[SR-Users] Older Sip Phone ia hangup after 60sec with Realtime asterisk
Daniel-Constantin Mierla
miconda at gmail.com
Mon Jan 6 12:55:44 CET 2014
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 at 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 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
> CSeq: 2 INVITE
> Session-Expires: 120;refresher=uas
> Contact: <sip:901 at 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 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 192.168.192.92
> CSeq: 2 INVITE
> Session-Expires: 120;refresher=uas
> Contact: <sip:901 at 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 at 192.168.192.92:5080 SIP/2.0
> From: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 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 at 192.168.192.190:5060>
> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901 at 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 at 192.168.192.92:5080 SIP/2.0
> From: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> To: <sip:901 at 192.168.192.92>;tag=as7cd1f3fc
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 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 at 192.168.192.190:5060>
> Proxy-Authorization: Digest username="99206", realm="192.168.192.92", nonce="UspTFFLKUehXxuLrJ9AbLwT69Jtg5MFQ", uri="sip:901 at 192.168.192.92", response="8d49c10f184381801aac43fc45b15117", algorithm=MD5
> Content-Length:0
>
> <------------->
> --- (11 headers 0 lines) ---
> sip1*CLI> -- Executing [901 at 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 at 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 at 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 at 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 at 192.168.192.92>;tag=as7cd1f3fc
> To: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> Contact: <sip:901 at 192.168.192.92:5080>
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 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 at 192.168.192.92>;tag=as7cd1f3fc
> To: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 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 at 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 at 192.168.192.92>;tag=as7cd1f3fc
> To: Richard Nough<sip:99206 at 192.168.192.92>;tag=bec0a8c0-13c4-52cad076-29850-7422
> Call-ID: 105b2ef4-bec0a8c0-13c4-52cad076-2984c-5404 at 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 at 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
More information about the sr-users
mailing list