asterisk/asterisk.git
9 years agoMerged revisions 319812 via svnmerge from
Mark Murawki [Thu, 19 May 2011 18:12:49 +0000 (18:12 +0000)]
Merged revisions 319812 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | 9 lines

  In cel_odbc, an uninitialized RWLIST is attempted to be locked.

  Added INIT and DESTROY for the RWLIST odbc_tables

  (closes issue #19331)
  Reported by: kobaz
  Patches:
        odbc_cel.patch uploaded by kobaz (license 834)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319813 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 319758 via svnmerge from
Richard Mudgett [Thu, 19 May 2011 16:52:47 +0000 (16:52 +0000)]
Merged revisions 319758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines

  CCSS generic agent with POTS and ISDN phones fail caller busy call-back test.

  If the following is true after a CCSS activation:
  * The generic agent is for an analog phone or ISDN phone.  (Caller party)
  * The called party becomes available.
  * The caller party is not available.

  When the caller party becomes available, the caller is not alerted to the
  called party being available.  The generic agent still thinks the caller
  is busy.

  * Fixed the generic agent device state event subscription to look for all
  device states that are considered available.

  * Encapsulated the device state test for CCSS generic device available in
  cc_generic_is_device_available().  Made the generic agent and monitor use
  the new function instead of the manually coded inline equivalent.

  JIRA AST-559
  JIRA SWP-3462
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 319654 via svnmerge from
Terry Wilson [Wed, 18 May 2011 23:18:32 +0000 (23:18 +0000)]
Merged revisions 319654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines

  Merged revisions 319653 via svnmerge from
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    r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines

    Merged revisions 319652 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

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      r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines

      Make sure everyone gets an unhold when a transfer succeeds

      Some phones, like the Snom phones, send a hold to the transfer target after
      before sending the REFER. We need to make sure that we unhold the parties
      that are being connected after the masquerade. If Local channels with the /nm
      option are used when dialing the parties, hold music would still be playing on
      the transfer target, even after being connected with the transferee.
    ........
  ................
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319661 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 319552 via svnmerge from
Terry Wilson [Wed, 18 May 2011 20:25:32 +0000 (20:25 +0000)]
Merged revisions 319552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines

  Unbreak the storing of registrations for restart

  The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
  on restart/reload. This patch tries to unbreak things while leaving the intent
  of the original fix intact.
  (closes issue #19318)
  Reported by: remiq
  Patches:
        diff.txt uploaded by twilson (license 396)
  Tested by: lmadsen, remiq
........

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9 years agoMerged revisions 319529 via svnmerge from
Terry Wilson [Wed, 18 May 2011 20:07:07 +0000 (20:07 +0000)]
Merged revisions 319529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines

  Merged revisions 319528 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines

    Merged revisions 319527 via svnmerge from
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      r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines

      Fix app_dial ring groups

      Revert part of r315643. We need to remove the datastore here as well.
      The code in bridging code will catch anything that app_dial might miss.

      (closes issue #19311)
      Reported by: mspuhler
      Patches:
            issue_19311_no_answer.diff uploaded by elguero (license 37)
    ........
  ................
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9 years agoMerged revisions 319469 via svnmerge from
Richard Mudgett [Tue, 17 May 2011 22:04:59 +0000 (22:04 +0000)]
Merged revisions 319469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines

  Merged revision 319468 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier

  ..........
    r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines

    The mISDN HDLC mode is prevented on dialed channels.

    The use of mISDN HDLC mode is prevented if the mISDN dial technology
    option 'h1' is used when config option astdtmf=yes.

    There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
    mode.  Instead of setting the channel to HDLC mode it is set to
    transparent(no dsp, no hdlc), although hdlc is not "no hdlc".  I.e the
    logging message is correct, but the if condition is not.

    Make check the nodsp and hdlc flags.

    JIRA ABE-2787
    JIRA SWP-3437
  ..........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRemove extraneous line variables.
Damien Wedhorn [Tue, 17 May 2011 21:59:55 +0000 (21:59 +0000)]
Remove extraneous line variables.

The vars were either explicitly or implicitly not used.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOption needed for Q931_IE_TIME_DATE to be optional in CONNECT message.
Richard Mudgett [Tue, 17 May 2011 20:13:27 +0000 (20:13 +0000)]
Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message.

The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.

Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.

(closes issue #19221)
Reported by: kenner

JIRA SWP-3396

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 319367 via svnmerge from
Leif Madsen [Tue, 17 May 2011 12:54:13 +0000 (12:54 +0000)]
Merged revisions 319367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) | 10 lines

  Don't create [general] voicemail context when using users.conf

  Prior to this patch, app_voicemail would create a [general] context when parsing users.conf.

  (closes issue #18891)
  Reported by: pdugas
  Patches:
        app_voicemail-ignore-general.patch uploaded by pdugas (license 1222)
        app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71)
  Tested by: pdugas
........

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9 years agoMerged revisions 319365 via svnmerge from
Leif Madsen [Tue, 17 May 2011 12:40:02 +0000 (12:40 +0000)]
Merged revisions 319365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) | 6 lines

  Make Debian init script lsb compliant

  (closes issue #18896)
  Reported by: manwe
  Patches:
        debian_init_lsb.patch uploaded by manwe (license 1223)
........

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9 years agoFix up skinny hints.
Damien Wedhorn [Mon, 16 May 2011 21:39:33 +0000 (21:39 +0000)]
Fix up skinny hints.

Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 319261 via svnmerge from
Jonathan Rose [Mon, 16 May 2011 21:08:50 +0000 (21:08 +0000)]
Merged revisions 319261 via svnmerge from
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  r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines

  Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths.
........

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9 years agoMerged revisions 319259 via svnmerge from
Richard Mudgett [Mon, 16 May 2011 20:41:31 +0000 (20:41 +0000)]
Merged revisions 319259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines

  Deadlock between generic CCSS agent and native ISDN CCSS.

  Deadlock can occur when the generic CCSS agent is deleting duplicate CC
  offers and the native ISDN CC driver is processing an incoming CC message.
  The cc_core_instances container lock cannot be held when an agent or
  monitor callback is invoked without the possibility of a deadlock.

  * Make kill_duplicate_offers() remove the reference in cc_core_instances
  outside of the container lock.

  JIRA AST-566
  JIRA SWP-3469
........

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9 years agoMerged revisions 319204 via svnmerge from
Terry Wilson [Mon, 16 May 2011 18:21:17 +0000 (18:21 +0000)]
Merged revisions 319204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319204 | twilson | 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) | 11 lines

  Merged revisions 319202 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines

    Unlink a peer from peers_by_ip when expiring a registration

    Review: https://reviewboard.asterisk.org/r/1218/
  ........
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9 years agoMerged revisions 319145 via svnmerge from
David Vossel [Mon, 16 May 2011 15:58:12 +0000 (15:58 +0000)]
Merged revisions 319145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319145 | dvossel | 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) | 9 lines

  Merged revisions 319144 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines

    Fixes issue with peer ref-counting during handle_request_subscribe.
    (closes issue #19293)
    Reported by: irroot
  ........
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9 years agoMerged revisions 319142 via svnmerge from
Matthew Nicholson [Mon, 16 May 2011 15:54:52 +0000 (15:54 +0000)]
Merged revisions 319142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) | 8 lines

  Make sure tcptls_session exists before dereferencing it.

  (closes issue #19192)
  Reported by: stknob
  Patches:
        10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723)
  Tested by: vois, Chainsaw
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years ago When a error in T.38 negotiation happens or its rejected on a channel the
Gregory Nietsky [Mon, 16 May 2011 14:56:53 +0000 (14:56 +0000)]
 When a error in T.38 negotiation happens or its rejected on a channel the
 state of the channel reverts to unknown this should be rejected.

 this is important for negotiating T.38 gateway see #13405

 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.

 Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.

 (closes issue #18889)
 Reported by: irroot
 Tested by: irroot, darkbasic,  mnicholson

 Review: https://reviewboard.asterisk.org/r/1115

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 319085 via svnmerge from
Paul Belanger [Mon, 16 May 2011 14:38:16 +0000 (14:38 +0000)]
Merged revisions 319085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) | 10 lines

  Support gmime-2.4

  (closes issue #18863)
  Reported by: tzafrir
  Patches:
        gmime-2.4-18.diff uploaded by tzafrir (license 46)
        Tested by: tzafrir

  Review: https://reviewboard.asterisk.org/r/1213/
........

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9 years agoMerged revisions 319083 via svnmerge from
David Vossel [Mon, 16 May 2011 14:29:06 +0000 (14:29 +0000)]
Merged revisions 319083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) | 5 lines

  Fixes Big Endian build issue.

  (closes issue #19298)
  Reported by: tzafrir
........

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9 years agoAdd activatesub and dialandactivate sub.
Damien Wedhorn [Sun, 15 May 2011 23:17:57 +0000 (23:17 +0000)]
Add activatesub and dialandactivate sub.

When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319024 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 318921 via svnmerge from
Brett Bryant [Fri, 13 May 2011 18:10:45 +0000 (18:10 +0000)]
Merged revisions 318921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) | 8 lines

  Fixes a segmentation fault in dynamic hints when a channel technology isn't
  loaded for a hint.

  (closes issue #18495)
  Reported by: bertrand
  Tested by: bertrand
........

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9 years agoMerged revisions 318919 via svnmerge from
Brett Bryant [Fri, 13 May 2011 18:06:27 +0000 (18:06 +0000)]
Merged revisions 318919 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines

  This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too
  much time has passed between sending audio.

  (closes issue #18206)
  Reported by: bernhardsi
  Patches:
        res_srtp_unhold.patch uploaded by bernhards (license 1138)
  Tested by: bernhards, notthematrix
........

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9 years agoMerged revisions 318917 via svnmerge from
Brett Bryant [Fri, 13 May 2011 17:58:53 +0000 (17:58 +0000)]
Merged revisions 318917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) | 11 lines

  This patch allows TCP peers into the ast_db where they were previously
  restricted.

  (closes issue #18882)
  Reported by: cmaj
  Patches:
        patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
        uploaded by cmaj (license 830)
  Tested by: cmaj
........

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9 years agoMerged revisions 318868 via svnmerge from
Richard Mudgett [Fri, 13 May 2011 16:30:29 +0000 (16:30 +0000)]
Merged revisions 318868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines

  CDR's are being written immediately on caller hangup.

  CDR's are being written immediately on caller hangup.  The dialplan is not
  able to modify it in the h exten.  The h exten in the initial context is
  not run before closing CDR's when the bridge is unlinked if a macro is
  active and does not have an h exten.

  * Make ast_bridge_call() check for an h exten in the current context and
  if a macro is active then the initial context.  The first h exten found is
  then run before closing the CDR.

  (closes issue #18212)
  Reported by: leearcher
  Patches:
        issue18212_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett

  Review: https://reviewboard.asterisk.org/r/1206/
........

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9 years agoMove exten used for dialing from device to subchannel.
Damien Wedhorn [Fri, 13 May 2011 08:33:35 +0000 (08:33 +0000)]
Move exten used for dialing from device to subchannel.

There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.

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9 years agoMerged revisions 318720 via svnmerge from
Matthew Nicholson [Fri, 13 May 2011 01:55:38 +0000 (01:55 +0000)]
Merged revisions 318720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) | 4 lines

  Handle ipv6 addresses in the sent-by Via: field.

  This change fixes a regression in via header parsing and ipv6 handling.

  (closes issue #18951)
........

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9 years agoMerged revisions 318783 via svnmerge from
Richard Mudgett [Fri, 13 May 2011 01:50:15 +0000 (01:50 +0000)]
Merged revisions 318783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines

  PRI early media won't ring.

  And another way to pass early media.  Don't indicate that there is inband
  information present, just assume that the B channel is connected.

  * Restore clearing the dialing flag Rx squelch unconditionally when a
  PROCEEDING message comes in.

  (closes issue #19268)
  Reported by: tbsky
  Patches:
        issue19268_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: tbsky
........

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9 years agoMerged revisions 318671 via svnmerge from
Alec L Davis [Thu, 12 May 2011 22:56:43 +0000 (22:56 +0000)]
Merged revisions 318671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines

  Fix directed group pickup feature code *8 with pickupsounds enabled

  Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.

  1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
  2). dialplan applications for directed_pickups shouldn't beep.
  3). feature code for directed pickup should beep on success/failure if configured.

  Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.

  Moved app_directed:pickup_do() to features:ast_do_pickup().

  Functions below, all now use the new ast_do_pickup()
  app_directed_pickup.c:
     pickup_by_channel()
     pickup_by_exten()
     pickup_by_mark()
     pickup_by_part()
  features.c:
     ast_pickup_call()

  (closes issue #18654)
  Reported by: Docent
  Patches:
        ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
  Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett

  Review: https://reviewboard.asterisk.org/r/1185/
........

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9 years agoConsolidate setsubstate_* into setsubstate and use a switch.
Damien Wedhorn [Thu, 12 May 2011 20:44:21 +0000 (20:44 +0000)]
Consolidate setsubstate_* into setsubstate and use a switch.

Consolidate the functions and add some debugging info. Allows to be
able to set a substate without explicitly knowing what the state is.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318635 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd setsubstate_onhook.
Damien Wedhorn [Thu, 12 May 2011 07:25:52 +0000 (07:25 +0000)]
Add setsubstate_onhook.

Add the setsubstate_onhook to complete the initial substate handling
procedures. Added dumpsub(sub, forcehangup) which is the common way of
calling setsubstate_onhook. Dumpsub attempts to activate another sub
after setting the current one onhook.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 318550 via svnmerge from
Terry Wilson [Wed, 11 May 2011 18:52:53 +0000 (18:52 +0000)]
Merged revisions 318550 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) | 2 lines

  Comment out the REF_DEBUG that slipped in during debugging
........

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9 years agoMerged revisions 318549 via svnmerge from
Terry Wilson [Wed, 11 May 2011 18:50:51 +0000 (18:50 +0000)]
Merged revisions 318549 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r318549 | twilson | 2011-05-11 13:39:48 -0500 (Wed, 11 May 2011) | 27 lines

  Merged revisions 318548 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) | 19 lines

    Clean up several chan_sip reference leaks

    Several situations in the code could lead to peers or sip_pvt references
    being leaked. This would cause RTP ports to never be destroyed (leading
    to exhaustion of all available RTP ports) and memory leaks.

    The original patch for this issue from rgagnon was the result of an
    obscene amount of testing and hard work, for which I am very grateful. I
    did some cleanup and added a few additional refcount fixes that I found.

    (closes issue #17255)
    Reported by: kvveltho
    Patches:
          tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by rgagnon (license 1202)
    Tested by: rgagnon, twilson, wdoekes, loloski

    Review: https://reviewboard.asterisk.org/r/1101/
    Review: https://reviewboard.asterisk.org/r/1207/
    Review: https://reviewboard.asterisk.org/r/1210/
  ........
................

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9 years agoMerged revisions 318499 via svnmerge from
Richard Mudgett [Tue, 10 May 2011 23:42:57 +0000 (23:42 +0000)]
Merged revisions 318499 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines

  Unable to pickup DAHDI/PRI call because call state is reported as DIALING.

  The channel state is not updated to RINGING when an ALERTING message is
  received.  Regression caused when sig_pri.c (also sig_ss7.c) extracted
  from chan_dahdi.c.

  * Added missing channel state update to RINGING when the
  AST_CONTROL_RINGING frame is queued for ISDN and SS7.

  (closes issue #19257)
  Reported by: alecdavis
  Patches:
        issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: alecdavis, rmudgett
........

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9 years agoMerged revisions 318436 via svnmerge from
Russell Bryant [Tue, 10 May 2011 15:16:34 +0000 (15:16 +0000)]
Merged revisions 318436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 May 2011) | 2 lines

  chan_iax2: change LOG_NOTICE to LOG_DEBUG in iax2_read().
........

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9 years agoMerged revisions 318337 via svnmerge from
Terry Wilson [Tue, 10 May 2011 00:22:02 +0000 (00:22 +0000)]
Merged revisions 318337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r318337 | twilson | 2011-05-09 15:23:15 -0500 (Mon, 09 May 2011) | 18 lines

  Merged revisions 318331 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines

    Don't offer video to directmedia callee unless caller offered it as well

    Make sure that when directmedia is enabled, that video is not offered to the
    callee even if it supports it. p->vrtp will not exist since the caller didn't
    offer video.

    (closes issue #19195)
    Reported by: one47
    Patches:
          sip_cant_add_video_rtp uploaded by one47 (license 23)
  ........
................

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9 years agoMerged revisions 318351 via svnmerge from
Richard Mudgett [Mon, 9 May 2011 23:16:12 +0000 (23:16 +0000)]
Merged revisions 318351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines

  Remove references to res_features and its export file.

  The contents of res/res_features.c was moved to into main/features.c
  awhile ago.  There is no longer any need for the res/Makefile to reference
  res_features or the res_features linker exports file to exist.
........

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9 years agoMerged revisions 318282 via svnmerge from
Richard Mudgett [Mon, 9 May 2011 19:09:16 +0000 (19:09 +0000)]
Merged revisions 318282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines

  Hangup extension executed twice.

  When a user hangs up a call, in certain circumstances, the hangup
  extension can end up being executed twice:

  1) If a call is bridged and the 'h' extension executes the Hangup
  application, then the 'h' extension will be executed twice.

  2) If a call is bridged within a macro (Dial or Queue), it has its own 'h'
  extension, the main context also has an 'h' extension, and the macro 'h'
  extension executes the Hangup application, then both 'h' extensions will
  be executed.

  * Revert originally commited fix for #16106 and just set
  AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call().  The
  bridge code just executed an 'h' extension so the main PBX loop does not
  need to execute one as well.

  (issue #16106)
  Reported by: ajohnson

  (issue #16548)
  Reported by: hajekd
........

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9 years agoMerged revisions 318233 via svnmerge from
David Vossel [Mon, 9 May 2011 17:13:01 +0000 (17:13 +0000)]
Merged revisions 318233 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r318233 | dvossel | 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) | 14 lines

  Merged revisions 318230 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ........
    r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) | 7 lines

    Fixes cases where sip_set_rtp_peer can return too early during media path reset.

    (closes issue #19225)
    Reported by: one47
    Patches:
          sip_set_rtp_peer.patch uploaded by one47 (license 23)
  ........
................

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9 years agoMerged revisions 318231 via svnmerge from
Richard Mudgett [Mon, 9 May 2011 17:00:05 +0000 (17:00 +0000)]
Merged revisions 318231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines

  Don't get early media for ISDN on outgoing calls.

  It looks to be a long-standing misinterpretation of the progress indicator
  ie values:
  1 - Call is not end-to-end ISDN; further call progress information may be
  available in-band.
  8 - In-band information or an appropriate pattern is now available.

  Only value 8 is handled by chan_dahdi/sig_pri.  The 1 value is not handled
  as early media probably because the meaning of the second half of it's
  description was overlooked.

  * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
  PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.

  (closes issue #18868)
  Reported by: isrl
  Patches:
        issue18868_19246_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: satish_lx

  ..........

  No inband progress on PRI_EVENT_RINGING even if inband flag set.

  My ISDN-PRI provider sends an ALERTING with "Inband information or
  appropriate pattern now available", but Asterisk only generates and passes
  the RING to the SIP extension, not the inband message.  Unfortunately, the
  inband message is not a ringback tone but a prompt that says the number is
  not in service.  The SIP extension then hears two rings and the call is
  hungup which confuses the caller.

  * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband
  audio is indicated with an ALERTING message.

  (closes issue #19246)
  Reported by: cristiandimache
  Patches:
        issue19246_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: cristiandimache
................

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9 years agoIncrease prepend filename length.
Leif Madsen [Mon, 9 May 2011 14:41:33 +0000 (14:41 +0000)]
Increase prepend filename length.

(closes issue #19238)
Reported by: byronclark
Patches:
      increase_prepend_filename_length.patch uploaded by byronclark (license 1200)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318194 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMinor change to 318141 to improve parsing behavior.
Jonathan Rose [Mon, 9 May 2011 14:37:10 +0000 (14:37 +0000)]
Minor change to 318141 to improve parsing behavior.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318193 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 318148 via svnmerge from
Jonathan Rose [Mon, 9 May 2011 14:21:33 +0000 (14:21 +0000)]
Merged revisions 318148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines

  Documenting an observed behavior of features in features.conf.  Since parkinglots use an
  integer for the parkinglot extensions, leading zeros specified in the configuration file
  are ignored.
........

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9 years agoMerged revisions 318142 via svnmerge from
Matthew Nicholson [Mon, 9 May 2011 14:11:57 +0000 (14:11 +0000)]
Merged revisions 318142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) | 9 lines

  Make indicate/control frames WRITE events on framehooks.  Also, if a framehook
  returns a non-control frame, don't forward it to the channel.

  (closes issue #19251)
  Reported by: irroot
  Patches:
        (modified) framehook_indicate.patch2 uploaded by irroot (license 52)
  Tested by: irroot
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAllows ParkedCall application to specify a parkinglot.
Jonathan Rose [Mon, 9 May 2011 13:56:32 +0000 (13:56 +0000)]
Allows ParkedCall application to specify a parkinglot.

When invoking the app parkedcall, the argument can now include '@parkinglot' after the
extension.

(closes issue #18777)
Reported by: cartama
Patches:
      0018777.diff uploaded by cartama (license 1157)

Review: https://reviewboard.asterisk.org/r/1209/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318141 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd setsubstate_callwait.
Damien Wedhorn [Mon, 9 May 2011 07:40:40 +0000 (07:40 +0000)]
Add setsubstate_callwait.

If a call is made to a line that already has a call and the device is
offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318106 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 318057 via svnmerge from
Russell Bryant [Sat, 7 May 2011 23:36:41 +0000 (23:36 +0000)]
Merged revisions 318057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 May 2011) | 8 lines

  res_config_curl: fix a crash with static realtime.

  (closes issue #18413)
  Reported by: jmls
  Patches:
        20101202__issue18413.diff.txt uploaded by tilghman (license 14)
  Tested by: jmls
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 318055 via svnmerge from
Russell Bryant [Sat, 7 May 2011 23:26:05 +0000 (23:26 +0000)]
Merged revisions 318055 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) | 7 lines

  chan_iax2: Don't overwrite port found with an SRV lookup.

  (closes issue #17291)
  Reported by: jcovert
  Patches:
        chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoOnly allow voicemail if substate is OFFHOOK or no channel active (UNSET).
Damien Wedhorn [Fri, 6 May 2011 23:07:55 +0000 (23:07 +0000)]
Only allow voicemail if substate is OFFHOOK or no channel active (UNSET).

(closes issue #17901)
Reported by: salecha

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoRename sub->parent to sub->line.
Damien Wedhorn [Fri, 6 May 2011 22:32:45 +0000 (22:32 +0000)]
Rename sub->parent to sub->line.

Improve readability of code, eg, (sub->parent == d->activeline) becomes
(sub->line == d->activeline).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMove the hookstate from line to device.
Damien Wedhorn [Fri, 6 May 2011 22:24:08 +0000 (22:24 +0000)]
Move the hookstate from line to device.

Long time coming, finally moving the hookstate from line to device.
This may fix some issues where a device has multiple lines. Previously
we had to run through all lines on a device to see if it was actually
onhook or not.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317996 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317969 via svnmerge from
Russell Bryant [Fri, 6 May 2011 21:49:47 +0000 (21:49 +0000)]
Merged revisions 317969 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines

  Use the right variable to print the time in a debug message.

  The original patch also increased some buffer sizes, but that was already
  done in this version.

  (closes issue #17034)
  Reported by: sysreq
  Patches:
        asterisk-issue-17034.patch uploaded by sysreq (license 1009)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317970 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317967 via svnmerge from
Russell Bryant [Fri, 6 May 2011 21:47:05 +0000 (21:47 +0000)]
Merged revisions 317967 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines

  Fix some more "set but unused" compiler warnings.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317968 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317918 via svnmerge from
David Vossel [Fri, 6 May 2011 21:10:30 +0000 (21:10 +0000)]
Merged revisions 317918 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 May 2011) | 7 lines

  Fixes missing colon from To/From headers in RTCP manager events.

  (closes issue #18221)
  Reported by: clegall_proformatique
  Patches:
        18221_1.patch uploaded by ebroad (license 878)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317917 via svnmerge from
Russell Bryant [Fri, 6 May 2011 21:07:49 +0000 (21:07 +0000)]
Merged revisions 317917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) | 7 lines

  Fix calculation of free RAM to make minmemfree option work.

  (closes issue #17124)
  Reported by: loic
  Patches:
        pbx_c.diff uploaded by loic (license 1020)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd a cdr_csv to MySQL import script to contrib/scripts.
Russell Bryant [Fri, 6 May 2011 20:47:37 +0000 (20:47 +0000)]
Add a cdr_csv to MySQL import script to contrib/scripts.

(closes issue #17036)
Reported by: precisenetworks
Patches:
      import-cdr-csv-mysql.pl uploaded by precisenetworks (license 1010)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317916 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd the Uniqueid header to Userevent.
Russell Bryant [Fri, 6 May 2011 20:44:53 +0000 (20:44 +0000)]
Add the Uniqueid header to Userevent.

(closes issue #16962)
Reported by: jlpedrosa
Patches:
      patch.diff uploaded by jlpedrosa (license 1002)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317915 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317867 via svnmerge from
Russell Bryant [Fri, 6 May 2011 20:02:31 +0000 (20:02 +0000)]
Merged revisions 317867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines

  chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer.

  Don't duplicate variables on the sip_pvt.  Just reset the variable list each
  time.

  (closes issue #19202)
  Reported by: wdoekes
  Patches:
        issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317865 via svnmerge from
Russell Bryant [Fri, 6 May 2011 19:48:06 +0000 (19:48 +0000)]
Merged revisions 317865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines

  chan_sip: fix a deadlock in check_rtp_timeout.

  Don't block doing silly deadlock avoidance.  Just return and try again later.
  The funciton gets called often enough that it's fine.  Also, this change was
  already made in trunk.

  (closes issue #18791)
  Reported by: irroot
  Patches:
        chan_sip.rtptimeout.patch uploaded by irroot (license 52)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 317861 via svnmerge
Russell Bryant [Fri, 6 May 2011 19:35:30 +0000 (19:35 +0000)]
Blocked revisions 317861 via svnmerge

........
  r317861 | russell | 2011-05-06 14:35:00 -0500 (Fri, 06 May 2011) | 11 lines

  URI encode less characters in the RPID and Contact headers.

  If this change causes any problems, we will need to backport the more extensive
  uri encoding and decoding handling changes that are in trunk/1.10.

  (closes issue #18686)
  Reported by: wolfgang
  Patches:
        quick-and-dirty.patch uploaded by wdoekes (license 717)
  Tested by: wdoekes, devellow, wolfgang, mav3rick
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317862 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoBlocked revisions 317858 via svnmerge
Matthew Nicholson [Fri, 6 May 2011 19:34:46 +0000 (19:34 +0000)]
Blocked revisions 317858 via svnmerge

........
  r317858 | mnicholson | 2011-05-06 14:31:50 -0500 (Fri, 06 May 2011) | 6 lines

  pbx_lua autoservice fixes

  Don't start an autoservice in pbx_lua if pbx_lua already started one and don't
  stop one if we didn't start one.  Also start and stop the autoservice when
  transferring control from and to the pbx.
........

This change is already implemented in trunk.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317860 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317837 via svnmerge from
Russell Bryant [Fri, 6 May 2011 19:25:35 +0000 (19:25 +0000)]
Merged revisions 317837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines

  Fix a crash in the MySQL() application.

  This code was not handling channel datastores safely.  The channel
  must be locked.

  (closes issue #17964)
  Reported by: wuwu
  Patches:
        issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71)
  Tested by: wuwu
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317843 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdated CHANGES to note the autoservice changes for pbx_lua
Matthew Nicholson [Fri, 6 May 2011 19:23:23 +0000 (19:23 +0000)]
Updated CHANGES to note the autoservice changes for pbx_lua

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdated the sample pbx_lua config file to reflect autoservice changes.
Matthew Nicholson [Fri, 6 May 2011 19:19:56 +0000 (19:19 +0000)]
Updated the sample pbx_lua config file to reflect autoservice changes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317818 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317805 via svnmerge from
Russell Bryant [Fri, 6 May 2011 19:15:45 +0000 (19:15 +0000)]
Merged revisions 317805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) | 7 lines

  Add a new sipfriends.sql for MySQL that has more fields in it.

  (closes issue #16399)
  Reported by: pabelanger
  Patches:
        sipfriends.sql.v3 uploaded by pabelanger (license 224)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317807 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDefault to starting an autoservice in pbx_lua. The autoservice is
Matthew Nicholson [Fri, 6 May 2011 19:14:39 +0000 (19:14 +0000)]
Default to starting an autoservice in pbx_lua.  The autoservice is
automatically stopped when applications are executed, so this shouldn't cause
any problems.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317806 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMake pbx_lua handle managing the autoservice better.
Matthew Nicholson [Fri, 6 May 2011 19:01:57 +0000 (19:01 +0000)]
Make pbx_lua handle managing the autoservice better.

Make autoservice_start() and autoservice_stop() return nothing.  Also check if
the autoservice flag is set before starting or stopping the autoservice and
stop and start the autoservice when returning control to and getting control
from the pbx engine.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317803 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdded note about changes in pbx_lua's behavior when applications do dialplan jumps
Matthew Nicholson [Fri, 6 May 2011 18:40:35 +0000 (18:40 +0000)]
Added note about changes in pbx_lua's behavior when applications do dialplan jumps

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317802 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUse two spaces after periods for the recent pbx_lua change descriptions
Matthew Nicholson [Fri, 6 May 2011 18:07:05 +0000 (18:07 +0000)]
Use two spaces after periods for the recent pbx_lua change descriptions

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317723 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoUpdated CHANGES for hints support in pbx_lua
Matthew Nicholson [Fri, 6 May 2011 18:05:52 +0000 (18:05 +0000)]
Updated CHANGES for hints support in pbx_lua

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317722 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoDetect Goto in pbx_lua.
Matthew Nicholson [Fri, 6 May 2011 18:04:23 +0000 (18:04 +0000)]
Detect Goto in pbx_lua.

This code will actually detect any dialplan jump from any application that
calls ast_explicit_goto().  This change is only being done in trunk as it may
change the way some dialplans execute.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317670 via svnmerge from
Richard Mudgett [Fri, 6 May 2011 16:23:14 +0000 (16:23 +0000)]
Merged revisions 317670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines

  Fix SIP connected line updates.

  This patch fixes a couple SIP connected line update problems:

  1) The connected line needs to be updated when the initial INVITE is sent
  if there is a peer callerid configured.  Previously, the connected line
  information did not get reported until the call was connected so SIP could
  not report connected line information in ringing or progress messages.

  2) The connected line should not be updated on initial connect if there is
  no connected line information.  Previously, all it did was wipe out any
  default preset CONNECTEDLINE information set by the dialplan with empty
  strings.

  (closes issue #18367)
  Reported by: GeorgeKonopacki
  Patches:
        issue18367_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett

  Review: https://reviewboard.asterisk.org/r/1199/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317584 via svnmerge from
Terry Wilson [Fri, 6 May 2011 08:21:22 +0000 (08:21 +0000)]
Merged revisions 317584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines

  Merged revisions 317575 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines

    Merged revisions 317574 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines

      Re-fix queue round-robin

      This part of the change for r315596 was incorrect. No bridge occurs
      when doing a roundrobin dial and no one answers, so this code shouldn't
      have been removed.
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317596 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317530 via svnmerge from
Russell Bryant [Thu, 5 May 2011 23:47:23 +0000 (23:47 +0000)]
Merged revisions 317530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) | 10 lines

  If the configure script runs, force a rebuild of menuselect-tree.

  Some contents in the menuselect tree are dependent on configure script
  parameters, namely --enable-dev-mode.

  (closes issue #17219)
  Reported by: Nick_Lewis
  Patches:
        issue_17219.rev1.txt uploaded by russell (license 2)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317486 via svnmerge from
Russell Bryant [Thu, 5 May 2011 23:16:16 +0000 (23:16 +0000)]
Merged revisions 317486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) | 9 lines

  Fix some more realtime MySQL schema issues.

  (closes issue #18537)
  Reported by: denzs
  Patches:
        sipfriends.sql.svndiff uploaded by denzs (license 1182)
        queue_log.sql.svndiff uploaded by denzs (license 1182)
        meetme.sql.svndiff uploaded by denzs (license 1182)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317487 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317484 via svnmerge from
Russell Bryant [Thu, 5 May 2011 23:13:04 +0000 (23:13 +0000)]
Merged revisions 317484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) | 8 lines

  Fix some errors in sample MySQL realtime schema files.

  (closes issue #18915)
  Reported by: Dovid
  Patches:
        sipfriends.patch uploaded by Dovid (license 652)
        meetme.patch uploaded by Dovid (license 652)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317485 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd "calendar show types" CLI command.
Russell Bryant [Thu, 5 May 2011 23:10:27 +0000 (23:10 +0000)]
Add "calendar show types" CLI command.

(closes issue #18246)
Reported by: junky
Patches:
      calendar_types.diff uploaded by junky (license 177)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd CEL extra field to cel_pgsql.
Russell Bryant [Thu, 5 May 2011 23:08:05 +0000 (23:08 +0000)]
Add CEL extra field to cel_pgsql.

(closes issue #18462)
Reported by: joscas
Patches:
      bug_18462.diff uploaded by snuffy (license 35)
      cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317482 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317480 via svnmerge from
Russell Bryant [Thu, 5 May 2011 23:02:11 +0000 (23:02 +0000)]
Merged revisions 317480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011) | 8 lines

  Don't lose cdr_syslog config on a reload.

  (closes issue #18679)
  Reported by: enegaard
  Patches:
        issue18679_seanbright.patch uploaded by seanbright (license 71)
  Tested by: enegaard
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317481 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317478 via svnmerge from
Russell Bryant [Thu, 5 May 2011 22:55:09 +0000 (22:55 +0000)]
Merged revisions 317478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines

  Fix some consistency issues with jitterbuffer config.

  Store the defaults noted in the sample config files in the jitterbuffer config
  data structure.  This makes the CLI commands that output these settings show
  the right thing.  Also only show the settings that are relevant in the settings
  CLI commands, based on which jitterbuffer is selected and whether it's enabled.

  (closes issue #19083)
  Reported by: rgagnon
  Patches:
        issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317476 via svnmerge from
Russell Bryant [Thu, 5 May 2011 22:49:36 +0000 (22:49 +0000)]
Merged revisions 317476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011) | 8 lines

  Add a datastore fixup to fix a pbx_lua crash.

  (closes issue #19055)
  Reported by: jamhed
  Patches:
        lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
  Tested by: mnicholson, jamhed
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317477 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317474 via svnmerge from
Russell Bryant [Thu, 5 May 2011 22:44:52 +0000 (22:44 +0000)]
Merged revisions 317474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) | 2 lines

  Fix more "set but unused" warnings.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317429 via svnmerge from
Russell Bryant [Thu, 5 May 2011 22:12:10 +0000 (22:12 +0000)]
Merged revisions 317429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) | 5 lines

  Only display inband DTMF warning if inband DTMF detection is enabled.

  (closes issue #18901)
  Reported by: irroot
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317430 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317427 via svnmerge from
Russell Bryant [Thu, 5 May 2011 22:02:31 +0000 (22:02 +0000)]
Merged revisions 317427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) | 7 lines

  Fix potential memory leak, and use of uninitialized memory.

  (closes issue #16476)
  Reported by: junky
  Patches:
        M16476.diff uploaded by junky (license 177)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317428 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317425 via svnmerge from
Russell Bryant [Thu, 5 May 2011 21:54:17 +0000 (21:54 +0000)]
Merged revisions 317425 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317425 | russell | 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) | 7 lines

  Add missing ActioID handling to Events action.

  (closes issue #18949)
  Reported by: edersohe
  Patches:
        0018949.patch uploaded by edersohe (license 1228)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317426 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd some new editline bindings by default, and allow for user specified configuration.
Sean Bright [Thu, 5 May 2011 21:20:00 +0000 (21:20 +0000)]
Add some new editline bindings by default, and allow for user specified configuration.

I excluded the part of this patch that used the HOME environment variable since
the built-in editline library goes to great lengths to disallow that.  Instead
only settings the EDITRC environment variable will use a user specified file.

Also, the default environment variable use to determine the edit more is
AST_EDITMODE instead of AST_EDITOR (although the latter is still supported).

(closes issue #15929)
Reported by: kkm
Patches:
      astcli-editrc-v2.diff uploaded by kkm (license 888)
      015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
Tested by: seanbright

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317395 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMove hold stuff to the setsubstate arrangement.
Damien Wedhorn [Thu, 5 May 2011 20:46:49 +0000 (20:46 +0000)]
Move hold stuff to the setsubstate arrangement.

skinny_hold moved to setsubstate_hold and skinny_unhold integrated into
setsubstate_connected. Removed sub->onhold and replaced with
SUBSTATE_HOLD.

Also fixed inbound call answering by queueing an AST_CONTROL_ANSWER on
answering a SUBSTATE_RINGIN sub (was a typo).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317382 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317370 via svnmerge from
Sean Bright [Thu, 5 May 2011 20:27:25 +0000 (20:27 +0000)]
Merged revisions 317370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May 2011) | 10 lines

  Don't duplicate our data on the stack and just use the MYSQL_ROW directly.

  With large result sets we were blowing out the stack.

  (closes issue #19090)
  Reported by: mickecarlsson
  Patches:
        issue19090_trunk_svn.patch uploaded by seanbright (license 71)
  Tested by: mickecarlsson
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317377 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317336 via svnmerge from
Russell Bryant [Thu, 5 May 2011 19:56:44 +0000 (19:56 +0000)]
Merged revisions 317336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) | 7 lines

  Increase buffer size to be PATH_MAX for a path.

  (closes issue #19239)
  Reported by: byronclark
  Patches:
        queue_announce_length.patch uploaded by byronclark (license 1200)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317337 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317283 via svnmerge from
Jonathan Rose [Thu, 5 May 2011 19:33:11 +0000 (19:33 +0000)]
Merged revisions 317283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | 10 lines

  Resolves a deadlock that occurs during sip_new

  This is based on an uncommitted patch by jpeeler for the issue.  Instead of
  relocking and then unlocking the channel though, we keep the lock on the channel
  until we are finished doing what we need to the channel.

  (closes issue #18441)
  Reported by: Alric
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317334 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317281 via svnmerge from
Russell Bryant [Thu, 5 May 2011 18:46:22 +0000 (18:46 +0000)]
Merged revisions 317281 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r317281 | russell | 2011-05-05 13:39:44 -0500 (Thu, 05 May 2011) | 29 lines

  Merged revisions 317255 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2

  ................
    r317255 | russell | 2011-05-05 13:29:53 -0500 (Thu, 05 May 2011) | 22 lines

    Merged revisions 317211 via svnmerge from
    https://origsvn.digium.com/svn/asterisk/branches/1.4

    ........
      r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) | 15 lines

      chan_sip: fix broken realtime peer count, fix memory leak

      This patch addresses two bugs in chan_sip:

      1) The count of realtime peers and users was off.  The increment checked the
      value of the caching option, while the decrement did not.

      2) Add a missing regfree() for a regex.

      (closes issue #19108)
      Reported by: vrban
      Patches:
            missing_regfree.patch uploaded by vrban (license 756)
            sip_object_counter.patch uploaded by vrban (license 756)
    ........
  ................
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317196 via svnmerge from
Matthew Nicholson [Thu, 5 May 2011 18:09:23 +0000 (18:09 +0000)]
Merged revisions 317196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May 2011) | 8 lines

  Set SO_KEEPALIVE on SIP TCP sockets so that they eventually go away when a peer
  abruptly disappears.  This mostly occurs after a successful registration.

  (closes issue #17544)
  Reported by: marcelloceschia
  Patches:
        (modified) tcptls.patch uploaded by st (license 907)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317198 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoFixes reliability issues with func_jitterbuffer's usage in the new ConfBridge applica...
David Vossel [Thu, 5 May 2011 18:08:42 +0000 (18:08 +0000)]
Fixes reliability issues with func_jitterbuffer's usage in the new ConfBridge application.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317104 via svnmerge from
Leif Madsen [Thu, 5 May 2011 15:06:59 +0000 (15:06 +0000)]
Merged revisions 317104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r317104 | lmadsen | 2011-05-05 11:04:24 -0400 (Thu, 05 May 2011) | 15 lines

  Merged revisions 317102 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

  ........
    r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) | 8 lines

    Disable console colourization inside safe_asterisk checks.

    (closes issue #19213)
    Reported by: lefoyer
    Patches:
          issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by wdoekes (license 717)
    Tested by: wdoekes, lefoyer
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 317058 via svnmerge from
Leif Madsen [Thu, 5 May 2011 12:28:40 +0000 (12:28 +0000)]
Merged revisions 317058 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) | 7 lines

  Remove unused directory and clear up some documentation.

  (closes issue #19193)
  Reported by: bchia
  Patches:
        cel-csv.diff uploaded by lathama (license 1028)
  Tested by: lathama, Marquis42
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317059 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd setsubstate_congestion and setsubstate_progress.
Damien Wedhorn [Thu, 5 May 2011 09:03:32 +0000 (09:03 +0000)]
Add setsubstate_congestion and setsubstate_progress.

Move handling of both state handling from skinny_indicate to it's own sub.
Also, modified behaviour to not hangup the sub and let the dialplan
have a chance in doing what it wants for congestion. Added various states to
substate2str and added these states where applicable for other set_substate_
procs.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317026 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd setsubstate_busy.
Damien Wedhorn [Thu, 5 May 2011 08:10:14 +0000 (08:10 +0000)]
Add setsubstate_busy.

Move handling of setting busy state from skinny_indicate to it's own sub.
Also, modified behaviour to not hangup the sub and let the dialplan
have a chance in doing what it wants (eg busy(10); hangup() in the dialplan
now gives a busy indication for 10 secs and then hangs up.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316994 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdding the Move to Front Hash functionality
Stefan Schmidt [Thu, 5 May 2011 07:09:20 +0000 (07:09 +0000)]
Adding the Move to Front Hash functionality

Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists.

(closes issue #19233)
Reported by: schmidts

Review: https://reviewboard.asterisk.org/r/1201/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 316917-316919 via svnmerge from
Sean Bright [Thu, 5 May 2011 02:34:29 +0000 (02:34 +0000)]
Merged revisions 316917-316919 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) | 5 lines

  Make sure that tcptls_session is properly initialized.

  (issue #18598)
  Reported by: ksn
........
  r316918 | seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 lines

  Look at the correct buffer for our digest info instead of an empty one.

  (issue #18598)
  Reported by: ksn
........
  r316919 | seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 lines

  Use the correct HTTP method when generating our digest, otherwise we always fail.

  When calculating the 'A2' portion of our digest for verification, we need the
  HTTP method that is currently in use.  Unfortunately our mapping function was
  incorrect, resulting in invalid hashes being generated and, in turn, failures
  in authentication.

  (closes issue #18598)
  Reported by: ksn
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316920 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoAdd setsubstate_ringout (equivalent to AST_STATE ringing).
Damien Wedhorn [Wed, 4 May 2011 21:44:06 +0000 (21:44 +0000)]
Add setsubstate_ringout (equivalent to AST_STATE ringing).

Renamed previous setsubstate_ringout to setsubstate_dialing for a state
when attempting to dial a number, substate ringout now for when core
has indicated that the channel is actually ringing on the other end.
Also added substate2str for debugging purposes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316885 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9 years agoMerged revisions 316831 via svnmerge from
Richard Mudgett [Wed, 4 May 2011 18:57:02 +0000 (18:57 +0000)]
Merged revisions 316831 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines

  Wait for leader with Music On Hold allows crosstalk between participants.

  Parenthesis in the wrong position.  Regression from issue #14365 when
  expanding conference flags to use 64 bits.

  (closes issue #18418)
  Reported by: MrHanMan
  Tested by: rmudgett
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316832 65c4cc65-6c06-0410-ace0-fbb531ad65f3