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f38db490-d61c-443f-a65b-d21fe96a405b
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next one
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is present. The interface has changed.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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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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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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Background application when using the m option.
(issue #9176, mjagdis)
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* 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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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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* Add support for private hold. By setting "hold=private" for a trunk, only
the station that put the call on hold will be able to retrieve it from hold.
Also, by setting "hold=private" for a station, any call that station puts
on hold can only be retrieved by that station.
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* Add support for the "barge=no" option for trunks. If this option is set,
then stations will not be able to join in on a call that is on progress
on this trunk.
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r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line
a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines
Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)
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* Add support for station ring delays. Ring delays can be set globally for a
station or for specific trunks on the station.
* Fix a few bugs in existing code.
* Restructure and Reorganize code to improve readability and maintainability.
* Improve formatting of the "sla show (trunks|stations)" CLI commands.
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played but before the timeout ends.
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call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code.
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endings and there was an extra one where it should not have been.
(issue #9128)
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via list)
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alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139)
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r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 lines
Obey the clearglobalvars option in extensions reload (or dialplan reload depending on your version). (issue #9146 reported by ramonpeek)
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formats. (issue #9151 reported by junky)
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present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis)
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(Problem brought up by Ben Smithurst on the asterisk-dev list)
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commands to work. (issue #9135 reported by mvanbaak)
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r56729 | russell | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines
Ensure that lock.h is included in utils.c with AST_API_MODULE defined so that
the implementations will be properly included when the AST_INLINE_API functions
are not going to be inlined. (issue #9124, festr)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines
Issue 9130 - If prev is the last item on the channel list, then evaluating
additional conditions (e.g. name prefix) will cause a NULL dereference.
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Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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and be deleted at close to the same time and unavoidable. If this happens this message creeps up.
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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voicemail is sent via email using something like sendmail. In the patch from
bug 8033 to fix various IMAP storage problems, the line endings in the email
file were changed in the code from "\n" to "\r\n". However, this breaks
sending regular voicemail to email. So, this change conditionally sets line
endings to "\r\n" only if IMAP_STORAGE is enabled.
(issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage)
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r56279 | file | 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines
Always defer Agent logoff if any channels are up until they hang up. (issue #9123 reported by arbrandes)
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This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines
Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.
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change here.
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r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines
If we receive a frame that is not in any of the negotiated formats, then drop
it. (potentially issue #8781 and SPD-12)
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