Hi again, Still Missing 200OK for this call. It'll be helpful to send a complete trace for the call coming in to the Asterisk at first place and then Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessen...@gmail.com> wrote: > > > <------------> > Dial (.......) in new stack > > > == Using SIP RTP CoS mark 5 > Audio is at 19614 > Adding codec 100003 (ulaw) to SDP > Adding codec 100002 (gsm) to SDP > Adding codec 100004 (alaw) to SDP > Adding codec 100017 (testlaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > > Reliably Transmitting (no NAT) to 2.2.2.2:5060: > INVITE sip:1...@sip1.domain.com.ua SIP/2.0 > Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299 > Max-Forwards: 70 > From: "Bomber" <sip:1...@sip1.domain.com.ua>;tag=as1b8070ba > To: <sip:1...@sip1.domain.com.ua> > Contact: <sip:101@2.2.2.101:5080> > Call-ID: 2ac37537499c919f0168358234952...@sip1.domain.com.ua > CSeq: 102 INVITE > User-Agent: Asterisk > Date: Tue, 06 Aug 2013 10:18:03 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 319 > > v=0 > o=root 1885227245 1885227245 IN IP4 2.2.2.101 > s=Asterisk > c=IN IP4 2.2.2.101 > t=0 0 > m=audio 19614 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > -- Called SIP/1...@sip1.domain.com.ua > > <--- Transmitting (no NAT) to 2.2.2.2:5060 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 > Via: SIP/2.0/UDP 192.168.10.240:52396 > ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- > Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes> > Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes> > From: "101" <sip:1...@sip1.domain.com.ua > >;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD > To: "Bomber" <sip:1...@sip1.domain.com.ua>;tag=as1bd39f9d > Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 > CSeq: 10050 INVITE > Server: Asterisk > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Session-Expires: 1800;refresher=uas > Contact: <sip:101@2.2.2.101:5080> > Content-Length: 0 > > > <------------> > > <--- SIP read from UDP:2.2.2.2:5060 ---> > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080 > From: "Bomber" <sip:1...@sip1.domain.com.ua>;tag=as1b8070ba > To: <sip:1...@sip1.domain.com.ua> > Call-ID: 2ac37537499c919f0168358234952...@sip1.domain.com.ua > CSeq: 102 INVITE > Server: kamailio (4.0.2 (x86_64/linux)) > Content-Length: 0 > > <-------------> > --- (8 headers 0 lines) --- > > <--- SIP read from UDP:2.2.2.2:5060 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299 > Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes> > Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes> > Call-ID: 2ac37537499c919f0168358234952...@sip1.domain.com.ua > From: "Bomber" <sip:1...@sip1.domain.com.ua>;tag=as1b8070ba > To: <sip:1...@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV > CSeq: 102 INVITE > Contact: "101" <sip:1...@client.gw.pub.ip:17303;ob> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, > MESSAGE, OPTIONS > Content-Length: 0 > > <-------------> > --- (11 headers 0 lines) --- > list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes> > list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes> > -- SIP/sip1.domain.com.ua-0000050f is ringing > > <--- Transmitting (no NAT) to 2.2.2.2:5060 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2 > Via: SIP/2.0/UDP 192.168.10.240:52396 > ;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5- > Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes> > Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes> > From: "101" <sip:1...@sip1.domain.com.ua > >;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD > To: "Bomber" <sip:1...@sip1.domain.com.ua>;tag=as1bd39f9d > Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9 > CSeq: 10050 INVITE > Server: Asterisk > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Session-Expires: 1800;refresher=uas > Contact: <sip:101@2.2.2.101:5080> > Content-Length: 0 > > > <------------> > > _______________________________________________ > 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 > >
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