Commit Graph

24729 Commits

Author SHA1 Message Date
Rusty Newton 873969d6c5 Broke the build! Forgot para tags within my description.
https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304
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Merged revisions 399064 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13 14:27:54 +00:00
David M. Lee 2a57f6ccf7 res_pjsip: Forward PJSIP logging to Asterisk logging
This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log
messages to Asterisk's logger. This is done in a new module:
res_pjsip_log_forwarder.so.

This patch sets defaultenabled on the existing res_pjsip_logger.so to
no, since logging every SIP packet seems a bit odd to do by default, and
is (hopefully) less necessary with regular PJSIP logging.

It also removes res_rtp_asterisk's disabling of PJSIP logging.

(closes issue ASTERISK-22360)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2830/
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Merged revisions 399049 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13 14:22:07 +00:00
David M. Lee f56796a539 ARI: Fix WebSocket response when subprotocol isn't specified
When I moved the ARI WebSocket from /ws to /ari/events, I added code to
allow a WebSocket to connect without specifying the subprotocol if
there's only one subprotocol handler registered for the WebSocket.

Naively, I coded it to always respond with the subprotocol in use.
Unfortunately, according to RFC 6455, if the server's response includes
a subprotocol header field that "indicates the use of a subprotocol that
was not present in the client's handshake [...], the client MUST _Fail
the WebSocket Connection_.", emphasis theirs.

This patch correctly omits the Sec-WebSocket-Protocol if one is not
specified by the client.

(closes issue ASTERISK-22441)
Review: https://reviewboard.asterisk.org/r/2828/
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Merged revisions 399039 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13 14:19:19 +00:00
Kinsey Moore 0ffcd11380 Fix several crashes in MeetMeAdmin
This change ensures that MeetMeAdmin commands requiring a user actually
get a user and fixes another issue where an extra dereference could
occur for a last-entered user being ejected if a user identifier was
also provided.

(closes issue ASTERISK-21907)
Reported by: Alex Epshteyn
Review: https://reviewboard.asterisk.org/r/2844/
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Merged revisions 399033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 399034 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 399035 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13 14:17:15 +00:00
Rusty Newton fc09e5eb66 'identify' configObject doesn't have a synopsis
Add a straightforward synopsis and description to the identify config object
in XML documentation.

(issue ASTERISK-22311)
(closes issue ASTERISK-22311)
Reported By: Rusty Newton
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Merged revisions 399031 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-13 13:28:26 +00:00
Richard Mudgett ef53242700 CLI bridge: Fix "bridge destroy <id>" and "bridge kick <id> <chan>" tab completion.
These two commands must deal with the live bridges container for tab
completion and not the stasis cache.
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Merged revisions 399021 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 23:42:23 +00:00
Richard Mudgett 94754227a6 astobj2: Register the bridges container for debug inspection.
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Merged revisions 399019 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 23:36:33 +00:00
Rusty Newton 1b777d8946 Documentation fix and improvements to XML configuration help res_pjsip_acl
*  One bug fix. Made the synopsis for "type" to accurate.
 *  changing the usage of "IP-domains" to "IP addresses"
 *  clarifying the usage for the options, by adding a relevant description for
    each
 *  modified other areas of the XML help for clarity, such as the module
    description and a few synopsis changes here and there. See the patch.

(issue ASTERISK-22458)
(closes issue ASTERISK-22458)
Reported By: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2823/
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Merged revisions 399017 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 23:23:12 +00:00
Jonathan Rose 039030f245 chan_sip: Revert r398835 due to failing tests involving originate
(issue ASTERISK-22424)
Reported by: Jonathan Rose
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Merged revisions 398977 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 20:27:56 +00:00
Richard Mudgett 5be186d7c5 core_local: Fix memory corruption race condition.
The masquerade super test is failing on v12 with high fence violations and
crashing.  The fence violations are showing that party id allocated memory
strings are somehow getting corrupted in the
bridge_reconfigured_connected_line_update() function.  The invalid string
values happen to be the freed memory fill pattern.

After much puzzling, I deduced that the
bridge_reconfigured_connected_line_update() is copying a string out of the
source channel's caller party id struct just as another thread is updating
it with a new value.  The copying thread is using the old string pointer
being freed by the updating thread.  A search of the code found the
unreal_colp_redirect_indicate() routine updating the caller party id's
without holding the channel lock.

A latent bug in v1.8 and v11 hatched in v12 because of the bridging and
connected line changes.  :)

(issue ASTERISK-22221)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2839/
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Merged revisions 398938 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 16:44:34 +00:00
David M. Lee 6ad74509f3 Fix symbol collision with pjsua.
We shouldn't be exporting any symbols that start with pjsip_.
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Merged revisions 398927 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398928 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 15:23:54 +00:00
Rusty Newton 4e3f78ad7b 'queue add member' help text correction
You are adding dial strings to the queue, not channels. An aribitrary string
could be used, but you are typically referencing a channel. Correcting the
command help text.

(issue ASTERISK-22263)
(closes issue ASTERISK-22263)
Reported By: Rusty Newton
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Merged revisions 398884 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398886 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-12 00:04:57 +00:00
Rusty Newton 7c346a31ef Documentation fix - waitfordialtone is not boolean, it's time in milliseconds
Changing text in chan_dahdi.conf sample to be accurate.

(issue ASTERISK-22308)
(closes issue ASTERISK-22308)
Reported By: Malcolm Davenport
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Merged revisions 398880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11 23:52:49 +00:00
Jonathan Rose 187802eeb2 chan_sip: Reject calls without prior SDP on 200 OK
If we receive a 200 OK without SDP, we will now check to see if
the remote address has been established for that channel's RTP
session and if the to tag for that channel has changed from
the most recent to tag in a response less than 200.
If either a change has been made since the last to-tag was
received or the remote address is unset, then we will drop
the call.

(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header
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Merged revisions 398835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11 20:03:19 +00:00
Russell Bryant 9b3e0b095e Fix typo in confbridge.conf.sample
The denoise filter requires func_speex, not codec_speex.  Fix this in the
description of the denoise=yes option in confbridge.conf.
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Merged revisions 398820 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398821 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11 18:03:30 +00:00
Kevin Harwell 4d35941891 pjsip: reinvite for connected line updates occurs when it should not
Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:

1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.

Also added an SDP when an update is sent out.

(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
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Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-11 14:23:28 +00:00
Richard Mudgett 83bf017db9 Fix incorrect usages of ast_realloc().
There are several locations in the code base where this is done:
buf = ast_realloc(buf, new_size);

This is going to leak the original buf contents if the realloc fails.

Review: https://reviewboard.asterisk.org/r/2832/
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Merged revisions 398757 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10 18:05:47 +00:00
David M. Lee 87cf916cdb Fixed utils directory breakage from r398748, this time with extra hate.
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Merged revisions 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10 17:50:13 +00:00
David M. Lee 6a1f3d626b Fixed utils directory breakage from r398648
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Merged revisions 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-10 17:26:19 +00:00
Richard Mudgett 35b5549df8 MALLOC_DEBUG: Change fence magic number to be completely different from the freed magic number.
Race conditions between freeing a nul terminated string and
ast_strdup()'ing it are more likely to be detected if the fence and freed
magic numbers are completely different.
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Merged revisions 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09 23:29:44 +00:00
Mark Michelson 9fb34542d1 Add extra debugging to res_pjsip_endpoint_identifier_ip
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Merged revisions 398694 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09 22:00:44 +00:00
David M. Lee c2e6e1ef49 Fix DEBUG_THREADS when lock is acquired in __constructor__
This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.

With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).

This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).

(closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
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Merged revisions 398648 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09 20:13:40 +00:00
David M. Lee 0bcc676d09 Multiple revisions 398638-398639
........
  r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line
  
  Added note about expected behavior of originate
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  r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line
  
  Added note about expected behavior of originate (the rest of the commit)
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Merged revisions 398638-398639 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09 19:09:21 +00:00
David M. Lee 4015f05dce Blocked revisions 398559,398578
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Multiple revisions 398559,398578

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  r398559 | kmoore | 2013-09-06 14:32:03 -0500 (Fri, 06 Sep 2013) | 20 lines
  
  Blocked revisions 398558
  
  ........
  Fix Jabber/XMPP distributed MWI
  
  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.
  
  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........
  
  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r398578 | kmoore | 2013-09-06 16:03:45 -0500 (Fri, 06 Sep 2013) | 1 line
  
  Unblock r398558


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-09 19:08:15 +00:00
Matthew Jordan 361d308b21 Update CDR Unit tests to reflect container changes in r398579
When a channel joins a multi-party bridge, the ordering of the CDRs that is
created is determined by the ordering of the channels who happen to be in that
bridge. When r398579 changed the number of buckets in the container to
something sensible, it changed the ordering that the CDRs was created in,
causing one of the multiparty tests to fail. This fixes the test with the
now expected ordering.
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Merged revisions 398628 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-08 23:30:39 +00:00
Kinsey Moore ad89dffb8a Prevent XMPP timeout on blank responses
Sometimes the Google Voice servers have a bad habit of sending out 1
byte replies to the xmpp resource. When a blank 1 byte reply is
received from the socket the buffer attempts to wait (endlessly) for
the rest of the reply from google which effectively blocks the socket
and google voice calls will no longer come into the server.

This patch allows the xmpp module to correctly detect empty packets and
send out ping replies to google. It also sets a socket timeout on the
default socket which prevents the xmpp socket from closing and
preventing future google voice calls from coming into the server.

Furthermore instead of sending an empty reply back to google we send a
proper xmpp ping reply back. This also adds several more
socket messages.

(closes issue ASTERISK-22347)
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/2771
Patches:
    xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-07 01:03:07 +00:00
Kinsey Moore 0bed76989a Multiple revisions 398558,398577
........
  r398558 | kmoore | 2013-09-06 14:28:16 -0500 (Fri, 06 Sep 2013) | 17 lines
  
  Fix Jabber/XMPP distributed MWI
  
  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.
  
  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........
  
  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | 10 lines
  
  Commit the remainder of r398523
  
  This is a missing part of the commit in revision 398523 that corrects
  the name of a variable.
  
  (issue ASTERISK-22435)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 21:23:02 +00:00
Richard Mudgett 5396198f16 cdr: Change the number of container buckets to be similar to the channels container.
* Fix the temporary cdr candidate containers to use a prime number of
buckets.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 21:17:45 +00:00
Richard Mudgett a4c18f4e10 core_local: Fix LocalOptimizationBegin AMI event missing Source channel snapshot.
* Fix the LocalOptimizationBegin AMI event by eliminating an artificial
buffer size limitation that is too small anyway.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 20:21:21 +00:00
Richard Mudgett 51bd4fe8fe cdr: Fix some ref leaks.
* Added missing unregister of the cdr container in cdr_engine_shutdown().

* Fixed ref leak in off nominal path of cdr_object_alloc().

* Removed some unnecessary NULL checks in cdr_object_dtor().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 20:03:01 +00:00
Richard Mudgett f5ae5e27c8 astobj2: Add warn unused attribute to some functions.
* Fixed resulting warnings with improper use of ao2_global_obj_replace().

* Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the
equivalent and more appropriate ao2_global_obj_release() call.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 19:26:48 +00:00
Kinsey Moore 53dbe10f5c Fix build warnings
When AST_DEVMODE is not defined, ast_asserts are not compiled into the
binary. In some cases, this means variables are not referenced or are
set but unused which causes warnings to show up.

(closes issue ASTERISK-22446)
Reported by: Jason Parker (qwell)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-06 18:53:32 +00:00
Kinsey Moore 5a3c17f91f Fix chan_h323 compilation
This fixes the things in chan_h323 that were missed or ignored in the
great channel opaquification and gets chan_h323 back into a compiling
state.

(closes issue ASTERISK-22365)
Reported by: Dmitry Melekhov
Patches:
    chan_h323.patch uploaded by Dmitry Melekhov
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2013-09-06 16:01:05 +00:00
Richard Mudgett ccfad032e4 astobj2: Only define ao2_bt() once.
* Make ao2_bt() not use single char variable names.

* Fix ao2_bt() formatting.
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2013-09-05 21:48:02 +00:00
Richard Mudgett 778d174126 chan_iax2: Reduce indentation in __attempt_transmit().
* Reduce indentation in __attempt_transmit().

* Don't update the static last error time variable every time in
__schedule_action() and socket_read().
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2013-09-05 19:18:10 +00:00
Richard Mudgett 5954da694e chan_iax2: Fix stray reference to worker thread idle_list.
* Fix stray reference to idle_list in cleanup_thread_list().  This may be
the reason for the note in iax2_process_thread() about threads not being
removed from the task lists.

* Move cleanup_thread_list(&idle_list) to after the other lists are
cleaned up.
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2013-09-05 17:31:29 +00:00
Richard Mudgett bdb0121769 chan_iax2: Fix bridgecallno deadlock avoidance.
* Fix bridgecallno deadlock avoidance.  When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.

* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.

* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list.  defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
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2013-09-05 17:17:53 +00:00
Mark Michelson 23ed2ca57f Clarify server_uri and client_uri registration settings.
Used some of Rusty's suggested language plus also included
more SIPesque descriptions of where the URIs are actually
used in an outgoing REGISTER.

(closes issue ASTERISK-22390)
reported by Rusty Newton
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2013-09-05 14:10:45 +00:00
Richard Mudgett 586a825325 chan_iax2: Add missing control frame names to debug frame decode output.
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2013-09-04 23:07:41 +00:00
Mark Michelson 517389b1e7 Give more detail regarding failures to create request with auth credentials.
(issue ASTERISK-22386)
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2013-09-04 22:49:25 +00:00
Jonathan Rose 3e6923c3f6 unit tests: test_voicemail_api leaks stringfields from snapshots
(closes issue ASTERISK-22414)
Reported by: Corey Farrell
Patches:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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2013-09-04 21:37:01 +00:00
Jonathan Rose 64924d72bf app_voicemail: Fix leaking config objects when msg_id doesn't match
(issues ASTERISK-22414)
Reported by: Corey Farrell
Patch:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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2013-09-04 21:20:02 +00:00
Richard Mudgett 2ce0c9f4a0 chan_misdn: Fix misdn debug output printed with arbitrary verbose levels.
Fix the misdn debug output to remote consoles.  chan_misdn uses
ast_console_puts() which doesn't know about verbose levels.  Better to use
ast_verbose() instead.  Without this patch the misdn debug messages are
appended to the verbose level which ever was set by the message sent to
the console before, i.e.  any undefined level.

(closes issue AST-1218)
Reported by: Guenther Kelleter
Patches:
      misdnlog.patch (license #6372) patch uploaded by Guenther Kelleter
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2013-09-04 16:03:14 +00:00
Kevin Harwell 1be7514a59 Debug messages for pjsip outbound registration
Added debug messages indicating that an outbound registration attempt was made
and it was successful in pjsip.

(closes issue ASTERISK-22388)
Reported by: Rusty Newton
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2013-09-04 14:32:25 +00:00
Alexandr Anikin b8f6f8ebe1 Fix remote tcs sequence handling on empty tcs received
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2013-09-03 20:28:01 +00:00
Kinsey Moore e5542ab1e7 Prevent a crash in res_pjsip_dtmf_info.c
This change makes sure that a content type header exists before
checking the contents of the header against known SIP INFO DTMF content
types.
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2013-09-03 18:09:02 +00:00
David M. Lee 32d325439f Fixed 'make clean' for wiki docs
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2013-09-03 17:19:30 +00:00
Walter Doekes 8b5d178c36 Be a little more verbose when loading cel_custom.conf.
Review: https://reviewboard.asterisk.org/r/2805/
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2013-09-03 14:29:52 +00:00
David M. Lee 2d1d5a98d5 Fix graceful shutdown crash.
The cleanup code for optional_api needs to happen after all of the optional
API users and providers have unused/unprovided. Unfortunately, regsitering the
atexit() handler at the beginning of main() isn't soon enough, since module
destructors run after that.
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2013-08-30 20:58:59 +00:00
Rusty Newton be219c9ec9 New pjsip.conf.sample
(issue ASTERISK-22145)
(closes issue ASTERISK-22145)
Reported By: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2811/
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2013-08-30 20:37:54 +00:00