Commit Graph

8 Commits

Author SHA1 Message Date
Joshua Colp
0b898073d1 Add an API call that steals the answered channel so that a destruction of the dialing structure does not hang it up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-25 02:52:10 +00:00
Joshua Colp
3aaf122439 Add support for call forwarding and timeouts to the dialing API.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-30 20:42:28 +00:00
Russell Bryant
ab31bec392 Add an option to the dial API for playing music instead of ringing to the caller.
I started this for use with SLA but ended up deciding not to use it.  However,
there is no reason not to put this part in, anyway.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 19:16:24 +00:00
Russell Bryant
8f8df3e5a9 Merged revisions 54103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | 2 lines

Change ast_set_state_callback() to ast_dial_set_state_callback()

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12 19:18:33 +00:00
Russell Bryant
2a5477b35e Merged revisions 54066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines

- Add the ability to register a callback to monitor state changes in an
  asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12 18:01:15 +00:00
Russell Bryant
5715b49c30 Merged revisions 53810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10 00:40:57 +00:00
Russell Bryant
65439dcbb3 Merged revisions 52107 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24 Jan 2007) | 3 lines

Fix the formatting of doxygen comments to properly indicate that the comment
documents the previous entity, as opposed to the next one.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24 21:43:22 +00:00
Joshua Colp
9826fc599b Merged revisions 52049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines

Merge in dialing API and the app_page that uses it. (issue #BE-118)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24 18:23:07 +00:00