[asterisk-dev] Regarding SIP direct media (new topic)

Brian Candler B.Candler at pobox.com
Tue Oct 17 00:54:50 MST 2006


On Mon, Oct 16, 2006 at 08:55:10PM +0200, Johansson Olle E wrote:
> >Please have a look at the SIP exchange which is attached as
> >sip-codec-mismatch.txt to http://bugs.digium.com/view.php?id=8152
> >and see if that makes the problem clearer.
> >
> As noted in the bug tracker - we need a SIP debug from Asterisk :-)

Sorry, I missed the notes, as I wasn't emailed when they were added. (Mantis
did this when notes were added to 8072, so I don't know why it didn't for
8152)

Anyway I have re-run the test and uploaded 'sip debug' output and sip.conf.

Ideally there would be a test harness for recreating cases like this; then
the test can be re-run against future versions of Asterisk too. I don't know
if such a thing exists for Asterisk.

Regards,

Brian.


More information about the asterisk-dev mailing list