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2008-03-16add missing break to case AST_CONTROL_SRCUPDATEmvanbaak1-0/+1
(closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@108961 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Add a control frame to indicate the source of media has changed. Depending ↵file1-0/+4
on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@106235 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27Two fixes:mmichelson1-5/+31
1. Make the list of ast_dial_channels a lockable list. This is because in some cases, the ast_dial may exist in multiple threads due to asynchronous execution of its application, and I found some cases where race conditions could exist. 2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been cleared yet. (closes issue #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by putnopvut (license 60) Tested by: jvandal git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@104841 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16Introduce a lock into the dialing API that protects it when destroying the ↵file1-0/+15
structure. (closes issue #11687) Reported by: callguy Patches: 11687.diff uploaded by file (license 11) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@98960 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21Since we are freeing list elements within a list traversal, we need to use ↵mmichelson1-1/+3
the safe traversal and remove the item from the list before freeing it. (closes issue 11612, reported by dtyoo) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@94468 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Fix issues with async dialing with an application executing. The application ↵file1-5/+24
has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@89610 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24Add a few more state changes in handle_frame_ownerless() so that the SLA coderussell1-0/+3
will get notified of these changes even when an owner channel is not provided. This isn't from a specific bug report, it's just something I noticed while poking around. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61774 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22Merge changes from team/russell/sla_updates.russell1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@56277 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12Change ast_set_state_callback() to ast_dial_set_state_callback()russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54103 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12- Add the ability to register a callback to monitor state changes in anrussell1-26/+39
asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54066 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-10Merge team/russell/sla_rewriterussell1-22/+31
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: http://svn.digium.com/svn/asterisk/branches/1.4@53810 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-24Merge in dialing API and the app_page that uses it. (issue #BE-118)file1-0/+802
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52049 f38db490-d61c-443f-a65b-d21fe96a405b
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