asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	3d6e6e07ef	Merged revisions 57364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	ae8c0f3fcb	Merged revisions 57207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57207 \| russell \| 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) \| 2 lines minor tweaks to the sla docs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	9138e53bc9	Merged revisions 56277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56277 \| russell \| 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) \| 18 lines Merge changes from team/russell/sla_updates. 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/trunk@56278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	5715b49c30	Merged revisions 53810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53810 \| russell \| 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) \| 24 lines Merge team/russell/sla_rewrite This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago

4 Commits (96f7036c99d8be1001ca9eaafd81a59a25879417)