I think the only thing you can do is (if this path is fixed), to simply ignore the Route headers and to do a static routing.
Regards, Bogdan Julien Chavanton wrote: > I think I will try the option to use the "textops" module to enforce > the correct order of Record-Route to validate this is my problem etc. > > > > ------------------------------------------------------------------------ > *From:* [email protected] on behalf of Julien Chavanton > *Sent:* Thu 30/04/2009 3:44 PM > *To:* Bogdan-Andrei Iancu > *Cc:* [email protected] > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route > > thank you, this is a problem as I do not control this proxy (2.2.2.2), > is there a suggested way of handling this problem ? > > Maybe there is something esle wrong on my side cusaing this problem so > I am including the SIP communication between the proxy this time > > > > # > U 1.1.1.1:5060 -> 2.2.2.2:5060 > INVITE sip:[email protected] SIP/2.0. > Record-Route: <sip:1.1.1.1;lr>. > Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0. > Via: SIP/2.0/UDP > 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366. > Max-Forwards: 69. > Contact: <sip:[email protected]:58366>. > To: "15141234567"<sip:[email protected]>. > From: "777"<sip:[email protected]>;tag=a030735d. > Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc.. > CSeq: 1 INVITE. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO. > Content-Type: application/sdp. > User-Agent: eyeBeam release 1003s stamp 31159. > Content-Length: 478. > P-hint: Route[6]: mediaproxy . > . > v=0. > o=- 8 2 IN IP4 10.0.1.74. > s=CounterPath eyeBeam 1.5. > c=IN IP4 1.1.1.1. > t=0 0. > m=audio 52550 RTP/AVP 0 8 18 101. > a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006. > a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006. > a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006. > a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006. > a=fmtp:18 annexb=no. > a=fmtp:101 0-15. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575. > a=direction:active. > # > U 2.2.2.2:5060 -> 1.1.1.1:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060. > Via: SIP/2.0/UDP > 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366. > To: "15141234567" <sip:[email protected]>. > From: "777" <sip:[email protected]>;tag=a030735d. > Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc.. > CSeq: 1 INVITE. > Contact: <sip:[email protected]>. > Content-Length: 0. > Record-Route: <sip:1.1.1.1;lr>. > User-Agent: Packetrino. > Supported: replaces. > Record-Route: <sip:2.2.2.2:5060;lr>. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > . > > > ------------------------------------------------------------------------ > *From:* Bogdan-Andrei Iancu [mailto:[email protected]] > *Sent:* Thu 30/04/2009 3:44 PM > *To:* Julien Chavanton > *Cc:* [email protected] > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route > > Hi Julian, > > Julien Chavanton wrote: > > > > > > UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA > > > > P1 --> P2 > > INVITE > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes> > > > > P2 --> P1 > > 100 Trying > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes> > > Record-Route: <sip:2.2.2.2:5060;lr> > > > ^^^^^^^^^^^^ > > This is not correct. The RR of P2 most me on top of RR of P1 - adding RR > headers works as a stack. > > Regards, > Bogdan > > > > Is there something wrong ? shouldn't proxy 2.2.2.2 add his > > Record-Route on top of the existing Record-Route ? > > > > ------------------------------------------------------------------------ > > *From:* Bogdan-Andrei Iancu [mailto:[email protected]] > > *Sent:* Thu 30/04/2009 8:12 AM > > *To:* Julien Chavanton > > *Cc:* [email protected] > > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route > > > > Hi Julien, > > > > I think Asterisk is doing the job properly. As you see the 200 OK has: > > Contact: <sip:[email protected]:5060>. > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > Record-Route: <sip:2.2.2.2:5060;lr>. > > > > So, Asterisk is generating the ACK with the Contact in RURI and the > > Route set in the reverted order (correct loose routing). > > -> RURI: sip:[email protected]:5060 > > Destination: sip:2.2.2.2:5060;lr > > Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes > > > > I think the problem here is who and why adding the bottom RR in 200 OK > > (why 2 of them ?) > > > > Regards, > > Bogdan > > > > Julien Chavanton wrote: > > > > > > Hi, > > > > > > I have a situation whit multiple proxy where ACK is not sent as I > > > would expect. > > > > > > if we look at the following "200 OK", I am expecting ACK to be sent to > > > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this > > > normal ? > > > > > > Do I have to handle Record-Route differently ? > > > > > > > > > > > > > > > > > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > > SIP/2.0 200 OK. > > > Via: SIP/2.0/UDP > > > > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > > To: <sip:[email protected]>;tag=as664de2c2. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 INVITE. > > > Content-Type: application/sdp. > > > Contact: <sip:[email protected]:5060>. > > > Content-Length: 241. > > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > > User-Agent: Packetrino. > > > Supported: replaces. > > > Record-Route: <sip:2.2.2.2:5060;lr>. > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > --------------------------------------------------------- > > > > > > complete SIP signaling > > > > > > --------------------------------------------------------- > > > > > > # > > > U 192.168.1.108:5060 -> 1.1.1.1:5060 > > > INVITE sip:[email protected] SIP/2.0. > > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport. > > > Max-Forwards: 70. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > To: <sip:[email protected]>. > > > Contact: <sip:[email protected]>. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 INVITE. > > > User-Agent: Asterisk PBX 1.6.0.6. > > > Date: Wed, 29 Apr 2009 15:38:18 GMT. > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > > Supported: replaces, timer. > > > Content-Type: application/sdp. > > > Content-Length: 265. > > > . > > > v=0. > > > o=root 1992389746 1992389746 IN IP4 192.168.1.108. > > > s=Asterisk PBX 1.6.0.6. > > > c=IN IP4 192.168.1.108. > > > t=0 0. > > > m=audio 11232 RTP/AVP 0 101. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:101 telephone-event/8000. > > > a=fmtp:101 0-16. > > > a=silenceSupp:off - - - -. > > > a=ptime:20. > > > a=sendrecv. > > > > > > # > > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > > SIP/2.0 100 Giving a try. > > > Via: SIP/2.0/UDP > > > > > > 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > To: <sip:[email protected]>. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 INVITE. > > > Server: OpenSIPS (1.4.4-notls (x86_64/linux)). > > > Content-Length: 0. > > > . > > > > > > # > > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > > SIP/2.0 183 Session Progress. > > > Via: SIP/2.0/UDP > > > > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > > To: <sip:[email protected]>;tag=as664de2c2. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 INVITE. > > > Content-Type: application/sdp. > > > Contact: <sip:[email protected]:5060>. > > > Content-Length: 241. > > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > > User-Agent: Packetrino. > > > Supported: replaces. > > > Record-Route: <sip:2.2.2.2:5060;lr>. > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > > . > > > v=0. > > > o=root 29378 29378 IN IP4 64.2.142.160. > > > s=session. > > > c=IN IP4 1.1.1.1. > > > t=0 0. > > > m=audio 52528 RTP/AVP 0 101. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:101 telephone-event/8000. > > > a=fmtp:101 0-16. > > > a=silenceSupp:off - - - -. > > > a=ptime:20. > > > a=sendrecv. > > > > > > # > > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > > SIP/2.0 180 Ringing. > > > Via: SIP/2.0/UDP > > > > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > > To: <sip:[email protected]>;tag=as664de2c2. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 INVITE. > > > Contact: <sip:[email protected]:5060>. > > > Content-Length: 0. > > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > > User-Agent: Packetrino. > > > Supported: replaces. > > > Record-Route: <sip:2.2.2.2:5060;lr>. > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > > . > > > > > > # > > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > > SIP/2.0 200 OK. > > > Via: SIP/2.0/UDP > > > > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > > To: <sip:[email protected]>;tag=as664de2c2. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 INVITE. > > > Content-Type: application/sdp. > > > Contact: <sip:[email protected]:5060>. > > > Content-Length: 241. > > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > > User-Agent: Packetrino. > > > Supported: replaces. > > > Record-Route: <sip:2.2.2.2:5060;lr>. > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > > . > > > v=0. > > > o=root 29378 29379 IN IP4 64.2.142.160. > > > s=session. > > > c=IN IP4 1.1.1.1. > > > t=0 0. > > > m=audio 52528 RTP/AVP 0 101. > > > a=rtpmap:0 PCMU/8000. > > > a=rtpmap:101 telephone-event/8000. > > > a=fmtp:101 0-16. > > > a=silenceSupp:off - - - -. > > > a=ptime:20. > > > a=sendrecv. > > > > > > # > > > U 192.168.1.108:5060 -> 2.2.2.2:5060 > > > ACK sip:[email protected]:5060 SIP/2.0. > > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport. > > > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>. > > > Max-Forwards: 70. > > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > > To: <sip:[email protected]>;tag=as664de2c2. > > > Contact: <sip:[email protected]>. > > > Call-ID: [email protected] > > > <mailto:[email protected]>. > > > CSeq: 102 ACK. > > > User-Agent: Asterisk PBX 1.6.0.6. > > > Content-Length: 0. > > > . > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > [email protected] > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
