Commit Graph

11 Commits

Author SHA1 Message Date
Paul Belanger 3a4520214a Merged revisions 329055 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/2.0

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  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
  
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    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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-20 21:31:29 +00:00
Richard Mudgett 0096238b52 Merged revisions 320823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25 17:14:11 +00:00
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