asterisk-tools:lbts-asterisk-with-uk-clid.git
8 years agoMerged revisions 273793 via svnmerge from svn_trunk
Tilghman Lesher [Sat, 3 Jul 2010 02:36:31 +0000 (02:36 +0000)]
Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines

  Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.

  (closes issue #17407)
   Reported by: pdf
   Patches:
         20100527__issue17407.diff.txt uploaded by tilghman (license 14)

  Review: https://reviewboard.asterisk.org/r/751/
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273830 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 273717 via svnmerge from
Tilghman Lesher [Fri, 2 Jul 2010 17:10:59 +0000 (17:10 +0000)]
Merged revisions 273717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines

  Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.

  (closes issue #17564)
   Reported by: ramonpeek
   Patches:
         20100630__issue17564.diff.txt uploaded by tilghman (license 14)
   Tested by: ramonpeek
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273718 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoBlocked revisions 273639 via svnmerge
Tilghman Lesher [Fri, 2 Jul 2010 16:57:50 +0000 (16:57 +0000)]
Blocked revisions 273639 via svnmerge

........
  r273639 | tilghman | 2010-07-02 10:46:27 -0500 (Fri, 02 Jul 2010) | 8 lines

  If all members are paused, the wrong status is indicated.

  (closes issue #17576)
   Reported by: ramonpeek
   Patches:
         diff.txt uploaded by ramonpeek (license 266)
   Tested by: ramonpeek
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273715 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoThe switch fallthrough could create some errorneous situations, so best to force...
Tilghman Lesher [Fri, 2 Jul 2010 16:57:28 +0000 (16:57 +0000)]
The switch fallthrough could create some errorneous situations, so best to force directly to the default case.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273714 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoFix various typos reported by Lintian
Tzafrir Cohen [Fri, 2 Jul 2010 15:57:02 +0000 (15:57 +0000)]
Fix various typos reported by Lintian

(Also fix the typos in the comments)

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 273565 via svnmerge from
Russell Bryant [Thu, 1 Jul 2010 22:16:23 +0000 (22:16 +0000)]
Merged revisions 273565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines

  Don't return a partially initialized datastore.

  If memory allocation fails in ast_strdup(), don't return a partially
  initialized datastore.  Bad things may happen.

  (related to ABE-2415)
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273566 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 273474 via svnmerge from
Jeff Peeler [Thu, 1 Jul 2010 20:28:15 +0000 (20:28 +0000)]
Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines

  Allow admin user to join conference without using admin mode and no user pin.

  Configuring the conference in meetme.conf like the following:
  conf => 2345,,6666
  did not prompt for pin when used without admin mode. This meant that the
  conference could not be joined as an admin even if the user knew the correct
  pin. The original bug report was submitted claiming that the blank user pin
  should deny entry into the conference. I think a better way to handle this
  would be with a feature enhancement that used the following syntax:
  conf => 2345,X,6666 - where X denotes no acceptable pin allowed

  (closes issue #15704)
  Reported by: modelnine
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273522 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoProperly handle failures of fax->start_session()
Matthew Nicholson [Thu, 1 Jul 2010 19:34:47 +0000 (19:34 +0000)]
Properly handle failures of fax->start_session()

FAX-177

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273464 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agocorrect handling of get_destination return values
David Vossel [Thu, 1 Jul 2010 16:40:17 +0000 (16:40 +0000)]
correct handling of get_destination return values

A failure when calling the get_destination can mean multiple things.  If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate.  This patch adds the
get_destination_result enum to differentiate between these and other failure
types.  The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized.  This indicates to the initiator of the INVITE to retry the request
with a correct URI.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273427 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 273354 via svnmerge from
Jeff Peeler [Thu, 1 Jul 2010 15:12:31 +0000 (15:12 +0000)]
Merged revisions 273354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines

  Ensure channel placed in meetme in ringing state is properly hung up.

  An outgoing channel placed in meetme while still ringing which was then hung up
  would not exit meetme and the channel was not properly destroyed. Specifically
  checking for this scenario by looking at the appropriate control frames resolves
  the issue.

  (closes issue #15871)
  Reported by: Ivan
  Patches:
        meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273355 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoFixed whitespace problems
Matthew Nicholson [Thu, 1 Jul 2010 14:37:37 +0000 (14:37 +0000)]
Fixed whitespace problems

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273352 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAltered my comment about TCP_NODELAY
Matthew Nicholson [Thu, 1 Jul 2010 14:34:31 +0000 (14:34 +0000)]
Altered my comment about TCP_NODELAY

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273350 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't free written frames in chan_mobile's mbl_write() function.
Matthew Nicholson [Thu, 1 Jul 2010 12:57:18 +0000 (12:57 +0000)]
Don't free written frames in chan_mobile's mbl_write() function.

(closes issue #16430)
Reported by: azbest
Tested by: azbest

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273312 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoSet TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets. This...
Matthew Nicholson [Wed, 30 Jun 2010 18:48:21 +0000 (18:48 +0000)]
Set TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets.  This regression was introduced in r48338.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273270 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoFix rt(c)p set debug ip taking wrong argument
Paul Belanger [Wed, 30 Jun 2010 17:28:04 +0000 (17:28 +0000)]
Fix rt(c)p set debug ip taking wrong argument

Also clean up some coding errors.

(closes issue #17469)
Reported by: wdoekes
Patches:
      astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, pabelanger

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273233 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoRemove unnecessary if test in CV_DSTR()
Richard Mudgett [Wed, 30 Jun 2010 17:17:05 +0000 (17:17 +0000)]
Remove unnecessary if test in CV_DSTR()

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273198 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMisc doxygen cleanup in config.h
Richard Mudgett [Wed, 30 Jun 2010 17:15:46 +0000 (17:15 +0000)]
Misc doxygen cleanup in config.h

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273197 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoPermission checking for the system application is backwards.
Tilghman Lesher [Wed, 30 Jun 2010 01:07:02 +0000 (01:07 +0000)]
Permission checking for the system application is backwards.

(closes issue #17550)
 Reported by: kenner
 Patches:
       manager.c.diff uploaded by kenner (license 1040)
 Tested by: kenner

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273144 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't attempt to proceed if our internal parser indicates an invalid file.
Tilghman Lesher [Wed, 30 Jun 2010 01:01:14 +0000 (01:01 +0000)]
Don't attempt to proceed if our internal parser indicates an invalid file.

(closes issue #17560)
 Reported by: Nick_Lewis

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273142 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 273060 via svnmerge from
Tilghman Lesher [Tue, 29 Jun 2010 23:20:40 +0000 (23:20 +0000)]
Merged revisions 273060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines

  Allow the "useragent" value to be restored into memory from the realtime backend.

  This value is purely informational.  It does not alter configuration at all.

  (closes issue #16029)
   Reported by: Guggemand
   Patches:
         realtime-useragent.patch uploaded by Guggemand (license 897)
   Tested by: Guggemand
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273078 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoRecorded merge of revisions 273057 via svnmerge from
Tilghman Lesher [Tue, 29 Jun 2010 22:59:51 +0000 (22:59 +0000)]
Recorded merge of revisions 273057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines

  _Really_ skip the channel... don't just retry for another 200 cycles.

  (Closes issue SWP-1652, ABE-2240)
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273058 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoExclude libical for insufficient versions.
Tilghman Lesher [Tue, 29 Jun 2010 22:40:00 +0000 (22:40 +0000)]
Exclude libical for insufficient versions.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273055 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoSend DialPlanComplete as a response, not as a separate event.
Tilghman Lesher [Tue, 29 Jun 2010 22:39:22 +0000 (22:39 +0000)]
Send DialPlanComplete as a response, not as a separate event.

Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.

(closes issue #17504)
 Reported by: rrb3942
 Patches:
       showdialplan_patch.diff uploaded by rrb3942 (license 1003)
 Tested by: rrb3942

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@273054 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agosend a 400 Bad Request on malformed sip request
David Vossel [Tue, 29 Jun 2010 20:44:05 +0000 (20:44 +0000)]
send a 400 Bad Request on malformed sip request

RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax.  Currently we
simply ignore a packet with a missing callid, to, from, or
via header.  Instead of ignoring we now send the 400 Bad request.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272981 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272925 via svnmerge from
Tilghman Lesher [Mon, 28 Jun 2010 21:50:57 +0000 (21:50 +0000)]
Merged revisions 272925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines

  Don't change ownership/group/permissions on run directory, if it already exists.

  (closes issue #17076)
   Reported by: stuarth
   Patches:
         20100324__issue17076.diff.txt uploaded by tilghman (license 14)
   Tested by: stuarth
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272926 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272921-272922 via svnmerge from
Tilghman Lesher [Mon, 28 Jun 2010 21:42:52 +0000 (21:42 +0000)]
Merged revisions 272921-272922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines

  Change the way that we read include files, to accommodate for changes in GCC 4.4.

  (closes issue #17472)
   Reported by: seandarcy
   Patches:
         config2.patch uploaded by nivan (license 1066)
   Tested by: nivan
........
  r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines

  Also trim trailing blanks on #includes
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272923 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agorfc compliant sip option parsing + new unit test
David Vossel [Mon, 28 Jun 2010 18:38:47 +0000 (18:38 +0000)]
rfc compliant sip option parsing + new unit test

RFC 3261 section 8.2.2.3 states that if any unsupported options
are found in the Require header field, a "420 (Bad Extension)"
response should be sent with an Unsupported header field containing
only the unsupported options.

This is not currently being done correctly.  Right now, if Asterisk
detects any unsupported sip options in a Require header the entire
list of options are returned in the Unsupported header even if some
of those options are in fact supported.  This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response.  A unit test verifying
this functionality has been created.  Some code refactoring was required.

Review: https://reviewboard.asterisk.org/r/680/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272880 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272804 via svnmerge from
Mark Michelson [Mon, 28 Jun 2010 17:33:12 +0000 (17:33 +0000)]
Merged revisions 272804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines

  Decode URI in contact header of 302 response.

  ABE-2352
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272805 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoUse the underscore package so that underscores do not need to be escaped.
Russell Bryant [Mon, 28 Jun 2010 15:33:32 +0000 (15:33 +0000)]
Use the underscore package so that underscores do not need to be escaped.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272684 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agocode guidelines cleanup for retrans_pkt() function
David Vossel [Mon, 28 Jun 2010 14:55:25 +0000 (14:55 +0000)]
code guidelines cleanup for retrans_pkt() function

I am doing work in this function.  I noticed a large number of
coding guidline fixes that needed to be made.  Rather than have
those changes distract from my functional changes I decided
to separate these into a separate patch.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272652 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272562 via svnmerge from
Tilghman Lesher [Fri, 25 Jun 2010 20:18:47 +0000 (20:18 +0000)]
Merged revisions 272562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines

  Make the structure of the table specified before match the queries and results.

  (closes issue #17557)
   Reported by: cmaj
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272568 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoImplemement support for handling multiple documents when sending.
Matthew Nicholson [Fri, 25 Jun 2010 19:42:54 +0000 (19:42 +0000)]
Implemement support for handling multiple documents when sending.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272558 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agochan_sip: more accurate retransmissions
David Vossel [Fri, 25 Jun 2010 19:39:53 +0000 (19:39 +0000)]
chan_sip: more accurate retransmissions

RFC3261 states that Timer A should start at 500ms (T1) by default.
In chan_sip this value initially started at 1000ms and I changed
it to 500ms recently. After doing that I noticed in my packet
captures that it still occasionally retransmitted starting at
1000ms instead of 500ms like I told it to.  This occurs because
the scheduler runs in the do_monitor thread.  If a new retransmission
is added while the do_monitor thread is sleeping then it may not
detect that retransmission for nearly 1000ms.  To fix this I just
poke the do_monitor thread to wake up when a new packet is sent
reliably requiring retransmits.  The thread then detects the new
scheduler entry and adjusts its sleep time to account for it.

Review: https://reviewboard.asterisk.org/r/747

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272557 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoSymlink sounds files, to save disk space, when multiple tarballs/checkouts are on...
Tilghman Lesher [Fri, 25 Jun 2010 19:17:16 +0000 (19:17 +0000)]
Symlink sounds files, to save disk space, when multiple tarballs/checkouts are on the same system.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272533 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272446 via svnmerge from
Richard Mudgett [Thu, 24 Jun 2010 22:11:26 +0000 (22:11 +0000)]
Merged revisions 272446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines

  ss_thread calls pri_grab without lock during overlap dial

  Recent changes to chan_dahdi with relation to overlap dialing call
  pri_grab without first obtaining a lock.

  (closes issue #17414)
  Reported by: pdf
  Patches:
        bug17414.patch uploaded by jpeeler (license 325)
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272447 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoResolve some errors produced during module unload of chan_iax2.
Russell Bryant [Wed, 23 Jun 2010 23:09:28 +0000 (23:09 +0000)]
Resolve some errors produced during module unload of chan_iax2.

The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down.  This patch addresses the following type of error
that comes from chan_iax2:

[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
                chan_iax2.c line 11371 (iax2_process_thread_cleanup):
                Error destroying mutex &thread->lock: Device or resource busy

For an example in the context of a build, see:

http://bamboo.asterisk.org/browse/AST-TRUNK-739/log

The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock.  While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.

Review: https://reviewboard.asterisk.org/r/736/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272370 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272367 via svnmerge from
Matthew Nicholson [Wed, 23 Jun 2010 22:36:49 +0000 (22:36 +0000)]
Merged revisions 272367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

This version of the patch only adds AgentComplete for attended transfers.  It was already present for blind transfers.

........
  r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines

  Send AgentComplete manager events in the event of blind and attended transfers.

  (closes issue #16819)
  Reported by: elbriga
  Patches:
        app_queue.diff uploaded by elbriga (license 482)
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272368 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoIf there is realtime configuration, it does not get re-read on reload unless the...
Tilghman Lesher [Wed, 23 Jun 2010 21:53:49 +0000 (21:53 +0000)]
If there is realtime configuration, it does not get re-read on reload unless the config file also changes.

(closes issue #16982)
 Reported by: dmitri
 Patches:
       res_musiconhold.patch uploaded by dmitri (license 1001)
 Tested by: atis

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272332 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoEnsure a NULL file while debugging cannot crash AEL.
Tilghman Lesher [Wed, 23 Jun 2010 21:06:40 +0000 (21:06 +0000)]
Ensure a NULL file while debugging cannot crash AEL.

(closes issue #17215)
 Reported by: vazir
 Patches:
       20100518__issue17215.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272260 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoFix previous merge. ast_test_flag != ast_test_flag64
Paul Belanger [Wed, 23 Jun 2010 21:06:15 +0000 (21:06 +0000)]
Fix previous merge. ast_test_flag != ast_test_flag64

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272259 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 272255 via svnmerge from
Paul Belanger [Wed, 23 Jun 2010 21:00:00 +0000 (21:00 +0000)]
Merged revisions 272255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines

  First caller into a dynamic conference now enter pin once.

  If MeetMe is configured to use dynamic conference
  numbers, then the first caller (which creates the
  conference) had to enter the PIN number twice.

  (closes issue #15878)
  Reported by: shawkris
  Patches:
        issue15878.patch uploaded by pabelanger (license 224)
  Tested by: pabelanger
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272257 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoUpdate configure when changing autconf m4 files...
Terry Wilson [Wed, 23 Jun 2010 20:59:17 +0000 (20:59 +0000)]
Update configure when changing autconf m4 files...

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272256 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoHonor the --with-${library}=path for AST_EXT_TOOL_CHECK
Terry Wilson [Wed, 23 Jun 2010 20:53:48 +0000 (20:53 +0000)]
Honor the --with-${library}=path for AST_EXT_TOOL_CHECK

(closes issue #16991)
Reported by: pprindeville
Patches:
      with_netsnmp.patch.txt uploaded by twilson (license 396)
Tested by: twilson

Review: https://reviewboard.asterisk.org/r/739/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272254 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoCorrect manager variable 'EventList' case.
Paul Belanger [Wed, 23 Jun 2010 20:35:45 +0000 (20:35 +0000)]
Correct manager variable 'EventList' case.

(closes issue #17520)
Reported by: kobaz
Patches:
      manager.patch uploaded by kobaz (license 834)
Tested by: lmadsen

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272252 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAdd localization support for Spanish
Paul Belanger [Wed, 23 Jun 2010 20:22:44 +0000 (20:22 +0000)]
Add localization support for Spanish

(closes issue #17548)
Reported by: cjacobsen
Patches:
      say.conf.sample.diff uploaded by cjacobsen (license 1029)

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272243 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAdd new AMI command LocalOptimizeAway.
Tim Ringenbach [Wed, 23 Jun 2010 19:59:43 +0000 (19:59 +0000)]
Add new AMI command LocalOptimizeAway.

This command lets you request a "/n" local channel
optimize itself out of the way anyway.

Review: https://reviewboard.asterisk.org/r/732/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272218 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoD'oh! Defaultenabled FTL.
Tilghman Lesher [Wed, 23 Jun 2010 18:45:18 +0000 (18:45 +0000)]
D'oh!  Defaultenabled FTL.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272150 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoRecorded merge of revisions 272147 via svnmerge from
Tilghman Lesher [Wed, 23 Jun 2010 18:41:18 +0000 (18:41 +0000)]
Recorded merge of revisions 272147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010) | 5 lines

  Backport part of revision 136715 to fix callerid in voicemail text files (IMAP only).

  (closes issue #16945)
   Reported by: mneuhauser
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272148 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't start the sla thread unless we realy need it
Terry Wilson [Wed, 23 Jun 2010 18:39:20 +0000 (18:39 +0000)]
Don't start the sla thread unless we realy need it

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272146 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoLoad all lines from realtime, not just the first one.
Tilghman Lesher [Wed, 23 Jun 2010 18:25:54 +0000 (18:25 +0000)]
Load all lines from realtime, not just the first one.

(closes issue #17144)
 Reported by: nahuelgreco
 Patches:
       20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14)
 Tested by: tilghman

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272145 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMake sure reload updates SLA config
Terry Wilson [Wed, 23 Jun 2010 17:21:40 +0000 (17:21 +0000)]
Make sure reload updates SLA config

Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.

(closes issue #16818)
Reported by: mbonin
Patches:
      sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272109 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAdd extra protection for reinvite glare scenario.
Mark Michelson [Wed, 23 Jun 2010 17:08:34 +0000 (17:08 +0000)]
Add extra protection for reinvite glare scenario.

Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.

The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.

The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272090 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't try to lock/unlock an uninitialized lock on a dahdi_pri.
Russell Bryant [Tue, 22 Jun 2010 23:20:37 +0000 (23:20 +0000)]
Don't try to lock/unlock an uninitialized lock on a dahdi_pri.

This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272052 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agofixes issue with 'dialplan remove extension blah' segfaulting with tab completion
David Vossel [Tue, 22 Jun 2010 22:11:50 +0000 (22:11 +0000)]
fixes issue with 'dialplan remove extension blah' segfaulting with tab completion

(closes issue #17440)
Reported by: kobaz

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@272014 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoignore CANCEL request after having already received final response to INVITE
David Vossel [Tue, 22 Jun 2010 20:37:05 +0000 (20:37 +0000)]
ignore CANCEL request after having already received final response to INVITE

RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response.  This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".

Review: https://reviewboard.asterisk.org/r/697/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271977 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agominor fixes for white/black event filters
David Vossel [Tue, 22 Jun 2010 17:57:28 +0000 (17:57 +0000)]
minor fixes for white/black event filters

This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271905 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 271902 via svnmerge from
Matthew Nicholson [Tue, 22 Jun 2010 17:35:17 +0000 (17:35 +0000)]
Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines

  Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set.  This is necessary to keep the ref count correct.

  (closes issue #16815)
  Reported by: rain
  Patches:
        chan_sip-unref-fix.diff uploaded by rain (license 327) (modified)
  Tested by: rain
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271903 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAdd regular expression filtering for manager events.
Jeff Peeler [Tue, 22 Jun 2010 16:29:18 +0000 (16:29 +0000)]
Add regular expression filtering for manager events.

This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.

(closes issue #14861)
Reported by: fnordian
Patches:
      eventfilter3.patch uploaded by fnordian (license 110),
      modified by me

Review: https://reviewboard.asterisk.org/r/673/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271868 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoResolve some errors that occur on a graceful shutdown.
Russell Bryant [Tue, 22 Jun 2010 16:28:03 +0000 (16:28 +0000)]
Resolve some errors that occur on a graceful shutdown.

Don't Finalize() if Initialize() did not succeed.  This resulted in an error
about trying to Finalize() an invalid handle.

Also trim some trailing whitespace while in the area.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271867 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoChange the method of retrieving the Asterisk version string.
Russell Bryant [Tue, 22 Jun 2010 16:17:14 +0000 (16:17 +0000)]
Change the method of retrieving the Asterisk version string.

Using this method makes it so res_fax doesn't have to be rebuilt on every
svn update.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271833 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agofixes attended transfer behavior when both transferee and transferer hung up
David Vossel [Tue, 22 Jun 2010 15:46:22 +0000 (15:46 +0000)]
fixes attended transfer behavior when both transferee and transferer hung up

If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to.  This mirrors the expected behavior used in 1.4.

(closes issue #17444)
Reported by: corruptor

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271831 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoUpdated the CHANGES file documenting the addition of a configurable port in the dundi...
Matthew Nicholson [Tue, 22 Jun 2010 15:08:39 +0000 (15:08 +0000)]
Updated the CHANGES file documenting the addition of a configurable port in the dundi config file.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271764 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 271761 via svnmerge from
Matthew Nicholson [Tue, 22 Jun 2010 14:54:58 +0000 (14:54 +0000)]
Merged revisions 271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines

  Allow users to specify a port for dundi peers.

  (closes issue #17056)
  Reported by: klaus3000
  Patches:
        dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
  Tested by: klaus3000
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271762 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 271689 via svnmerge from
Matthew Nicholson [Tue, 22 Jun 2010 12:58:28 +0000 (12:58 +0000)]
Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines

  Modify chan_sip's packet generation api to automatically calculate the Content-Length.  This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated.  This change was made to ensure that the Content-Length is always correct.

  (closes issue #17326)
  Reported by: kenner
  Tested by: mnicholson, kenner

  Review: https://reviewboard.asterisk.org/r/693/
........

This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271690 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoConflict kqueue on OS X, since it doesn't work there yet, anyway.
Tilghman Lesher [Mon, 21 Jun 2010 22:41:00 +0000 (22:41 +0000)]
Conflict kqueue on OS X, since it doesn't work there yet, anyway.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271657 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoadd speex 16khz sample frame so codec cost can be calculated
David Vossel [Mon, 21 Jun 2010 21:58:33 +0000 (21:58 +0000)]
add speex 16khz sample frame so codec cost can be calculated

(closes issue #17534)
Reported by: fabled
Patches:
      speex-wb-sample.diff uploaded by fabled (license 448)

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271625 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 271552 via svnmerge from
Jeff Peeler [Mon, 21 Jun 2010 20:46:53 +0000 (20:46 +0000)]
Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines

  Do not use sizeof to calculate size of a heap allocated character array.

  Change left out from 271399.

  (closes issue #16053)
  Reported by: diLLec
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271554 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agofixes crash when From header URI is missing "sip:"
David Vossel [Mon, 21 Jun 2010 20:46:22 +0000 (20:46 +0000)]
fixes crash when From header URI is missing "sip:"

(closes issue #17437)
Reported by: klaus3000
Patches:
      sip_crash uploaded by dvossel (license 671)
Tested by: klaus3000

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271553 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agofixes logic error introduced by slin16 sip support
David Vossel [Mon, 21 Jun 2010 20:33:41 +0000 (20:33 +0000)]
fixes logic error introduced by slin16 sip support

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271551 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAdd new application for declining counting words in multiple languages.
Tilghman Lesher [Mon, 21 Jun 2010 05:10:06 +0000 (05:10 +0000)]
Add new application for declining counting words in multiple languages.

(closes issue #16869)
 Reported by: chappell
 Patches:
       app_say_counted-20100317.c uploaded by chappell (license 8)
 Tested by: chappell

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271520 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 271399 via svnmerge from
Jeff Peeler [Fri, 18 Jun 2010 21:32:09 +0000 (21:32 +0000)]
Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines

  Fix crash when parsing some heavily nested statements in AEL on reload.

  Due to the recursion used when compiling AEL in gen_prios, all the stack space
  was being consumed when parsing some AEL that contained nesting 13 levels deep.
  Changing a few large buffers to be heap allocated fixed the crash, although I
  did not test how many more levels can now be safely used.

  (closes issue #16053)
  Reported by: diLLec
  Tested by: jpeeler
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271483 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agofile.c was truncating audio file formats to the lower 32bits.
David Vossel [Fri, 18 Jun 2010 18:59:05 +0000 (18:59 +0000)]
file.c was truncating audio file formats to the lower 32bits.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271341 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoRecorded merge of revisions 271335 via svnmerge from
Jeff Peeler [Fri, 18 Jun 2010 18:36:55 +0000 (18:36 +0000)]
Recorded merge of revisions 271335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) | 13 lines

  Eliminate deadlock potential in dahdi_fixup().

  (This is a backport of 269307, committed to trunk by rmudgett.)

  Calling dahdi_indicate() when the channel private lock is already
  held can cause a deadlock if the PRI lock is needed because
  dahdi_indicate() will also get the channel private lock.  The pri_grab()
  function assumes that the channel private lock is held once to avoid
  deadlock.

  (closes issue #17261)
  Reported by: aragon
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271336 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agofixes some coding guideline issue
David Vossel [Thu, 17 Jun 2010 21:23:41 +0000 (21:23 +0000)]
fixes some coding guideline issue

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271300 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoretransmit response to BYE requests until timer J expires
David Vossel [Thu, 17 Jun 2010 18:45:32 +0000 (18:45 +0000)]
retransmit response to BYE requests until timer J expires

According to RFC 3261 section 17.2.2, which describes non-INVITE server
transaction, when a dialog enters the Completed state it must destroy
the dialog after Timer J (T1*64) fires.  For a BYE transaction Asterisk
terminates the dialog immediately during sip_hangup() when it should be
waiting T1*64 ms.  This results in some odd behavior.  For instance if
Asterisk receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which Asterisk
responds with a "481 Call leg/transaction does not exist" because the
dialog is already gone.

To resolve this I made a function called sip_scheddestroy_final().  This
differs slightly from sip_schedestroy() in that it enables a flag that
will prevent the destruction from ever being rescheduled or canceled
afterwards.  It also prevents the pvt's needdestroy flag from being set
which triggers the destruction of the dialog within the do_monitor thread().
By using this function we are guaranteed destruction will not occur
until the scheduled time.  This allows Asterisk to respond to any possible
retransmits for a dialog after we process the initial BYE request for T1*64 ms.

Other changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy.  sip_scheddestroy always calls
sip_cancel_destroy before scheduling the new destruction so it is completely
unnecessary.

Review: https://reviewboard.asterisk.org/r/694/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271262 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoadds support for slin16 in sip
David Vossel [Thu, 17 Jun 2010 18:36:06 +0000 (18:36 +0000)]
adds support for slin16 in sip

(closes issue #16153)
Reported by: kfister
Patches:
      16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
      slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271261 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoadds speex 16khz audio support
David Vossel [Thu, 17 Jun 2010 17:23:43 +0000 (17:23 +0000)]
adds speex 16khz audio support

(closes issue #17501)
Reported by: fabled
Patches:
      asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448)
Tested by: malcolmd, fabled, dvossel

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271231 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoChange expected operation from error to debug message
Jeff Peeler [Thu, 17 Jun 2010 15:34:08 +0000 (15:34 +0000)]
Change expected operation from error to debug message

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271192 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoBlocked revisions 271123 via svnmerge
Matthew Nicholson [Thu, 17 Jun 2010 15:11:55 +0000 (15:11 +0000)]
Blocked revisions 271123 via svnmerge

........
  r271123 | mnicholson | 2010-06-17 10:11:27 -0500 (Thu, 17 Jun 2010) | 7 lines

  Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct.

  (closes issue #15827)
  Reported by: DennisD
  Patches:
        (modified) dnsmgr_15827.patch uploaded by chappell (license 8)
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271124 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agooption w[(secs)] incorrectly capitalized in xmldoc
Paul Belanger [Thu, 17 Jun 2010 00:30:51 +0000 (00:30 +0000)]
option w[(secs)] incorrectly capitalized in xmldoc

(closes issue #17516)
Reported by: karlfife

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271089 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoaddition of more parse_uri test cases
David Vossel [Wed, 16 Jun 2010 22:37:45 +0000 (22:37 +0000)]
addition of more parse_uri test cases

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@271056 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 270979 via svnmerge from
Paul Belanger [Wed, 16 Jun 2010 21:17:39 +0000 (21:17 +0000)]
Merged revisions 270979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines

  Fixed typo in macro-page

  Reported to #asterisk-dev by a student of jsmith.
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270987 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoFix the actual place that was pointed out, for previous commit.
Jason Parker [Wed, 16 Jun 2010 21:12:25 +0000 (21:12 +0000)]
Fix the actual place that was pointed out, for previous commit.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270983 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 270980 via svnmerge from
Jason Parker [Wed, 16 Jun 2010 21:10:48 +0000 (21:10 +0000)]
Merged revisions 270980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines

  Need to lock the agent chan before access its internal bits.

  Pointed out by russellb on asterisk-dev mailing list.
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270981 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoSet sin_family to AF_INET when doing lookups, also reset sin_port the first time...
Matthew Nicholson [Wed, 16 Jun 2010 20:34:31 +0000 (20:34 +0000)]
Set sin_family to AF_INET when doing lookups, also reset sin_port the first time the ip address changes.

(closes issue #15827)
Reported by: DennisD
Patches:
      dnsmgr_15827.patch uploaded by chappell (license 8)
Tested by: DennisD, gentlec, damage, wimpy

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270974 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoaddition of G.719 pass-through support
David Vossel [Wed, 16 Jun 2010 19:03:24 +0000 (19:03 +0000)]
addition of G.719 pass-through support

(closes issue #16293)
Reported by: malcolmd
Patches:
      g719.passthrough.patch.7 uploaded by malcolmd (license 924)
      format_g719.c uploaded by malcolmd (license 924)

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMSG_OOB flag on HANGUP packet removed.
Paul Belanger [Wed, 16 Jun 2010 18:43:22 +0000 (18:43 +0000)]
MSG_OOB flag on HANGUP packet removed.

Per Tilghman's request on IRC (#asterisk-bugs).

(closes issue #17506)
Reported by: brycebaril
Tested by: pabelanger, tilghman

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270936 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 270866 via svnmerge from
David Vossel [Wed, 16 Jun 2010 17:36:51 +0000 (17:36 +0000)]
Merged revisions 270866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines

  fixes chan_iax2 race condition

  There is code in chan_iax2.c that attempts to guarantee that only a single
  active thread will handle a call number at a time.  This code works once
  the thread is added to an active_list of threads, but we are not currently
  guaranteed that a newly activated thread will enter the active_list immediately
  because it is left up to the thread to add itself after frames have been
  queued to it.  This means that if two frames come in for the same call number
  at the same time, it is possible for them to grab two separate threads because
  the first thread did not add itself to the active_list fast enough.  This
  causes some pretty complex problems.

  This patch resolves this race condition by immediately adding an activated
  thread to the active_list within the network thread and only depending on
  the thread to remove itself once it is done processing the frames queued to
  it.  By doing this we are guaranteed that if another frame for the same call
  number comes in at the same time, that this thread will immediately be found
  in the active_list of threads.

  Review: https://reviewboard.asterisk.org/r/720/
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270867 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoFix no call waiting caller ID
Jeff Peeler [Wed, 16 Jun 2010 16:45:07 +0000 (16:45 +0000)]
Fix no call waiting caller ID

Clearing the callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270836 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoUpdate formatting for channelvariables.tex
Paul Belanger [Wed, 16 Jun 2010 15:05:11 +0000 (15:05 +0000)]
Update formatting for channelvariables.tex

(closes issue #17511)
Reported by: klaus3000
Patches:
      channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270801 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't blow up if an ast_channel doesn't get allocated.
Russell Bryant [Tue, 15 Jun 2010 22:48:12 +0000 (22:48 +0000)]
Don't blow up if an ast_channel doesn't get allocated.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270726 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't continue sending the file when there has been an error
Terry Wilson [Tue, 15 Jun 2010 21:42:33 +0000 (21:42 +0000)]
Don't continue sending the file when there has been an error

If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.

(closes issue #16682)
Reported by: lmadsen

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270692 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoDon't send files twice and remove extra \r\n from header
Terry Wilson [Tue, 15 Jun 2010 21:10:15 +0000 (21:10 +0000)]
Don't send files twice and remove extra \r\n from header

After the manager http auth changes, we forgot to remove the manual
sending of the file. Also, ast_http_send adds two \r\n to the header that
is passed to it, so a trailing \r\n is removed from the Content-type
header. It might be better to change ast_http_send, but I don't like changing
the behavior of an API function.

(closes issue #17239)
Reported by: cjacobsen
Patches:
      patch2.diff uploaded by cjacobsen (license 1029)
Tested by: lathama, cjacobsen

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270660 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMake contactdeny apply to src ip when nat=yes
Terry Wilson [Tue, 15 Jun 2010 20:18:04 +0000 (20:18 +0000)]
Make contactdeny apply to src ip when nat=yes

chan_sip's "contactdeny" feature screens the "to be registered contact".
In case of nat=yes it should not use the address information from the
Contact header (which is not used at all for routing), but the source
IP address of the request.

Thus, if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the screening.

This commit makes contactdeny apply to the src ip when nat=yes instead.

(closes issue #17276)
Reported by: klaus3000
Patches:
      patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000

Review: [full review board URL with trailing slash]

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270658 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 270583 via svnmerge from
Tilghman Lesher [Tue, 15 Jun 2010 18:26:26 +0000 (18:26 +0000)]
Merged revisions 270583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines

  Variables have always been case-sensitive, so we should not be removing case-insensitive matches.

  Bug reported via the -dev list.  See
  http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270584 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoArgh, mixed declarations and code.
Tilghman Lesher [Tue, 15 Jun 2010 18:16:04 +0000 (18:16 +0000)]
Argh, mixed declarations and code.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270552 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoAdd distributed devicestate via the XMPP protocol.
Tilghman Lesher [Tue, 15 Jun 2010 17:06:23 +0000 (17:06 +0000)]
Add distributed devicestate via the XMPP protocol.

(closes issue #15757)
 Reported by: Marquis
 Patches:
       distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
 Tested by: Marquis, lmadsen, marcelloceschia

Review: https://reviewboard.asterisk.org/r/351/

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270519 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 270442 via svnmerge from
Leif Madsen [Tue, 15 Jun 2010 12:51:37 +0000 (12:51 +0000)]
Merged revisions 270442 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line

  Move information about zonemessages into the [zonemessages] section.
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270443 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoMerged revisions 270331 via svnmerge from
Paul Belanger [Mon, 14 Jun 2010 21:33:55 +0000 (21:33 +0000)]
Merged revisions 270331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) | 14 lines

  Properly play first file in sort list.

  When using sort=alpha we would always skip the first file
  in the list first time through.  We now check for that
  properly.

  (closes issue #17470)
  Reported by: pabelanger
  Patches:
        sort.aplha.patch uploaded by pabelanger (license 224)
  Tested by: lmadsen

  Review: https://reviewboard.asterisk.org/r/703/
........

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270332 f38db490-d61c-443f-a65b-d21fe96a405b

8 years agoExtract sig_ss7_init_linkset() to sig_ss7.
Richard Mudgett [Mon, 14 Jun 2010 20:51:09 +0000 (20:51 +0000)]
Extract sig_ss7_init_linkset() to sig_ss7.

Also found a place where sig_pri_init_pri() was inlined and called it
instead.

git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@270298 f38db490-d61c-443f-a65b-d21fe96a405b