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
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@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@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@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@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@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@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
Nough<sip: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
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
_______________________________________________
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sr-users@lists.sip-router.org
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