[asterisk-users] pjsip direct_media=yes and "unknown" endpoints

Joshua Colp jcolp at digium.com
Wed Apr 26 11:33:12 CDT 2017

On Wed, Apr 26, 2017, at 01:25 PM, Daniel Tryba wrote:


> The request now gets routed based on a fully qualified domainname (with
> NAPTR/SRV records), which ultimately resolves to an ip that is matched in
> the
> endpoint SBC used above to originate a call.  But now the asterisk stays
> in the
> loop regarding RTP, a simple bridge is created but never switches to
> direct
> media.
> SIP: enduser <-> uplink <-> asterisk 13 <-> pathfinder (302 redirect)
> SIP: enduser <-> uplink <-> asterisk 13 <-> sip.xxxxxx.nl
> RTP: enduser <-> uplink <-> asterisk 13 <-> sip.xxxxxx.nl
> Anybody got an idea why the last scenario fails to work?

If you turn up core debug (core set debug 2) and ensure it is going to
the CLI then the bridge_native_rtp module will tell you why exactly it
can't native bridge. You might also want to do a core show channel on
both channels to see what the codecs are.

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

More information about the asterisk-users mailing list