]> git.ipfire.org Git - thirdparty/asterisk.git/log
thirdparty/asterisk.git
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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117507 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117135 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@117086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116466 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116088 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115341 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115333 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115327 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115320 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115285 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115276 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@115196 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114848 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114823 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114708 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114649 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

17 years agoStore the manager session ID explicitly as 4 byte ID instead of a ulong. The
Russell Bryant [Wed, 23 Apr 2008 17:55:31 +0000 (17:55 +0000)] 
Store the manager session ID explicitly as 4 byte ID instead of a ulong.  The
mansession_id cookie is coded to be limited to 8 characters of hex, and this
could break logins from 64-bit machines in some cases.
(inspired by AST-20)

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17 years agoFix find_callno_locked() to actually return the callno locked in some more cases.
Russell Bryant [Wed, 23 Apr 2008 17:16:32 +0000 (17:16 +0000)] 
Fix find_callno_locked() to actually return the callno locked in some more cases.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAdd 502 support for both directions, not only one... (see r114571)
Olle Johansson [Wed, 23 Apr 2008 16:51:41 +0000 (16:51 +0000)] 
Add 502 support for both directions, not only one...  (see r114571)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoInstead of stopping dialplan execution when SayNumber attempts to say a large number...
Joshua Colp [Wed, 23 Apr 2008 14:54:11 +0000 (14:54 +0000)] 
Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on.
(closes issue #12502)
Reported by: bcnit

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17 years agoTreat a 502 just like a 503, when it comes to processing a response code
Tilghman Lesher [Tue, 22 Apr 2008 23:51:44 +0000 (23:51 +0000)] 
Treat a 502 just like a 503, when it comes to processing a response code

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114571 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoWhen we receive a full frame that is supposed to contain our call number,
Russell Bryant [Tue, 22 Apr 2008 22:15:36 +0000 (22:15 +0000)] 
When we receive a full frame that is supposed to contain our call number,
ensure that it has the correct one.
(closes issue #10078)
(AST-2008-006)

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17 years agoI thought I was going to be able to leave 1.4 alone, but that was not the case.
Russell Bryant [Tue, 22 Apr 2008 21:14:55 +0000 (21:14 +0000)] 
I thought I was going to be able to leave 1.4 alone, but that was not the case.
I ran into some problems with G.722 in 1.4, so I have merged in all of the fixes
in this area that I have made in trunk/1.6.0, and things are happy again.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114550 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoTrivial change to read the number of samples from a frame before calling ast_write()
Russell Bryant [Tue, 22 Apr 2008 19:45:00 +0000 (19:45 +0000)] 
Trivial change to read the number of samples from a frame before calling ast_write()

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114545 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoAfter a parked call times out, allow the call back to the parker to time out.
Russell Bryant [Tue, 22 Apr 2008 18:29:56 +0000 (18:29 +0000)] 
After a parked call times out, allow the call back to the parker to time out.
(closes issue #10890)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoIf the dial string passed to the call channel callback does not indicate an
Russell Bryant [Tue, 22 Apr 2008 18:03:33 +0000 (18:03 +0000)] 
If the dial string passed to the call channel callback does not indicate an
extension, then consider the extension on the channel before falling back
to the default.

(closes issue #12479)
Reported by: darren1713
Patches:
      exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoMerge changes from team/russell/issue_9520
Russell Bryant [Tue, 22 Apr 2008 15:20:37 +0000 (15:20 +0000)] 
Merge changes from team/russell/issue_9520

These changes make sure that the reference count for sip_peer objects properly
reflects the fact that the peer is sitting in the scheduler for a scheduled
callback for qualifying peers or for expiring registrations.  Without this, it
was possible for these callbacks to happen at the same time that the peer was
being destroyed.  This was especially likely to happen with realtime peers, and
for people making use of the realtime prune CLI command.

(closes issue #9520)
Reported by: kryptolus
Committed patch by me

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoOnly drop audio if we receive it without a progress indication. We allow other frames...
Joshua Colp [Mon, 21 Apr 2008 14:39:32 +0000 (14:39 +0000)] 
Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call.
(closes issue #12440)
Reported by: aragon

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114322 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoEnsure that help text terminates with a newline
Tilghman Lesher [Sat, 19 Apr 2008 13:57:17 +0000 (13:57 +0000)] 
Ensure that help text terminates with a newline

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoMOH usage information needs a terminating newline, or else
Tilghman Lesher [Sat, 19 Apr 2008 13:49:50 +0000 (13:49 +0000)] 
MOH usage information needs a terminating newline, or else
"asterisk -rx 'help moh reload'" will hang.  Reported via
-dev list, fixed by me.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoDon't destroy a manager session if poll() returns an error of EAGAIN.
Russell Bryant [Fri, 18 Apr 2008 21:48:06 +0000 (21:48 +0000)] 
Don't destroy a manager session if poll() returns an error of EAGAIN.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoensure directories are created before we try to install stuff into them
Russell Bryant [Fri, 18 Apr 2008 20:01:09 +0000 (20:01 +0000)] 
ensure directories are created before we try to install stuff into them

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoSUBDIRS_INSTALL is already listed as a subtarget for bininstall
Russell Bryant [Fri, 18 Apr 2008 19:58:55 +0000 (19:58 +0000)] 
SUBDIRS_INSTALL is already listed as a subtarget for bininstall

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114275 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoClearing up error messages so they make a bit more sense. Also removing a redundant...
Mark Michelson [Fri, 18 Apr 2008 17:44:29 +0000 (17:44 +0000)] 
Clearing up error messages so they make a bit more sense. Also removing a redundant error
message.

Issue AST-15

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17 years agoEnsure that we don't ast_strdupa(NULL)
Russell Bryant [Fri, 18 Apr 2008 15:24:09 +0000 (15:24 +0000)] 
Ensure that we don't ast_strdupa(NULL)
(closes issue #12476)
Reported by: davidw
Patch by me

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114248 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoOnly complete the SIP channel name once for 'sip show channel <channel>'
Sean Bright [Fri, 18 Apr 2008 13:33:32 +0000 (13:33 +0000)] 
Only complete the SIP channel name once for 'sip show channel <channel>'

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114245 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoFor consistency sake, ensure that the values that ${CALLINGPRES} returns are valid...
Tilghman Lesher [Fri, 18 Apr 2008 06:49:16 +0000 (06:49 +0000)] 
For consistency sake, ensure that the values that ${CALLINGPRES} returns are valid as an
input to SetCallingPres.  (Closes issue #12472)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114242 65c4cc65-6c06-0410-ace0-fbb531ad65f3

17 years agoRemove redundant safety net. The check for the autoservice channel list state
Russell Bryant [Thu, 17 Apr 2008 22:15:43 +0000 (22:15 +0000)] 
Remove redundant safety net.  The check for the autoservice channel list state
accomplishes the same goal in a better way.

(issue #12470)
Reported By: atis

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