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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines

Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)

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(issue #9105, junky)


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r58165 | russell | 2007-03-06 18:25:19 -0600 (Tue, 06 Mar 2007) | 12 lines

Merged revisions 58164 via svnmerge from 
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines

If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them.  It could lead to weird behavior,
including crashes.  (issue #8977)

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(inspired by issue #9215)


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addition to where it is already sent if either side hangs up.
(issue #9219, rgollent)
In passing, I put this code in a function so it would not be duplicated
a third time.


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r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines

Merged revisions 58115 via svnmerge from 
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line

Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) | 3 lines

Clarify the documentation of the dialout and sendvoicemail options.
(issue #9000, caio1982 and serge-v)

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build here, too.


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r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines

Merged revisions 58052 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines

Change error message to proper message

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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines

Return an error of transmit_response is called without a session.
(issue #9002)

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r57591 | russell | 2007-03-02 18:02:29 -0600 (Fri, 02 Mar 2007) | 1 line

add missing configuration template.  Thanks to Lacy Moore on asterisk-users for pointing this out\!
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Add zap show version CLI command. This pulls the version/echo canceller in use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)


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r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines

Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)

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r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2 lines

Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r)

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r57870 | file | 2007-03-05 12:52:03 -0500 (Mon, 05 Mar 2007) | 10 lines

Merged revisions 57869 via svnmerge from 
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r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines

Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares)

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r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon, 05 Mar 2007) | 9 lines

Merged revisions 57825 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line

Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2 lines

Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)

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r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2 lines

Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)

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r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines

Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)

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the conversion of IAX2 variables from the current custom method to ast_storage.


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r57649 | tilghman | 2007-03-03 00:45:00 -0600 (Sat, 03 Mar 2007) | 10 lines

Merged revisions 57648 via svnmerge from 
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r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines

Memory leak of a list, if call recording was abandoned

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r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007) | 1 line

submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri
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Thanks to Lacy Moore on the asterisk-users list for pointing out that this
was missing!


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r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02 Mar 2007) | 3 lines

Update the check that is used to determine whether zaptel transcoder support
is present.  The interface has changed.

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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines

Merged revisions 57475 via svnmerge from 
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines

If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.

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r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri, 02 Mar 2007) | 9 lines

Merged revisions 57458 via svnmerge from 
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r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line

further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction
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application.  (issue #9187, Junky)


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r57426 | murf | 2007-03-01 22:21:36 -0700 (Thu, 01 Mar 2007) | 1 line

I almost had comma escapes right, but 9184 points out the problem-- the escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL.
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r57396 | russell | 2007-03-01 18:20:44 -0600 (Thu, 01 Mar 2007) | 4 lines

Return the correct digit that interrupted the stream.  This fixes exiting the
Background application when using the m option.
(issue #9176, mjagdis)

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r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines

Merge changes from svn/asterisk/team/russell/sla_updates

* Originally, I put in the documentation that only Zap interfaces would be
  supported on the trunk side.  However, after a discussion with Qwell, we came
  up with a way to make IP trunks work as well, using some things already in
  Asterisk.  So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
  voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
  in SLA.  The station's channel needs to be passed to the dial API when
  dialing the trunk.
* Change a WARNING message to DEBUG in channel.h.  This message is of no use
  to users.

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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines

Merged revisions 57317 via svnmerge from 
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines

Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)

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