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r336316 | twilson | 2011-09-16 17:11:39 -0500 (Fri, 16 Sep 2011) | 9 lines
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r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 Sep 2011) | 2 lines
Whitespace fix
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r336313 | twilson | 2011-09-16 17:07:00 -0500 (Fri, 16 Sep 2011) | 12 lines
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r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) | 5 lines
Add missing frame types to func_frame_trace
Also casts control frames to the proper enum so that the compile will catch
new additions.
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r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines
Merged revisions 336294 via svnmerge from
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r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a new type of control frame
so that our RTP bridge loop can properly detect when these situations occur and check to see
if peers need to be updated in order to send their media to the proper location.
(Closes issue ASTERISK-18340)
Reported by: Thomas Arimont
(Closes issue ASTERISK-17725)
Reported by: kwk
Tested by: twilson, jrose
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r336235 | seanbright | 2011-09-16 15:10:39 -0400 (Fri, 16 Sep 2011) | 9 lines
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r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep 2011) | 2 lines
Make a note that inotify won't work with an NFS mounted spooler directory.
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r336167 | irroot | 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) | 22 lines
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r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | 16 lines
The round robin routing routine in chan_misdn.c is broken.
it rotates between ports but never checks the channels in the ports.
i have extensivly tested it and verified it works on 1 upto 4 ports.
before the patch only 1 out of each port was used now all are used as
expected.
(closes issue ASTERISK-18413)
Reported by: irroot
Tested by: irroot
Reviewed by: irroot
Review: https://reviewboard.asterisk.org/r/1410/
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r336094 | irroot | 2011-09-15 17:54:46 +0200 (Thu, 15 Sep 2011) | 26 lines
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r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | 20 lines
Locking order in app_queue.c causes deadlocks.
a channel lock must never be held with the queues container lock held.
the deadlock occured on masquerade.
the queues container lock is a relic of the past the old queue module lock.
with ao2 there is no need to hold this lock when dealing with members this
patch removes unneeded locks.
(closes issue ASTERISK-18101)
(closes issue ASTERISK-18487)
Reported by: Paul Rolfe, Jason Legault
Tested by: irroot, Jason Legault, Paul Rolfe
Reviewed by: Matthew Nicholson
Review: https://reviewboard.asterisk.org/r/1402/
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r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011) | 2 lines
Removes some no-op code found in format_cap.c.
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r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 lines
Meetme: Introducing a new option "k" to kill a conference if there's only a single member left.
When using Meetme as a modular call bridge from third party applications, it's handy to make
it behave like a normal call bridge. When the second to last person exists, the last person
will be kicked out of the conference when this option is enabled.
(closes issue ASTERISK-18234)
Review: https://reviewboard.asterisk.org/r/1376/
Patch by oej, sponsored by ClearIT, Solna, Sweden
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r335991 | irroot | 2011-09-15 10:29:12 +0200 (Thu, 15 Sep 2011) | 17 lines
Merged revisions 335978 via svnmerge from
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r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | 11 lines
lock the channel before calling ast_bridged_channel() to prevent a seg fault.
AMI agents list called on shutdown causes a segfault, introducing proper locking
will prevent this.
(closes issue ASTERISK-18092)
Reported by: agustina
Patches: chan_agent.patch (License #5041) patch uploaded by irroot
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r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines
Merged revisions 335911 via svnmerge from
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r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines
Remove unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated an error
about not having PRI_L2_PERSISTENCE if you did not have the absolute
latest libpri SVN checkout installed.
The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to
be for libraries that are dependent upon other libraries and not
necessarily for optional/added features within a library.
(closes issue ASTERISK-18535)
Reported by: Michael Keuter
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r335852 | rmudgett | 2011-09-14 11:00:37 -0500 (Wed, 14 Sep 2011) | 18 lines
Merged revisions 335851 via svnmerge from
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r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) | 11 lines
Fixed cut-n-paste regression using the wrong variable.
Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
sections for channel configuration.
(closes issue ASTERISK-18496)
Reported by: Sean Darcy
Patches:
jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Sean Darcy, rmudgett
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r335791 | mnicholson | 2011-09-14 08:28:50 -0500 (Wed, 14 Sep 2011) | 11 lines
Merged revisions 335790 via svnmerge from
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r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep 2011) | 4 lines
The tech and data members of fast_originate_helper are not string fields.
ASTERISK-17709
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r335721 | rmudgett | 2011-09-13 17:10:44 -0500 (Tue, 13 Sep 2011) | 9 lines
Merged revisions 335720 via svnmerge from
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r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011) | 1 line
Remove obsolete todo comment about PICKUPRESULT.
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Do parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the build-time
default language (normally "en") is always the default one.
Review: https://reviewboard.asterisk.org/r/1342/
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>
Original-Commit: http://svn.digium.com/svn/asterisk/branches/1.8@335716
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r335656 | tilghman | 2011-09-13 13:55:33 -0500 (Tue, 13 Sep 2011) | 11 lines
Merged revisions 335655 via svnmerge from
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r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) | 4 lines
Move mandatory checks closer to the beginning of the file.
If these are going to fail, they should fail as quickly as possible.
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r335653 | mnicholson | 2011-09-13 13:47:57 -0500 (Tue, 13 Sep 2011) | 12 lines
Merged revisions 335618 via svnmerge from
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r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep 2011) | 5 lines
Don't limit the size of appdata for manager originate actions.
ASTERISK-17709
Patch by: tilghman (with modifications)
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Review: https://reviewboard.asterisk.org/r/1434/
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Review: https://reviewboard.asterisk.org/r/1427/
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Review: https://reviewboard.asterisk.org/r/1432/
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r335510 | russell | 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) | 22 lines
Merged revisions 335497 via svnmerge from
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r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines
Fix a crash in res_ais.
This patch resolves a crash observed in a load testing environment that
involved the use of the res_ais module. I observed some crashes where
the event delivery callback would get called, but the length parameter
incidcating how much data there was to read was 0. The code assumed
(with good reason I would think) that if this callback got called, there
was an event available to read. However, if the rare case that there's
nothing there, catch it and return instead of blowing up.
More specifically, the change always ensure that the size of the received
event in the cluster is always big enough to be a real ast_event.
Review: https://reviewboard.asterisk.org/r/1423/
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r335434 | mnicholson | 2011-09-12 10:55:48 -0500 (Mon, 12 Sep 2011) | 13 lines
Merged revisions 335433 via svnmerge from
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r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep 2011) | 6 lines
Properly set caller_warning and callee_warning before we try to use them.
ASTERISK-18199
Patch by: elguero
Testing by: rtang
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r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines
Merged revisions 335341 via svnmerge from
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r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines
Ensure frames are not written to dialed channel if ringback is requested
When a single channel was dialed and there was media to be forwarded to the
calling channel, the media was written without regard for ringback causing
silence to be heard in some circumstances. This regression was introduced
when the meaning of "single" changed to mean only the number of channels
dialed.
(closes issue ASTERISK-18083)
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devices
Review: https://reviewboard.asterisk.org/r/1429/
(closes issue ASTERISK-18497)
Thanks to russellb for peer review.
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r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines
Merged revisions 335319 via svnmerge from
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r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines
Lock the peer->mvipvt to avoid crashes with SIP history enabled
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
which cause issues with SIP history additions in combination with the max limit for
number of history entries.
Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288)
Thanks to irrot for peer review. Work with this bug funded by IPvision AS
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r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines
Merged revisions 335320 via svnmerge from
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r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines
Prevent IAX2 from getting IPv6 addresses via DNS
IAX2 does not support IPv6 and getting such addresses from DNS can cause error
messages on the remote end involving bad IPv4 address casts in the presence of
IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6
addresses via DNS queries.
(closes issue ASTERISK-18090)
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r335260 | schmidts | 2011-09-12 11:11:45 +0000 (Mon, 12 Sep 2011) | 12 lines
Merged revisions 335259 via svnmerge from
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r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines
build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value.
adding an ao2_unlink from the peers_by_ip container fix it.
Review: https://reviewboard.asterisk.org/r/1428/
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Review: https://reviewboard.asterisk.org/r/1426/
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) | 9 lines
Move code for VALID_EXTEN from app_readexten to func_dialplan
Mark VALID_EXTEN deprecated.
Review: https://reviewboard.asterisk.org/r/1396/
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r334954 | rmudgett | 2011-09-08 17:28:56 -0500 (Thu, 08 Sep 2011) | 17 lines
Merged revisions 334953 via svnmerge from
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r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines
Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used.
Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to
unregister its logger level.
* Make ast_logger_unregister_level() use ast_free() instead of free().
When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call
to free(). Therefore, if you allocated memory with a form of ast_malloc
you must free it with ast_free.
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r334844 | pabelanger | 2011-09-07 15:37:24 -0400 (Wed, 07 Sep 2011) | 11 lines
Merged revisions 334843 via svnmerge from
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r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) | 4 lines
Cleanup chan_iax2.c log messages
Review: https://code.asterisk.org/code/cru/CR-AST-11
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r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines
Merged revisions 334840 via svnmerge from
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r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines
Fix AMI action Park crash.
* Made AMI action Park not say anything to the parker channel (AMI header
Channel2) since the AMI action is a third party parking the call. (This
is a change in behavior that cannot be preserved without a lot of effort.)
* Made not play pbx-parkingfailed if the Park 's' option is used.
JIRA AST-660
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r334747 | schmidts | 2011-09-07 15:10:37 +0000 (Wed, 07 Sep 2011) | 9 lines
Merged revisions 334682 via svnmerge from
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r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) | 3 lines
Adding the Feature to sent a Reason Header in a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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the flag AST_FLAG_ANSWERED_ELSEWHERE before doing a masquerade in the pickup function.
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r334621 | alecdavis | 2011-09-07 20:14:50 +1200 (Wed, 07 Sep 2011) | 9 lines
Merged revisions 334620 via svnmerge from
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r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 Sep 2011) | 2 lines
peroid typo
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Allow tracking of previous versions through log file records to be tracked.
Each time log file is created or opened, log Asterisk Version, Buildinfo. etc.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1409/
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r334617 | alecdavis | 2011-09-07 19:45:00 +1200 (Wed, 07 Sep 2011) | 17 lines
Merged revisions 334616 via svnmerge from
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r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep 2011) | 10 lines
Prevent segfault if call arrives before Asterisk is fully booted.
Prevent ast_pbx_start and ast_run_start from starting a new thread unless asterisk
is fully booted.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1407/
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keyword.
This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
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r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines
Merged revisions 334453 via svnmerge from
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r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines
Make SQL query in app_voicemail.c portable LIMIT is not portable.
Regression from r312212
(closes issue ASTERISK-18255)
Reported by: Leif Madsen
Tested by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1415/
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r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep 2011) | 6 lines
authdebug is now disabled by default
To enable this functionaility again set authdebug = yes in iax.conf
Review: https://reviewboard.asterisk.org/r/1414/
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r334455 | irroot | 2011-09-06 15:58:56 +0200 (Tue, 06 Sep 2011) | 18 lines
Merged revisions 334453 via svnmerge from
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r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | 13 lines
Make SQL query in app_voicemail.c portable LIMIT is not portable.
Regression from r312212
(closes issue ASTERISK-18255)
Reported by: Leif Madsen
Tested by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1415/
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