +0100 (Tor, 15 Jan 2009) | 10 lines Merged revisions 168721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines Meetme actually has realtime but wasn't documented ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168727 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Merged revisions 168711 via svnmerge from oej1-4/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168715 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-12Merged revisions 168481 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r168481 | russell | 2009-01-12 08:57:49 -0600 (Mon, 12 Jan 2009) | 10 lines Merged revisions 168480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168484 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-07Merged revisions 167477 via svnmerge from lmadsen1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 Jan 2009) | 8 lines Update queues.conf.sample documentation. Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so. (closes issue #14179) Reported by: CrashHD Tested by: CrashHD ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@167491 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163213 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r163213 | mmichelson | 2008-12-11 14:57:44 -0600 (Thu, 11 Dec 2008) | 9 lines Add an option to voicemail.conf to allow urgent messages to be forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163214 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-09Merged revisions 155671 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r155671 | seanbright | 2008-11-09 11:30:29 -0500 (Sun, 09 Nov 2008) | 1 line Fix this as well. Pointed out by tzafrir. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@155672 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-07Merged revisions 155360 via svnmerge from mmichelson1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, 07 Nov 2008) | 8 lines Remove one more instance of the sample configuration lying about what's possible. The tz cannot be set in a context like this. It can only be set in the general section or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing this out ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@155362 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-06Merged revisions 155012 via svnmerge from mmichelson1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines The documentation listed the ability to set 'maxmsg' per context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@155014 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30Merged revisions 152765 via svnmerge from tilghman1-39/+43
https://origsvn.digium.com/svn/asterisk/trunk ........ r152765 | tilghman | 2008-10-29 23:26:34 -0500 (Wed, 29 Oct 2008) | 5 lines Set up an example stdexten that preserves the original context and extension in the CDR. (Related to issue #13799) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@152777 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-29Merged revisions 152605 via svnmerge from murf1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r152605 | murf | 2008-10-28 23:47:13 -0600 (Tue, 28 Oct 2008) | 22 lines Merged revisions 152538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@152606 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-15Merged revisions 149756 via svnmerge from bweschke1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r149756 | bweschke | 2008-10-15 16:14:20 -0400 (Wed, 15 Oct 2008) | 10 lines Merged revisions 149683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) | 4 lines An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c (closes issue #13709) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@149758 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Merged revisions 148120 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r148120 | tilghman | 2008-10-09 18:25:53 -0500 (Thu, 09 Oct 2008) | 6 lines Fix example schema (closes issue #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn (license 503) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@148128 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Merged revisions 147896 via svnmerge from tilghman1-22/+29
https://origsvn.digium.com/svn/asterisk/trunk ........ r147896 | tilghman | 2008-10-09 12:46:15 -0500 (Thu, 09 Oct 2008) | 4 lines Remove "second form" of extensions, as it no longer applies. Also, cleanup the grammar, formatting, and introduce several clarifications to the text. (Closes issue #13654) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@147898 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-04Merged revisions 146312 via svnmerge from seanbright1-15/+28
https://origsvn.digium.com/svn/asterisk/trunk ........ r146312 | seanbright | 2008-10-03 21:54:44 -0400 (Fri, 03 Oct 2008) | 8 lines Add ability to remotely reboot snom phones. Also cleaned up and reorganized sip_notify.conf.sample a bit as well. Tested snom reboot on snom 360 and verified snom-check-cfg worked as well. (closes issue #13601) Reported by: mjc Tested by: seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@146314 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-26Merged revisions 144829 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r144829 | file | 2008-09-26 20:12:13 -0300 (Fri, 26 Sep 2008) | 2 lines Update documentation to include default setting. This is for you jtodd! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@144832 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-25Merged revisions 144523 via svnmerge from murf1-9/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r144523 | murf | 2008-09-25 15:18:12 -0600 (Thu, 25 Sep 2008) | 13 lines I added a little verbage to hashtab for the hashtab_destroy func. It was pretty sparsely documented. This update fleshes out the pbx_lua module, to add the switch statements to the extensions in the extensions.lua file, as well as removing them when the module is unloaded. Many thanks to Matt Nicholson for his fine contribution! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@144529 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Merged revisions 142866 via svnmerge from tilghman1-0/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines Merged revisions 142865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines Create rules for disallowing contacts at certain addresses, which may improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@142868 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-18Merged revisions 138694 via svnmerge from mmichelson1-5/+13
https://origsvn.digium.com/svn/asterisk/trunk ........ r138694 | mmichelson | 2008-08-18 15:23:11 -0500 (Mon, 18 Aug 2008) | 10 lines Change the queue timeout priority logic into less ugly and confusing code pieces. Clarify the logic within queues.conf.sample. (closes issue #12690) Reported by: atis Patches: queue_timeoutpriority.patch uploaded by atis (license 242) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@138698 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-16Merged revisions 138442 via svnmerge from seanbright1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, cdr_tds has *never* read the port configuration option from cdr_tds.conf. So go ahead and remove it from the sample config. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@138444 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-15Merged revisions 138260 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@138261 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-14Merged revisions 137732 via svnmerge from russell1-435/+435
https://origsvn.digium.com/svn/asterisk/trunk ................ r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@137813 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Merged revisions 136241 via svnmerge from rmudgett1-4/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines * The allowed_bearers setting in misdn.conf misspelled one of its options: digital_restricted. * Fixed some other spelling errors and typos. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136594 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04Merged revisions 135536 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines fix a config sample typo ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135537 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-04Merged revisions 135473 via svnmerge from russell1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines Add a minor clarification to the documentation of mohinterpret and mohsuggest ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135474 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-01Merge changes from team/bbryant/keyrotationrussell1-1/+21
This set of changes enhances IAX2 encryption support by adding key rotation to provide enhanced security. The key used for encryption is rotated right after the call gets set up, and then again every few minutes. This was discussed at the last AstriDevCon. For interoperability with older versions of Asterisk, there is an option that disables key rotation. (closes issue #13018) Reported by: bbryant Patches: 07072008__iax2_key_rotation.diff uploaded by bbryant (license 36) Tested by: russell, bbryant git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135158 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-01SIP should use the transport type set in the Moved Temporarily for the nexttilghman1-1/+1
invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135126 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-01IMAP storage functioned under the assumption that foldersmmichelson1-0/+5
such as "Work" and "Family" would be subfolders of the INBOX. This is an invalid assumption to make, but it could be desirable to set up folders in this manner, so a new option for voicemail.conf, "imapparentfolder" has been added to allow for this. (closes issue #13142) Reported by: jaroth Patches: parentfolder.patch uploaded by jaroth (license 50) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135067 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-30Move implementation of an attended-transfer-complete sound from one channeltilghman4-4/+17
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28remove remaining Zaptel references in various placeskpfleming1-7/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134086 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-22Merged revisions 132713 via svnmerge from tilghman1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132778 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-22Merged revisions 132641 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132643 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Update configuration files to add missing options for jingle, gtalk, bbryant4-2/+8
manager.conf, and features.conf. (closes issue #13128) Reported by: caio1982 Patches: missing_options1.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132514 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-15Additional option for videosupport (always) that disables the optimization totilghman1-5/+11
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Merged revisions 130039 via svnmerge from kpfleming1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130040 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-07Update a few instances of "extensions reload" to "dialplan reload"mmichelson1-1/+1
in the documentation. Patch provided by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128599 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵oej1-5/+2
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-06- Fixing issues with "sip show settings"oej1-1/+2
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128524 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Make TCP disabled by default (it's considered experimental)oej1-3/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128237 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Reformatting the config sampleoej1-16/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128236 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-05Add option to wait to be able to explicitly send ACM via the Proceeding() ↵mattf1-0/+11
application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128122 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03Added a new option, "timeoutpriority" to queues.conf. A detailedmmichelson1-4/+41
explanation of the change may be found in configs/queues.conf.sample (closes issue #12690) Reported by: atis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127720 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-02The ackcall and endcall options in agents.conf now have supplemental optionsmmichelson1-4/+10
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable instead of being hardcoded to '#' and '*'. (AST-86) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127558 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Add a configuration option so the global outboundproxy can use tcptls ↵bbryant1-0/+3
without it being defined by each sip user. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127154 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-01Merged revisions 126844 via svnmerge from oej1-4/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126845 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-30rename zapata.conf.sample to chan_dahdi.conf.samplejpeeler1-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126675 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-27Change the way that the transport option works for sip users. transport will ↵bbryant1-0/+7
now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125891 f38db490-d61c-443f-a65b-d21fe96a405b
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