Next time please send only the trace of the relevant SIP dialog (between provider and Kamailio/Asterisk). Ther seconds dialog started by Asterisk is not relevant.
The problem is rather simple:
U 2011/03/15 15:43:48.237614 6.1.1.1:5060 -> 5.1.1.1:5060 INVITE sip:1231234@domain.com SIP/2.0 Record-Route: sip:6.1.1.1;lr=on;ftag=B0432A3C-37B Via: SIP/2.0/UDP 6.1.1.1;branch=z9hG4bK4e1f.614446c3.0 Via: SIP/2.0/UDP 6.1.1.2:5060;branch=z9hG4bK47ABD036B Remote-Party-ID: sip:1231000@6.1.1.2;party=calling;screen=yes;privacy=off From: "1231000" sip:1231000@6.1.1.2;tag=B0432A3C-37B To: sip:1231234@ire.e164.org.uk Date: Tue, 15 Mar 2011 15:43:48 gmt Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B@6.1.1.2 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 14 Timestamp: 1300203828 Contact: sip:1231000@6.1.1.2:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 417
v=0 o=CiscoSystemsSIP-GW-UserAgent 4797 428 IN IP4 6.1.1.2 s=SIP Call c=IN IP4 6.1.1.2 t=0 0 m=audio 23382 RTP/AVP 8 18 4 3 98 0 101 c=IN IP4 6.1.1.2 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16
U 2011/03/15 15:43:48.246612 5.1.1.1:5060 -> 6.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 6.1.1.1;branch=z9hG4bK4e1f.614446c3.0;rport=5060 Via: SIP/2.0/UDP 6.1.1.2:5060;branch=z9hG4bK47ABD036B From: "1231000" sip:1231000@6.1.1.2;tag=B0432A3C-37B To: sip:1231234@ire.e164.org.uk Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B@6.1.1.2 CSeq: 101 INVITE Server: kamailio (3.1.2 (i386/linux)) Content-Length: 0
U 2011/03/15 15:43:48.248371 1.2.3.3:5060 -> 1.2.3.1:5060 INVITE sip:1231234@domain.com SIP/2.0 Record-Route: sip:1.2.3.3;r2=on;lr=on;ftag=B0432A3C-37B Record-Route: sip:5.1.1.1;r2=on;lr=on;ftag=B0432A3C-37B Record-Route: sip:6.1.1.1;lr=on;ftag=B0432A3C-37B Via: SIP/2.0/UDP 1.2.3.3;branch=z9hG4bK4e1f.576ffdd1.0 Via: SIP/2.0/UDP 6.1.1.1;rport=5060;branch=z9hG4bK4e1f.614446c3.0 Via: SIP/2.0/UDP 6.1.1.2:5060;branch=z9hG4bK47ABD036B Remote-Party-ID: sip:1231000@6.1.1.2;party=calling;screen=yes;privacy=off From: "1231000" sip:1231000@6.1.1.2;tag=B0432A3C-37B To: sip:1231234@ire.e164.org.uk Date: Tue, 15 Mar 2011 15:43:48 gmt Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B@6.1.1.2 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: MSSGW Allow: INVITE, BYE, CANCEL, ACK CSeq: 101 INVITE Max-Forwards: 13 Timestamp: 1300203828 Contact: sip:1231000@6.1.1.1:5060
^^^^^^^^^^^^^^^^
Here, Kamailio changed the received contact. As there is another proxy between the UAC and Kamailio, Kamailio must not modify the contact. (remove fix_nated_contact() for requests coming from the service provider)
By changing the contact, the BYE gets looped in the provider's Openser proxy until the message gets rejected due to the size.
regards Klaus
On 15.03.2011 18:16, Asgaroth wrote:
On 15/03/2011 14:29, Klaus Darilion wrote:
I prefer for ngrep traces (you could replace usernames/IP-addresses)
I have attached 2 text files of ngrep traces. Both are of the same call, one trace was performed at the asterisk media server, and the other was performed at the proxy. I've replaced all user names and ip address.
Just for info the IP are as follows:
1.2.3.1 = Asterisk media server (running asterisk 1.8.3) 1.2.3.2 = Kamailio location server (running kamailio 3.1.2) 1.2.3.3 = Kamailio proxy server (internal interface) (running kamailio 3.1.2) 5.1.1.1 = Kamailio proxy server (external interface) (running kamailio 3.1.2) 6.1.1.1 = Provider proxy server 6.1.1.2 = Provider media gateway
Please let me know if you require any additional information.
Thank you for taking the time to take a look at the traces.
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