asterisk/asterisk.git
9 years agoMerged revisions 309445 via svnmerge from
Richard Mudgett [Fri, 4 Mar 2011 15:28:20 +0000 (15:28 +0000)]
Merged revisions 309445 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309403 via svnmerge from
David Ruggles [Fri, 4 Mar 2011 01:52:21 +0000 (01:52 +0000)]
Merged revisions 309403 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines

  Merged revisions 309356 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines

    Merged revisions 309355 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines

      fix small memory leak

      fix small memory leak caused by a string allocation that wasn't freed

      (closes issue #18907)
      Reported by: andy11
      Patches:
            asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224)
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309404 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd HangupRequest manager event, to specify when/where a channel gets hung up.
Jason Parker [Wed, 2 Mar 2011 21:08:39 +0000 (21:08 +0000)]
Add HangupRequest manager event, to specify when/where a channel gets hung up.

(closes issue #18226)
Reported by: clegall_proformatique
Patches:
      asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309256 via svnmerge from
Jason Parker [Wed, 2 Mar 2011 19:54:43 +0000 (19:54 +0000)]
Merged revisions 309256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309256 | qwell | 2011-03-02 13:54:20 -0600 (Wed, 02 Mar 2011) | 15 lines

  Merged revisions 309255 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | 8 lines

    Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.

    Since it's a duplicate, nothing is going to be done, so delme doesn't need to
    be set at all.  Strangely, when this was added, this was being set to 1 in 1.6,
    and 0 in trunk.

    (issue AST-439)
  ........
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9 years agoMerged revisions 309204 via svnmerge from
Jason Parker [Tue, 1 Mar 2011 22:26:37 +0000 (22:26 +0000)]
Merged revisions 309204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines

  Fix consistency of CRLFs on HTTP headers that get sent out.

  (closes issue #18186)
  Reported by: nivaldomjunior
  Patches:
        18186-httpheadernewline.diff uploaded by qwell (license 4)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309209 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309170 via svnmerge from
Richard Mudgett [Tue, 1 Mar 2011 21:57:58 +0000 (21:57 +0000)]
Merged revisions 309170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 Mar 2011) | 7 lines

  Document CHANNEL(keypad_digits) and CHANNEL(no_media_path).

  * Added XML documentation for CHANNEL(keypad_digits) and
  CHANNEL(no_media_path).

  * Tweaked XML documentation for CHANNEL(reversecharge).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309171 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309126 via svnmerge from
Richard Mudgett [Tue, 1 Mar 2011 18:50:07 +0000 (18:50 +0000)]
Merged revisions 309126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 Mar 2011) | 16 lines

  Chan_dahdi does not retain CID when detecting DTMF CID without polarity reversal.

  Looks like an unintended change when sig_analog.c was extracted from
  chan_dahdi.c.

  Removed useless conditional around needed code and fixed resulting
  compiler warning.

  (closes issue #18667)
  Reported by: enegaard
  Patches:
        issue18667.patch uploaded by enegaard (license 1197)
  Tested by: enegaard

  JIRA SWP-2965
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309127 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309084 via svnmerge from
David Vossel [Tue, 1 Mar 2011 16:22:27 +0000 (16:22 +0000)]
Merged revisions 309084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309084 | dvossel | 2011-03-01 10:09:11 -0600 (Tue, 01 Mar 2011) | 15 lines

  Merged revisions 309083 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) | 9 lines

    Fixes thread blocking issue in the sip TCP/TLS implementation.

    (closes issue #18497)
    Reported by: vois
    Patches:
          issues_18497.diff uploaded by dvossel (license 671)
    Tested by: vois, rossbeer, kowalma, Freddi_Fonet
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 309035 via svnmerge from
Tilghman Lesher [Mon, 28 Feb 2011 11:16:06 +0000 (11:16 +0000)]
Merged revisions 309035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines

  Merged revisions 309033-309034 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines

    A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.

    Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
  ........
    r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines

    Clarify meaning, removing double negative (stupid!)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309036 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308991 via svnmerge from
Tilghman Lesher [Mon, 28 Feb 2011 09:34:16 +0000 (09:34 +0000)]
Merged revisions 308991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r308991 | tilghman | 2011-02-28 03:33:22 -0600 (Mon, 28 Feb 2011) | 14 lines

  Merged revisions 308990 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) | 7 lines

    Statements updating zero rows may return SQL_NO_DATA.  This is fine; it's handled.

    (closes issue #18815)
     Reported by: irroot
     Patches:
           func_odbc.insert_nodata.patch uploaded by irroot (license 52)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308945 via svnmerge from
Alec L Davis [Fri, 25 Feb 2011 18:58:10 +0000 (18:58 +0000)]
Merged revisions 308945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21 lines

  Fix Deadlock with attended transfer of SIP call

  Call path
    sip_set_rtp_peer (locks chan then pvt)
     transmit_reinvite_with_sdp
      try_suggested_sip_codec
       pbx_builtin_getvar_helper (locks p->owner)

  But by the time p->owner lock was attempted, seems as though chan and p->owner were different.

  So in sip_set_rtp_peer, lock pvt first then lock p->owner using deadlocking methods.

  (closes issue #18837)
  Reported by: alecdavis
  Patches:
        bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis, Irontec, ZX81, cmaj

  Review: [https://reviewboard.asterisk.org/r/1126/]
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308946 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308903 via svnmerge from
Richard Mudgett [Thu, 24 Feb 2011 21:43:32 +0000 (21:43 +0000)]
Merged revisions 308903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines

  Invalid read in ast_channel_set_caller_event().

  Valgrind reported that ast_channel_set_caller_event() was reading data
  from a freed buffer when using the pre_set structure.

  Rearange things to pre-calculate the name and number pointer before
  updating the caller party structure to see if the name or number was
  changed.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308815 via svnmerge from
Terry Wilson [Thu, 24 Feb 2011 17:59:32 +0000 (17:59 +0000)]
Merged revisions 308815 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines

  Merged revisions 308814 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines

    Merged revisions 308813 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines

      Don't broadcast FullyBooted to every AMI connection

      The FullyBooted event should not be sent to every AMI connection every
      time someone connects via AMI. It should only be sent to the user who
      just connected.

      (closes issue #18168)
      Reported by: FeyFre
      Patches:
            bug0018168.patch uploaded by FeyFre (license 1142)
      Tested by: FeyFre, twilson
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308723 via svnmerge from
Matthew Nicholson [Thu, 24 Feb 2011 15:10:58 +0000 (15:10 +0000)]
Merged revisions 308723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines

  Merged revisions 308722 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines

    Merged revisions 308721 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines

      silence gcc 4.2 compiler warning
    ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308679 via svnmerge from
Terry Wilson [Thu, 24 Feb 2011 03:49:07 +0000 (03:49 +0000)]
Merged revisions 308679 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r308679 | twilson | 2011-02-23 21:41:34 -0600 (Wed, 23 Feb 2011) | 15 lines

  Merged revisions 308678 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines

    Use remotesecret to authenticate with a remote party

    The remotesecret option was only being used for outbound registration
    and not for placing calls. This patch uses remotesecret on outbound
    calls if it is set, otherwise secret is still used.

    Review: https://reviewboard.asterisk.org/r/1107/
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix compiler warning.
Richard Mudgett [Wed, 23 Feb 2011 23:55:58 +0000 (23:55 +0000)]
Fix compiler warning.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308622 via svnmerge from
Richard Mudgett [Wed, 23 Feb 2011 23:45:02 +0000 (23:45 +0000)]
Merged revisions 308622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) | 9 lines

  sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.

  (closes issue #18874)
  Reported by: cmaj
  Patches:
        patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830)

  JIRA SWP-3172
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308623 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMedia Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio...
David Vossel [Tue, 22 Feb 2011 23:04:49 +0000 (23:04 +0000)]
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse ast_debug for console logging
Andrew Latham [Tue, 22 Feb 2011 15:33:56 +0000 (15:33 +0000)]
Use ast_debug for console logging

Guessed the log levels based on info that level 3
is the soft roof.  Can we create a page / document
to define the levels?

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308416 via svnmerge from
Matthew Nicholson [Mon, 21 Feb 2011 15:04:19 +0000 (15:04 +0000)]
Merged revisions 308416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines

  Merged revisions 308414 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines

    Merged revisions 308413 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines

      Properly check the bounds of arrays when decoding UDPTL packets.  Also, remove broken support for receiving UDPTL packets larger than 16k.  That shouldn't ever happen anyway.

      AST-2011-002
      FAX-281
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308417 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd HTTP URI Debug logging and update notice
Andrew Latham [Mon, 21 Feb 2011 14:14:41 +0000 (14:14 +0000)]
Add HTTP URI Debug logging and update notice

enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308372 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofix a memory leak in device state
Tzafrir Cohen [Mon, 21 Feb 2011 13:58:18 +0000 (13:58 +0000)]
fix a memory leak in device state

The callback handle_statechange (pbx.c) fails to release its data
pointer, leaking memory in the process.

Reported by: tzafrir
Patches:
      18735_pbx_free_callback.diff uploaded by tzafrir (license 46)

Review: https://reviewboard.asterisk.org/r/1110/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308371 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd CSS MIME Type
Andrew Latham [Sat, 19 Feb 2011 14:07:38 +0000 (14:07 +0000)]
Add CSS MIME Type

Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308288 via svnmerge from
Tilghman Lesher [Sat, 19 Feb 2011 11:03:44 +0000 (11:03 +0000)]
Merged revisions 308288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) | 2 lines

  A few more (copies of) files to ignore in this directory.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308242 via svnmerge from
Alexandr Anikin [Fri, 18 Feb 2011 00:11:06 +0000 (00:11 +0000)]
Merged revisions 308242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines

  added g729onlyA option for announce only AnnexA g.729 codec in
  h.323 capabilities. Option can be global or per user/peer.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308243 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd more verbage to CLI command 'pri show channels' usage.
Richard Mudgett [Thu, 17 Feb 2011 20:21:56 +0000 (20:21 +0000)]
Add more verbage to CLI command 'pri show channels' usage.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308150 via svnmerge from
Paul Belanger [Wed, 16 Feb 2011 22:02:41 +0000 (22:02 +0000)]
Merged revisions 308150 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb 2011) | 2 lines

  Fix FreeBSD builds.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308157 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308098 via svnmerge from
Alexandr Anikin [Wed, 16 Feb 2011 08:06:01 +0000 (08:06 +0000)]
Merged revisions 308098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 lines

  ifdef __linux__ keepalive variables also
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308099 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 308010 via svnmerge from
Jason Parker [Tue, 15 Feb 2011 23:34:27 +0000 (23:34 +0000)]
Merged revisions 308010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines

  Merged revisions 308007 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines

    Merged revisions 308002 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines

      Fix regression that changed behavior of queues when ringing a queue member.

      This reverts r298596, which was to fix a highly bizarre and contrived issue
      with a queue member that called into his own queue being transferred back
      into his own queue.  I couldn't reproduce that issue in any way.  I think one
      of the other recent transfer fixes actually fixed this.

      (closes issue #18747)
      Reported by: vrban
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308013 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoinclude tcp keepalive socket calls only on linux, freebsd and others
Alexandr Anikin [Tue, 15 Feb 2011 23:07:47 +0000 (23:07 +0000)]
include tcp keepalive socket calls only on linux, freebsd and others
don't have these options on sockets.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd CLI "pri show channels" command.
Richard Mudgett [Tue, 15 Feb 2011 21:42:55 +0000 (21:42 +0000)]
Add CLI "pri show channels" command.

List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting.  Calls on hold or call-waiting
are not associated with any B channel.

JIRA LIBPRI-27
JIRA SWP-2547

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307962 via svnmerge from
Richard Mudgett [Tue, 15 Feb 2011 19:53:32 +0000 (19:53 +0000)]
Merged revisions 307962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line

  Don't crash when forcing caller id.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes compile error in chan_phone for big endian
David Vossel [Tue, 15 Feb 2011 18:09:25 +0000 (18:09 +0000)]
Fixes compile error in chan_phone for big endian

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307879 via svnmerge from
Richard Mudgett [Tue, 15 Feb 2011 16:18:43 +0000 (16:18 +0000)]
Merged revisions 307879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines

  No response sent for SIP CC subscribe/resubscribe request.

  Asterisk does not send a response if we try to subscribe for call
  completion after we have received a 180 Ringing.  You can only subscribe
  for call completion when the call has been cleared.

  When we receive the 180 Ringing, for this call, its call-completion state
  is 'CC_AVAILABLE'.  If we then send a subscribe message to Asterisk, it
  trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
  Because this is an invalid state change, it just ignores the message.  The
  only state Asterisk will accept our subscribe message is in the
  'CC_CALLER_OFFERED' state.

  Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
  the call by sending a CANCEL.

  Asterisk should always send a response.  Even if its a negative one.

  The fix is to allow for the CCSS core to notify a CC agent that a failure
  has occurred when CC is requested.  The "ack" callback is replaced with a
  "respond" callback.  The "respond" callback has a parameter indicating
  either a successful response or a specific type of failure that may need
  to be communicated to the requester.

  (closes issue #18336)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson, rmudgett

  JIRA SWP-2633

  (closes issue #18337)
  Reported by: GeorgeKonopacki
  Tested by: mmichelson

  JIRA SWP-2634
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307837 via svnmerge from
Tilghman Lesher [Tue, 15 Feb 2011 07:03:44 +0000 (07:03 +0000)]
Merged revisions 307837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307837 | tilghman | 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) | 15 lines

  Merged revisions 307836 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) | 8 lines

    Need to retrieve the rows affected before using the associated variable.

    (closes issue #18795)
     Reported by: irroot
     Patches:
           20110211__issue18795.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307793 via svnmerge from
Tilghman Lesher [Mon, 14 Feb 2011 20:18:02 +0000 (20:18 +0000)]
Merged revisions 307793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307793 | tilghman | 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) | 15 lines

  Merged revisions 307792 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) | 8 lines

    Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once.

    (issue #18156)
     Reported by: asgaroth
     Patches:
           20110214__issue18156.diff.txt uploaded by tilghman (license 14)
     Tested by: tilghman
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307795 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMaking trunk compile again.
Tilghman Lesher [Mon, 14 Feb 2011 07:01:46 +0000 (07:01 +0000)]
Making trunk compile again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307752 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307750 via svnmerge from
Tilghman Lesher [Mon, 14 Feb 2011 06:54:08 +0000 (06:54 +0000)]
Merged revisions 307750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines

  Calling a gosub routine defined in AEL from Dial/Queue ceased to work.

  A bug in AEL did not distinguish between the "s" extension generated by
  AEL and an "s" extension that was required to exist by the chan_dahdi
  (or another channel) that was not supplied with a starting extension.
  Therefore, AEL made incorrect assumptions about what commands were
  permissable in the context.  This was fixed by making AEL generate a
  different extension name.  However, Dial and Queue make additional
  assumptions about the name of the default gosub extension.  Therefore,
  they needed to be brought into line with a "macro" rendered by AEL (as
  a gosub), without breaking traditional dialplans written without the
  aid of AEL.

  Related to (issue #18480)
   Reported by: nivek

  (closes issue #18729)
   Reported by: kkm
   Patches:
         20110209__issue18729.diff.txt uploaded by tilghman (license 14)
         018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
   Tested by: kkm
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agolc not found - it's warning, not error,
Alexandr Anikin [Sun, 13 Feb 2011 10:50:22 +0000 (10:50 +0000)]
lc not found - it's warning,  not error,
change malloc to ast_calloc again

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agochange malloc to ast_calloc calls to prevent crash of asterisk
Alexandr Anikin [Sat, 12 Feb 2011 23:25:58 +0000 (23:25 +0000)]
change malloc to ast_calloc calls to prevent crash of asterisk

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307677 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307536 via svnmerge from
Jason Parker [Thu, 10 Feb 2011 22:43:51 +0000 (22:43 +0000)]
Merged revisions 307536 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307536 | qwell | 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) | 22 lines

  Merged revisions 307535 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines

    Merged revisions 307534 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines

      Remove color when executing commands via a remote console.

      Essentially this makes '-x' imply '-n' on rasterisk.  This was done in a
      different and incomplete way previously, which I'm reverting here.

      (issue #18776)
      Reported by: alecdavis
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307467 via svnmerge from
Mark Michelson [Thu, 10 Feb 2011 17:45:24 +0000 (17:45 +0000)]
Merged revisions 307467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) | 5 lines

  Fix a gaffe in the CCSS sample configuration.

  Discovered by Philippe Lindheimer and pointed out on #asterisk-dev
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307468 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes bug in chan_sip where nativeformats are not set correctly.
David Vossel [Thu, 10 Feb 2011 17:12:10 +0000 (17:12 +0000)]
Fixes bug in chan_sip where nativeformats are not set correctly.

The nativeformats field was being overwritten when it should have been
appended too.  This caused some format capabilities to be lost briefly and
some log warnings to be output.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoCorrections for properly work with H.323v2 (older) endpoints and other
Alexandr Anikin [Thu, 10 Feb 2011 13:29:19 +0000 (13:29 +0000)]
Corrections for properly work with H.323v2 (older) endpoints and other
small fixes.

Interpret remote side H.225 version.

Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.

Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.

(closes issue #18542)
Reported by: vmikhelson
Patches:
      issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd new manager action MeetmeListRooms.
Jeff Peeler [Wed, 9 Feb 2011 22:48:02 +0000 (22:48 +0000)]
Add new manager action MeetmeListRooms.

From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.

(closes issue #17905)
Reported by: rcasas
Patches:
      app_meetme.c.patch uploaded by rcasas (license 641)

Review: https://reviewboard.asterisk.org/r/874/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDisable color during running test
Andrew Latham [Wed, 9 Feb 2011 21:46:24 +0000 (21:46 +0000)]
Disable color during running test

(closes issue #18776)
Reported by: alecdavis
Patches:
     ast_deb_init.diff uploaded by lathama (license 1028)
Tested by: andrel, lathama

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307315 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307273 via svnmerge from
Jeff Peeler [Wed, 9 Feb 2011 21:08:22 +0000 (21:08 +0000)]
Merged revisions 307273 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) | 8 lines

  Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback.

  (closes issue #18758)
  Reported by: rgagnon
  Patches:
        branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
        trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307274 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllow parkedmusicclass to be settable for non-default parking lots.
Jeff Peeler [Wed, 9 Feb 2011 20:11:11 +0000 (20:11 +0000)]
Allow parkedmusicclass to be settable for non-default parking lots.

(closes issue #17946)
Reported by: bluecrow76
Patches:
      asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307228 via svnmerge from
Jeff Peeler [Wed, 9 Feb 2011 19:53:28 +0000 (19:53 +0000)]
Merged revisions 307228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r307228 | jpeeler | 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) | 17 lines

  Merged revisions 307227 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines

    Make sure to set parking dial context for non-default parking lots.

    Since parking_con_dial isn't settable, set all parking lots to "park-dial".

    (closes issue #17946)
    Reported by: bluecrow76
    Patches:
          asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270)
          modified by me
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoclarify warning when no loadable module support
Tzafrir Cohen [Wed, 9 Feb 2011 19:17:01 +0000 (19:17 +0000)]
clarify warning when no loadable module support

Clarify warning message when LOADABLE_MODULES is disabled but we still
try to load a module.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307142 via svnmerge from
Tilghman Lesher [Wed, 9 Feb 2011 05:53:29 +0000 (05:53 +0000)]
Merged revisions 307142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) | 3 lines

  Initialize tracking variable in structure properly.  Fixes a memory leak.
  (Reported by The_Boy_Wonder on IRC, fixed by me.)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307092 via svnmerge from
Jason Parker [Tue, 8 Feb 2011 21:24:57 +0000 (21:24 +0000)]
Merged revisions 307092 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | 9 lines

  Fix issue with verbose messages not showing on remote console.

  This code was reworked recently, and since the logchannel list hadn't been
  created yet at this point, and it was a verbose message, it was being dropped
  on the floor.  Now it'll continue on to where it should be handled.

  (closes issue #18580)
  Reported by: pabelanger
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307097 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 307065 via svnmerge from
Mark Michelson [Tue, 8 Feb 2011 21:18:26 +0000 (21:18 +0000)]
Merged revisions 307065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines

  Add a couple of useful channel variables for the CC recall macro.

  CC_EXTEN and CC_CONTEXT will allow you to determine the channel
  and context that will be called when the recall occurs.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307071 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306979 via svnmerge from
Terry Wilson [Tue, 8 Feb 2011 20:42:44 +0000 (20:42 +0000)]
Merged revisions 306979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306979 | twilson | 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) | 16 lines

  Merged revisions 306973 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306973 | twilson | 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) | 9 lines

    Merged revisions 306972 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines

      Fix comparison for REFER Replaces tags with pedantic=yes
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307061 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDocumentation Updates
Andrew Latham [Tue, 8 Feb 2011 20:31:13 +0000 (20:31 +0000)]
Documentation Updates

Note default polling setting in voicemail.conf
Add missing config to asterisk.conf
Update manpage

(issue #16505)
Reported by: tzafrir
Patches:
     asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306967 via svnmerge from
Jeff Peeler [Tue, 8 Feb 2011 19:42:03 +0000 (19:42 +0000)]
Merged revisions 306967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306967 | jpeeler | 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) | 16 lines

  Merged revisions 306966 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines

    Merged revisions 306965 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line

      fix this line again
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306968 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306962 via svnmerge from
Jeff Peeler [Tue, 8 Feb 2011 19:26:05 +0000 (19:26 +0000)]
Merged revisions 306962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines

  Merged revisions 306961 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines

    Merged revisions 306960 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines

      Backup file storing message duration is not used with IMAP_STORAGE, remove code.

      The message duration is stored in the body of the email when using IMAP_STORAGE,
      so nothing needs to happen with the backup file.

      (closes issue #18718)
      Reported by: kerframil
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306866 via svnmerge from
Jeff Peeler [Tue, 8 Feb 2011 16:22:07 +0000 (16:22 +0000)]
Merged revisions 306866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306866 | jpeeler | 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) | 16 lines

  Merged revisions 306865 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines

    Merged revisions 306864 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line

      make this safer and fully correct, pointed out by Steve Davis
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306867 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDocumentation Updates.
Andrew Latham [Tue, 8 Feb 2011 02:05:03 +0000 (02:05 +0000)]
Documentation Updates.

Start updates to the man pages.

(issue #16505)
Reported by: tzafrir
Tested by: lathama

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306827 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDefine the MCID acronym in chan_dahdi.conf.sample.
Richard Mudgett [Tue, 8 Feb 2011 00:43:34 +0000 (00:43 +0000)]
Define the MCID acronym in chan_dahdi.conf.sample.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306793 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse correct conditional for MCID send.
Richard Mudgett [Tue, 8 Feb 2011 00:26:01 +0000 (00:26 +0000)]
Use correct conditional for MCID send.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306791 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoPass a MCID request to the bridged channel.
Richard Mudgett [Mon, 7 Feb 2011 23:33:44 +0000 (23:33 +0000)]
Pass a MCID request to the bridged channel.

Pass a MCID request to the bridged channel so the bridged channel can send
it to the network.

The ability to send the MCID request on an ISDN span is enabled with the
new chan_dahdi.conf mcid_send option.

JIRA SWP-2845
JIRA ABE-2736

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306674 via svnmerge from
Terry Wilson [Mon, 7 Feb 2011 22:46:07 +0000 (22:46 +0000)]
Merged revisions 306674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306674 | twilson | 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) | 24 lines

  Merged revisions 306673 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines

    Merged revisions 306672 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines

      Don't try to pickup a call in the middle of a masquerade

      If A calls B which doesn't answer and C & D both try to do a call pickup, it is
      possible for ast_pickup_call to answer the call, then fail to masquerade one of
      the calls because the other one is already in the process of masquerading. This
      patch checks to see if the channel is in the process of masquerading before
      call before selecting it for a pickup.

      Review: https://reviewboard.asterisk.org/r/1094/
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306675 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306619 via svnmerge from
Terry Wilson [Mon, 7 Feb 2011 22:31:25 +0000 (22:31 +0000)]
Merged revisions 306619 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306619 | twilson | 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) | 24 lines

  Merged revisions 306618 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306618 | twilson | 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) | 17 lines

    Merged revisions 306617 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines

      Don't allow a REFER w/replaces to replace its own dialog

      Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces
      header that matches the dialog of the REFER. This would be a situation like A
      calls B, A calls C, A transfers B to A, which is just silly. This patch makes
      the transfer fail instead of making Asterisk freak out and forget to hang other
      channels up.

      Review: https://reviewboard.asterisk.org/r/1093/
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306670 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306575 via svnmerge from
Mark Michelson [Mon, 7 Feb 2011 17:55:38 +0000 (17:55 +0000)]
Merged revisions 306575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines

  Rearrange a bit of code in the generic CC recall operation.

  By waiting to call the callback macro after the CC_INTERFACES,
  extension, priority, and context have been set, this information
  can be accessed more easily within the callback macro.

  Reported by Philippe Lindheimer.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306576 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes use of ast_format_cap_append where ast_format_cap_copy is necessary.
David Vossel [Mon, 7 Feb 2011 16:33:43 +0000 (16:33 +0000)]
Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306541 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agofix trivial issue after dvossel patch, initial zero fill user and peer
Alexandr Anikin [Sat, 5 Feb 2011 22:16:07 +0000 (22:16 +0000)]
fix trivial issue after dvossel patch, initial zero fill user and peer
structure before cap structure allocated.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoIgnore voice frames in chan_dahdi native bridging. Hardware is handling them.
Richard Mudgett [Sat, 5 Feb 2011 02:55:50 +0000 (02:55 +0000)]
Ignore voice frames in chan_dahdi native bridging.  Hardware is handling them.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306464 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSend manager event for blackfilter only if it DOES NOT match.
Jeff Peeler [Fri, 4 Feb 2011 22:37:11 +0000 (22:37 +0000)]
Send manager event for blackfilter only if it DOES NOT match.

The logic got reversed, oops. Works properly now when multiple blackfilters are
present.

(closes issue #18283)
Reported by: telecos82
Patches:
      ast_managereventfilter.patch uploaded by telecos82 (license 687)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd ISDN display ie text handling options to chan_dahdi.conf.
Richard Mudgett [Fri, 4 Feb 2011 20:30:48 +0000 (20:30 +0000)]
Add ISDN display ie text handling options to chan_dahdi.conf.

The display ie handling can be controlled independently in the send and
receive directions with the following options:

* Block display text data.

* Use display text in SETUP/CONNECT messages for name.

* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).

* Pass arbitrary display text during a call.  Sent in INFORMATION
messages.  Received from any message that the display text was not used as
a name.

If the display options are not set then the options default to legacy
behavior.

The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.

To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.

JIRA SWP-2688
JIRA ABE-2693

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306356 via svnmerge from
Jason Parker [Fri, 4 Feb 2011 19:24:54 +0000 (19:24 +0000)]
Merged revisions 306356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines

  Merged revisions 306346 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines

    Don't fallthrough to 'unknown' in the 'ringing' case.

    This could cause improper exits from the queue.

    (closes issue #18499)
    Reported by: zaltar
    Patches:
          app_queue.patch uploaded by zaltar (license 1148)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix compiler warning.
Richard Mudgett [Fri, 4 Feb 2011 19:09:00 +0000 (19:09 +0000)]
Fix compiler warning.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306326 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306324 via svnmerge from
Richard Mudgett [Fri, 4 Feb 2011 18:57:39 +0000 (18:57 +0000)]
Merged revisions 306324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines

  Don't send redirecting updates to the caller if the dialplan forked the call.

  Each fork in the dial could be redirected and confuse the caller.  For
  ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN
  redirects calls in sequence not in parallel.

  * Also fixed a formatting inconsistency in app_dial.c and make a warning
  message more useful about what frame type could not be written.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306325 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRevert changes to extconf.c
Paul Belanger [Fri, 4 Feb 2011 18:16:16 +0000 (18:16 +0000)]
Revert changes to extconf.c

It seems extconf.c already defines some local ast_debug() functions.  Theses
should be removed and replaced with logger.h.  A patch will be added to
reviewboard shortly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoReplace ast_log(LOG_DEBUG, ...) with ast_debug()
Paul Belanger [Fri, 4 Feb 2011 16:55:39 +0000 (16:55 +0000)]
Replace ast_log(LOG_DEBUG, ...) with ast_debug()

(closes issue #18556)
Reported by: kkm

Review: https://reviewboard.asterisk.org/r/1071/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFix compile error in codec ilbc translator.
David Vossel [Fri, 4 Feb 2011 16:42:15 +0000 (16:42 +0000)]
Fix compile error in codec ilbc translator.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306215 via svnmerge from
Jeff Peeler [Thu, 3 Feb 2011 23:50:08 +0000 (23:50 +0000)]
Merged revisions 306215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) | 20 lines

  Fix SIP deadlock involving state changes.

  Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper)
  has caused locking problems. Both of these functions lock the channel when
  the channel argument is passed in!

  In this case, the suspected problem (the backtrace makes it impossible to tell)
  was the private being locked in sip_set_rtp_peer and then:
  transmit_reinvite_with_sdp
   try_suggested_sip_codec
     pbx_builtin_getvar_helper
  (Traced to verify that the fix was only required in 1.8 and later.)

  (closes issue #18491)
  Reported by: cmaj
  Patches:
        chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830)
  Tested by: cmaj
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306216 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306127 via svnmerge from
Terry Wilson [Thu, 3 Feb 2011 21:13:11 +0000 (21:13 +0000)]
Merged revisions 306127 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines

  Merged revisions 306126 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines

    Merged revisions 306119 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines

      Set hangup cause in local_hangup

      When a call involves a local channel (like SIP -> Local -> SIP), the hangup
      cause was not being set. This resulted in SIP channels sometimes getting a
      503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+
      this also can cause issues with CCSS that involve a local channel. This patch
      sets the hangupcause for one side of the local channel to the other in
      local_hangup for outbound calls.
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306128 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 306124 via svnmerge from
Jeff Peeler [Thu, 3 Feb 2011 20:51:09 +0000 (20:51 +0000)]
Merged revisions 306124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r306124 | jpeeler | 2011-02-03 14:50:48 -0600 (Thu, 03 Feb 2011) | 17 lines

  Merged revisions 306123 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines

    Set exception on channel in parking thread when POLLPRI event detected.

    This is done just to make the code be equivalent to the old select code. As
    noted in 303106 the same issue was already fixed in this branch, but the
    exception was not set on the channel in the case of POLLPRI. The reason that
    this did not cause a problem here is because in 122923 the check in __ast_read
    to check the exception flag was removed.

    (related to #18637)
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306125 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoModify alignment of 'core show codecs', since the ID is no longer a huge int.
Jason Parker [Thu, 3 Feb 2011 18:37:06 +0000 (18:37 +0000)]
Modify alignment of 'core show codecs', since the ID is no longer a huge int.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes output of "core show codecs" to display image types correctly.
David Vossel [Thu, 3 Feb 2011 18:12:57 +0000 (18:12 +0000)]
Fixes output of "core show codecs" to display image types correctly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306053 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAsterisk media architecture conversion - no more format bitfields
David Vossel [Thu, 3 Feb 2011 16:22:10 +0000 (16:22 +0000)]
Asterisk media architecture conversion - no more format bitfields

This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agores_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
Andrew Latham [Thu, 3 Feb 2011 16:13:40 +0000 (16:13 +0000)]
res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support

(issue #18713)
Reported by: lathama
Patches:
     snom_dir.diff uploaded by lathama (license 1028)
Tested by: lathama

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305923 via svnmerge from
Richard Mudgett [Thu, 3 Feb 2011 00:29:46 +0000 (00:29 +0000)]
Merged revisions 305923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines

  Merged revisions 305889 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines

    Merged revisions 305888 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines

      Minor AST_FRAME_TEXT related issues.

      * Include the null terminator in the buffer length.  When the frame is
      queued it is copied.  If the null terminator is not part of the frame
      buffer length, the receiver could see garbage appended onto it.

      * Add channel lock protection with ast_sendtext().

      * Fixed AMI SendText action ast_sendtext() return value check.
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305844 via svnmerge from
Tilghman Lesher [Wed, 2 Feb 2011 20:06:33 +0000 (20:06 +0000)]
Merged revisions 305844 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) | 5 lines

  Eliminate a file descriptor leak when using the FILE() dialplan function.

  (closes issue #18731)
  Reported by: marioabajo
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoReplacing doc/* and asterisk.pdf with wiki links
Andrew Latham [Wed, 2 Feb 2011 19:30:49 +0000 (19:30 +0000)]
Replacing doc/* and asterisk.pdf with wiki links

Adding links to http(s)://wiki.asterisk.org

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305843 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoReplacing doc/* with wiki links
Andrew Latham [Wed, 2 Feb 2011 18:59:29 +0000 (18:59 +0000)]
Replacing doc/* with wiki links

Adding links to http(s)://wiki.asterisk.org

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoReplace link to old doc with new wiki page.
Andrew Latham [Wed, 2 Feb 2011 15:25:12 +0000 (15:25 +0000)]
Replace link to old doc with new wiki page.

Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305692 via svnmerge from
Jason Parker [Tue, 1 Feb 2011 22:48:55 +0000 (22:48 +0000)]
Merged revisions 305692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb 2011) | 7 lines

  Reverse sense of an error test when reading from astdb.

  (closes issue #18545)
  Reported by: jcovert
  Patches:
        chan_iax2.c.patch uploaded by jcovert (license 551)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoSIP Configuration Documentation
Andrew Latham [Tue, 1 Feb 2011 21:16:31 +0000 (21:16 +0000)]
SIP Configuration Documentation

sip show settings reports qualifyfreq in milliseconds.
sip.conf configures qualifyfreg in seconds.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305650 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305603 via svnmerge from
Brett Bryant [Tue, 1 Feb 2011 19:27:23 +0000 (19:27 +0000)]
Merged revisions 305603 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r305603 | bbryant | 2011-02-01 14:23:20 -0500 (Tue, 01 Feb 2011) | 4 lines

  Add a possible solution to a customer problem with reloading cel_pgsql.so
  quickly.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agodoc/tex dir removed, but corresponding entries still exists
Andrew Latham [Tue, 1 Feb 2011 18:03:48 +0000 (18:03 +0000)]
doc/tex dir removed, but corresponding entries still exists

Update README, CHANGES, and Makefile.  Direct users to
http://wiki.asterisk.org for documentation or to the
AST.txt and AST.pdf included in the tarball.

(closes issue #18443)
Reported by: bas
Patches:
      changes.diff uploaded by lathama (license 1028)
      readme.diff uploaded by lathama (license 1028)
Tested by: lathama bas

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305561 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305473 via svnmerge from
Jason Parker [Tue, 1 Feb 2011 17:05:38 +0000 (17:05 +0000)]
Merged revisions 305473 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305473 | qwell | 2011-02-01 11:04:23 -0600 (Tue, 01 Feb 2011) | 23 lines

  Merged revisions 305472 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r305472 | qwell | 2011-02-01 11:02:09 -0600 (Tue, 01 Feb 2011) | 16 lines

    Merged revisions 305471 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) | 9 lines

      Close file descriptor for timing source when a MOH class gets destroyed.

      (closes issue #18457)
      Reported by: mcallist
      Patches:
            18457-closetimer.diff uploaded by qwell (license 4)
            18457-closetimer_trunk.diff uploaded by qwell (license 4)
      Tested by: qwell, loloski
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305474 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd's two features to confbridge: confbridge kick, and confbridge list.
Brett Bryant [Tue, 1 Feb 2011 16:05:23 +0000 (16:05 +0000)]
Add's two features to confbridge: confbridge kick, and confbridge list.

(closes issue #14389)
(closes issue #18007)
Reported by: jcollie
Patches:
      0001-Fix-up-bridging-module-so-that-menuselect-works.patch uploaded by jcollie (license 412)
      0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch uploaded by jcollie (license 412)
Tested by: file

Review: https://reviewboard.asterisk.org/r/1084/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305433 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305343 via svnmerge from
Richard Mudgett [Tue, 1 Feb 2011 00:07:30 +0000 (00:07 +0000)]
Merged revisions 305343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305343 | rmudgett | 2011-01-31 18:01:09 -0600 (Mon, 31 Jan 2011) | 21 lines

  Merged revisions 305342 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r305342 | rmudgett | 2011-01-31 17:50:10 -0600 (Mon, 31 Jan 2011) | 14 lines

    Merged revisions 305341 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines

      Obtain the pri lock for PRI queue counters.

      Need to obtain the pri lock when calling pri_dump_info_str() to avoid a
      reentrancy problem when calculating the Q.921 Q count statistic.

      JIRA AST-484
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305254 via svnmerge from
Jason Parker [Mon, 31 Jan 2011 23:08:38 +0000 (23:08 +0000)]
Merged revisions 305254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines

  Merged revisions 305253 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines

    Merged revisions 305252 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines

      Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))

      chan_iax2 and other channel drivers already had code to prevent this.  The
      attempt that app_dial was making to prevent it was not correct, so I fixed that.

      (closes issue #18371)
      Reported by: gbour
      Patches:
            18371.patch uploaded by gbour (license 1162)
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305255 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305247 via svnmerge from
Jason Parker [Mon, 31 Jan 2011 22:26:06 +0000 (22:26 +0000)]
Merged revisions 305247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | 7 lines

  Add alternative name for config option.

  The SIP sample configuration had "tlscadir" as the option name, but chan_sip
  used the more correct "tlscapath".  Now both are accepted.

  Discovered (sort of) by a user on IRC in #asterisk
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305198 via svnmerge from
Jason Parker [Mon, 31 Jan 2011 21:31:31 +0000 (21:31 +0000)]
Merged revisions 305198 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r305198 | qwell | 2011-01-31 15:30:44 -0600 (Mon, 31 Jan 2011) | 2 lines

  Fix compile error.  pseudofd no longer exists.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305203 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 305131 via svnmerge from
Jason Parker [Mon, 31 Jan 2011 21:01:28 +0000 (21:01 +0000)]
Merged revisions 305131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305131 | qwell | 2011-01-31 15:00:25 -0600 (Mon, 31 Jan 2011) | 16 lines

  Merged revisions 305130 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r305130 | qwell | 2011-01-31 14:59:37 -0600 (Mon, 31 Jan 2011) | 9 lines

    Merged revisions 305129 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan 2011) | 2 lines

      Set file descriptors to -1 on creation, so that we don't see weirdness later.
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAsterisk HTTP response Content-type
Andrew Latham [Mon, 31 Jan 2011 13:57:53 +0000 (13:57 +0000)]
Asterisk HTTP response Content-type

Address content type for BSD and other platforms

(closes issue #18456)
Reported by: alexo
Patches:
    asterisk18_http.patch uploaded by alexo (license 1175)
Tested by: alexo

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305084 65c4cc65-6c06-0410-ace0-fbb531ad65f3