Commit Graph

9 Commits

Author SHA1 Message Date
Richard Mudgett 928ec2b990 Merged revisions 309445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04 15:28:20 +00:00
Russell Bryant 2a0983d0c5 Merged revisions 294535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-10 14:15:53 +00:00
Russell Bryant dd1e62c095 Merged revisions 287193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-16 22:00:15 +00:00
David Vossel bcf5988caf Merged revisions 283493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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".
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-24 20:36:35 +00:00
David Vossel eca5209181 Merged revisions 282302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-13 22:27:20 +00:00
Russell Bryant 1990c4347e Merged revisions 281650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-08-10 21:50:24 +00:00
Paul Belanger a84347029b Merged revisions 279689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines
  
  Updated documentation for FAX logger level.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-26 23:35:03 +00:00
Paul Belanger 4bd366a926 Merged revisions 279566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-26 19:58:12 +00:00
Russell Bryant c61b87c5f6 Shuffle UPGRADE.txt files for 1.10.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23 19:17:30 +00:00