I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures.
For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC matches an endpoint based on ip and dials the uplink: -- Executing [+31xxxxxxxxx@outgoingrr:9] Dial("PJSIP/sbcs-00000092", "PJSIP/+31xxxxxxxxx@uplink") in new stack -- Called PJSIP/+31xxxxxxxxx@uplink -- PJSIP/uplink-00000093 is making progress passing it to PJSIP/sbcs-00000092 -- PJSIP/uplink-00000093 answered PJSIP/sbcs-00000092 -- Channel PJSIP/uplink-00000093 joined 'simple_bridge' basic-bridge <3b25c543-13a3-4d74-b2fe-7122a1cfe4a4> -- Channel PJSIP/sbcs-00000092 joined 'simple_bridge' basic-bridge <3b25c543-13a3-4d74-b2fe-7122a1cfe4a4> > Bridge 3b25c543-13a3-4d74-b2fe-7122a1cfe4a4: switching from simple_bridge technology to native_rtp > Remotely bridged 'PJSIP/sbcs-00000092' and 'PJSIP/uplink-00000093' - media will flow directly between them > Remotely bridged 'PJSIP/sbcs-00000092' and 'PJSIP/uplink-00000093' - media will flow directly between them Whoever when a terminating call comes in from the uplink provider, a sip request is send to a redirector. The redirector has redirect_method=uri_core configured (the only method that works for me). -- Executing [+31xxxxxxxxx@incoming:11] Dial("PJSIP/uplink-00000094", "PJSIP/+31xxxxxxxxx@pathfinder") in new stack -- Called PJSIP/+31xxxxxxxxx@pathfinder -- Now forwarding PJSIP/uplink-00000094 to 'PJSIP/pathfinder/sip:+31xxxxxx...@sip.xxxxxx.nl' (thanks to PJSIP/pathfinder-00000095) ... -- PJSIP/pathfinder-00000096 answered PJSIP/uplink-00000094 -- Channel PJSIP/pathfinder-00000096 joined 'simple_bridge' basic-bridge <1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a> -- Channel PJSIP/uplink-00000094 joined 'simple_bridge' basic-bridge <1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a> ... -- Channel PJSIP/pathfinder-00000096 left 'simple_bridge' basic-bridge <1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a> -- Channel PJSIP/uplink-00000094 left 'simple_bridge' basic-bridge <1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a> 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? -- _____________________________________________________________________ -- 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