Re: [asterisk-users] No external audio on SIP => PJSIP both behind same NAT
Le 08/04/2019 à 20:06, Administrator TOOTAI a écrit : Hi, I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk 13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also with PJSIP. Both LAN Asteriks are also connected via IAX. Everything is working fine except SIP call from 1.4 to external number: there is no audio. SIP call to eg demo@Asterisk13 is OK. If I replace the SIP link between 1.4 and 13 with the IAX trunk it's OK. Other way is OK in full SIP. I tried few parameters on both Asteriks, no luck. The RTP port range is the same on both instance. If someone had a clue on this, welcome ;) and thanks in advance. Sorry for noise, I had a typo direct-media=no insteed of direct_media=no Daniel -- _ -- 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
[asterisk-users] No external audio on SIP => PJSIP both behind same NAT
Hi, I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk 13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also with PJSIP. Both LAN Asteriks are also connected via IAX. Everything is working fine except SIP call from 1.4 to external number: there is no audio. SIP call to eg demo@Asterisk13 is OK. If I replace the SIP link between 1.4 and 13 with the IAX trunk it's OK. Other way is OK in full SIP. I tried few parameters on both Asteriks, no luck. The RTP port range is the same on both instance. If someone had a clue on this, welcome ;) and thanks in advance. Daniel -- _ -- 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
