as well as some typos.


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roster is received.


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and some patches (all disclaimed).

- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments

Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.



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- constify some arguments to functions in callerid.c / callerid.h


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my concerns are listed in issue #6305


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Convert chan_iax2 to use dnsmgr in order to deal with hostnames that can change their resolved IP (aka dynamic dns setups) (issue #6305 reported and fixed by ivanfm)


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   Thanks to everyone involved in working with this!


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https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r31738 | bweschke | 2006-06-03 10:48:13 -0400 (Sat, 03 Jun 2006) | 3 lines

 Fix doxygen comment about AST_LIST_HEAD_INIT_NOLOCK


........


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https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r31520 | kpfleming | 2006-06-01 15:27:50 -0500 (Thu, 01 Jun 2006) | 2 lines

handle Zap transfers behind chan_agent properly so the agent doesn't get stuck waiting for the call to hang up

........


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as well as being more efficient, patch from jcollie's
base64 branch


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and fixes bug with pruneregister


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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)

Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!


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output to remove consoles. The prototypes added to logger.h still need
doxygen documentation, as well.

- Add the new command line option to the man page
- make the mute option a flag instead of an int since it is only a binary
  option
- remove useless extern keywords for prototypes added to logger.h
- rename ast_console_mute() to ast_console_toggle_mute() since that is what
  it actually does
- actually apply the mute option to newly created remote consoles instead of
  only working when the CLI command is used
- don't imply the NO_FORK option if the mute command line option is provided
- place the new CLI command in the correct place in the list which has to be
  in alphabetical order
- Finally, clean up a few spacing issues to conform to the coding guidelines


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Merge in branch which gives you the ability to set the hangup causecode using the Hangup application. (issue #7160 reported by kmilitzer branch by jcollie)


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is an error executing the AGI script, or the AGI script itself returns a
non-zero value, the AGISTATUS variable will now be set to FAILURE instead of
SUCCESS.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30328 65c4cc65-6c06-0410-ace0-fbb531ad65f3

update ast_mutex_init to allow mutexes that are all zero bytes to be initialized (in the case of a dynamically-allocated structure containing a mutex)


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 This is part 2/2 of the patches for #7090. Adds one-step call parking to /trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340.



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 - with contributions from miconda, jcollie, and sb
 - branch maintained by oej
Thanks everyone!


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allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)


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   So, I have removed all of the uses of AST_LIST_HEAD_INIT and replaced them
   with the equivalent static initializations.
 - On passing, fix a memory leak in the unload_module() function of chan_agent.
   The agents list mutex was never destroyed, and the elements in the agents
   list were not freed.


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(Thanks Dinesh for tracking the bug and fixing it)



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support in SIP.


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instead of being added to the compiler commands.  This header file will be
installed and modules built outside of the main tree will be able to use the
same build options used to build the rest of Asterisk.


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As it turns out, all of these checks were useless, because alloca will never
return NULL.


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update iax2_indicate to pass control frame payload to the connected channel
add an API call for sending an indication with payload, and use it for control frames with payload


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