Re: [SR-Users] Older Sip Phone ia hangup after 60sec with Realtime asterisk

Hello, I am beginner using kamailio with much appreciated. Only one sip-phone is hang up after 60 seconds problem. This sip phone has no nat function at all.(SANYO SIP-2100) Grand Stream is works fine with kamailio. I would like give me your great advice with much appreciated. Environment. CentOS5.10, Asterisk-11.6.0 with PostgreSQL-9.2.5 as Realtime. Kamailio-4.1.0 Only Asterisk and PostgreSQL with older sip phone works fine. If Kamailio is running that registered is OK, But meetme(example) is hangup after 60 sec. I do not know "reINVITE or RTP" problem. [...] *** Test call to meetme Logs. **** sip1*CLI> sip set debug on sip1*CLI> SIP Debugging re-enabled sip1*CLI> sip set debug on sip1*CLI> Name/username Host Dyn Forcerport ACL Port Status Description Realtime 99206/99206 192.168.192.92 D N 5060 OK (515 ms) Cached RT 1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline] sip1*CLI> -- Executing [901@99:1] Answer("SIP/99206-00000000", "") Audio is at 15506 sip1*CLI> Adding codec 100003 (ulaw) to SDP sip1*CLI> Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP sip1*CLI> <--- Reliably Transmitting (NAT) to 192.168.192.92:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9 Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes> From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 INVITE Session-Expires: 120;refresher=uas Contact: <sip:901@192.168.192.92:5080> Content-Type: application/sdp Require: timer Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> sip1*CLI> Retransmitting #1 (NAT) to 192.168.192.92:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.538bb4a63719b3291fac2770ec3b5b31.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29932-3be9 Record-Route: <sip:192.168.192.92;lr=on;ftag=bec0a8c0-13c4-52cad076-29850-7422;nat=yes> From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 INVITE Session-Expires: 120;refresher=uas Contact: <sip:901@192.168.192.92:5080> Content-Type: application/sdp Require: timer Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> ACK sip:901@192.168.192.92:5080 SIP/2.0 From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e Max-Forwards: 16 Contact: <sip:99206@192.168.192.190:5060> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901@192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5 Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> ACK sip:901@192.168.192.92:5080 SIP/2.0 From: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 To: <sip:901@192.168.192.92>;tag=as7cd1f3fc Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.192.92;branch=z9hG4bKac02.9f229e8f27e7aa8e3fad6cc83135f434.0 Via: SIP/2.0/UDP 192.168.192.190:5060;rport=5060;branch=z9hG4bK-52cad077-29a6e-306e Max-Forwards: 16 Contact: <sip:99206@192.168.192.190:5060> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901@192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5 Content-Length:0 <-------------> --- (11 headers 0 lines) --- sip1*CLI> -- Executing [901@99:2] Wait("SIP/99206-00000000", "1") sip1*CLI> > 0x17aa0bd0 -- Probation passed - setting RTP source address to 192.168.192.190:17096 sip1*CLI> -- Executing [901@99:3] Authenticate("SIP/99206-00000000", "5963") sip1*CLI> -- <SIP/99206-00000000> Playing 'agent-pass.gsm' (language 'ja') sip1*CLI> -- <SIP/99206-00000000> Playing 'auth-thankyou.gsm' (language 'ja') sip1*CLI> -- Executing [901@99:4] MeetMe("SIP/99206-00000000", "99901,pM") == Parsing '/etc/asterisk/meetme.conf': Found sip1*CLI> -- Created MeetMe conference 1023 for conference '99901' sip1*CLI> -- <SIP/99206-00000000> Playing 'conf-onlyperson.gsm' (language 'ja') sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000 sip1*CLI> -- Stopped music on hold on SIP/99206-00000000 sip1*CLI> -- Started music on hold, class 'default', on SIP/99206-00000000 sip1*CLI> Audio is at 15506 Adding codec 100003 (ulaw) to SDP Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.192.92:5060: INVITE sip:99206@192.168.192.190:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport Max-Forwards: 70 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Contact: <sip:901@192.168.192.92:5080> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.6.0 Session-Expires: 120;refresher=uac Min-SE: 90 Allow: INVITE, ACK, CANCEL, BYE X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 284 v=0 o=root 729993436 729993436 IN IP4 192.168.192.92 s=Asterisk PBX 11.6.0 c=IN IP4 192.168.192.92 t=0 0 m=audio 15506 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- sip1*CLI> <--- SIP read from UDP:192.168.192.92:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK71165cd1;rport=5080 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 102 INVITE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0 --- <--- SIP read from UDP:192.168.192.92:5060 ---> SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.192.92:5080;branch=z9hG4bK339a9eb8;rport=5080 From: <sip:901@192.168.192.92>;tag=as7cd1f3fc To: Richard Nough<sip:99206@192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404(a)192.168.192.92 CSeq: 103 BYE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0 I hope you have a great 2014. Kind regards, Nori _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users