Hello,
Thank you for quick reply. Now I have rpmbuilt ngrep and installed. I usually use tcpdump. I will take the log day after tommorow and send. Because I must work on client office tommorow.
Kind regrads, Nori
On Mon, 06 Jan 2014 12:55:44 +0100 Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
can you get the ngrep output on kamailio server? From asterisk log I see that an INVITE with To-tag has no Route header, which should be there if run though kamailio.
Cheers, Daniel
On 06/01/14 08:50, Noriyuki Hayashi wrote:
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 Noughsip: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@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 Noughsip: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@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 Noughsip: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@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 Noughsip: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@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 Noughsip: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@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 Noughsip:99206@192.168.192.92;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404@192.168.192.92 CSeq: 102 INVITE Server: kamailio (4.1.0 (x86_64/linux)) Content-Length: 0
> [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid") VALUES ('2014-01-06 15:49:12','"Richard Nough" <99206>','99206','901','99','SIP/99206-00000000','MeetMe','99901,pM',60,60,'ANSWERED',3,'nori@wats','1388990952.0')]<--- 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 Noughsip:99206@192.168.192.92;tag=bec0a8c0-13c4-52cad076-29850-7422 Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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