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2011-07-20Merged revisions 329055 via svnmerge from pabelanger1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/2.0 ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for PRI support. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@329056 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-25Merged revisions 320823 via svnmerge from rmudgett1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320825 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-04Merged revisions 309445 via svnmerge from rmudgett1-0/+9
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309446 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-10Merged revisions 294535 via svnmerge from russell1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) | 5 lines Tweak a couple of CLI commands back to their original form. The "module" in this case is two parts, so there are two words before the verb of the CLI command. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@294536 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16Merged revisions 287193 via svnmerge from russell1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf. Review: https://reviewboard.asterisk.org/r/922/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-24Merged revisions 283493 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283494 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-13Merged revisions 282302 via svnmerge from dvossel1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines remove current STUN support from chan_sip.c This patch removes the current broken/useless stun support from chan_sip. (closes issue #17622) Reported by: philipp2 Review: https://reviewboard.asterisk.org/r/855/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@282304 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-10Merged revisions 281650 via svnmerge from russell1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines Change the default value for alwaysauthreject in sip.conf to "yes". (closes issue #17756) Reported by: oej ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281651 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Merged revisions 279689 via svnmerge from pabelanger1-3/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines Updated documentation for FAX logger level. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279692 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Merged revisions 279566 via svnmerge from pabelanger1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) | 8 lines Add documentation for FAX logger level. (closes issue #17715) Reported by: vrban Patches: 17715.patch uploaded by pabelanger (license 224) Tested by: vrban ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279567 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Shuffle UPGRADE.txt files for 1.10.russell1-0/+264
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@279118 f38db490-d61c-443f-a65b-d21fe96a405b
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