Russell Bryant [Mon, 24 Jan 2011 20:41:17 +0000 (20:41 +0000)]
Add gsm-devel as a package to install on redhat based systems.
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Matthew Nicholson [Mon, 24 Jan 2011 18:59:22 +0000 (18:59 +0000)]
According to section 19.1.2 of RFC 3261:
For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
The unit tests for these functions have also been updated.
ABE-2705
Review: https://reviewboard.asterisk.org/r/1081/
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Jason Parker [Mon, 24 Jan 2011 17:21:12 +0000 (17:21 +0000)]
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r303467 | qwell | 2011-01-24 11:20:03 -0600 (Mon, 24 Jan 2011) | 22 lines
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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Russell Bryant [Sat, 22 Jan 2011 04:13:15 +0000 (04:13 +0000)]
Revert default compiler change.
If someone wishes to do so, it is trivial to set your own default when running
the configure script.
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Jason Parker [Fri, 21 Jan 2011 23:11:34 +0000 (23:11 +0000)]
Temporarily revert r303288
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Jason Parker [Fri, 21 Jan 2011 21:51:06 +0000 (21:51 +0000)]
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r303286 | qwell | 2011-01-21 15:50:11 -0600 (Fri, 21 Jan 2011) | 22 lines
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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Tilghman Lesher [Fri, 21 Jan 2011 09:09:54 +0000 (09:09 +0000)]
Really use llvm-gcc, when available.
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Tilghman Lesher [Fri, 21 Jan 2011 08:13:18 +0000 (08:13 +0000)]
Add DB_KEYS.
Discussion on #asterisk on 2011-01-19:
(02:07:03 PM) boch: i wonder how to cycle all entries in a tree
(02:07:11 PM) leifmadsen: use While()
(02:07:17 PM) leifmadsen: you need to know the tree structure already though
(02:07:36 PM) boch: what you mean?
(02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan
(02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything
(02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script
(02:10:13 PM) boch: for example i need to know all entries in the tree
(02:10:15 PM) boch: got it
(02:10:20 PM) leifmadsen: exactly
(02:10:22 PM) leifmadsen: that's the problem
(02:10:22 PM) boch: thank you
(02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over.
(02:15:35 PM) leifmadsen: database shows everything
(02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>'
(02:16:41 PM) leifmadsen: guess no one has found that important enough to program :)
(02:16:52 PM) leifmadsen: at that point you should probably just use a relational database...
(02:17:10 PM) mateu: i dunno
(02:17:16 PM) mateu: seems pretty basic to me.
(02:17:16 PM) leifmadsen: me either
(02:17:19 PM) leifmadsen: sure does
(02:17:24 PM) leifmadsen: no one has programmed it though
(02:17:28 PM) ***leifmadsen shrugs
(02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen
(02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ?
(02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT()
(02:30:31 PM) leifmadsen: although HASHKEYS() might work
(02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS()
(02:31:01 PM) leifmadsen: DBKEYS() I guess?
(02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family?
(02:34:02 PM) leifmadsen: ya
(02:34:16 PM) leifmadsen: how would you iterate through layers of them?
(02:34:30 PM) leifmadsen: i.e. family/key/key/key ?
(02:34:43 PM) Corydon76-home: Essentially, yes
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Richard Mudgett [Thu, 20 Jan 2011 20:35:50 +0000 (20:35 +0000)]
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r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines
Merged revision 303098 from
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r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines
CC_INTERFACES does not get built correctly with local channels.
If local channels are used with CCSS, CC_INTERFACES gets garbage prepended
to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended
to it.
* Initialize the buffer to eliminate the prepended garbage.
* Filter out the empty interface strings to eliminate the latter.
* Added a diagnostic message if the CC_INTERFACES is ever empty.
JIRA ABE-2740
JIRA SWP-2848
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Shaun Ruffell [Thu, 20 Jan 2011 19:58:54 +0000 (19:58 +0000)]
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r303107 | sruffell | 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) | 23 lines
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r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) | 15 lines
main/features: Use POLLPRI when waiting for events on parked channels.
This change resolves a regression in the 1.6.2 when converting from
select to poll. The DAHDI timers use POLLPRI to indicate that the timer
fired, but features was not waiting for that flag. The result was no
audio for MOH when a call was parked and res_timing_dahdi was in use.
This patch is slightly modified from the one on the mantis issue. It does
not set an exception on the channel if the POLLPRI flag is set.
(closes issue #18262)
Reported by: francesco_r
Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
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Jeff Peeler [Thu, 20 Jan 2011 17:14:01 +0000 (17:14 +0000)]
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r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines
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r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines
Add new queue strategy to preserve behavior for when queue members moved to ao2.
Add queue strategy called "rrordered" to mimic old behavior from when queue
members were stored in a linked list.
ABE-2707
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Russell Bryant [Thu, 20 Jan 2011 16:12:35 +0000 (16:12 +0000)]
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r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines
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r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines
Resolve a compiler warning.
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Leif Madsen [Thu, 20 Jan 2011 15:46:24 +0000 (15:46 +0000)]
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r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines
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r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds
> as the value is in milliseconds.
>
> (closes issue #18264)
> Reported by: jacco
> Patches:
> app_dial_patch.txt uploaded by lmadsen (license 10)
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Tilghman Lesher [Thu, 20 Jan 2011 09:07:27 +0000 (09:07 +0000)]
On systems which have LLVM, use that compiler. Should result in a massive speed increase.
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Russell Bryant [Wed, 19 Jan 2011 23:57:27 +0000 (23:57 +0000)]
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r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2 lines
Only check container count if it exists.
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Sean Bright [Wed, 19 Jan 2011 23:53:44 +0000 (23:53 +0000)]
Clarify a source comment about configuration template categories.
(closes issue #18578)
Reported by: astmiv
Patches:
asterisk.main.config.2.patch uploaded by astmiv (license 1189)
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Sean Bright [Wed, 19 Jan 2011 23:49:54 +0000 (23:49 +0000)]
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r302834 | seanbright | 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) | 14 lines
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r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) | 7 lines
Support greetingsfolder as documented in voicemail.conf.sample.
(closes issue #17870)
Reported by: edhorton
Patches:
__20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10)
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Paul Belanger [Wed, 19 Jan 2011 23:33:42 +0000 (23:33 +0000)]
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r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan 2011) | 2 lines
Add binutils-dev for BETTER_BACKTRACES
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Russell Bryant [Wed, 19 Jan 2011 23:07:22 +0000 (23:07 +0000)]
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r302789 | russell | 2011-01-19 17:06:46 -0600 (Wed, 19 Jan 2011) | 11 lines
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r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) | 4 lines
Turn a noisy verbose message into a debug message.
This can drown your console if you're using the AMI over HTTP.
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Russell Bryant [Wed, 19 Jan 2011 22:36:30 +0000 (22:36 +0000)]
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r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines
Resolve a memory leak with the manager interface is disabled.
The intent of this check as it stands in previous versions of Asterisk was to
check if there are any active sessions. If there were no sessions, then the
function would return immediately and not bother with queueing up the manager
event to be processed. Since the conversion of storing sessions in an astobj2
container, this check will always pass. I changed it to go back to checking
what was intended.
The side effect of this was that if the AMI is disabled, the manager event
queue is populated anyway, but the code that runs to clear out the queue
never runs. A producer with no consumer is a bad thing.
Reported internally by kmorgan.
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Richard Mudgett [Wed, 19 Jan 2011 21:35:28 +0000 (21:35 +0000)]
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r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines
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r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines
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r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines
DTMF transfer plays the wrong sounds for wrong number or other call failure.
* Set the default for features.conf.sample xferfailsound option to "beeperr"
as documented instead of "pbx-invalid" and corrected the use of it in DTMF
blind transfer (#1).
* Improved DTMF blind transfer handling of wrong numbers.
Most of the concerns in this issue were taken care of by the patch for
issue 17999: Issues with DTMF triggered attended transfers.
(closes issue #18379)
Reported by: gincantalupo
Tested by: rmudgett
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Tilghman Lesher [Wed, 19 Jan 2011 21:24:25 +0000 (21:24 +0000)]
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r302680 | tilghman | 2011-01-19 15:23:31 -0600 (Wed, 19 Jan 2011) | 16 lines
Merged revisions 302675 via svnmerge from
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r302675 | tilghman | 2011-01-19 15:22:45 -0600 (Wed, 19 Jan 2011) | 9 lines
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r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 Jan 2011) | 2 lines
Add some API documentation
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Tilghman Lesher [Wed, 19 Jan 2011 20:33:30 +0000 (20:33 +0000)]
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r302634 | tilghman | 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) | 22 lines
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r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines
Kill zombies.
When we ast_safe_fork() with a non-zero argument, we're expected to reap our
own zombies. On a zero argument, however, the zombies are only reaped when
there aren't any non-zero forked children alive. At other times, we
accumulate zombies. This code is forward ported from res_agi in 1.4, so that
forked children are always reaped, thus preventing an accumulation of zombie
processes.
(closes issue #18515)
Reported by: ernied
Patches:
20101221__issue18515.diff.txt uploaded by tilghman (license 14)
Tested by: ernied
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Jason Parker [Wed, 19 Jan 2011 20:15:54 +0000 (20:15 +0000)]
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r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) | 1 line
Fix typo pointed out on asterisk-users list.
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Sean Bright [Wed, 19 Jan 2011 19:04:25 +0000 (19:04 +0000)]
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r302555 | seanbright | 2011-01-19 14:03:32 -0500 (Wed, 19 Jan 2011) | 14 lines
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r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines
Don't call strlen() when we only need to look at the next character or two.
(closes issue #18042)
Reported by: wdoekes
Patches:
astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717)
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Sean Bright [Wed, 19 Jan 2011 18:55:43 +0000 (18:55 +0000)]
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r302552 | seanbright | 2011-01-19 13:54:47 -0500 (Wed, 19 Jan 2011) | 14 lines
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r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan 2011) | 7 lines
Remove an extraneous \r\n at the end of a parking manager events.
(closes issue #18363)
Reported by: clegall_proformatique
Patches:
asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139)
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Sean Bright [Wed, 19 Jan 2011 18:45:44 +0000 (18:45 +0000)]
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r302549 | seanbright | 2011-01-19 13:43:11 -0500 (Wed, 19 Jan 2011) | 17 lines
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r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines
Properly handle partial reads from fgets() when handling AGIs.
When fgets() failed with EAGAIN, we were continually decrementing the available
space left in our buffer, resulting in botched command handling.
(closes issue #16032)
Reported by: notahat
Patches:
agi_buffer_patch2.diff uploaded by fnordian (license 110)
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Sean Bright [Wed, 19 Jan 2011 17:59:18 +0000 (17:59 +0000)]
Merged revisions 302505 via svnmerge from
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r302505 | seanbright | 2011-01-19 12:58:11 -0500 (Wed, 19 Jan 2011) | 14 lines
Merged revisions 302504 via svnmerge from
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r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan 2011) | 7 lines
Make sure that h_length is set when we short-circuit out of ast_gethostbyname.
(closes issue #16135)
Reported by: thedavidfactor
Patches:
utils.patch uploaded by thedavidfactor (license 903)
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Paul Belanger [Wed, 19 Jan 2011 17:15:40 +0000 (17:15 +0000)]
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r302462 | pabelanger | 2011-01-19 12:09:35 -0500 (Wed, 19 Jan 2011) | 9 lines
Merged revisions 302461 via svnmerge from
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r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed, 19 Jan 2011) | 2 lines
Handle 'Resource temporarily unavailable' error more gracefully.
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Sean Bright [Wed, 19 Jan 2011 15:54:22 +0000 (15:54 +0000)]
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r302417 | seanbright | 2011-01-19 10:53:20 -0500 (Wed, 19 Jan 2011) | 16 lines
Merged revisions 302416 via svnmerge from
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r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan 2011) | 9 lines
Remove references to priorityjumping from the sample extensions.conf.
Priority jumping was removed from pbx_config in r68970.
(closes issue #18622)
Reported by: kshumard
Patches:
extensions.conf.sample.patch uploaded by kshumard (license 92)
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Sean Bright [Wed, 19 Jan 2011 15:46:56 +0000 (15:46 +0000)]
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r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan 2011) | 7 lines
Initialize an uninitialized variable.
(closes issue #18640)
Reported by: jcovert
Patches:
chan_sip.c.patch uploaded by jcovert (license 551)
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Sean Bright [Wed, 19 Jan 2011 15:34:07 +0000 (15:34 +0000)]
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r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines
Use appropriate type for requested format in chan_local.
We were passing and storing the requested format as an int instead of format_t
resulting in truncation.
(closes issue #18238)
Reported by: whizemen
Patches:
0018238_speex16.patch uploaded by whizemen (license 1143)
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Richard Mudgett [Tue, 18 Jan 2011 22:06:55 +0000 (22:06 +0000)]
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r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) | 1 line
Use the expanded format type instead of plain int.
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Matthew Nicholson [Tue, 18 Jan 2011 21:44:49 +0000 (21:44 +0000)]
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r302314 | mnicholson | 2011-01-18 15:43:21 -0600 (Tue, 18 Jan 2011) | 18 lines
Merged revisions 302313 via svnmerge from
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r302313 | mnicholson | 2011-01-18 15:40:03 -0600 (Tue, 18 Jan 2011) | 11 lines
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r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan 2011) | 4 lines
URI encode the user part of the contact header.
ABE-2705
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Jeff Peeler [Tue, 18 Jan 2011 20:40:59 +0000 (20:40 +0000)]
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r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines
Merged revisions 302265 via svnmerge from
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r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines
Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip.
Lock scenario presented here:
Thread 1
holds ast_rdlock_contexts &conlock
holds handle_statechange hints
holds handle_statechange hint
waiting for cb_extensionstate
Locked Here: chan_sip.c line 7428 (find_call)
Thread 2
holds handle_request_do &netlock
holds find_call sip_pvt_ptr
waiting for ast_rdlock_contexts &conlock
Locked Here: pbx.c line 9911 (ast_rdlock_contexts)
Chan_sip has an established locking order of locking the sip_pvt and then
getting the context lock. So the as stated by the summary, the operations in
thread 2 have been modified to no longer require the context lock.
(closes issue #18310)
Reported by: one47
Patches:
statecbs_ao2.mk2.patch uploaded by one47 (license 23),
modified by me
Review: https://reviewboard.asterisk.org/r/1072/
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Russell Bryant [Tue, 18 Jan 2011 20:21:29 +0000 (20:21 +0000)]
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r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 5 lines
Don't enable AO2_DEBUG by default if AST_DEVMODE is on.
AO2_DEBUG is not important and is causing a false compiler warning to be
generated on my Ubuntu Natty dev box.
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Richard Mudgett [Tue, 18 Jan 2011 18:17:01 +0000 (18:17 +0000)]
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r302174 | rmudgett | 2011-01-18 12:11:43 -0600 (Tue, 18 Jan 2011) | 102 lines
Merged revisions 302173 via svnmerge from
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r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines
Merged revisions 302172 via svnmerge from
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r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines
Issues with DTMF triggered attended transfers.
Issue #17999
1) A calls B. B answers.
2) B using DTMF dial *2 (code in features.conf for attended transfer).
3) A hears MOH. B dial number C
4) C ringing. A hears MOH.
5) B hangup. A still hears MOH. C ringing.
6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
For v1.4 C will ring forever until C answers the dead line. (Issue #17096)
Problem: When A and B hangup, C is still ringing.
Issue #18395
SIP call limit of B is 1
1. A call B, B answered
2. B *2(atxfer) call C
3. B hangup, C ringing
4. Timeout waiting for C to answer
5. Recall to B fails because B has reached its call limit.
Because B reached its call limit, it cannot do anything until the transfer
it started completes.
Issue #17273
Same scenario as issue 18395 but party B is an FXS port. Party B cannot
do anything until the transfer it started completes. If B goes back off
hook before C answers, B hears ringback instead of the expected dialtone.
**********
Note for the issue #17273 and #18395 fix:
DTMF attended transfer works within the channel bridge. Unfortunately,
when either party A or B in the channel bridge hangs up, that channel is
not completely hung up until the transfer completes. This is a real
problem depending upon the channel technology involved.
For chan_dahdi, the channel is crippled until the hangup is complete.
Either the channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.
For chan_sip, a call limit of one is going to block that endpoint from any
further calls until the hangup is complete.
For party A this is a minor problem. The party A channel will only be in
this condition while party B is dialing and when party B and C are
conferring. The conversation between party B and C is expected to be a
short one. Party B is either asking a question of party C or announcing
party A. Also party A does not have much incentive to hangup at this
point.
For party B this can be a major problem during a blonde transfer. (A
blonde transfer is our term for an attended transfer that is converted
into a blind transfer. :)) Party B could be the operator. When party B
hangs up, he assumes that he is out of the original call entirely. The
party B channel will be in this condition while party C is ringing, while
attempting to recall party B, and while waiting between call attempts.
WARNING:
The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will
replace the party B channel technology with a NULL channel driver to
complete hanging up the party B channel technology. The consequences of
this code is that the 'h' extension will not be able to access any channel
technology specific information like SIP statistics for the call.
ATXFER_NULL_TECH is not defined by default.
**********
(closes issue #17999)
Reported by: iskatel
Tested by: rmudgett
JIRA SWP-2246
(closes issue #17096)
Reported by: gelo
Tested by: rmudgett
JIRA SWP-1192
(closes issue #18395)
Reported by: shihchuan
Tested by: rmudgett
(closes issue #17273)
Reported by: grecco
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1047/
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Terry Wilson [Mon, 17 Jan 2011 16:38:21 +0000 (16:38 +0000)]
Merged revisions 293493 via svnmerge from
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r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines
Only offer codecs both sides support for directmedia
When using directmedia, Asterisk needs to limit the codecs offered to just
the ones that both sides recognize, otherwise they may end up sending audio
that the other side doesn't understand.
(closes issue #17403)
Reported by: one47
Patches:
sip_codecs_simplified4 uploaded by one47 (license 23)
Tested by: one47, falves11
Review: https://reviewboard.asterisk.org/r/967/
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Terry Wilson [Mon, 17 Jan 2011 15:06:10 +0000 (15:06 +0000)]
Merged revisions 302005 via svnmerge from
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r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 Jan 2011) | 2 lines
Document "encryption" option in sip.conf.sample
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Richard Mudgett [Fri, 14 Jan 2011 21:13:08 +0000 (21:13 +0000)]
Merged revisions 301946 via svnmerge from
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r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) | 13 lines
Deadlock between dahdi_request() and pri_dchannel() processing an incomming call.
The sig_pri_new_ast_channel() is called with the channel private lock held
when pri_dchannel() calls it and no channel private lock held when
dahdi_request() calls it. The use of pri_grab() in
sig_pri_new_ast_channel() could leave the channel private lock held when
it returns if the lock was not held before calling it.
Make sig_pri_new_ast_channel() just lock the PRI span lock instead of
using pri_grab(). It is safe to do this because dahdi_request() does not
have the channel private lock and the deadlock potential with the PRI span
lock is only between pri_dchannel() and other threads.
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Brett Bryant [Fri, 14 Jan 2011 20:18:26 +0000 (20:18 +0000)]
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r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) | 6 lines
Changing previous revisions 301845/301847 to use ast_sockaddr_setnull() instead
of setting the field manually to avoid uninitialized data.
Review: https://reviewboard.asterisk.org/r/1076/
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Andrew Latham [Fri, 14 Jan 2011 20:07:02 +0000 (20:07 +0000)]
Add relationships to function documentation.
Fix amatuer type mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301850
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Brett Bryant [Fri, 14 Jan 2011 19:44:11 +0000 (19:44 +0000)]
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r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) | 9 lines
Fix for a consistent MulticastRTP channel driver crash due to use of unitilized
data.
(closes issue #18290)
(closes issue #18602)
Reported by: voipgate, wybecom
Review: https://reviewboard.asterisk.org/r/1076/
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Andrew Latham [Fri, 14 Jan 2011 19:39:22 +0000 (19:39 +0000)]
Add relationships to function documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301846
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Jeff Peeler [Fri, 14 Jan 2011 17:34:28 +0000 (17:34 +0000)]
Merged revisions 301790 via svnmerge from
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r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) | 42 lines
Resolve deadlock involving REFER.
Two fixes:
1) One must always have the private unlocked before calling
pbx_builtin_setvar_helper to not invalidate locking order since it locks the
channel.
2) Unlock the channel before calling pbx_find_extension, which starts and stops
autoservice during the lookup. The problem scenario as illustrated by the
reporter:
Thread: do_monitor
-----------------------
handle_request_do
handle_incoming
handle_request_refer
ast_parking_ext_valid
pbx_find_extension
ast_autoservice_stop
while (chan_list_state == as_chan_list_state) { usleep(1000); }
Thread: autoservice_run
-----------------------
autoservice_run
chan = ast_waitfor_n
ast_waitfor_nandfds
ast_waitfor_nandfds_classic / simple / complex (depending on your system)
ast_channel_lock(c[x]);
handle_request_do and schedule_process_request_queue locks the owner
if it exists. The autoservice thread is waiting for the channel lock, which
wasn't ever released since the do_monitor thread was waiting for autoservice
operations to complete. Solved by unlocking the channel but keeping a reference
to guarantee safety.
(closes issue #18403)
Reported by: jthurman
Patches:
20110103-blind_deadlock.diff uploaded by jthurman (license 614)
issue18403.patch uploaded by jpeeler (license 325)
Tested by: jthurman
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Leif Madsen [Thu, 13 Jan 2011 17:02:34 +0000 (17:02 +0000)]
Merged revisions 301731 via svnmerge from
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r301731 | lmadsen | 2011-01-13 11:01:43 -0600 (Thu, 13 Jan 2011) | 15 lines
Merged revisions 301730 via svnmerge from
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r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) | 7 lines
Add static entry for split Polycom 332 firmware.
(closes issue #18607)
Reported by: cjacobsen
Patches:
polycom_331.diff uploaded by cjacobsen (license 1029)
Tested by: lathama
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Paul Belanger [Thu, 13 Jan 2011 16:27:22 +0000 (16:27 +0000)]
Add dialplan variables for asterisk.conf directories
Review: https://reviewboard.asterisk.org/r/1075/
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Terry Wilson [Wed, 12 Jan 2011 21:24:18 +0000 (21:24 +0000)]
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r301683 | twilson | 2011-01-12 15:19:48 -0600 (Wed, 12 Jan 2011) | 15 lines
Merged revisions 301682 via svnmerge from
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r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) | 9 lines
Don't reject all SUBSCRIBE auth requests
When merging another SUBSCRIBE fix from 1.4, some braces were put in
the wrong place. This patch fixes that.
(closes issue #18597)
Reported by: thsgmbh
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Matthew Nicholson [Wed, 12 Jan 2011 18:52:30 +0000 (18:52 +0000)]
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r301595 | mnicholson | 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) | 22 lines
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r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines
Removed a usleep(1) that shouldn't be necessary in session_do, and removed the
ms_t member from the mansession_session structure.
Merged revisions 301591 via svnmerge from
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r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines
Don't store the thread id for the manager session in the structure we pass to
the thread for the manager session.
ABE-2543
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Jeff Peeler [Wed, 12 Jan 2011 18:12:31 +0000 (18:12 +0000)]
Merged revisions 301504 via svnmerge from
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r301504 | jpeeler | 2011-01-12 12:12:08 -0600 (Wed, 12 Jan 2011) | 26 lines
Merged revisions 301503 via svnmerge from
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r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines
Merged revisions 301502 via svnmerge from
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r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines
Fix CPU spike when pressing DTMF after agent login.
The problem here is that DTMF was being continuously deferred and requeued
since ast_safe_sleep is called in a loop. There are serveral other places in the
code that sleeps and then loops in a similar fashion. Because of this fact I
opted to not defer DTMF any more, which will not affect the original fix:
https://reviewboard.asterisk.org/r/674
(closes issue #18130)
Reported by: rgj
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David Vossel [Wed, 12 Jan 2011 16:05:58 +0000 (16:05 +0000)]
Merged revisions 301446 via svnmerge from
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r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) | 2 lines
Removal of unused variables so Asterisk will compile.
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Stefan Schmidt [Wed, 12 Jan 2011 15:59:28 +0000 (15:59 +0000)]
fix wrong text of rerun menuselect after user interface warning
the warning, if no user interface for menuselect warning was found is not right.
you have to rerun configure before make menuselect after installing a proper user interface.
(closes issue 0018594)
Reported by: Dovid
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Tilghman Lesher [Wed, 12 Jan 2011 00:27:30 +0000 (00:27 +0000)]
Merged revisions 301402 via svnmerge from
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r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) | 7 lines
Call execl() directly for a better solution for paths with spaces.
(closes issue #18600)
Reported by: ebroad
Patches:
20110111__issue18600__2.diff.txt uploaded by tilghman (license 14)
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Paul Belanger [Tue, 11 Jan 2011 19:19:01 +0000 (19:19 +0000)]
Merged revisions 301311 via svnmerge from
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r301311 | pabelanger | 2011-01-11 14:16:06 -0500 (Tue, 11 Jan 2011) | 9 lines
Merged revisions 301310 via svnmerge from
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r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, 11 Jan 2011) | 2 lines
Fix a logic issue when passing context ARG
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Matthew Nicholson [Tue, 11 Jan 2011 18:55:16 +0000 (18:55 +0000)]
Merged revisions 301308 via svnmerge from
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r301308 | mnicholson | 2011-01-11 12:51:40 -0600 (Tue, 11 Jan 2011) | 18 lines
Merged revisions 301307 via svnmerge from
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r301307 | mnicholson | 2011-01-11 12:42:05 -0600 (Tue, 11 Jan 2011) | 11 lines
Merged revisions 301305 via svnmerge from
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r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines
Prevent buffer overflows in ast_uri_encode()
ABE-2705
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Tilghman Lesher [Mon, 10 Jan 2011 22:40:23 +0000 (22:40 +0000)]
Merged revisions 301263 via svnmerge from
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r301263 | tilghman | 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011) | 8 lines
Little endian machines were not converted properly.
(closes issue #18583)
Reported by: jcovert
Patches:
20110110__issue18583.diff.txt uploaded by tilghman (license 14)
Tested by: jcovert
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Paul Belanger [Sun, 9 Jan 2011 21:42:47 +0000 (21:42 +0000)]
Merged revisions 301221 via svnmerge from
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r301221 | pabelanger | 2011-01-09 16:40:34 -0500 (Sun, 09 Jan 2011) | 21 lines
Merged revisions 301220 via svnmerge from
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r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan 2011) | 14 lines
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still override the setting
using the --with-sounds-cache option.
(closes issue #18589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/1074/
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Paul Belanger [Sat, 8 Jan 2011 22:02:39 +0000 (22:02 +0000)]
Merged revisions 301177 via svnmerge from
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r301177 | pabelanger | 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) | 14 lines
Merged revisions 301176 via svnmerge from
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r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) | 7 lines
Indicate log level argument for Log() is not optional
(closes issue #18586)
Reported by: kshumard
Patches:
app_verbose.c.patch uploaded by kshumard (license 92)
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Richard Mudgett [Sat, 8 Jan 2011 01:13:58 +0000 (01:13 +0000)]
Merged revisions 301134 via svnmerge from
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r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 Jan 2011) | 7 lines
The DTMF attended transfer feature cannot callback a chan_dahdi BRI phone.
The DAHDI ISDN channel name is not dialable.
Make a channel name like DAHDI/i3/400-12 dialable when the sequence number
is stripped off of the name.
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Jason Parker [Fri, 7 Jan 2011 20:53:45 +0000 (20:53 +0000)]
Merged revisions 301090 via svnmerge from
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r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines
Merged revisions 301089 via svnmerge from
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r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines
Initialize useropts/adminopts in case there is no column in the realtime DB.
(closes issue #18182)
Reported by: dimas
Patches:
v1-18182.patch uploaded by dimas (license 88)
Tested by: dimas
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Jeff Peeler [Fri, 7 Jan 2011 19:58:52 +0000 (19:58 +0000)]
Merged revisions 301047 via svnmerge from
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r301047 | jpeeler | 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) | 15 lines
Merged revisions 301046 via svnmerge from
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r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines
Fix regression causing forwarding voicemails to not work with file storage.
I had actually already fixed this in 295200 in 1.4 and thought it wasn't
missing in the other branches for some reason.
(closes issue #18358)
Reported by: cabal95
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Tilghman Lesher [Fri, 7 Jan 2011 18:23:52 +0000 (18:23 +0000)]
Oops, missed the actual decoding part.
(closes issue #18046)
Reported by: wdoekes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301008
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Jeff Peeler [Fri, 7 Jan 2011 17:24:52 +0000 (17:24 +0000)]
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r300955 | jpeeler | 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) | 21 lines
Merged revisions 300951 via svnmerge from
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r300951 | jpeeler | 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) | 14 lines
Merged revisions 300918 via svnmerge from
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r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines
Ensure good bye prompt in voicemail is played at the correct time.
Specifically in the case of timing out but not leaving voicemail nothing
should be heard. And when leaving voicemail it should be heard.
ABE-2647
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Mark Murawki [Fri, 7 Jan 2011 07:47:36 +0000 (07:47 +0000)]
Added support for postgres database retry query on disconnection to res_config_pgsql
If your postgres connection died suddenly in between res_config_pgsql
queries, the next query will fail because the query is executed on a
disconnected/disconnecting handle. The query is abandoned and is
returned from in error.
Now we will reconnect and try again if a query was run on a
disconnected connection.
(closes issue #18071)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300882
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Tilghman Lesher [Thu, 6 Jan 2011 17:50:57 +0000 (17:50 +0000)]
XML validation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300841
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Tilghman Lesher [Thu, 6 Jan 2011 17:28:32 +0000 (17:28 +0000)]
Add a hashcompat mode called "legacy", which translates a literal plus sign to a space.
(closes issue #18046)
Reported by: wdoekes
Patches:
20100930__issue18046.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300840
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Tilghman Lesher [Thu, 6 Jan 2011 06:29:08 +0000 (06:29 +0000)]
Merged revisions 300798 via svnmerge from
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r300798 | tilghman | 2011-01-06 00:28:18 -0600 (Thu, 06 Jan 2011) | 8 lines
Don't destroy handle not created by use (because the caller will).
(closes issue #18526)
Reported by: makoto
Patches:
res-config-mysql-include.patch uploaded by makoto (license 38)
Tested by: makoto
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David Ruggles [Thu, 6 Jan 2011 01:41:57 +0000 (01:41 +0000)]
update safe_asterisk script
change defaults to make a little more sense. Default log location is now asterisk log location and default email notification has been changed to root on the local machine
Review: https://reviewboard.asterisk.org/r/1067/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300761
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Richard Mudgett [Wed, 5 Jan 2011 21:07:40 +0000 (21:07 +0000)]
Merged revisions 300714 via svnmerge from
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r300714 | rmudgett | 2011-01-05 14:54:21 -0600 (Wed, 05 Jan 2011) | 21 lines
Merged revision 300711 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, 05 Jan 2011) | 14 lines
A call retrieved from hold may wind up with no audio.
If the retrieved call is natively bridged then the call may not have any
audio path. The following warning message is given:
"Failed to add <dfd> to conference <chan>/<chan>: Invalid argument".
* Open the media on a B channel when pri_fixup_principle() moves the call
from a no_b_channel channel to a real channel.
* Added lock protection while pri_fixup_principle() moves a call from one
private structure to another.
* Made some pri_fixup_principle() messages more meaningful.
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Tilghman Lesher [Wed, 5 Jan 2011 18:57:05 +0000 (18:57 +0000)]
Merged revisions 300623 via svnmerge from
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r300623 | tilghman | 2011-01-05 12:56:12 -0600 (Wed, 05 Jan 2011) | 24 lines
Merged revisions 300622 via svnmerge from
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r300622 | tilghman | 2011-01-05 12:54:58 -0600 (Wed, 05 Jan 2011) | 17 lines
Merged revisions 300621 via svnmerge from
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r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines
Use the sanity check in place of the disconnect/connect cycle.
The disconnect/connect cycle has the potential to cause random crashes.
(closes issue #18243)
Reported by: ks3
Patches:
res_odbc.patch uploaded by ks3 (license 1147)
Tested by: ks3
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Paul Belanger [Wed, 5 Jan 2011 16:30:56 +0000 (16:30 +0000)]
Merged revisions 300575 via svnmerge from
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r300575 | pabelanger | 2011-01-05 11:29:19 -0500 (Wed, 05 Jan 2011) | 13 lines
Merged revisions 300574 via svnmerge from
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r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan 2011) | 6 lines
Change deprecated message to LOG_WARNING
Also removed latter part of message
Discussed on #asterisk-dev
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Leif Madsen [Tue, 4 Jan 2011 21:54:20 +0000 (21:54 +0000)]
Merged revisions 300521 via svnmerge from
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r300521 | lmadsen | 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) | 17 lines
Merged revisions 300520 via svnmerge from
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r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines
Fix backwards and broken XML documentation.
(closes issue #18547)
Reported by: jcovert
Patches:
xmldoc.c.patch uploaded by jcovert (license 551)
chan_iax2.c.doc.patch uploaded by jcovert (license 551)
chan_sip.c.patch uploaded by jcovert (license 551)
chan_agent.c.patch uploaded by jcovert (license 551)
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Leif Madsen [Tue, 4 Jan 2011 21:01:30 +0000 (21:01 +0000)]
Merged revisions 300433 via svnmerge from
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r300433 | lmadsen | 2011-01-04 15:00:55 -0600 (Tue, 04 Jan 2011) | 15 lines
Merged revisions 300431 via svnmerge from
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r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011) | 7 lines
Add some documentation to users.conf.sample.
(closes issue #18531)
Reported by: lathama
Patches:
users.conf.sample2.diff uploaded by lathama (license 1028)
Tested by: lathama
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Russell Bryant [Tue, 4 Jan 2011 21:00:36 +0000 (21:00 +0000)]
Merged revisions 300430 via svnmerge from
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r300430 | russell | 2011-01-04 15:00:16 -0600 (Tue, 04 Jan 2011) | 18 lines
Merged revisions 300429 via svnmerge from
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r300429 | russell | 2011-01-04 14:59:56 -0600 (Tue, 04 Jan 2011) | 11 lines
Merged revisions 300428 via svnmerge from
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r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) | 4 lines
Update the autosupport script from Digium support.
(closes AST-395)
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Leif Madsen [Tue, 4 Jan 2011 19:45:59 +0000 (19:45 +0000)]
Merged revisions 300384 via svnmerge from
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r300384 | lmadsen | 2011-01-04 13:45:22 -0600 (Tue, 04 Jan 2011) | 7 lines
Update STAT() to use the comma instead of the pipe.
(closes issue #18503)
Reported by: cjacobsen
Patches:
old_separator.diff uploaded by cjacobsen (license 1029)
Tested by: lathama
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Moises Silva [Tue, 4 Jan 2011 18:51:58 +0000 (18:51 +0000)]
Update MFC-R2 code to use new DTMF-R2 functionality in OpenR2
(closes issue #18576)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300345
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Terry Wilson [Tue, 4 Jan 2011 18:06:46 +0000 (18:06 +0000)]
Merged revisions 300301 via svnmerge from
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r300301 | twilson | 2011-01-04 11:54:41 -0600 (Tue, 04 Jan 2011) | 29 lines
Merged revisions 300298 via svnmerge from
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r300298 | twilson | 2011-01-04 11:37:26 -0600 (Tue, 04 Jan 2011) | 22 lines
Merged revisions 300216 via svnmerge from
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r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) | 15 lines
Don't authenticate SUBSCRIBE re-transmissions
This only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer...
(closes issue #18075)
Reported by: mdu113
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, mdu113
Review: https://reviewboard.asterisk.org/r/1005/
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Jan Kalab [Tue, 4 Jan 2011 17:04:14 +0000 (17:04 +0000)]
Merged revisions 300214 via svnmerge from
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r300214 | pitel | 2011-01-04 18:01:52 +0100 (Út, 04 led 2011) | 7 lines
Memory leaking in calendars
ne_request_destroy() was missing in icalendar and exchange calendar modules, causing memory leak.
(closes issue #18521)
Review: https://reviewboard.asterisk.org/r/1068/
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Richard Mudgett [Tue, 4 Jan 2011 16:38:28 +0000 (16:38 +0000)]
Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off hold.
Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212
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Richard Mudgett [Mon, 3 Jan 2011 23:18:20 +0000 (23:18 +0000)]
Merged revisions 300166 via svnmerge from
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r300166 | rmudgett | 2011-01-03 17:14:55 -0600 (Mon, 03 Jan 2011) | 11 lines
Merged revisions 300165 via svnmerge from
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r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) | 4 lines
Use correct variable for atxfercallbackretries config option.
* Misc formatting changes.
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David Ruggles [Mon, 3 Jan 2011 14:09:29 +0000 (14:09 +0000)]
initialize playing_silence in struct initialization
playing_silence was not initialized with the struct
was initialized, it was being set after the fact
which caused problems if something that relied on
playing_silence being set was called too quickly
(closes issue #18430)
Reported by: stevebrandli
Patches:
externalivr.patch uploaded by thedavidfactor (license 903)
Tested by: thedavidfactor, stevebrandli
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300121
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Leif Madsen [Mon, 3 Jan 2011 13:15:38 +0000 (13:15 +0000)]
Merged revisions 300082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011) | 11 lines
Increase side of mapping response field.
I've increased the size of the response field in a DUNDi mapping because of
some documentation I'm writing. Previously it was set to AST_MAX_EXTENSION which
is only 80 characters, which is far too small when you're using some dialplan
functions to craft a response. The example I'm using is:
extensions =>
RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial
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Tilghman Lesher [Fri, 31 Dec 2010 09:29:10 +0000 (09:29 +0000)]
Support negative filters.
(closes issue #17979)
Reported by: tilghman
Patches:
20100911__for_blitzrage.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300045
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Tilghman Lesher [Fri, 31 Dec 2010 09:21:47 +0000 (09:21 +0000)]
Support an alternate configuration file for the 'logger reload' command.
(closes issue #17668)
Reported by: tilghman
Patches:
20100718__logger_reload_altconf__2.diff.txt uploaded by tilghman (license 14)
Review: (by lmadsen, russell within comments on issue tracker)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300044
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Sean Bright [Wed, 29 Dec 2010 22:19:26 +0000 (22:19 +0000)]
Remove some trailing whitespace and steal revision 300000.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300000
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Tilghman Lesher [Wed, 29 Dec 2010 22:03:50 +0000 (22:03 +0000)]
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r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines
Quote arguments, just in case there's a space in a pathname.
(Diagnosed by pabelanger on #asterisk-dev, fixed by me.)
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Paul Belanger [Wed, 29 Dec 2010 19:29:51 +0000 (19:29 +0000)]
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r299948 | pabelanger | 2010-12-29 14:28:36 -0500 (Wed, 29 Dec 2010) | 2 lines
Only remove /tmp/astdatadir, not /var/lib/asterisk
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Paul Belanger [Wed, 29 Dec 2010 18:24:19 +0000 (18:24 +0000)]
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r299907 | pabelanger | 2010-12-29 13:22:23 -0500 (Wed, 29 Dec 2010) | 2 lines
Properly quote varibles for MAC OS X
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Paul Belanger [Tue, 28 Dec 2010 19:00:04 +0000 (19:00 +0000)]
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r299865 | pabelanger | 2010-12-28 13:53:37 -0500 (Tue, 28 Dec 2010) | 9 lines
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r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) | 2 lines
Documentation typo
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Tilghman Lesher [Mon, 27 Dec 2010 21:23:55 +0000 (21:23 +0000)]
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r299820 | tilghman | 2010-12-27 15:23:10 -0600 (Mon, 27 Dec 2010) | 2 lines
More space-in-pathname issues.
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Tilghman Lesher [Mon, 27 Dec 2010 20:42:01 +0000 (20:42 +0000)]
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r299794 | tilghman | 2010-12-27 14:41:04 -0600 (Mon, 27 Dec 2010) | 2 lines
Mac OS X spaces-in-pathnames fix.
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Tilghman Lesher [Sun, 26 Dec 2010 21:17:02 +0000 (21:17 +0000)]
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r299752 | tilghman | 2010-12-26 15:15:58 -0600 (Sun, 26 Dec 2010) | 2 lines
Properly quote path on Darwin.
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Alexandr Anikin [Sat, 25 Dec 2010 16:35:09 +0000 (16:35 +0000)]
Change order of sending TCS and MSD packets
Change order of sending Terminal Capability Set and MasterSlave
Determination packets, MSD send when TCS exchange procedure is done
(we send tcs ack to remote and we have remote tcs ack already
or we receive tcs ack from remote and we have send our tcs ack to
remote already). Some endpoints can work in this sequence only,
i suggest they can't work with both (tcs and msd) exchange procedures
simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls.
(closes issue #18433)
Reported by: MrHanMan
Patches:
tcs-msd-h245-3.patch uploaded by may213 (license 454)
Tested by: MrHanMan, may213
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Tilghman Lesher [Sat, 25 Dec 2010 10:08:04 +0000 (10:08 +0000)]
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r299626 | tilghman | 2010-12-25 04:07:15 -0600 (Sat, 25 Dec 2010) | 19 lines
Merged revisions 299625 via svnmerge from
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r299625 | tilghman | 2010-12-25 04:05:00 -0600 (Sat, 25 Dec 2010) | 12 lines
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r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) | 5 lines
Move check for extension existence below variable inheritance, due to the possible use of an eswitch.
(closes issue #16228)
Reported by: jlaguilar
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Tilghman Lesher [Fri, 24 Dec 2010 17:59:23 +0000 (17:59 +0000)]
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r299583 | tilghman | 2010-12-24 11:58:30 -0600 (Fri, 24 Dec 2010) | 7 lines
Reset 'first' variable after usage.
(closes issue #18525)
Reported by: makoto
Patches:
res-config-mysql-update2.patch uploaded by makoto (license 38)
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Moises Silva [Thu, 23 Dec 2010 02:55:28 +0000 (02:55 +0000)]
Blocked revisions 299531 via svnmerge
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r299531 | moy | 2010-12-22 21:53:02 -0500 (Wed, 22 Dec 2010) | 13 lines
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r299530 | moy | 2010-12-22 21:28:37 -0500 (Wed, 22 Dec 2010) | 7 lines
Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are accepted
(closes issue #18438)
Reported by: mariner7
Tested by: moy
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Moises Silva [Thu, 23 Dec 2010 01:46:16 +0000 (01:46 +0000)]
Enqueue AST_CONTROL_PROGRESS after AST_CONTROL_RINGING when MFC-R2 calls are accepted
(closes issue #18438)
Reported by: mariner7
Tested by: moy
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Tilghman Lesher [Wed, 22 Dec 2010 20:10:34 +0000 (20:10 +0000)]
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r299449 | tilghman | 2010-12-22 14:05:02 -0600 (Wed, 22 Dec 2010) | 15 lines
Merged revisions 299448 via svnmerge from
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r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) | 8 lines
Resolve warnings by disambiguating the "s" extension as used by chan_dahdi from the "s" extension as used by the AEL macros.
(closes issue #18480)
Reported by: nivek
Patches:
20101215__issue18480__2.diff.txt uploaded by tilghman (license 14)
Tested by: nivek
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Richard Mudgett [Wed, 22 Dec 2010 02:12:01 +0000 (02:12 +0000)]
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r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) | 17 lines
Chan_dahdi sends an empty COLP on the bridged channel.
Chan_dahdi always inserts a connected party IE when you call from one
dahdi channel to another dahdi channel, even if no such information was
received on the 2nd channel. This clears the display of many phones.
* Removed leftover artifact from before the valid flag was added.
* Updated all of the channel's caller id information with the new
connected line information instead of just the string parts.
(closes issue #18508)
Reported by: wimpy
Patches:
issue18508_trunk.patch uploaded by rmudgett (license 664)
Tested by: wimpy, rmudgett
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Matthew Nicholson [Tue, 21 Dec 2010 16:02:52 +0000 (16:02 +0000)]
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r299353 | mnicholson | 2010-12-21 09:25:03 -0600 (Tue, 21 Dec 2010) | 30 lines
Merged revisions 299242 via svnmerge from
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r299242 | mnicholson | 2010-12-20 15:25:35 -0600 (Mon, 20 Dec 2010) | 23 lines
Merged revisions 299194,299198,299220 via svnmerge from
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r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec 2010) | 6 lines
Respond as soon as possible with a 202 Accepted to refer requests.
This change also plugs a few memory leaks that can occur when parking sip calls.
ABE-2656
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r299198 | mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 lines
Remove changes to via processing that were not supposed to go into the last commit.
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r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines
Use ast_free() instead of free()
ABE-2656
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