Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...
Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and we now have the groupcount system to implement call-limits in the dialplan. You
can use the "setvar" option in realtime/sip.conf to set limits per device.
- Implement "callcounter" as a new option to enable the call counting we need to
report device status to queue, manager and SIP subscriptions.
The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If needed, the code to extract this information should be implemented
in some generic header or library and the function called here.
(closed bug #11362)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a NOP as far as the current code is concerned,
but there is already support in ./configure and the
Makefiles for the various libraries used by console_video.c
(not yet in the tree) so addition is trivial.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
make the format explicit in a debug message;
print the audio instead of aggregated peer capability in a debugging msg.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
writing to the wrong byte. Also, remove some non-thread safe test code.
(closes issue #11317)
Reported by: IgorG
Patches:
unistim-2.patch uploaded by IgorG (license 20)
- additional changes by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
with the SIPPEER() argument of the same name. The deprecation procedure is not
being used here since this is a trunk-only option.
(closes issue #11307, reported by pj, patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this, you can control the peer in the dialplan, so you avoid placing outbound
calls when the device has reached busy-level.
Reported by pj.
Closes bug #11180
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the bug tracker. Since we don't authenticate devices (peers/users) on OPTIONS we don't
have the proper context set for the user/peer.
However, we might not want to process an authentication for every OPTIONS, so we could
have a config option for this, "optionsforceok" to always answer 200 OK on the request
and not check device or destination, nor add a SDP. If Asterisk sends the OPTIONs request,
it doesn't care about the reply. Some devices use OPTIONs to discover capabilities,
since we should answer like an INVITE from the device and we need to support that properly
too, which we don't today.
So much to do :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(this should be handled with the dependencies generated by
configure and menuselect, but will be fixed later)
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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