allocated. These changes caused crashes when using a channel type that did
not support the jitterbuffer. Instead of fixing why it's crashing, I'm going
to implement this in a better way next week. The way I did it caused a
jitterbuffer to be allocated on every channel where the channel type supported
jitterbuffers, even if they were disabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that channels not using a jitterbuffer don't waste as much memory
- ensure that the channel drivers that use jitterbuffers can handle a failure
from configuring a jitterbuffer on a new channel because of a memory
allocation error
- On passing through these channel drivers, configure the jitterbuffer before
starting the PBX thread instead of afterwards. If the pbx fails to start for
whatever reason, this would have caused a crash.
- Also on passing, move the increase of the usecount to after all of the
possible failure conditions in the function
- fix a place where ast_update_use_count() was not called
- ensure that the owner channel pointer of the channel pvt strcutures is set to
NULL in failure conditions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Block fix from 1.2
- Implement part of that fix that was not already implemented, but in a different way
basically, don't cancel destruction when we receive re-transmits.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bug reported in the t38 issue report, but by mistake ignored before commit.
Thanks to everyone informing me about this, and Corydon for helping me sort
out sscanf :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
bug 5090 by josh colp. Thanks to everyone who help get this patch through
especially to the author Steven Underwood.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
created as global variables. (The fact that these were getting created on
my system probably means that these are in the wrong place so oej, you may
want to look at this again.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Change variable name to make register_verify more readable (p -> peer not pvt in this function)
- Get Contact: header only once instead of twice
- Add some comments to register_verify
Caused by issue #7327... :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Change/add comments
- Declare internal function as static
- Remove functionname: in descriptions of functions
- Move Enums to top of file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
