On Tue, Mar 10, 2015 at 6:11 PM, Chirag Desai <[email protected]> wrote: > OK, it stopped working. > > It turns out the transport and endpoints in PJSIP are ok. I can send an > invite from my unregistered snom phone and I can see some activity in the > CLI. > > However, when I dial from my snom to Kamailio and have it pass the message > to asterisk, PJSIP seems to ignore the sip messages even though they are > there. > > Is there something wrong in the invite that I'm missing? > > U 2015/03/10 22:47:43.539208 [kamailio public ip]:5060 -> [asterisk public > ip]:5061 > INVITE sip:[email protected];user=phone SIP/2.0. > Record-Route: <sip:[kamailio public ip];r2=on;lr=on;nat=yes>. > Record-Route: <sip:[kamailio public ip];transport=tcp;r2=on;lr=on;nat=yes>. > Via: SIP/2.0/UDP 1 > [kamailio public > ip];branch=z9hG4bKc10a.a307d27e5d7581c259704fcd865a69e2.0;i=1. > Via: SIP/2.0/TCP > [snomprivateip]:47153;received=[snompublicip];branch=z9hG4bK-cc9ldmdhvvdi;rport=47473. > From: <sip:[email protected]>;tag=tu0if9akzq. > To: <sip:[email protected];user=phone>. > Call-ID: 8d74ff54e076-hajfjxwp1crj. > CSeq: 2 INVITE. > Max-Forwards: 16. > Contact: > <sip:1000@[snom_private_ip]:47153;alias=[snom_public_ip]~47473~2;transport=tcp;line=snl8cukk>;reg-id=1. > X-Serialnumber: [snom_mac_address]. > P-Key-Flags: resolution="31x13", keys="4". > User-Agent: snom760/8.7.3.25. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO, UPDATE. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Session-Expires: 3600;refresher=uas. > Min-SE: 90. > Content-Type: application/sdp. > Content-Length: 598. > > . > v=0. > o=root 1667335791 1667335791 IN IP4 [snom_private_ip]. > s=call. > c=IN IP4 [snom_private_ip]. > t=0 0. > m=audio 59358 RTP/AVP 9 0 8 3 97 98 99 100 106 107 108 109 18 101. > a=rtpmap:9 G722/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 G726-16/8000. > a=rtpmap:98 G726-24/8000. > a=rtpmap:99 G726-32/8000. > a=rtpmap:100 G726 > > My transports are: > > [transport-udp] > type=transport > protocol=udp > bind:0.0.0.0:5061 > > > [transport-tcp] > type=transport > protocol=tcp > bind=0.0.0.0:5061 >
If the INVITE request is not shown in the CLI with 'pjsip set logger on', then Asterisk is not actually receiving the request. Does a pcap show the message being sent to the correct IP/port? If you change the transports to bind to port 5060, does that change anything? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- 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
