asterisk/asterisk.git
11 years agoAdd lastms to the require API call.
Tilghman Lesher [Wed, 8 Apr 2009 05:06:22 +0000 (05:06 +0000)]
Add lastms to the require API call.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186899 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186841 via svnmerge from
Mark Michelson [Wed, 8 Apr 2009 00:09:28 +0000 (00:09 +0000)]
Merged revisions 186841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines

  Fix a few typos of the word "frequency."

  (closes issue #14842)
  Reported by: jvandal
  Patches:
        frequency-typo.diff uploaded by jvandal (license 413)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186842 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix bad merge from fix for issue 13867.
Mark Michelson [Wed, 8 Apr 2009 00:01:49 +0000 (00:01 +0000)]
Fix bad merge from fix for issue 13867.

(closes issue #14686)
Reported by: davidw

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186837 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186832 via svnmerge from
Mark Michelson [Tue, 7 Apr 2009 23:50:56 +0000 (23:50 +0000)]
Merged revisions 186832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines

  Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY.

  Without this flag set, warning sounds will not be properly played to either party
  of the bridge.

  (closes issue #14845)
  Reported by: adomjan
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186833 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186775 via svnmerge from
Tilghman Lesher [Tue, 7 Apr 2009 22:23:46 +0000 (22:23 +0000)]
Merged revisions 186775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines

  Fix Macro documentation to match current (and intended) behavior.
  (See -dev mailing list)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186799 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186719 via svnmerge from
Mark Michelson [Tue, 7 Apr 2009 20:46:18 +0000 (20:46 +0000)]
Merged revisions 186719 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines

  Ensure that \r\n is printed after the ActionID in an OriginateResponse.

  (closes issue #14847)
  Reported by: kobaz
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186720 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a log message getting output when it should not have been.
Joshua Colp [Mon, 6 Apr 2009 23:11:13 +0000 (23:11 +0000)]
Fix a log message getting output when it should not have been.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186687 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix problem when authenticating a non-RTP dialog.
Joshua Colp [Mon, 6 Apr 2009 17:03:07 +0000 (17:03 +0000)]
Fix problem when authenticating a non-RTP dialog.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186653 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd support for changing the outbound codec on a SIP call using
Joshua Colp [Mon, 6 Apr 2009 16:15:30 +0000 (16:15 +0000)]
Add support for changing the outbound codec on a SIP call using
a dialplan variable.

This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.

(closes issue #13243)
Reported by: samdell3
Patches:
      13243.diff uploaded by file (license 11)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186624 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoSilly svn. These files didn't get merged over in the merge of the issue8824 branch.
Mark Michelson [Mon, 6 Apr 2009 16:06:25 +0000 (16:06 +0000)]
Silly svn. These files didn't get merged over in the merge of the issue8824 branch.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 186565 via svnmerge
Mark Michelson [Mon, 6 Apr 2009 13:57:39 +0000 (13:57 +0000)]
Blocked revisions 186565 via svnmerge

........
  r186565 | mmichelson | 2009-04-06 08:54:41 -0500 (Mon, 06 Apr 2009) | 3 lines

  Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoPass the correct value to sizeof when copying address information.
Joshua Colp [Mon, 6 Apr 2009 13:23:12 +0000 (13:23 +0000)]
Pass the correct value to sizeof when copying address information.

(issue #14827)
Reported by: pj
Patches:
      14827.diff uploaded by file (license 11)
Tested by: pj

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove merged branch properties accidentally merged to trunk.
Richard Mudgett [Sat, 4 Apr 2009 00:13:50 +0000 (00:13 +0000)]
Remove merged branch properties accidentally merged to trunk.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoThis commit introduces COLP/CONP and Redirecting party information into Asterisk.
Mark Michelson [Fri, 3 Apr 2009 22:41:46 +0000 (22:41 +0000)]
This commit introduces COLP/CONP and Redirecting party information into Asterisk.

The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186458 via svnmerge from
Kevin P. Fleming [Fri, 3 Apr 2009 20:20:01 +0000 (20:20 +0000)]
Merged revisions 186458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines

  Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested

  Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later).
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186461 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186445 via svnmerge from
Tilghman Lesher [Fri, 3 Apr 2009 19:59:55 +0000 (19:59 +0000)]
Merged revisions 186445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines

  Found a conflict in the last commit, due to multiple targets
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186447 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186415 via svnmerge from
Tilghman Lesher [Fri, 3 Apr 2009 19:30:34 +0000 (19:30 +0000)]
Merged revisions 186415 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines

  Distinguish in a sent email between simple sends and forwards.
  (closes issue #11678)
   Reported by: jamessan
   Patches:
         20090330__bug11678.diff.txt uploaded by tilghman (license 14)
   Tested by: tilghman, lmadsen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186444 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd better support for relaying success or failure of the ast_transfer() API call.
Joshua Colp [Fri, 3 Apr 2009 16:47:27 +0000 (16:47 +0000)]
Add better support for relaying success or failure of the ast_transfer() API call.

This API call now waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to truly convey whether
the transfer worked or not. In the case of the Transfer() dialplan application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.

(closes issue #12713)
Reported by: davidw
Tested by: file

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186382 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoaudio_audiohook_write_list() did not correctly update sample size after ast_translate.
David Vossel [Fri, 3 Apr 2009 16:29:47 +0000 (16:29 +0000)]
audio_audiohook_write_list() did not correctly update sample size after ast_translate.

audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz.  the sample size is now updated after translating to reflect this possibility.  This caused the audio on the receiving end to sound terrible.  Thanks to jcolp and mmichelson for helping me work this out.

(issue AST-197)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186379 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186320 via svnmerge from
Joshua Colp [Fri, 3 Apr 2009 15:52:50 +0000 (15:52 +0000)]
Merged revisions 186320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines

  Fix a problem with the crypto variable definitions not actually being defined properly.

  (closes issue #14804)
  Reported by: jvandal
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186321 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoCompatibility fix for glibc 2.4
Tilghman Lesher [Fri, 3 Apr 2009 15:18:28 +0000 (15:18 +0000)]
Compatibility fix for glibc 2.4
(Closes issue #14820)
Reported by: phsultan

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix the ability to retrieve voicemail messages from IMAP.
Mark Michelson [Fri, 3 Apr 2009 14:32:05 +0000 (14:32 +0000)]
Fix the ability to retrieve voicemail messages from IMAP.

A recent change made interactive vm_states no longer get
added to the list of vm_states and instead get stored in
thread-local storage.

In trunk and all the 1.6.X branches, the problem is that
when we search for messages in a voicemail box, we would
attempt to update the appropriate vm_state struct by directly
searching in the list of vm_states instead of using the
get_vm_state_by_imap_user function. This meant we could not
find the interactive vm_state that we wanted.

(closes issue #14685)
Reported by: BlargMaN
Patches:
      14685.patch uploaded by mmichelson (license 60)
Tested by: BlargMaN, qualleyiv, mmichelson

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186286 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186229 via svnmerge from
Russell Bryant [Fri, 3 Apr 2009 02:03:48 +0000 (02:03 +0000)]
Merged revisions 186229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines

Fix a memory leak in cdr_radius.

I came across this while doing some testing of my ast_channel_ao2 branch.
After running a test overnight that generated over 5 million calls, Asterisk
had taken up about 1 GB of my system memory.  So, I re-ran the test with
MALLOC_DEBUG turned on.  However, it showed no leaks in Asterisk during the
test, even though Asterisk was still consuming it somehow.

Instead, I turned to valgrind, which when run with --leak-check=full, told
me exactly where the leak came from, which was from allocations inside the
radiusclient-ng library.  This explains why MALLOC_DEBUG did not report it.

After a bit of analysis, I found that we were leaking a little bit of memory
every time a CDR record was passed to cdr_radius.

I don't actually have a radius server set up to receive CDR records.  However,
I always have my development systems compile and install all modules.  In
addition to making sure there are not build errors across modules, always
loading modules helps find bugs like this, too, so it is strongly recommend for
all developers.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186230 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186174 via svnmerge from
Mark Michelson [Thu, 2 Apr 2009 21:56:21 +0000 (21:56 +0000)]
Merged revisions 186174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines

  Fix instructions in one-step parking comment to make more sense.

  Changed a capital K to a lowercase k.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186175 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186081 via svnmerge from
Kevin P. Fleming [Thu, 2 Apr 2009 17:26:07 +0000 (17:26 +0000)]
Merged revisions 186081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines

  ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186101 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerge in the RTP engine API.
Joshua Colp [Thu, 2 Apr 2009 17:20:52 +0000 (17:20 +0000)]
Merge in the RTP engine API.

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 186059 via svnmerge from
Tilghman Lesher [Thu, 2 Apr 2009 17:10:28 +0000 (17:10 +0000)]
Merged revisions 186059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
  r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines

  Merged revisions 186056 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.2

  ........
    r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines

    Fix for AST-2009-003
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 186057 via svnmerge
Tilghman Lesher [Thu, 2 Apr 2009 17:04:40 +0000 (17:04 +0000)]
Blocked revisions 186057 via svnmerge

........
  r186057 | tilghman | 2009-04-02 12:03:59 -0500 (Thu, 02 Apr 2009) | 2 lines

  Avoid multiple warning messages in SIP, due to this column not existing
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186058 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMissed a common case for needing to extend the buffer.
Tilghman Lesher [Thu, 2 Apr 2009 15:14:22 +0000 (15:14 +0000)]
Missed a common case for needing to extend the buffer.
(closes issue #14716)
 Reported by: sum
 Patches:
       20090402__bug14716.diff.txt uploaded by tilghman (license 14)
 Tested by: sum

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186021 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 185952 via svnmerge from
Kevin P. Fleming [Thu, 2 Apr 2009 13:51:44 +0000 (13:51 +0000)]
Merged revisions 185952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines

  the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior.

  this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerge changes from str_substitution that are unrelated to that branch.
Tilghman Lesher [Wed, 1 Apr 2009 20:13:28 +0000 (20:13 +0000)]
Merge changes from str_substitution that are unrelated to that branch.
Included is a small bugfix to an ast_str helper, but most of these changes
are simply doxygen fixes.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185912 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 185845 via svnmerge from
David Vossel [Wed, 1 Apr 2009 19:03:32 +0000 (19:03 +0000)]
Merged revisions 185845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines

  Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491

  Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno.  Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries.  Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct.  When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite.  In this case, it is in response to the glare invite and must be dealt with or the call is dropped.  I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261.  Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well.

  (closes issue #12013)
  Reported by: alx

  Review: http://reviewboard.digium.com/r/213/
........

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11 years agoAddress Russell's comments regarding rev 185704.
Mark Michelson [Wed, 1 Apr 2009 13:59:34 +0000 (13:59 +0000)]
Address Russell's comments regarding rev 185704.

Use ast_debug and ast_softhangup_nolock.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 185771 via svnmerge from
Russell Bryant [Wed, 1 Apr 2009 13:48:26 +0000 (13:48 +0000)]
Merged revisions 185771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines

Fix a case where DTMF could bypass audiohooks.

This change fixes a situation where an audiohook that wants DTMF would not
actually get it.  This is in the code path where we end DTMF digit length
emulation while handling a NULL frame.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185772 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix dev-mode build on my box.
Russell Bryant [Wed, 1 Apr 2009 12:13:16 +0000 (12:13 +0000)]
Fix dev-mode build on my box.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185741 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAllow the AMI Hangup command to accept a Cause header.
Mark Michelson [Wed, 1 Apr 2009 00:39:01 +0000 (00:39 +0000)]
Allow the AMI Hangup command to accept a Cause header.

(closes issue #14695)
Reported by: mneuhauser
Patches:
      cause-for-hangup-manager-action.patch uploaded by mneuhauser (license 425)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185704 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoignore copied (generated) file
Kevin P. Fleming [Tue, 31 Mar 2009 22:35:07 +0000 (22:35 +0000)]
ignore copied (generated) file

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185664 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix trunk's compilation.
Mark Michelson [Tue, 31 Mar 2009 22:12:52 +0000 (22:12 +0000)]
Fix trunk's compilation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 185599 via svnmerge from
Mark Michelson [Tue, 31 Mar 2009 22:02:48 +0000 (22:02 +0000)]
Merged revisions 185599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines

  Fix crash that would occur if an empty member was specified in queues.conf.

  (closes issue #14796)
  Reported by: pida
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoOptimizations to the stringfields API
Kevin P. Fleming [Tue, 31 Mar 2009 21:29:50 +0000 (21:29 +0000)]
Optimizations to the stringfields API

This patch provides a number of optimizations to the stringfields API, focused around saving (not wasting) memory whenever possible. Thanks to Mark Michelson for inspiring this work and coming up with the first two optimizations that are represented here:

Changes:

- Cleanup of some code, fix incorrect doxygen comments

- When a field is emptied or replaced with a new allocation, decrease the amount of 'active' space in the pool it was held in; if that pool reaches zero active space, and is not the current pool, then free it as it is no longer in use

- When allocating a pool, try to allocate a size that will fit in a 'standard' malloc() allocation without wasting space

- When allocating space for a field, store the amount of space in the two bytes immediately preceding the field; this eliminates the need to call strlen() on the field when overwriting it, and more importantly it 'remembers' the amount of space the field has available, even if a shorter string has been stored in it since it was allocated

- Don't automatically double the size of each successive pool allocated; it's wasteful

http://reviewboard.digium.com/r/165/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 185531 via svnmerge
Mark Michelson [Tue, 31 Mar 2009 20:56:46 +0000 (20:56 +0000)]
Blocked revisions 185531 via svnmerge

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  r185531 | mmichelson | 2009-03-31 15:55:47 -0500 (Tue, 31 Mar 2009) | 3 lines

  Use AST_SCHED_DEL_SPINLOCK instead of manually using the logic.
........

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11 years agoMerged revisions 185468 via svnmerge from
Mark Michelson [Tue, 31 Mar 2009 19:46:18 +0000 (19:46 +0000)]
Merged revisions 185468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines

  Fix Russian voicemail intro to say the word "messages" properly.

  (closes issue #14736)
  Reported by: chappell
  Patches:
        voicemail_no_messages.diff uploaded by chappell (license 8)
........

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11 years agoImprove performance of the code handling the frame queue in chan_iax2.
Russell Bryant [Tue, 31 Mar 2009 19:07:58 +0000 (19:07 +0000)]
Improve performance of the code handling the frame queue in chan_iax2.

In my tests that exercised full frame handling in chan_iax2, the version with
these changes took 30% to 40% of the CPU time compared to the same test of
Asterisk trunk before these modifications.

While doing some profiling for <http://reviewboard.digium.com/r/205/>,
one function that caught my eye was network_thread() in chan_iax2.c.
After the things that I was working on there, it was the next target
for analysis and optimization.  I used oprofile's source annotation
functionality and found that the loop traversing the frame queue in
network_thread() was to blame for the excessive CPU cycle consumption.

The frame_queue in chan_iax2 previously held all frames that either were
pending transmission or had been transmitted and are still pending
acknowledgment.

In network_thread(), the previous code would go back through the main
for loop after reading a single incoming frame or after being signaled
because a frame had been queued up for initial transmission.  In each
iteration of the loop, it traverses the entire frame queue looking for
frames that need to be transmitted.  On a busy server, this could easily
be quite a few entries.

This patch is actually quite simple.  The frame_queue has become only a list
of frames pending acknowledgment.  Frames that need to be transmitted are
queued up to a dedicated transmit thread via the taskprocessor API.

As a result, the code in network_thread() becomes much simpler, as its only
job is to read incoming frames.

In addition to the previously described changes, this patch includes some
additional changes to the frame_queue.  Instead of one big frame_queue, now
there is a list per call number to further reduce wasted list traversals.
The biggest impact of this change is in socket_process().

For additional details on testing and test results, see the review request.

Review: http://reviewboard.digium.com/r/212/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 185362 via svnmerge from
David Brooks [Tue, 31 Mar 2009 16:46:57 +0000 (16:46 +0000)]
Merged revisions 185362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines

  Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces

  To drill into the xmpp to find the capabilities between channels, chan_gtalk
  calls iks_child() and iks_next(). iks_child() and iks_next() are functions in
  the iksemel xml parsing library that traverse xml nodes. The bug here is that
  both iks_child() and iks_next() will return the next iks_struct node
  *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG,
  which in most cases, it is, but in this case (a call being made from the
  Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data
  (they are extraneous whitespaces), and chan_gtalk doesn't handle that case,
  so capabilities don't match, and a call cannot be made.

  iks_first_tag() and iks_next_tag(), on the other hand, will not return the
  very next iks_struct, but will check to see if the next iks_struct is of
  type IKS_TAG. If it isn't, it will be skipped, and the next struct of type
  IKS_TAG it finds will be returned. This assures that chan_gtalk will find
  the iks_struct it is looking for.

  This fix simply changes all calls to iks_child() and iks_next() to become
  calls to iks_first_tag() and iks_next_tag(), which resolves the capability
  matching.

  The following is a payload listing from Empathy, which, due to the extraneous
  whitespace, will not be parsed correctly by iksemel:

  <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/>
   <payload-type clockrate='8000' name='PCMA' id='8'/>
   <payload-type clockrate='8000' name='PCMU' id='0'/>
   <payload-type clockrate='90000' name='MPA' id='97'/>
   <payload-type clockrate='16000' name='SIREN' id='98'/>
   <payload-type clockrate='8000' name='telephone-event' id='99'/>
  </description>
  </session>
  </iq>

Review: http://reviewboard.digium.com/r/181/
........

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11 years agoBlocked revisions 185298 via svnmerge
Mark Michelson [Tue, 31 Mar 2009 15:34:29 +0000 (15:34 +0000)]
Blocked revisions 185298 via svnmerge

........
  r185298 | mmichelson | 2009-03-31 10:34:05 -0500 (Tue, 31 Mar 2009) | 10 lines

  Fix some state_interface stuff that was in trunk but not in the backport to 1.4.

  Issue #14359 was fixed between the time that I posted the review of the backport
  of the state interface change for 1.4. This merges the changes from that issue
  back into 1.4.

  (closes issue #14359)
  Reported by: francesco_r
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoDon't free() an astobj2 object.
Russell Bryant [Tue, 31 Mar 2009 14:53:45 +0000 (14:53 +0000)]
Don't free() an astobj2 object.

(closes issue #14672)
Reported by: makoto

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185261 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 185196 via svnmerge from
Joshua Colp [Tue, 31 Mar 2009 14:07:36 +0000 (14:07 +0000)]
Merged revisions 185196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines

  Fix crash when moving audiohooks between channels.

  Handle the scenario where we are called to move audiohooks between channels
  and the source channel does not actually have any on it.

  (closes issue #14734)
  Reported by: corruptor
........

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11 years agoMerged revisions 185121 via svnmerge from
Richard Mudgett [Mon, 30 Mar 2009 20:42:14 +0000 (20:42 +0000)]
Merged revisions 185121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line

  Update the channel allocation method documentation.
........

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11 years agoMerged revisions 185120 via svnmerge from
Richard Mudgett [Mon, 30 Mar 2009 20:41:24 +0000 (20:41 +0000)]
Merged revisions 185120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines

  Make chan_misdn BRI TE side normally defer channel selection to the NT side.

  Channel allocation collisions are not handled by chan_misdn very well.
  This patch simply avoids the problem for BRI only.

  For PRI, allocation collisions are still possible but less likely since
  there are simply more channels available and each end could use a different
  allocation strategy.

  misdn.conf options available:
  te_choose_channel - Use to force the TE side to allocate channels.
  method - Specify the channel allocation strategy.

  (closes issue #13488)
  Reported by: Christian_Pinedo
  Patches:
        isdn_lib.patch.txt uploaded by crich
  Tested by: crich, siepkes, festr
........

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11 years agoMerged revisions 185031 via svnmerge from
Mark Michelson [Mon, 30 Mar 2009 16:26:48 +0000 (16:26 +0000)]
Merged revisions 185031 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines

  Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked.

  (This is copied and pasted from the review request I made for this patch)

  Asterisk has some odd behavior when queue weights are used. The current logic used when
  potentially calling a queue member is:

  If the member we are going to call is part of another queue and _that other queue has any
  callers in it_ and has a higher weight than the queue we are calling from, then don't try
  to contact that member. The issue here is what I have marked with underscores. If the
  higher-weighted queue has any callers in it at all, then the queue member will be unreachable
  from the lower-weighted queue. This has the potential to be really really bad if using a
  queue strategy, such as leastrecent or fewestcalls, with the potential to call the same
  member repeatedly.

  The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works
  well for this situation. With this set of changes, the logic used becomes:

  If the member we are going to call is part of another queue, the other queue has a higher
  weight than the queue we are calling from, and the higher weight queue has at least as many
  callers as available members, then do not try to contact the queue member. If the higher
  weighted queue has fewer callers than available members, then there is no reason to deny
  the call to this member since the other queue can afford to spare a member.

  Since the fix involved writing a generic function for determining the number of available
  members in the queue, I also modified the is_our_turn function to make use of the new
  num_available_members function to determine if it is our turn to try calling a member. There
  is one small behavior change. Before writing this patch, if you had autofill disabled, then
  if you were the head caller in a queue, you would automatically be told that it was your
  turn to try calling a member. This did not take into account whether there were actually any
  queue members available to take the call. Now we actually make sure there is at least one
  member available to take the call if autofill is disabled.

  (closes issue #13220)
  Reported by: garychen

  Review: http://reviewboard.digium.com/r/202/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185072 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoBlocked revisions 184980 via svnmerge
Mark Michelson [Mon, 30 Mar 2009 15:25:04 +0000 (15:25 +0000)]
Blocked revisions 184980 via svnmerge

........
  r184980 | mmichelson | 2009-03-30 10:23:59 -0500 (Mon, 30 Mar 2009) | 22 lines

  Backport state interface changes to app_queue from trunk.

  After several issues raised on the Asterisk bugtracker against
  the 1.4 branch were determined to be fixable with the state interface
  change available in the 1.6.X series, it finally came time to just
  suck it up and backport the change.

  For a detailed explanation of what this change entails, the original
  trunk commit for this feature may be found here:

  http://svn.digium.com/view/asterisk?view=revision&revision=97203

  In addition, the details for the use of this change to fix the problems
  stated in issue #12970 may be found in the review request I made for
  this change. It is linked below.

  (closes issue #12970)
  Reported by: edugs15

  Review: http://reviewboard.digium.com/r/116
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184986 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184947 via svnmerge from
Joshua Colp [Mon, 30 Mar 2009 14:37:47 +0000 (14:37 +0000)]
Merged revisions 184947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines

  Improve our handling of T38 in the initial INVITE from a device.

  We now answer with matching media streams to what is requested. If an INVITE
  is received with both a T38 and RTP media stream this means we answer with both.
  For any outgoing calls created as a result of this inbound one no T38 is requested
  in the initial INVITE. Instead if we start receiving udptl packets we trigger a
  reinvite on the outbound side.

  (closes issue #12437)
  Reported by: marsosa
  Tested by: pinga-fogo, okrief, file, afu

  Review: http://reviewboard.digium.com/r/208/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184948 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix build error when chan_h323 is not being built.
Russell Bryant [Mon, 30 Mar 2009 13:55:44 +0000 (13:55 +0000)]
Fix build error when chan_h323 is not being built.

(reported by cai1982 in #asterisk-dev)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184842 via svnmerge from
Russell Bryant [Sun, 29 Mar 2009 05:52:20 +0000 (05:52 +0000)]
Merged revisions 184842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines

Ensure targs variable is fully initialized.

(closes issue #14758)
Reported by: tim_ringenbach

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184843 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoSimplify chan_h323 build to not require a second run of "make".
Russell Bryant [Sun, 29 Mar 2009 05:32:04 +0000 (05:32 +0000)]
Simplify chan_h323 build to not require a second run of "make".

(closes issue #14715)
Reported by: jthurman
Patches:
      h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614)
Tested by: tzafrir, russell

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184838 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a typo in app_ices.
Leif Madsen [Fri, 27 Mar 2009 20:08:44 +0000 (20:08 +0000)]
Fix a typo in app_ices.

(closes issue #14765)
Reported by: timeshell
Patches:
      app_ices.svn-1.6.0.diff uploaded by timeshell (license 399)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184801 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoUpdate commit message guidelines in re: to punctuation.
Leif Madsen [Fri, 27 Mar 2009 19:31:04 +0000 (19:31 +0000)]
Update commit message guidelines in re: to punctuation.
The doxygen documentation has now been updated to state explicitly that I want
punctuation atthe end of the first sentence in a commit message. :).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184798 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoImprove timing interface to remember which provider provided a timer
Kevin P. Fleming [Fri, 27 Mar 2009 19:10:32 +0000 (19:10 +0000)]
Improve timing interface to remember which provider provided a timer

The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.

This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.

(closes issue #14697)
Reported by: moy

Review: http://reviewboard.digium.com/r/211/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoUse ast_random() instead of rand() to ensure we use the best RNG available.
Russell Bryant [Fri, 27 Mar 2009 18:04:43 +0000 (18:04 +0000)]
Use ast_random() instead of rand() to ensure we use the best RNG available.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoChange global_app_buf to ast_str_thread_global_buf.
Russell Bryant [Fri, 27 Mar 2009 16:21:10 +0000 (16:21 +0000)]
Change global_app_buf to ast_str_thread_global_buf.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184693 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a potential timer leak in bridge_softmix.
Joshua Colp [Fri, 27 Mar 2009 15:57:28 +0000 (15:57 +0000)]
Fix a potential timer leak in bridge_softmix.

It is possible for a bridge to be created without actually being used.
In that scenario a timing file descriptor would be opened and not
closed. To fix this the timing file descriptor is now closed in the
destroy callback, not the thread function.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184677 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix speech structure leak in the AGI speech recognition integration.
Joshua Colp [Fri, 27 Mar 2009 15:46:46 +0000 (15:46 +0000)]
Fix speech structure leak in the AGI speech recognition integration.

The AGI dialplan applications did not destroy the speech structure automatically
if it was not destroyed by the running AGI script. They will now do this.

(issue LUMENVOX-15)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184673 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove a cast that is not needed.
Joshua Colp [Fri, 27 Mar 2009 14:18:40 +0000 (14:18 +0000)]
Remove a cast that is not needed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoChange g_eid to ast_eid_default.
Russell Bryant [Fri, 27 Mar 2009 14:00:18 +0000 (14:00 +0000)]
Change g_eid to ast_eid_default.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184630 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a potential race condition when creating a software based mixing bridge.
Joshua Colp [Fri, 27 Mar 2009 13:57:29 +0000 (13:57 +0000)]
Fix a potential race condition when creating a software based mixing bridge.

It was possible for no timer to become available between creating the bridge
and starting it. We now open a timer when creating it and keep it open until the
bridge is destroyed.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184565 via svnmerge from
Joshua Colp [Fri, 27 Mar 2009 13:15:26 +0000 (13:15 +0000)]
Merged revisions 184565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines

  Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls.

  If calls were placed using an IP address or hostname the global nat setting was copied over
  but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP
  actions.

  (closes issue #14546)
  Reported by: acunningham
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184566 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix some issues with rwlock corruption that caused deadlock like symptoms.
Russell Bryant [Fri, 27 Mar 2009 02:20:23 +0000 (02:20 +0000)]
Fix some issues with rwlock corruption that caused deadlock like symptoms.

When dvossel and I were doing some load testing last week, we noticed that we
could make Asterisk trunk lock up instantly when we started generating a bunch
of calls.  The backtraces of locked threads were bizarre, and many were stuck
on an _unlock_ of an rwlock.

The changes are:

1) Fix a number of places where a backtrace would be loaded into an invalid
   index of the backtrace array.  It's an off by one error, which ends up
   writing over the rwlock itself.

2) Ensure that in the array of held locks, we NULL out an index once it is
   not being used so that it's not confusing when analyzing its contents.

3) Remove a bunch of logging referring to an rwlock operating being done
   with "deep reentrancy".  It is normal for _many_ threads to hold a
   read lock on an rwlock.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184531 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoDon't act surprised if we get a -1 indication.
Russell Bryant [Fri, 27 Mar 2009 01:40:28 +0000 (01:40 +0000)]
Don't act surprised if we get a -1 indication.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoPass more useful information through to lock tracking when DEBUG_THREADS is on.
Russell Bryant [Fri, 27 Mar 2009 01:35:56 +0000 (01:35 +0000)]
Pass more useful information through to lock tracking when DEBUG_THREADS is on.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184447 via svnmerge from
Kevin P. Fleming [Thu, 26 Mar 2009 22:18:14 +0000 (22:18 +0000)]
Merged revisions 184447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines

  use new, improved 8kHz prompts
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184388 via svnmerge from
David Vossel [Thu, 26 Mar 2009 21:09:37 +0000 (21:09 +0000)]
Merged revisions 184388 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines

  pri loop TestClient/TestServer fails: server SEND DTMF 8

  app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent.  During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up.

  (closes issue #12442)
  Reported by: tzafrir
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184389 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove unneeded AST_LIST_ENTRY() and comment on the purpose of ast_event_ref.
Russell Bryant [Wed, 25 Mar 2009 22:11:35 +0000 (22:11 +0000)]
Remove unneeded AST_LIST_ENTRY() and comment on the purpose of ast_event_ref.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184344 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoImprove performance of the ast_event cache functionality.
Russell Bryant [Wed, 25 Mar 2009 21:57:19 +0000 (21:57 +0000)]
Improve performance of the ast_event cache functionality.

This code comes from svn/asterisk/team/russell/event_performance/.

Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.

1) Asterisk 1.6.1 introduces some additional logic to be able to handle
   distributed device state.  This functionality comes at a cost.
   One relatively minor change in this patch is that the extra processing
   required for distributed device state is now completely bypassed if
   it's not needed.

2) One of the things that I noticed when profiling this code was that a
   _lot_ of time was spent doing string comparisons.  I changed the way
   strings are represented in an event to include a hash value at the front.
   So, before doing a string comparison, we do an integer comparison on the
   hash.

3) Finally, the code that handles the event cache has been re-written.
   I tried to do this in a such a way that it had minimal impact on the API.
   I did have to change one API call, though - ast_event_queue_and_cache().
   However, the way it works now is nicer, IMO.  Each type of event that
   can be cached (MWI, device state) has its own hash table and rules for
   hashing and comparing objects.  This by far made the biggest impact on
   performance.

For additional details regarding this code and how it was tested, please see the
review request.

(closes issue #14738)
Reported by: russell

Review: http://reviewboard.digium.com/r/205/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix issue with a T38 reinvite being sent even if not configured to do so.
Joshua Colp [Wed, 25 Mar 2009 19:22:06 +0000 (19:22 +0000)]
Fix issue with a T38 reinvite being sent even if not configured to do so.

If we receive a T38 request negotiate control frame we should only attempt to do so
if the option is enabled on the dialog.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184280 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184188 via svnmerge from
Eliel C. Sardanons [Wed, 25 Mar 2009 14:38:19 +0000 (14:38 +0000)]
Merged revisions 184188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines

  Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete.

  When moving the cursor backward and pressing TAB to autocomplete, a NULL is put
  in the line and we are loosing what we have already wrote after the actual
  cursor position.

  (closes issue #14373)
  Reported by: eliel
  Patches:
        asterisk.c.patch uploaded by eliel (license 64)
        Tested by: lmadsen
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184220 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoInclude poll-compat.h
Russell Bryant [Wed, 25 Mar 2009 14:33:32 +0000 (14:33 +0000)]
Include poll-compat.h

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoChange poll() to ast_poll().
Russell Bryant [Wed, 25 Mar 2009 02:03:13 +0000 (02:03 +0000)]
Change poll() to ast_poll().

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184151 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix build issues on Mac OSX.
Russell Bryant [Wed, 25 Mar 2009 01:42:10 +0000 (01:42 +0000)]
Fix build issues on Mac OSX.

(closes issue #14714)
Reported by: ygor

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184147 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 184078 via svnmerge from
Mark Michelson [Tue, 24 Mar 2009 22:40:39 +0000 (22:40 +0000)]
Merged revisions 184078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines

  Change NULL pointer check to be ast_strlen_zero.

  The 'digit' variable is guaranteed to be non-NULL, so the if
  statement could never evaluate true. Changing to ast_strlen_zero
  makes the logic correct.

  This was found while reviewing ast_channel_ao2 code review.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoPut siren7 and siren14 in ast_best_codec() just so they're in there somewhere.
Russell Bryant [Tue, 24 Mar 2009 22:00:58 +0000 (22:00 +0000)]
Put siren7 and siren14 in ast_best_codec() just so they're in there somewhere.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoExclude slin16, siren7, and siren14 from bandwidth=low and =medium
Russell Bryant [Tue, 24 Mar 2009 21:40:44 +0000 (21:40 +0000)]
Exclude slin16, siren7, and siren14 from bandwidth=low and =medium

The default codec configuration for chan_iax2 is bandwidth=low.  I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoSIP preferred codec only feature
David Vossel [Tue, 24 Mar 2009 20:01:29 +0000 (20:01 +0000)]
SIP preferred codec only feature

Added an option to respond to a SIP invite with only the single most preferred joint codec.  This limits the options of what codecs the other side can use.

(closes issue #12485)
Reported by: bamby
Review: http://reviewboard.digium.com/r/206/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183913 via svnmerge from
Tilghman Lesher [Tue, 24 Mar 2009 15:26:42 +0000 (15:26 +0000)]
Merged revisions 183913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines

  Additionally note that the operator option needs an 'o' extension.
  (Related to issue #14731)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183914 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAllow browsers to cache images and other static content.
Tilghman Lesher [Mon, 23 Mar 2009 23:28:20 +0000 (23:28 +0000)]
Allow browsers to cache images and other static content.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183865 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemoved trailing whitespace in chan_misdn files.
Richard Mudgett [Mon, 23 Mar 2009 22:35:02 +0000 (22:35 +0000)]
Removed trailing whitespace in chan_misdn files.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183700 via svnmerge from
Mark Michelson [Mon, 23 Mar 2009 18:58:03 +0000 (18:58 +0000)]
Merged revisions 183700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines

  Fix a memory leak in res_monitor.c

  The only way that this leak would occur is if Monitor were started
  using the Manager interface and no File: header were given. Discovered
  while reviewing the ast_channel_ao2 review request.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183766 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008.
Leif Madsen [Mon, 23 Mar 2009 18:06:40 +0000 (18:06 +0000)]
Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008.

(closes issue #14655)
Reported by: ulogic
Patches:
      chan_dahdi.patch uploaded by ulogic (license 728)
Tested by: lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183701 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a minor logic flaw with the bridge generic thread.
Joshua Colp [Sun, 22 Mar 2009 21:00:28 +0000 (21:00 +0000)]
Fix a minor logic flaw with the bridge generic thread.

We only want to move the channel pointers that are actually present.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183652 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183559 via svnmerge from
Russell Bryant [Fri, 20 Mar 2009 17:00:58 +0000 (17:00 +0000)]
Merged revisions 183559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines

Fix a crash in IAX2 registration handling found during load testing with dvossel.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183560 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix chan_sip so it builds.
Mark Michelson [Fri, 20 Mar 2009 16:25:17 +0000 (16:25 +0000)]
Fix chan_sip so it builds.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183555 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove symbols I just added to main/asterisk.exports and instead rename the functions.
Mark Michelson [Fri, 20 Mar 2009 16:24:20 +0000 (16:24 +0000)]
Remove symbols I just added to main/asterisk.exports and instead rename the functions.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183554 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoAdd some missing symbols to main/asterisk.exports
Mark Michelson [Fri, 20 Mar 2009 16:19:53 +0000 (16:19 +0000)]
Add some missing symbols to main/asterisk.exports

Hey! chan_sip.so loads now!

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183553 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRemove duplicate <description> inside the xml documentation.
Eliel C. Sardanons [Fri, 20 Mar 2009 12:12:49 +0000 (12:12 +0000)]
Remove duplicate <description> inside the xml documentation.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183511 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183386 via svnmerge from
David Vossel [Thu, 19 Mar 2009 20:30:39 +0000 (20:30 +0000)]
Merged revisions 183386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines

  Cleaning up a few things in detect disconnect patch

  Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory.  Cleaned up /param tags in features.h.  No longer send dynamic features in ast_feature_detect.

  issue #11583
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183436 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoRecorded merge of revisions 183342 via svnmerge from
Tilghman Lesher [Thu, 19 Mar 2009 19:22:12 +0000 (19:22 +0000)]
Recorded merge of revisions 183342 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r183342 | tilghman | 2009-03-19 14:21:30 -0500 (Thu, 19 Mar 2009) | 2 lines

  Reordering, to change prior to unlocking
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183345 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183319 via svnmerge from
Tilghman Lesher [Thu, 19 Mar 2009 19:17:31 +0000 (19:17 +0000)]
Merged revisions 183319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines

  Delay signalling progress until a PRI channel really signals progress.
  (closes issue #13034)
   Reported by: klaus3000
   Patches:
         20090316__bug13034.diff.txt uploaded by tilghman (license 14)
         patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65)
   Tested by: klaus3000
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183321 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183291 via svnmerge from
Jason Parker [Thu, 19 Mar 2009 18:34:11 +0000 (18:34 +0000)]
Merged revisions 183291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r183291 | qwell | 2009-03-19 13:28:16 -0500 (Thu, 19 Mar 2009) | 1 line

  Export some more required symbols.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183312 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoFix a memory leak associated with queues.
Mark Michelson [Thu, 19 Mar 2009 18:10:34 +0000 (18:10 +0000)]
Fix a memory leak associated with queues.

For every attempt that app_queue made to place an outbound call to a queue member,
we would allocate a queue_end_bridge structure. When the bridge for the call had
completed, we would free the structure. Unfortunately not all call attempts actually
end up bridged to a member, so we need to be more selective of when to allocate
the structure. With this change, the allocation occurs in an area where we can
guarantee that the call will be bridged.

(closes issue #14680)
Reported by: caspy
Patches:
      14680.patch uploaded by mmichelson (license 60)
Tested by: caspy

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183244 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183241 via svnmerge from
Russell Bryant [Thu, 19 Mar 2009 18:00:15 +0000 (18:00 +0000)]
Merged revisions 183241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines

Remove the use of RTLD_NOLOAD, as it is not behaving like expected.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183242 65c4cc65-6c06-0410-ace0-fbb531ad65f3

11 years agoMerged revisions 183238 via svnmerge from
Russell Bryant [Thu, 19 Mar 2009 17:42:06 +0000 (17:42 +0000)]
Merged revisions 183238 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r183238 | russell | 2009-03-19 12:41:39 -0500 (Thu, 19 Mar 2009) | 1 line

Allow the AES API to work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183239 65c4cc65-6c06-0410-ace0-fbb531ad65f3