* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
