Commit Graph

10 Commits

Author SHA1 Message Date
Mark Michelson
dce6a54a4a Trunk implementation of setting an alternate RTP source.
This contains the interface by which we can let an rtp instance know
that it might start receiving audio from a new source. This is similar
in nature to revision 197588 of Asterisk 1.4.

Review: https://reviewboard.asterisk.org/r/276



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18 15:20:17 +00:00
Kevin P. Fleming
e6b2e9a750 Const-ify the world (or at least a good part of it)
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21 21:13:09 +00:00
Joshua Colp
1179ecf165 Merged revisions 194208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines
  
  Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over.
  
  (closes issue #14815)
  Reported by: geoff2010
  Patches:
        v1-14815.patch uploaded by dimas (license 88)
  Tested by: geoff2010, file, dimas, ZX81, moliveras
  (closes issue #14460)
  Reported by: moliveras
  Tested by: moliveras
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-13 13:39:10 +00:00
Joshua Colp
973b36a3c7 Fix an incorrect clock rate when sending T140 text.
(closes issue #14029)
Reported by: epicac


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-14 17:40:50 +00:00
Joshua Colp
aaf1566222 Change how we set the local and remote address.
The code will now only change the address and port. It will not overwrite any other values.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 18:14:47 +00:00
Joshua Colp
8e4b5df187 Fix some uninitialized memory notices that appeared under valgrind.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-10 18:02:44 +00:00
Joshua Colp
0ab599bf94 Turn a warning message into a debug message and do not treat two situations as errors when they are not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-08 16:27:36 +00:00
Joshua Colp
c02b56f7bc Fix a log message getting output when it should not have been.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-06 23:11:13 +00:00
Mark Michelson
6f53ed4c67 This commit introduces COLP/CONP and Redirecting party information into Asterisk.
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03 22:41:46 +00:00
Joshua Colp
63de834395 Merge in the RTP engine API.
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02 17:20:52 +00:00