I'm using v11.11
I tried setting: force_avp=yes in a SIP peer in sip.conf and it seems to be ignored. The WebRTC client sends an INVITE with "RTP/SAVPF" and Asterisk is still sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string Are there some limitations with this option or does it depend on any other settings? Is there any debugging I can enable to understand what is going wrong? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
