[SR-Users] Older Sip Phone ia hangup after 60sec with Realtime asterisk
Noriyuki Hayashi
nhayashi at wats.gr.jp
Mon Jan 6 16:50:11 CET 2014
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 at 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 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
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PC-Mail：nhayashi at wats.gr.jp
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