]> git.ipfire.org Git - thirdparty/asterisk.git/log
thirdparty/asterisk.git
17 years agoCall waiting tone occurs too often, because it's getting serviced by both
Tilghman Lesher [Thu, 29 May 2008 20:24:11 +0000 (20:24 +0000)] 
Call waiting tone occurs too often, because it's getting serviced by both
subchannels.
(closes issue #11354)
 Reported by: cahen
 Patches:
       20080512__bug11354.diff.txt uploaded by Corydon76 (license 14)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@119071 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years ago - Fix a typo in the argument to Playtones
Russell Bryant [Thu, 29 May 2008 19:04:52 +0000 (19:04 +0000)] 
 - Fix a typo in the argument to Playtones
 - use ast_safe_sleep() instead of calling the wait application
(thanks to tilghman for pointing these out!)

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17 years agoMerged revisions 119008 via svnmerge from
Russell Bryant [Thu, 29 May 2008 18:49:12 +0000 (18:49 +0000)] 
Merged revisions 119008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines

Merge changes from team/russell/iax2-another-fix-to-the-fix

As described in the following post to the asterisk-dev mailing list, only
enforce destination call numbers when processing an ACK.

http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html

(closes issue #12631)
........

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17 years ago - Mark app_milliwatt dependent on res_indications (thanks to jsmith)
Russell Bryant [Thu, 29 May 2008 17:51:29 +0000 (17:51 +0000)] 
 - Mark app_milliwatt dependent on res_indications (thanks to jsmith)
 - fix a typo in a log message (thanks to qwell)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118961 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoChange milliwatt to use the proper tone by default (1004 Hz) instead of 1000 Hz.
Russell Bryant [Thu, 29 May 2008 17:38:38 +0000 (17:38 +0000)] 
Change milliwatt to use the proper tone by default (1004 Hz) instead of 1000 Hz.
An option is there to use 1000 Hz for anyone that might want it.

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17 years agoDefine also when not DEBUG_THREADS
Tilghman Lesher [Thu, 29 May 2008 17:33:01 +0000 (17:33 +0000)] 
Define also when not DEBUG_THREADS

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17 years agoAdd some debugging code that ensures that when we do deadlock avoidance, we
Tilghman Lesher [Thu, 29 May 2008 17:20:16 +0000 (17:20 +0000)] 
Add some debugging code that ensures that when we do deadlock avoidance, we
don't lose the information about how a lock was originally acquired.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years ago(closes issue #10668)
Steve Murphy [Thu, 29 May 2008 00:25:28 +0000 (00:25 +0000)] 
(closes issue #10668)
(closes issue #11721)
(closes issue #12726)
Reported by: arkadia
Tested by: murf

These changes:

1. revert the changes made via bug 10668;
   I should have known that such changes,
   even tho they made sense at the time,
   seemed like an omission, etc, were actually
   integral to the CDR system via forkCDR.
   It makes sense to me now that forkCDR didn't
   natively end any CDR's, but rather depended
   on natively closing them all at hangup time
   via traversing and closing them all, whether
   locked or not. I still don't completely
   understand the benefits of setvar and answer
   operating on locked cdrs, but I've seen
   enough to revert those changes also, and
   stop messing up users who depended on that
   behavior. bug 12726 found reverting the changes
   fixed his changes, and after a long review
   and working on forkCDR, I can see why.
2. Apply the suggested enhancements proposed
   in 10668, but in a completely compatible
   way. ForkCDR will behave exactly as before,
   but now has new options that will allow some
   actions to be taken that will slightly
   modify the outcome and side-effects of
   forkCDR. Based on conversations I've had
   with various people, these small tweaks
   will allow some users to get the behavior
   they need. For instance, users executing
   forkCDR in an AGI script will find the
   answer time set, and DISPOSITION set,
   a situation not covered when the routines
  were first written.
3. A small problem in the cdr serializer
   would output answer and end times even
   when they were not set. This is now
   fixed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118858 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agomerge revision 118702 from trunk to 1.4 -- Fixes a bug in chan_iax that uses send_com...
Brett Bryant [Wed, 28 May 2008 16:10:48 +0000 (16:10 +0000)] 
merge revision 118702 from trunk to 1.4 -- Fixes a bug in chan_iax that uses send_command to poke a peer while a channel is unlocked in some cases, and because it can cause seemingly random failures could be related to some bugs in the tracker...

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118716 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAdd an option to use the source IP address of RTP as the destination IP address of...
Joshua Colp [Wed, 28 May 2008 14:23:34 +0000 (14:23 +0000)] 
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann

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17 years agoFix an issue where codec preferences were not set on dialogs that were not authentica...
Joshua Colp [Tue, 27 May 2008 19:32:38 +0000 (19:32 +0000)] 
Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP.
(closes issue #12501)
Reported by: slimey

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17 years agoWhen showing an error message for a command, don't shorten the command output,
Tilghman Lesher [Tue, 27 May 2008 19:15:27 +0000 (19:15 +0000)] 
When showing an error message for a command, don't shorten the command output,
as it tends to confuse the user (it's fine for suggesting other commands,
however).
 Reported by: seanbright (on #asterisk-dev)
 Fixed by: me

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17 years agoRussell noted to me that in the case that separate threads use their
Mark Michelson [Tue, 27 May 2008 19:07:26 +0000 (19:07 +0000)] 
Russell noted to me that in the case that separate threads use their
own addressing system, the fix I made for issue 12376 does not guarantee
uniqueness to the datastores' uids. Though I know of no system that works
this way, I am going to change this right now to prevent trying to track
down some future bug that may occur and cause untold hours of debugging
time to track down.

The change involves using a global counter which increases with each new
chanspy_ds which is created. This guarantees uniqueness.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118509 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoNULL character should terminate only commands back to the core, not log
Tilghman Lesher [Tue, 27 May 2008 18:58:09 +0000 (18:58 +0000)] 
NULL character should terminate only commands back to the core, not log
messages to the console.
(closes issue #12731)
 Reported by: seanbright
 Patches:
       20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
 Tested by: seanbright

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118465 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agosmall update to the g() option of app_voicemail to
Michiel van Baak [Tue, 27 May 2008 17:17:20 +0000 (17:17 +0000)] 
small update to the g() option of app_voicemail to
note that gain changes only work on zap channels right now.

issue #12578 shows it's not clear right now.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118416 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAdd a unique id to the datastore allocated in app_chanspy since
Mark Michelson [Tue, 27 May 2008 16:38:38 +0000 (16:38 +0000)] 
Add a unique id to the datastore allocated in app_chanspy since
it is possible that multiple spies may be listening to the same
channel.

(closes issue #12376)
Reported by: DougUDI
Patches:
      12376_chanspy_uid.diff uploaded by putnopvut (license 60)
Tested by: destiny6628

(closes issue #12243)
Reported by: atis

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAdd a note that pbx_config.so is needed for Local channels.
Tilghman Lesher [Tue, 27 May 2008 15:45:37 +0000 (15:45 +0000)] 
Add a note that pbx_config.so is needed for Local channels.
(Closes issue #12671)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118358 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoRealtime flag affects construction in multiple ways, so consulting whether
Tilghman Lesher [Sun, 25 May 2008 16:02:04 +0000 (16:02 +0000)] 
Realtime flag affects construction in multiple ways, so consulting whether
rtcachefriends was set was done too soon (needed to be done inside build_peer,
not just as a flag to build_peer).
Also, fullcontact needed to be reconstructed, because realtime separates the
embedded ';' into multiple fields.
(closes issue #12722)
 Reported by: barthpbx
 Patches:
       20080525__bug12722.diff.txt uploaded by Corydon76 (license 14)
 Tested by: barthpbx
 (Much of the discussion happened on #asterisk-dev for diagnosing this issue)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118251 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoFix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf
Jeff Peeler [Fri, 23 May 2008 21:21:35 +0000 (21:21 +0000)] 
Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118163 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAdd format type checking for recently de-inlined function
Tilghman Lesher [Fri, 23 May 2008 13:18:44 +0000 (13:18 +0000)] 
Add format type checking for recently de-inlined function

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17 years agoAdd information on using the Asterisk console, including tab command line
Tilghman Lesher [Fri, 23 May 2008 12:59:16 +0000 (12:59 +0000)] 
Add information on using the Asterisk console, including tab command line
completion.  (Closes issue #12681)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118052 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoDon't declare a function that takes variable arguments as inline, because it's
Russell Bryant [Fri, 23 May 2008 12:30:53 +0000 (12:30 +0000)] 
Don't declare a function that takes variable arguments as inline, because it's
not valid, and on some compilers, will emit a warning.

http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline

(closes issue #12289)
Reported by: francesco_r
Patches by Tilghman, final patch by me

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17 years agoAlso remove preamble from asynchronous events (reported by jsmith on #asterisk-dev)
Tilghman Lesher [Thu, 22 May 2008 18:53:53 +0000 (18:53 +0000)] 
Also remove preamble from asynchronous events (reported by jsmith on #asterisk-dev)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117899 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoTake into account the length of delimiters when calculating result string length.
Tilghman Lesher [Thu, 22 May 2008 16:47:03 +0000 (16:47 +0000)] 
Take into account the length of delimiters when calculating result string length.
(closes issue #12696)
 Reported by: adomjan
 Patches:
       func_realtime.c-longdelimiter.patch uploaded by adomjan (license 487)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117809 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoEnsure that passed in zt_chan_conf structure is not modified in mkintf.
Jeff Peeler [Wed, 21 May 2008 20:11:14 +0000 (20:11 +0000)] 
Ensure that passed in zt_chan_conf structure is not modified in mkintf.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117582 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoApply the autoframing setting to dialogs that do not get matched against a user or...
Joshua Colp [Wed, 21 May 2008 19:38:28 +0000 (19:38 +0000)] 
Apply the autoframing setting to dialogs that do not get matched against a user or peer.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoRevert accidental commit of the last change
Tilghman Lesher [Wed, 21 May 2008 18:44:53 +0000 (18:44 +0000)] 
Revert accidental commit of the last change

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17 years agoStrip the preamble from the output also when -rx is not being used
Tilghman Lesher [Wed, 21 May 2008 18:40:14 +0000 (18:40 +0000)] 
Strip the preamble from the output also when -rx is not being used
(Related to issue #12702)

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17 years agoDon't filter the magic character in the network verboser. It gets filtered
Russell Bryant [Wed, 21 May 2008 18:28:46 +0000 (18:28 +0000)] 
Don't filter the magic character in the network verboser.  It gets filtered
once it reaches the client.
(related to issue #12702, pointed out by tilghman)

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17 years ago1) Don't print the verbose marker in front of every message from ast_verbose()
Russell Bryant [Wed, 21 May 2008 18:19:34 +0000 (18:19 +0000)] 
1) Don't print the verbose marker in front of every message from ast_verbose()
   being sent to remote consoles.

2) Fix pbx_gtkconsole to filter out the verbose marker.

(related to issue #12702)

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17 years agoDon't display the verbose marker for calls to ast_verbose() that do not include
Russell Bryant [Wed, 21 May 2008 18:11:51 +0000 (18:11 +0000)] 
Don't display the verbose marker for calls to ast_verbose() that do not include
a VERBOSE_PREFIX in front of the message.
(closes issue #12702)
Reported by: johnlange
Patched by me

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17 years agoPass a pointer for the conf parameter to the function mkintf rather than the whole...
Jeff Peeler [Wed, 21 May 2008 16:58:40 +0000 (16:58 +0000)] 
Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoUse the right pthread lock and condition when waiting.
Joshua Colp [Mon, 19 May 2008 16:50:52 +0000 (16:50 +0000)] 
Use the right pthread lock and condition when waiting.
(closes issue #12664)
Reported by: tomo1657
Patches:
      res_smdi.c.patch uploaded by tomo1657 (license 484)

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17 years agoThe addition of usleep(2) within ast_assert requires the inclusion of the unistd...
Tilghman Lesher [Mon, 19 May 2008 16:05:05 +0000 (16:05 +0000)] 
The addition of usleep(2) within ast_assert requires the inclusion of the unistd.h header

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17 years agoMake chan_h323 work with pwlib 1.12.0
Joshua Colp [Mon, 19 May 2008 15:22:10 +0000 (15:22 +0000)] 
Make chan_h323 work with pwlib 1.12.0
(closes issue #12682)
Reported by: bamby
Patches:
      pwlib_nopipe.diff uploaded by bamby (license 430)

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17 years agoAvoid access of uninitialized memory. This caused a bunch of crashes for me
Russell Bryant [Mon, 19 May 2008 03:44:04 +0000 (03:44 +0000)] 
Avoid access of uninitialized memory.  This caused a bunch of crashes for me
while doing load testing of development branch where I'm working on some
performance improvements.

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17 years agoCheck to make sure an RTP structure exists before calling ast_rtp_new_source on it.
Joshua Colp [Fri, 16 May 2008 20:28:11 +0000 (20:28 +0000)] 
Check to make sure an RTP structure exists before calling ast_rtp_new_source on it.
(closes issue #12669)
Reported by: sbisker

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17 years agoAvoid zombies when the channel exits before the AGI.
Tilghman Lesher [Wed, 14 May 2008 21:38:09 +0000 (21:38 +0000)] 
Avoid zombies when the channel exits before the AGI.
(closes issue #12648)
 Reported by: gkloepfer
 Patches:
       20080514__bug12648.diff.txt uploaded by Corydon76 (license 14)
 Tested by: gkloepfer

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17 years agoAdd ast_assert(), which can be used to handle fatal errors. It is only compiled
Russell Bryant [Wed, 14 May 2008 21:32:00 +0000 (21:32 +0000)] 
Add ast_assert(), which can be used to handle fatal errors.  It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)

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17 years agoDocument exitcontext in app_voicemail sample config
Jason Parker [Wed, 14 May 2008 20:43:08 +0000 (20:43 +0000)] 
Document exitcontext in app_voicemail sample config

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116409 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAdd linux-gnueabi in.
Joshua Colp [Wed, 14 May 2008 18:53:39 +0000 (18:53 +0000)] 
Add linux-gnueabi in.
(closes issue #12529)
Reported by: tzafrir

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17 years agoDetect another way for a connection to have gone away.
Tilghman Lesher [Wed, 14 May 2008 16:46:48 +0000 (16:46 +0000)] 
Detect another way for a connection to have gone away.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116296 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAccept text messages even with
Olle Johansson [Wed, 14 May 2008 12:51:06 +0000 (12:51 +0000)] 
Accept text messages even with
Content-Type: text/plain;charset=Södermanländska

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17 years agoA change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
Mark Michelson [Tue, 13 May 2008 23:47:49 +0000 (23:47 +0000)] 
A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.

After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS
is enabled in menuselect, the actual origin of channel locks is obscured
by the fact that all channel locks appear to happen in the function
ast_channel_lock(). This code change redefines ast_channel_lock to be a
macro which maps to __ast_channel_lock(), which then relays the proper
file name, line number, and function name information to the core lock
functions so that this information will be displayed in the case that
there is some sort of locking error or core show locks is issued.

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17 years agoFix a deadlock involving channel autoservice and chan_local that was debugged
Russell Bryant [Tue, 13 May 2008 21:17:23 +0000 (21:17 +0000)] 
Fix a deadlock involving channel autoservice and chan_local that was debugged
and fixed by mmichelson and me.

We observed a system that had a bunch of threads stuck in ast_autoservice_stop().
The reason these threads were waiting around is because this function waits to
ensure that the channel list in the autoservice thread gets rebuilt before the
stop() function returns.  However, the autoservice thread was also locked, so
the autoservice channel list was never getting rebuilt.

The autoservice thread was stuck waiting for the channel lock on a local channel.
However, the local channel was locked by a thread that was stuck in the autoservice
stop function.

It turned out that the issue came down to the local_queue_frame() function in
chan_local.  This function assumed that one of the channels passed in as an
argument was locked when called.  However, that was not always the case.  There
were multiple cases in which this channel was not locked when the function was
called.  We fixed up chan_local to indicate to this function whether this channel
was locked or not.  The previous assumption had caused local_queue_frame() to
improperly return with the channel locked, where it would then never get unlocked.

(closes issue #12584)
(related to issue #12603)

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17 years agoFix an issue that I noticed in autoservice while mmichelson and I were debugging
Russell Bryant [Tue, 13 May 2008 21:05:57 +0000 (21:05 +0000)] 
Fix an issue that I noticed in autoservice while mmichelson and I were debugging
a different problem.  I noticed that it was theoretically possible for two threads
to attempt to start the autoservice thread at the same time.  This change makes the
process of starting the autoservice thread, thread-safe.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115990 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoUse the right flag to open the audio in non-blocking.
Joshua Colp [Tue, 13 May 2008 20:28:23 +0000 (20:28 +0000)] 
Use the right flag to open the audio in non-blocking.
(closes issue #12616)
Reported by: nicklewisdigiumuser

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17 years agoIf the socket dies (read returns 0=EOF), return immediately.
Tilghman Lesher [Tue, 13 May 2008 18:36:13 +0000 (18:36 +0000)] 
If the socket dies (read returns 0=EOF), return immediately.
(Closes issue #12637)

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17 years agoIf a thread holds no locks, do not print any information on the thread when issuing
Mark Michelson [Mon, 12 May 2008 17:51:14 +0000 (17:51 +0000)] 
If a thread holds no locks, do not print any information on the thread when issuing
a core show locks command. This will help to de-clutter output somewhat.

Russell said it would be fine to place this improvement in the 1.4 branch, so that's
why it's going here too.

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17 years agoImprove res_ninit and res_ndestroy autoconf logic on the Darwin platform.
Joshua Colp [Fri, 9 May 2008 16:34:08 +0000 (16:34 +0000)] 
Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform.

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17 years agoRemove debug output.
Russell Bryant [Thu, 8 May 2008 19:19:50 +0000 (19:19 +0000)] 
Remove debug output.

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17 years agoMerged revisions 115564 via svnmerge from
Russell Bryant [Thu, 8 May 2008 19:15:25 +0000 (19:15 +0000)] 
Merged revisions 115564 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines

Fix a race condition that bbryant just found while doing some IAX2 testing.
He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes,
however, the audio was extremely choppy.  We looked at a packet trace and saw
a storm of INVAL and VNAK frames being sent from one box to another.

It turned out that what had happened was that one box tried to send a CONTROL
frame before the 3 way handshake had completed.  So, that frame did not include
the destination call number, because it didn't have it yet.  Part of our recent
work for security issues included an additional check to ensure that frames that
are supposed to include the destination call number have the correct one.  This
caused the frame to be rejected with an INVAL.  The frame would get retransmitted
for forever, rejected every time ...

This race condition exists in all versions that got the security changes,
in theory.  However, it is really only likely that this would cause a problem in
Asterisk trunk.  There was a control frame being sent (SRCUPDATE) at the _very_
beginning of the call, which does not exist in 1.2 or 1.4.  However, I am fixing
all versions that could potentially be affected by the introduced race condition.

These changes are what bbryant and I came up with to fix the issue.  Instead of
simply dropping control frames that get sent before the handshake is complete,
the code attempts to wait a little while, since in most cases, the handshake
will complete very quickly.  If it doesn't complete after yielding for a little
while, then the frame gets dropped.

........

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17 years agoDon't give up on attempting an outbound registration if we receive a 408 Timeout.
Russell Bryant [Thu, 8 May 2008 16:11:33 +0000 (16:11 +0000)] 
Don't give up on attempting an outbound registration if we receive a 408 Timeout.
(closes issue #12323)

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17 years agoremove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well
Russell Bryant [Thu, 8 May 2008 15:37:49 +0000 (15:37 +0000)] 
remove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well
(closes issue #9676)

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17 years agoDon't exit the script if Asterisk is not running.
Russell Bryant [Thu, 8 May 2008 15:32:08 +0000 (15:32 +0000)] 
Don't exit the script if Asterisk is not running.
(closes issue #12611)

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17 years agoDon't use a channel before checking for channel allocation failure.
Russell Bryant [Thu, 8 May 2008 15:24:54 +0000 (15:24 +0000)] 
Don't use a channel before checking for channel allocation failure.
(closes issue #12609)
Reported by: edantie

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17 years agoUse the same method for executing Asterisk as the rest of the script.
Russell Bryant [Thu, 8 May 2008 14:40:53 +0000 (14:40 +0000)] 
Use the same method for executing Asterisk as the rest of the script.
(closes issue #12611)
Reported by: b_plessis

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17 years agoTrack peer references when stored in the sip_pvt struct as the peer related to
Russell Bryant [Wed, 7 May 2008 18:17:19 +0000 (18:17 +0000)] 
Track peer references when stored in the sip_pvt struct as the peer related to
a qualify ping or a subscription.  This fixes some realtime related crashes.
(closes issue #12588)
(closes issue #12555)

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17 years agoMerged revisions 115511 via svnmerge from
Russell Bryant [Wed, 7 May 2008 16:24:09 +0000 (16:24 +0000)] 
Merged revisions 115511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines

Remove remnants of dlinkedlists.  I didn't actually use them in the final version
of my IAX2 improvements.

........

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17 years agoMerged revisions 115421 via svnmerge from
Jason Parker [Tue, 6 May 2008 19:55:29 +0000 (19:55 +0000)] 
Merged revisions 115421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | 7 lines

read requires an argument on some non-bash shells

(closes issue #12593)
Reported by: bkruse
Patches:
      getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)

........

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17 years agoSwitch to using ast_random() rather than just rand().
Jason Parker [Tue, 6 May 2008 19:34:58 +0000 (19:34 +0000)] 
Switch to using ast_random() rather than just rand().
This does not fix the bug reported, but I believe it is correct.

(from issue #12446)
Patches:
      bug_12446.diff uploaded by snuffy (license 35)

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17 years agoDon't print the terminating NUL. (Closes issue #12589)
Tilghman Lesher [Tue, 6 May 2008 19:31:39 +0000 (19:31 +0000)] 
Don't print the terminating NUL.  (Closes issue #12589)

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17 years agoAdd in missing argument.
Joshua Colp [Tue, 6 May 2008 13:54:15 +0000 (13:54 +0000)] 
Add in missing argument.

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17 years agoSeparate verbose output from CLI output, by using a preamble.
Tilghman Lesher [Mon, 5 May 2008 22:50:31 +0000 (22:50 +0000)] 
Separate verbose output from CLI output, by using a preamble.
(closes issue #12402)
 Reported by: Corydon76
 Patches:
       20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 (license 14)
       20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by Corydon76 (license 14)

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17 years agoMake sure that either the main speex library contains preprocess functions or that...
Joshua Colp [Mon, 5 May 2008 22:10:05 +0000 (22:10 +0000)] 
Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built.

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17 years agoDon't consider a caller "handled" until the caller is bridged with
Mark Michelson [Mon, 5 May 2008 21:41:34 +0000 (21:41 +0000)] 
Don't consider a caller "handled" until the caller is bridged with
a queue member. There was too much of an opportunity for the member
to hang up (either during a delay, announcement, or overly long
agi) between the time that he answered the phone and the time when
he actually was bridged with the caller. The consequence of this
was that if the member hung up in that interval, then proper
abandonment details would not be noted in the queue log if the caller
were to hang up at any point after the member hangup.

(closes issue #12561)
Reported by: ablackthorn

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17 years agoReverse order, such that user configs override default selections
Tilghman Lesher [Mon, 5 May 2008 20:17:55 +0000 (20:17 +0000)] 
Reverse order, such that user configs override default selections

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115312 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoErr, the documentation on the return value of ast_odbc_backslash_is_escape is exactly...
Tilghman Lesher [Mon, 5 May 2008 19:55:55 +0000 (19:55 +0000)] 
Err, the documentation on the return value of ast_odbc_backslash_is_escape is exactly backwards.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115308 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAvoid putting opaque="" in Digest authentication. This patch came from switchvox.
Russell Bryant [Mon, 5 May 2008 19:49:25 +0000 (19:49 +0000)] 
Avoid putting opaque="" in Digest authentication.  This patch came from switchvox.
It fixes authentication with Primus in Canada, and has been in use for a very long
time without causing problems with any other providers.
(closes issue AST-36)

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17 years agoBlocked revisions 115296 via svnmerge
Russell Bryant [Mon, 5 May 2008 17:53:54 +0000 (17:53 +0000)] 
Blocked revisions 115296 via svnmerge

........
r115296 | russell | 2008-05-05 12:53:26 -0500 (Mon, 05 May 2008) | 28 lines

Merge changes from team/russell/iax2_find_callno_1.2

These changes address a critical performance issue introduced in the latest
release.  The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers.  However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls.  On a small embedded platform, it would not be
able to handle a single call.  On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels.  Ouch.

These changes address some performance issues of the find_callno() function
that have bothered me for a very long time.  On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call.  This involved a mutex lock and unlock for each call number
checked.  So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks.  Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.

A second container for IAX2 pvt structs has been added.  It is an astobj2
hash table.  When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number.  Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.

In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.

........

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17 years agoWhen starting Asterisk, bug out if Asterisk is already running.
Tilghman Lesher [Mon, 5 May 2008 03:22:25 +0000 (03:22 +0000)] 
When starting Asterisk, bug out if Asterisk is already running.
(closes issue #12525)
 Reported by: explidous
 Patches:
       20080428__bug12525.diff.txt uploaded by Corydon76 (license 14)
 Tested by: mvanbaak

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17 years agoExpand the test function for GCC attributes so that more complex attributes are prope...
Joshua Colp [Sun, 4 May 2008 02:09:44 +0000 (02:09 +0000)] 
Expand the test function for GCC attributes so that more complex attributes are properly recognized.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoFor my next trick I will make these work with what our autoconf header file gives us.
Joshua Colp [Sun, 4 May 2008 01:50:59 +0000 (01:50 +0000)] 
For my next trick I will make these work with what our autoconf header file gives us.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115279 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoTreat warnings as errors when checking if a GCC attribute exists. We have to do this...
Joshua Colp [Sun, 4 May 2008 01:43:26 +0000 (01:43 +0000)] 
Treat warnings as errors when checking if a GCC attribute exists. We have to do this as GCC will just ignore the attribute and pop up a warning, it won't actually fail to compile.

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17 years agoAdd new "pri show version" command to show the libpri version for support reasons.
Brett Bryant [Fri, 2 May 2008 20:25:42 +0000 (20:25 +0000)] 
Add new "pri show version" command to show the libpri version for support reasons.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoClarify a comment that was, well, just wrong. It turns out that
Mark Michelson [Fri, 2 May 2008 14:28:19 +0000 (14:28 +0000)] 
Clarify a comment that was, well, just wrong. It turns out that
ignoring the way that macros expand. Instead, I have clarified in the
comment why the macro will work even if the scheduler id for the
task to be deleted changes during the execution of the macro.

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17 years agoChange the comment of deprecated to an actual compiler deprecation
Tilghman Lesher [Thu, 1 May 2008 23:20:25 +0000 (23:20 +0000)] 
Change the comment of deprecated to an actual compiler deprecation

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years ago'#' is another reserved character for URIs that also needs to be escaped.
Tilghman Lesher [Thu, 1 May 2008 18:59:08 +0000 (18:59 +0000)] 
'#' is another reserved character for URIs that also needs to be escaped.
(closes issue #10543)
 Reported by: blitzrage
 Patches:
       20080418__bug10543.diff.txt uploaded by Corydon76 (license 14)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoMerge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4
Russell Bryant [Wed, 30 Apr 2008 16:30:01 +0000 (16:30 +0000)] 
Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4

These changes address a critical performance issue introduced in the latest
release.  The fix for the latest security issue included a change that made
Asterisk randomly choose call numbers to make them more difficult to guess by
attackers.  However, due to some inefficient (this is by far, an understatement)
code, when Asterisk chose high call numbers, chan_iax2 became unusable after
just a small number of calls.  On a small embedded platform, it would not be
able to handle a single call.  On my Intel Core 2 Duo @ 2.33 GHz, I couldn't
run more than about 16 IAX2 channels.  Ouch.

These changes address some performance issues of the find_callno() function
that have bothered me for a very long time.  On every incoming media frame,
it iterated through every possible call number trying to find a matching
active call.  This involved a mutex lock and unlock for each call number
checked.  So, if the random call number chosen was 20000, then every media
frame would cause 20000 locks and unlocks.  Previously, this problem was
not as obvious since Asterisk always chose the lowest call number it could.

A second container for IAX2 pvt structs has been added.  It is an astobj2
hash table.  When we know the remote side's call number, the pvt goes into
the hash table with a hash value of the remote side's call number.  Then,
lookups for incoming media frames are a very fast hash lookup instead of an
absolutely insane array traversal.

In a quick test, I was able to get more than 3600% more IAX2 channels
on my machine with these changes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoDon't crash on bad SIP replys.
Olle Johansson [Wed, 30 Apr 2008 16:23:17 +0000 (16:23 +0000)] 
Don't crash on bad SIP replys.
Fix created in Huntsville together with Mark M (putnopvut)

(closes issue #12363)
Reported by: jvandal
Tested by: putnopvut, oej

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114890 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agouse the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size...
Kevin P. Fleming [Wed, 30 Apr 2008 14:46:57 +0000 (14:46 +0000)] 
use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined

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17 years agopay attention to *all* header files for dependency tracking, not just the local ones...
Kevin P. Fleming [Wed, 30 Apr 2008 12:14:07 +0000 (12:14 +0000)] 
pay attention to *all* header files for dependency tracking, not just the local ones (inspired by r578 of asterisk-addons by tilghman)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114875 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoUse the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's...
Mark Michelson [Tue, 29 Apr 2008 19:40:06 +0000 (19:40 +0000)] 
Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext
and macroexten fields. This is needed because if macros are daisy-chained, the incorrect
context and extension are placed on the new channel. I also added locking to the channel prior
to accessing these variables as noted in trunk's janitor project file.

(closes issue #12549)
Reported by: darren1713
Patches:
      app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
       (with modifications from me)
Tested by: putnopvut

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17 years agoChange warning message to debug, since there are cases where 0 results is perfectly...
Jason Parker [Tue, 29 Apr 2008 17:08:55 +0000 (17:08 +0000)] 
Change warning message to debug, since there are cases where 0 results is perfectly fine.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114829 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoMerged revisions 114822 via svnmerge from
Kevin P. Fleming [Tue, 29 Apr 2008 12:53:12 +0000 (12:53 +0000)] 
Merged revisions 114822 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines

stop script from appending source code if run multiple times

........

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17 years agoWhen modules are embedded, they take on a different name, without the ".so"
Tilghman Lesher [Mon, 28 Apr 2008 04:47:39 +0000 (04:47 +0000)] 
When modules are embedded, they take on a different name, without the ".so"
extension.  Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)

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17 years agoWhen we don't explicitly pass a path to the --with-tds configure option, we
Sean Bright [Sun, 27 Apr 2008 01:26:15 +0000 (01:26 +0000)] 
When we don't explicitly pass a path to the --with-tds configure option, we
may end up finding tds.h in /usr/local/include instead of /usr/include.  If
this happens, the grep that looks for the version (from tdsver.h) will fail
and we'll have some problems during the build.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoClicking forward without selecting a message leaves an errant .lock file.
Tilghman Lesher [Sat, 26 Apr 2008 13:15:21 +0000 (13:15 +0000)] 
Clicking forward without selecting a message leaves an errant .lock file.
(closes issue #12528)
 Reported by: pukepail
 Patches:
       patch.diff uploaded by pukepail (license 431)

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17 years agoUse consistent logic for checking to see if a call number has been chosen yet.
Russell Bryant [Fri, 25 Apr 2008 21:54:40 +0000 (21:54 +0000)] 
Use consistent logic for checking to see if a call number has been chosen yet.
Also, remove some redundant logic I recently added in a fix.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114673 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoMove the unlock of the spyee channel to outside the start_spying() function so that
Mark Michelson [Fri, 25 Apr 2008 19:32:02 +0000 (19:32 +0000)] 
Move the unlock of the spyee channel to outside the start_spying() function so that
the channel is not unlocked twice when using whisper mode.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114662 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoReference documentation files that actually exist.
Tilghman Lesher [Fri, 25 Apr 2008 15:53:52 +0000 (15:53 +0000)] 
Reference documentation files that actually exist.
(closes issue #12516)
 Reported by: linuxmaniac
 Patches:
       diff_rev114611.patch uploaded by linuxmaniac (license 472)

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17 years agoRe-invite RTP during a masquerade so that, for instance, an AMI
Mark Michelson [Thu, 24 Apr 2008 21:35:08 +0000 (21:35 +0000)] 
Re-invite RTP during a masquerade so that, for instance, an AMI
redirect of two channels which are natively bridged will preserve audio
on both channels. This prevents a problem with Asterisk not re-inviting
due to one of the channels having being a zombie.

(closes issue #12513)
Reported by: mneuhauser
Patches:
      asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114632 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoOutput of channel variables when eventwhencalled=vars was set
Mark Michelson [Thu, 24 Apr 2008 20:43:03 +0000 (20:43 +0000)] 
Output of channel variables when eventwhencalled=vars was set
was being truncated two characters. This patch corrects the
problem.

(closes issue #12493)
Reported by: davidw

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoResolve a deadlock in chan_local by releasing the channel lock
Mark Michelson [Thu, 24 Apr 2008 20:04:24 +0000 (20:04 +0000)] 
Resolve a deadlock in chan_local by releasing the channel lock
temporarily.

(closes issue #11712)
Reported by: callguy
Patches:
      11712.patch uploaded by putnopvut (license 60)
Tested by: acunningham

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoEnsure that when we set the accountcode, it actually shows up in the CDR.
Tilghman Lesher [Thu, 24 Apr 2008 19:53:36 +0000 (19:53 +0000)] 
Ensure that when we set the accountcode, it actually shows up in the CDR.
(Fix for AMI Originate)
(Closes issue #12007)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114621 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoFix a silly mistake in a change I made yesterday that caused chan_iax2 to blow
Russell Bryant [Thu, 24 Apr 2008 15:55:21 +0000 (15:55 +0000)] 
Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow
up very quickly.
(issue #12515)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoOnly have one max-forwards header in outbound REFERs.
Olle Johansson [Thu, 24 Apr 2008 14:55:18 +0000 (14:55 +0000)] 
Only have one max-forwards header in outbound REFERs.
Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe!

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoImprove some broken cookie parsing code. Previously, manager login over HTTP
Russell Bryant [Wed, 23 Apr 2008 22:18:12 +0000 (22:18 +0000)] 
Improve some broken cookie parsing code.  Previously, manager login over HTTP
would only work if the mansession_id cookie was first.  Now, the code builds
a list of all of the cookies in the Cookie header.  This fixes a problem
observed by users of the Asterisk GUI.
(closes AST-20)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoFix an issue that caused getting the correct next channel to not always work.
Russell Bryant [Wed, 23 Apr 2008 20:49:18 +0000 (20:49 +0000)] 
Fix an issue that caused getting the correct next channel to not always work.
Also, remove setting the amount of time to wait for a digit from 5 seconds back
down to 1/10 of a second.  I believe this was so the beep didn't get played over
and over really fast, but a while back I put in another fix for that issue.

(closes issue #12498)
Reported by: jsmith
Patches:
      app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114597 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoFix reload/unload for res_musiconhold module.
Jason Parker [Wed, 23 Apr 2008 18:28:44 +0000 (18:28 +0000)] 
Fix reload/unload for res_musiconhold module.

(closes issue #11575)
Reported by: sunder
Patches:
      M11575_14_rev3.diff uploaded by junky (license 177)
      bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114594 65c4cc65-6c06-0410-ace0-fbb531ad65f3