asterisk/asterisk.git
12 years agoAdd AST_FORMAT_SLINEAR16 to the list for ast_best_codec()
Russell Bryant [Thu, 17 Jan 2008 23:20:37 +0000 (23:20 +0000)]
Add AST_FORMAT_SLINEAR16 to the list for ast_best_codec()

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 99004 via svnmerge from
Russell Bryant [Thu, 17 Jan 2008 22:50:13 +0000 (22:50 +0000)]
Merged revisions 99004 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines

Have IAX2 optimize the codec translation path just like chan_sip does it.  If
the caller's codec is in our codec list, move it to the top to avoid transcoding.

(closes issue #10500)
Reported by: stevedavies
Patches:
      iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
      iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99006 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoFixing trunk IMAP build
Mark Michelson [Thu, 17 Jan 2008 22:22:02 +0000 (22:22 +0000)]
Fixing trunk IMAP build

(closes issue #11788)
Reported by: DEA
Patches:
      vm-imap-build-fix.txt uploaded by DEA (license 3)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99002 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd several busy detection related defines to menuselect.
Jason Parker [Thu, 17 Jan 2008 20:51:26 +0000 (20:51 +0000)]
Add several busy detection related defines to menuselect.
Allow better busy detect debugging (with BUSYDETECT_DEBUG).

Remove very old BUSYDETECT and BUSY_DETECT_MARTIN defines.

(closes issue #11107)
Patches:
      busydetect_enhancement.patch uploaded by agx (license 298)
      busydetect-r94975.diff uploaded by sergee (license 138)

Additional changes/cleanup by me.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98998 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agostate_interface could be NULL, so use the never-NULL cur->state_interface for this...
Mark Michelson [Thu, 17 Jan 2008 16:33:24 +0000 (16:33 +0000)]
state_interface could be NULL, so use the never-NULL cur->state_interface for this check

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98994 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoGet the device state of the state interface instead of the interface when creating...
Mark Michelson [Thu, 17 Jan 2008 16:26:41 +0000 (16:26 +0000)]
Get the device state of the state interface instead of the interface when creating a new queue member.
Thanks to Atis Lezdins for bringing this up on the Asterisk-Dev mailing list.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98993 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98991 via svnmerge from
Jason Parker [Thu, 17 Jan 2008 16:21:38 +0000 (16:21 +0000)]
Merged revisions 98991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

(Closes issue #11784)
........
r98991 | qwell | 2008-01-17 10:19:46 -0600 (Thu, 17 Jan 2008) | 4 lines

Add a clarification about the immediate= option of zapata.conf

Issue 11784, patch by klaus3000.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98992 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agomajor reliability and performance improvement in VWMI monitoring for FXO ports (code...
Kevin P. Fleming [Thu, 17 Jan 2008 16:17:52 +0000 (16:17 +0000)]
major reliability and performance improvement in VWMI monitoring for FXO ports (code by markster, me and dbailey)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98990 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoresolve (valid) compiler warning about variable that could be used before being initi...
Kevin P. Fleming [Thu, 17 Jan 2008 15:04:54 +0000 (15:04 +0000)]
resolve (valid) compiler warning about variable that could be used before being initialized

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoUpdate res_phoneprov to default to setting the SERVER variable to the IP
Terry Wilson [Thu, 17 Jan 2008 03:09:32 +0000 (03:09 +0000)]
Update res_phoneprov to default to setting the SERVER variable to the IP
the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf.  I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.

Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport.  Tested on Linux and OS X.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoChange the way the new filter feature works, by allowing it to be a column NOT
Tilghman Lesher [Thu, 17 Jan 2008 00:13:32 +0000 (00:13 +0000)]
Change the way the new filter feature works, by allowing it to be a column NOT
logged into the database.  This will allow more granularity of a decision
evaluated in the dialplan, then takes effect when posting the CDR.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98987 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd support for an easy way to automatically execute some Asterisk CLI commands
Russell Bryant [Thu, 17 Jan 2008 00:05:13 +0000 (00:05 +0000)]
Add support for an easy way to automatically execute some Asterisk CLI commands
immediately at startup.  Any commands in the startup_commands file in the Asterisk
config diretory will get executed.

(closes issue #11781)
Reported by: jamesgolovich
Patches:
      asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176)
    -- With some changes by me.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98986 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoChange AST_EXT_TOOL_CHECK to attempt to build against <package>_LIB, per recommendati...
Jason Parker [Wed, 16 Jan 2008 23:08:11 +0000 (23:08 +0000)]
Change AST_EXT_TOOL_CHECK to attempt to build against <package>_LIB, per recommendations from Russell.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoInfo about res_config_curl
Tilghman Lesher [Wed, 16 Jan 2008 22:36:58 +0000 (22:36 +0000)]
Info about res_config_curl

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98984 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBlocked revisions 98982 via svnmerge
Russell Bryant [Wed, 16 Jan 2008 22:36:47 +0000 (22:36 +0000)]
Blocked revisions 98982 via svnmerge

........
r98982 | russell | 2008-01-16 16:36:24 -0600 (Wed, 16 Jan 2008) | 5 lines

Add an unused pointer to the ast_channel struct.  This makes the ast_channel structure
retain the same size as it had in previous 1.4 releases.  Also, all of the offsets for
members in the structure are still the same (except for the two pointers that got replaced
for the new spy/whisper architecture.)

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98983 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoNew module res_config_curl (closes issue #11747)
Tilghman Lesher [Wed, 16 Jan 2008 22:20:45 +0000 (22:20 +0000)]
New module res_config_curl (closes issue #11747)
 Reported by: Corydon76
 Patches:
       res_config_curl.c uploaded by Corydon76 (license 14)
       20080116__bug11747.diff.txt uploaded by Corydon76 (license 14)
 Tested by: jmls

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98981 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerge the changes from issue #10665 from the team/group/sip_session_timers branch.
Russell Bryant [Wed, 16 Jan 2008 21:53:10 +0000 (21:53 +0000)]
Merge the changes from issue #10665 from the team/group/sip_session_timers branch.

This set of changes introduces SIP session timers support (RFC 4028).  In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.

To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."

(closes issue #10665)
Reported by: rjain
Patches:
      chan_sip.c.1.diff uploaded by rjain (license 226)
      chan_sip.c.diff uploaded by rjain (license 226)
      sip.conf.sample.diff uploaded by rjain (license 226)
      proc_422_rsp_comment.diff uploaded by rjain (license 226)
      chan_sip.c.cache.diff uploaded by rjain (license 226)
      chan_sip.memalloc uploaded by rjain (license 226)
      chan_sip.memalloc.bugfix uploaded by rjain (license 226)

      Patches tracked in team/group/sip_session_timers, with some additional fixes
      by russell and oej.

Tested by: jtodd, rjain, loloski

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBlocked revisions 98973 via svnmerge
Joshua Colp [Wed, 16 Jan 2008 20:36:21 +0000 (20:36 +0000)]
Blocked revisions 98973 via svnmerge

........
r98973 | file | 2008-01-16 16:34:30 -0400 (Wed, 16 Jan 2008) | 2 lines

Bump up cleancount due to previous commit that changed the channel structure.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98975 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBlocked revisions 98972 via svnmerge
Joshua Colp [Wed, 16 Jan 2008 20:35:49 +0000 (20:35 +0000)]
Blocked revisions 98972 via svnmerge

........
r98972 | file | 2008-01-16 16:33:47 -0400 (Wed, 16 Jan 2008) | 2 lines

Replace current spy architecture with backport of audiohooks. This should take care of current known spy issues.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoPartially revert r93898, because it broke the way netsnmp was being detected.
Jason Parker [Wed, 16 Jan 2008 19:41:25 +0000 (19:41 +0000)]
Partially revert r93898, because it broke the way netsnmp was being detected.

rizzo, do you want to discuss so we can rethink this, or do you have another way?

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98971 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd note about new update.log to CHANGES, by request of jmls and further prodding...
Jason Parker [Wed, 16 Jan 2008 18:34:19 +0000 (18:34 +0000)]
Add note about new update.log to CHANGES, by request of jmls and further prodding by jsmith.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98969 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd logging for 'make update' command (also fixes updates in some places). Issue...
Jason Parker [Wed, 16 Jan 2008 18:06:14 +0000 (18:06 +0000)]
Add logging for 'make update' command (also fixes updates in some places).  Issue #11766, initial patch by jmls.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98968 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98966 via svnmerge from
Joshua Colp [Wed, 16 Jan 2008 17:51:52 +0000 (17:51 +0000)]
Merged revisions 98966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 lines

Add missing NULLs at end of two ast_load_realtimes.
(closes issue #11769)
Reported by: tequ
Patches:
      chaniax.patch uploaded by dimas (license 88)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98967 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98964 via svnmerge from
Mark Michelson [Wed, 16 Jan 2008 17:21:49 +0000 (17:21 +0000)]
Merged revisions 98964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines

Fix a deadlock in chan_local in local_hangup. There was contention because
the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.

(closes issue #11730)
Reported by: UDI-Doug
Patches:
      11730.patch uploaded by putnopvut (license 60)
  Tested by: UDI-Doug

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98965 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMake users list static
Terry Wilson [Wed, 16 Jan 2008 16:06:06 +0000 (16:06 +0000)]
Make users list static

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98962 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98960 via svnmerge from
Joshua Colp [Wed, 16 Jan 2008 15:09:37 +0000 (15:09 +0000)]
Merged revisions 98960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 lines

Introduce a lock into the dialing API that protects it when destroying the structure.
(closes issue #11687)
Reported by: callguy
Patches:
      11687.diff uploaded by file (license 11)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98961 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98958 via svnmerge from
Joshua Colp [Wed, 16 Jan 2008 15:04:08 +0000 (15:04 +0000)]
Merged revisions 98958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 lines

Add two more SDP names for ulaw and alaw.
(closes issue #11777)
Reported by: tootai

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98959 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98955 via svnmerge from
Joshua Colp [Wed, 16 Jan 2008 03:09:11 +0000 (03:09 +0000)]
Merged revisions 98955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines

Don't drop the old record route information when dealing with packets related to a reinvite.
(closes issue #11545)
Reported by: kebl0155
Patches:
      reinvite-patch.txt uploaded by kebl0155 (license 356)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98956 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoRemove DNS lookup from sip_devicestate. This seems to come from way back when and...
Joshua Colp [Wed, 16 Jan 2008 02:30:13 +0000 (02:30 +0000)]
Remove DNS lookup from sip_devicestate. This seems to come from way back when and I can't think of a reason for it being here, plus it could cause needless DNS lookups.
(closes issue #10983)
Reported by: jtodd

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98954 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoTerry found this problem with running the expr2 parser on OSX. Make the #defines...
Steve Murphy [Wed, 16 Jan 2008 01:35:10 +0000 (01:35 +0000)]
Terry found this problem with running the expr2 parser on OSX. Make the #defines come out the same between the parser & lexer.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98951 via svnmerge from
Joshua Colp [Wed, 16 Jan 2008 01:17:25 +0000 (01:17 +0000)]
Merged revisions 98951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines

Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex.
(closes issue #11693)
Reported by: yzg

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98952 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98946 via svnmerge from
Russell Bryant [Tue, 15 Jan 2008 23:53:28 +0000 (23:53 +0000)]
Merged revisions 98946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines

Change a buffer in check_auth() to be a thread local dynamically allocated
buffer, instead of a massive buffer on the stack.  This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.

On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless.  BUFSIZ is a system specific define.  On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB.  We don't even support SIP messages
larger than 4 kB!  Further usage of this define should be avoided, unless it
is used in the proper context.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd the "filter" keyword
Tilghman Lesher [Tue, 15 Jan 2008 23:52:11 +0000 (23:52 +0000)]
Add the "filter" keyword

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98947 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoClean up something I did for ABI compatability in 1.4
Russell Bryant [Tue, 15 Jan 2008 23:35:29 +0000 (23:35 +0000)]
Clean up something I did for ABI compatability in 1.4

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98945 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98943 via svnmerge from
Russell Bryant [Tue, 15 Jan 2008 23:31:53 +0000 (23:31 +0000)]
Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines

Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.

The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed.  Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code.  The reason this
happens is that the channel might get masqueraded during this time.  During a
masquerade, existing translation paths get destroyed.

So, this patch fixes the issue in an API and ABI compatible way.  (This one is
 for you, paravoid!)

It changes an int in ast_frame to be used as flag bits.  The 1 bit is still used
to indicate that the frame contains timing information.  Also, a second flag has
been added to indicate that the frame came from a translator.  When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed.  At this point, the flag gets
cleared.  Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.

Admittedly, this feels like a hack.  But, it does fix the issue, and I was not able
to think of a better solution ...

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98934 via svnmerge from
Joshua Colp [Tue, 15 Jan 2008 20:10:20 +0000 (20:10 +0000)]
Merged revisions 98934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 lines

Based on the boundary found move over the correct amount.
(closes issue #11750)
Reported by: tasker

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98935 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98894 via svnmerge from
Joshua Colp [Mon, 14 Jan 2008 22:44:20 +0000 (22:44 +0000)]
Merged revisions 98894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 lines

Accept "; boundary=" not just ";boundary=" in the multipart mixed content type.
(closes issue #11750)
Reported by: tasker

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98895 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd backupdeleted option to app_voicemail
Jason Parker [Mon, 14 Jan 2008 22:19:40 +0000 (22:19 +0000)]
Add backupdeleted option to app_voicemail

(closes issue #10740)
Reported by: ruffle
Patches:
      app_voicemail.diff uploaded by ruffle (license 201)
      10740-voicemail.diff uploaded by qwell (license 4)
      20080113_bug10740.diff.txt uploaded by mvanbaak (license 7)
Tested by: blitzrage, mvanbaak, qwell

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBig improvement for app_directory. This patch breaks the do_directory function up
Mark Michelson [Mon, 14 Jan 2008 22:11:50 +0000 (22:11 +0000)]
Big improvement for app_directory. This patch breaks the do_directory function up
so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote
dimas from the original bug description:

"app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences.

1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be.
2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa).
3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message.
4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list.
5. Alot of duplicated code as already mentioned."

This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen
in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is
well worth it.

Huge thanks to dimas for this wonderful submission.

(closes issue #11744)
Reported by: dimas
Patches:
      dir3.patch uploaded by dimas (license 88)
  Tested by: putnopvut, dimas

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98888 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBlocked revisions 98849 via svnmerge
Mark Michelson [Mon, 14 Jan 2008 21:01:56 +0000 (21:01 +0000)]
Blocked revisions 98849 via svnmerge

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r98849 | mmichelson | 2008-01-14 14:59:26 -0600 (Mon, 14 Jan 2008) | 4 lines

Adding in appropriate unlocks for the locks I added. Thanks to joetester on IRC
for pointing this out.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98850 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMake sure the user's manager secret exists, even if it is blank.
Joshua Colp [Mon, 14 Jan 2008 20:01:19 +0000 (20:01 +0000)]
Make sure the user's manager secret exists, even if it is blank.
(closes issue #11749)
Reported by: srt

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd description of TOUPPER and TOLOWER dialplan functions to CHANGES.
Terry Wilson [Mon, 14 Jan 2008 18:42:16 +0000 (18:42 +0000)]
Add description of TOUPPER and TOLOWER dialplan functions to CHANGES.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98811 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd proper call forwarding (all and busy) support for chan_skinny.
Jason Parker [Mon, 14 Jan 2008 17:40:36 +0000 (17:40 +0000)]
Add proper call forwarding (all and busy) support for chan_skinny.
Note: NoAnswer support is currently not implemented, as it would take a
 significant amount of work to figure out how to do correctly.

Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98774 via svnmerge from
Russell Bryant [Mon, 14 Jan 2008 17:39:31 +0000 (17:39 +0000)]
Merged revisions 98774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines

Revert a change that introduces an unacceptable performance hit and is causing
memory leaks ... (from rev 97973)

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoFix for potential crash with vmexten
Jason Parker [Mon, 14 Jan 2008 17:18:35 +0000 (17:18 +0000)]
Fix for potential crash with vmexten

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98773 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98737 via svnmerge from
Mark Michelson [Mon, 14 Jan 2008 16:36:41 +0000 (16:36 +0000)]
Merged revisions 98737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan 2008) | 3 lines

Fixing another compilation error. I'm a bit off today :(

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98738 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBlocked revisions 98734 via svnmerge
Mark Michelson [Mon, 14 Jan 2008 16:34:00 +0000 (16:34 +0000)]
Blocked revisions 98734 via svnmerge

........
r98734 | mmichelson | 2008-01-14 10:30:33 -0600 (Mon, 14 Jan 2008) | 3 lines

Oops. Last commit had compilation error.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98736 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98733 via svnmerge from
Mark Michelson [Mon, 14 Jan 2008 16:33:06 +0000 (16:33 +0000)]
Merged revisions 98733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines

Adding explicit defaults for missing options to init_queue. This is necessary because
if a user either removes or comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the value from prior to the
reload.

Thanks to John Bigelow for pointing this error out to me.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoPrint out a warning when spaces are used in the variable name in Set and MSet. It...
Joshua Colp [Mon, 14 Jan 2008 15:07:30 +0000 (15:07 +0000)]
Print out a warning when spaces are used in the variable name in Set and MSet. It is extremely hard to debug this issue so this should make it easier.
(closes issue #11759)
Reported by: caio1982
Patches:
      setvar_space_warning1.diff uploaded by caio1982 (license 22)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98714 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoUpdate documentation.
Joshua Colp [Mon, 14 Jan 2008 14:33:17 +0000 (14:33 +0000)]
Update documentation.
(closes issue #11763)
Reported by: IgorG
Patches:
      docupd.v1.diff uploaded by IgorG (license 20)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd another small option for the JACK app and JACK_HOOK function. The 'n'
Russell Bryant [Mon, 14 Jan 2008 04:53:08 +0000 (04:53 +0000)]
Add another small option for the JACK app and JACK_HOOK function.  The 'n'
option tells JACK not to start jackd automatically if it is not already
running.  Otherwise, the default is that jackd will get started for you if
it isn't running already.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98676 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years ago- Break up the Misc. section a bit with a new section for Misc. New Modules
Russell Bryant [Sun, 13 Jan 2008 23:43:06 +0000 (23:43 +0000)]
- Break up the Misc. section a bit with a new section for Misc. New Modules
- Change spacing a bit in some places for consistent indentation

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98656 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoBring in the code from team/russell/jack/.
Russell Bryant [Sun, 13 Jan 2008 19:19:57 +0000 (19:19 +0000)]
Bring in the code from team/russell/jack/.

Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function.  Both
interfaces create an input and output JACK port.  The application makes
these ports the endpoint of the call.  The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
audiohook on the channel.  This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio.  This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.

In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/).  I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd configure script check for JACK.
Russell Bryant [Sun, 13 Jan 2008 18:08:50 +0000 (18:08 +0000)]
Add configure script check for JACK.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98605 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoRemove KDE configure script check that isn't used
Russell Bryant [Sun, 13 Jan 2008 18:01:56 +0000 (18:01 +0000)]
Remove KDE configure script check that isn't used

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoRemove a duplicate lock of the audiohook lock when destroying manipulate
Russell Bryant [Sun, 13 Jan 2008 00:10:00 +0000 (00:10 +0000)]
Remove a duplicate lock of the audiohook lock when destroying manipulate
audiohooks.  This causes an error when we attempt to destroy the lock later
when freeing the audiohook.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd a new CLI command, "core set chanvar", which allows you to set a channel
Russell Bryant [Sat, 12 Jan 2008 19:34:38 +0000 (19:34 +0000)]
Add a new CLI command, "core set chanvar", which allows you to set a channel
variable (or function) on an active channel from the CLI.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98558 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoConversion to load manager.conf into memory did not convert the password
Tilghman Lesher [Sat, 12 Jan 2008 18:12:56 +0000 (18:12 +0000)]
Conversion to load manager.conf into memory did not convert the password
functions correctly.  (Closes issue #11749)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98536 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agomerging a comment added in 1.4
Pari Nannapaneni [Sat, 12 Jan 2008 05:13:04 +0000 (05:13 +0000)]
merging a comment added in 1.4

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98514 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets...
Kevin P. Fleming [Sat, 12 Jan 2008 00:20:55 +0000 (00:20 +0000)]
Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets generated.

(closes issue #11212)
Reported by: tzafrir
Patches:
      zap_dnd.diff uploaded by tzafrir (modified by me) (license 46)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98488 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98467 via svnmerge from
Tilghman Lesher [Sat, 12 Jan 2008 00:17:26 +0000 (00:17 +0000)]
Merged revisions 98467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines

Add a connection timeout attribute, as that was what was intended with the
login timeout, but ODBC divides it up into 2 different timeouts.
(Closes issue #11745)

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98487 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd some extra checking to help out with a potential error when trying to
Russell Bryant [Fri, 11 Jan 2008 23:57:01 +0000 (23:57 +0000)]
Add some extra checking to help out with a potential error when trying to
run "make asterisk.pdf" when not all of the right packages are installed.

(closes issue #10763)
Reported by: Corydon76
Patches:
      20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98454 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd 'auto' signalling mode for Zaptel channels.
Kevin P. Fleming [Fri, 11 Jan 2008 23:10:57 +0000 (23:10 +0000)]
Add 'auto' signalling mode for Zaptel channels.
(closes issue #11690)
Reported by: tzafrir
Patches:
      signaling_to_signalling.diff uploaded by tzafrir (license 46)
      signalling_cleanup.diff uploaded by tzafrir (license 46)
      zap_auto_default.diff uploaded by tzafrir (license 46)
      zap_no_default_sig.diff uploaded by tzafrir (license 46)
      zap_signal_auto.diff uploaded by tzafrir (license 46)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoGoodbye again drumkilla.
Joshua Colp [Fri, 11 Jan 2008 23:09:31 +0000 (23:09 +0000)]
Goodbye again drumkilla.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98435 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agodrumkilla ftw.
Joshua Colp [Fri, 11 Jan 2008 23:00:21 +0000 (23:00 +0000)]
drumkilla ftw.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98434 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoI am no longer Rockin'
Joshua Colp [Fri, 11 Jan 2008 22:59:13 +0000 (22:59 +0000)]
I am no longer Rockin'

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoTesting something...
Joshua Colp [Fri, 11 Jan 2008 22:57:39 +0000 (22:57 +0000)]
Testing something...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98390 via svnmerge from
Russell Bryant [Fri, 11 Jan 2008 22:52:41 +0000 (22:52 +0000)]
Merged revisions 98390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | 9 lines

Fix up setting the EID on BSD based systems.

(closes issue #11646)
Reported by: caio1982
Patches:
      dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
      dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22)
Tested by: caio1982, mvanbaak

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98400 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 98325 via svnmerge from
Joshua Colp [Fri, 11 Jan 2008 19:53:01 +0000 (19:53 +0000)]
Merged revisions 98325 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines

If the incoming RTP stream changes codec force the bridge to break if the other side does not support it.
(closes issue #11729)
Reported by: tsearle
Patches:
      new_codec_patch_udiff.patch uploaded by tsearle (license 373)

........

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12 years agoMerged revisions 98317 via svnmerge from
Joshua Colp [Fri, 11 Jan 2008 19:30:34 +0000 (19:30 +0000)]
Merged revisions 98317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines

If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else.
(closes issue #11743)
Reported by: davevg
Patches:
      res_agi.diff uploaded by davevg (license 209)

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12 years agoMerged revisions 98315 via svnmerge from
Mark Michelson [Fri, 11 Jan 2008 19:12:05 +0000 (19:12 +0000)]
Merged revisions 98315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines

Properly report the hangup cause as no answer when someone does not answer

(closes issue #10574, reported by boch, patched by moy)

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12 years agoKevin noted that the thing that I _actually_ changed here was that I converted
Russell Bryant [Fri, 11 Jan 2008 19:05:24 +0000 (19:05 +0000)]
Kevin noted that the thing that I _actually_ changed here was that I converted
a value from a double, to a float, back to a double.  Sure enough, when I changed
my interim variable back to a double, it still blows up.  Switching all of these
to a float fixes the problem.  This seems like a compiler bug where a double passed
as an argument isn't getting properly aligned, so I'll have to see if I can replicate
it with a small test program.

(related to issue #11725)

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12 years agoFix a bus error that happened when asterisk was built with optimizations on
Russell Bryant [Fri, 11 Jan 2008 18:48:07 +0000 (18:48 +0000)]
Fix a bus error that happened when asterisk was built with optimizations on
with platforms that explode on unaligned access.  I'm not exactly sure why
this fixes it, but it fixed it on the machine I was testing on.  If it makes
sense to you, feel free to enlighten me.  :)

(closes issue #11725, patched by me)

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12 years agoPort Nick Gorham's timestamp patch to adaptive_odbc, too
Tilghman Lesher [Fri, 11 Jan 2008 18:35:30 +0000 (18:35 +0000)]
Port Nick Gorham's timestamp patch to adaptive_odbc, too

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12 years agoCommit Nick Gorham's suggestion for timestamp fix
Tilghman Lesher [Fri, 11 Jan 2008 18:30:00 +0000 (18:30 +0000)]
Commit Nick Gorham's suggestion for timestamp fix

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12 years agoBlocked revisions 98265 via svnmerge
Russell Bryant [Fri, 11 Jan 2008 18:26:32 +0000 (18:26 +0000)]
Blocked revisions 98265 via svnmerge

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r98265 | russell | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines

Backport the ability to set the ToS bits on Linux when not running as root.
Normally, we would not backport features into 1.4, but, I was convinced by the
justification supplied by the supplier of this patch.  He pointed out that this
patch removes a requirement for running as root, thus reducing the potential
impacts of security issues.

(closes issue #11742)
Reported by: paravoid
Patches:
      libcap.diff uploaded by paravoid (license 200)

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12 years agoMerged revisions 98219 via svnmerge from
Joshua Colp [Fri, 11 Jan 2008 17:27:58 +0000 (17:27 +0000)]
Merged revisions 98219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines

Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely

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12 years agoAt one point during working on this module, I had the lin/lin16 versions of the
Russell Bryant [Fri, 11 Jan 2008 17:17:54 +0000 (17:17 +0000)]
At one point during working on this module, I had the lin/lin16 versions of the
framein callbacks different.  However, they are now the same again, so remove
the duplicate code and use the same functions for the lin/lin16 versions.

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12 years agoMerged revisions 98164 via svnmerge from
Tilghman Lesher [Fri, 11 Jan 2008 16:08:43 +0000 (16:08 +0000)]
Merged revisions 98164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines

Back out changes from revision 97077, since it wasn't perfect

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12 years agoDocumentation updates
Tilghman Lesher [Fri, 11 Jan 2008 15:12:33 +0000 (15:12 +0000)]
Documentation updates

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98152 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAscom phones send Flash events as SIP INFO using '!' as the 'digit'
Kevin P. Fleming [Fri, 11 Jan 2008 12:51:21 +0000 (12:51 +0000)]
Ascom phones send Flash events as SIP INFO using '!' as the 'digit'

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12 years agoBlocked revisions 98082 via svnmerge
Russell Bryant [Fri, 11 Jan 2008 03:40:58 +0000 (03:40 +0000)]
Blocked revisions 98082 via svnmerge

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r98082 | russell | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines

Fix samples vs. length calculations for g722

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12 years ago - Fix the last set of places where incorrect assumptions were made about the
Russell Bryant [Fri, 11 Jan 2008 03:37:19 +0000 (03:37 +0000)]
 - Fix the last set of places where incorrect assumptions were made about the
   sample length with g722.  It is _2_ samples per byte, not 1.  This was all
   over the place, and I believed it, and it is what caused me to take so long
   to figure out what was broken.
 - Update copyright information on codec_g722.

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12 years agoFix "core show translation" to not output information for "unknown" codecs.
Mark Michelson [Fri, 11 Jan 2008 00:54:54 +0000 (00:54 +0000)]
Fix "core show translation" to not output information for "unknown" codecs.
This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.

(closes issue #11722, reported and initially patched by caio1982, final patch by me)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98047 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoAdd a new global and per-peer option to chan_sip, qualifyfreq, which allows you
Russell Bryant [Fri, 11 Jan 2008 00:38:23 +0000 (00:38 +0000)]
Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you
to set the qualify frequency.

(closes issue #11597)
Reported by: wilder
Patches:
      qualifyfreq5.patch uploaded by wilder (license 362)
   -- with some mods by me

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12 years agoBlocked revisions 98025 via svnmerge
Russell Bryant [Fri, 11 Jan 2008 00:15:58 +0000 (00:15 +0000)]
Blocked revisions 98025 via svnmerge

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r98025 | russell | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines

Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.

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12 years agoSimplify this code with a suggestion from Luigi on the asterisk-dev list.
Russell Bryant [Fri, 11 Jan 2008 00:12:22 +0000 (00:12 +0000)]
Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.

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12 years agoMerged revisions 97973 via svnmerge from
Tilghman Lesher [Thu, 10 Jan 2008 23:40:13 +0000 (23:40 +0000)]
Merged revisions 97973 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines

1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running.  Don't just ignore it.
(Closes issue #11698)

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12 years agoMerged revisions 97976 via svnmerge from
Russell Bryant [Thu, 10 Jan 2008 23:33:24 +0000 (23:33 +0000)]
Merged revisions 97976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines

Fix various timing calculations that made assumptions that the audio being
processed was at a sample rate of 8 kHz.

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12 years agoFix various issues in codec_g722.
Russell Bryant [Thu, 10 Jan 2008 23:16:09 +0000 (23:16 +0000)]
Fix various issues in codec_g722.
 - The most common fix being made here is to fix all of the places where the
   number of output samples and output bytes gets updated in the translator
   state structure.
 - Fix a number of other places where the number of samples provided as an
   initialization value to a struct was incorrect.

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12 years agoFix the buffer_samples value. For signed linear, the number of samples needed
Russell Bryant [Thu, 10 Jan 2008 23:10:00 +0000 (23:10 +0000)]
Fix the buffer_samples value.  For signed linear, the number of samples needed
to fill the buffer is half the buffer size.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoMerged revisions 97925 via svnmerge from
Mark Michelson [Thu, 10 Jan 2008 21:58:47 +0000 (21:58 +0000)]
Merged revisions 97925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan 2008) | 6 lines

Let us leave a voicemail for ourself if we have logged into VoiceMailMain and chosen
to leave a message.

(closes issue #11735, reported and patched by jamessan)

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12 years agoMerged revisions 97889 via svnmerge from
Steve Murphy [Thu, 10 Jan 2008 21:46:56 +0000 (21:46 +0000)]
Merged revisions 97889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line

Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro.
........

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12 years agoMerged revisions 97849 via svnmerge from
Steve Murphy [Thu, 10 Jan 2008 20:45:05 +0000 (20:45 +0000)]
Merged revisions 97849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line

This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least.
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12 years agoMerged revisions 97847 via svnmerge from
Jason Parker [Thu, 10 Jan 2008 20:13:24 +0000 (20:13 +0000)]
Merged revisions 97847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan 2008) | 1 line

Fix a comment that is no longer true.
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12 years agoUse the appropriate line ending for the X-Asterisk-VM-Message-Type header.
Mark Michelson [Thu, 10 Jan 2008 20:05:43 +0000 (20:05 +0000)]
Use the appropriate line ending for the X-Asterisk-VM-Message-Type header.

(closes issue #11734, reported and patched by jaroth)

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12 years agoheh, remove patch to generated file.
Terry Wilson [Thu, 10 Jan 2008 19:07:36 +0000 (19:07 +0000)]
heh, remove patch to generated file.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97826 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoCheck pointers before freeing (was getting WARNINGS under MALLOC_DEBUG)
Terry Wilson [Thu, 10 Jan 2008 19:03:04 +0000 (19:03 +0000)]
Check pointers before freeing (was getting WARNINGS under MALLOC_DEBUG)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97825 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agoFix problem with timestr going out of scope
Tilghman Lesher [Thu, 10 Jan 2008 17:38:36 +0000 (17:38 +0000)]
Fix problem with timestr going out of scope
(Closes issue #11726, closes issue #11731)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97805 65c4cc65-6c06-0410-ace0-fbb531ad65f3

12 years agominor formatting changes
Russell Bryant [Thu, 10 Jan 2008 17:30:24 +0000 (17:30 +0000)]
minor formatting changes

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