On Wed, Apr 26, 2017, at 01:25 PM, Daniel Tryba wrote: <snip>
> 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
