[asterisk-dev] [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more

[Joshua Colp]

Michael Maier wrote:
> Hello Joshua!
>
>
> I attached the sip debug without the progressinband=never set. The
> caller didn't get a ring back tone as expected.

Please keep this on list so that anyone who may run into a similar 
problem in the future has a chance of finding this discussion.

As for your log there's nothing of note really, it's just expecting to 
send the ringing as inband audio instead of out of band. Does "rtp set 
debug on" show the RTP traffic going to the other side?

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org