Hi Brad, I guess you are doing something funny in the script like allowing the 302 reply to be relaid out, but having the 503 generated by opensips - by chance, do you send the 503 in stateless mode ?
Regards, Bogdan Brad Bendy wrote: > Hi Bogdan, > > Here is a full trace, breakdown is like this > > .2 INVITES to .164 > .164 INVITES TO .168 > .168 sends a 302 to .164 > .164 sends .2 a 503 followed by a 302 > > .2 should never know about the 302 at all, but it's still getting back > to the originating proxy. > > We are not using get_redirects() to do anything with the 302 - from > some Googling and such it appears that might be needed, just not sure > how it would be used. > > Thanks for looking at this. > > 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060 > INVITE sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>. > Contact: <sip:[email protected]>. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > User-Agent: None. > Max-Forwards: 70. > Remote-Party-ID: "Test" <sip:[email protected]>;privacy=off;screen=no. > Date: Tue, 24 Aug 2010 12:01:35 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 281. > . > v=0. > o=root 2921 2921 IN IP4 69.xxx.xxx.2. > s=session. > c=IN IP4 69.xxx.xxx.2. > t=0 0. > m=audio 12570 RTP/AVP 18 0 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > 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 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060 > SIP/2.0 100 Giving a try. > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport=5060. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > Server: OpenSIPS (1.6.2-notls (x86_64/freebsd)). > Content-Length: 0. > . > > > U 72.xxx.xxx.164:5060 -> 72.xxx.xxx.168:5060 > INVITE sip:[email protected] SIP/2.0. > Record-Route: <sip:72.xxx.xxx.164;lr=on;ftag=as4f36ab60;did=36f.3f41d571>. > Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0. > Via: SIP/2.0/UDP > 69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>. > Contact: <sip:[email protected]>. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > User-Agent: None. > Max-Forwards: 69. > Remote-Party-ID: "Test" <sip:[email protected]>;privacy=off;screen=no. > Date: Tue, 24 Aug 2010 12:01:35 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 281. > . > v=0. > o=root 2921 2921 IN IP4 69.xxx.xxx.2. > s=session. > c=IN IP4 69.xxx.xxx.2. > t=0 0. > m=audio 12570 RTP/AVP 18 0 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > 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 72.xxx.xxx.168:5060 -> 72.xxx.xxx.164:5060 > SIP/2.0 100 Giving a try. > Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0. > Via: SIP/2.0/UDP > 69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > Server: OpenSIPS (1.6.2-notls (i386/freebsd)). > Content-Length: 0. > . > > > U 72.xxx.xxx.168:5060 -> 72.xxx.xxx.164:5060 > SIP/2.0 302 Moved Temporarily. > Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0. > Via: SIP/2.0/UDP > 69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>;tag=1235203116. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > Content-Type: application/csv. > Contact: sip:rn=6024810000;npdi;[email protected]. > User-Agent: eXosip/3.1.0. > Content-Length: 0. > . > > > U 72.xxx.xxx.164:5060 -> 72.xxx.xxx.168:5060 > ACK sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > Call-ID: [email protected]. > <mailto:[email protected].> > To: <sip:[email protected]>;tag=1235203116. > CSeq: 102 ACK. > Max-Forwards: 70. > User-Agent: OpenSIPS (1.6.2-notls (x86_64/freebsd)). > Content-Length: 0. > . > > > U 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060 > SIP/2.0 503 No more routes > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport=5060. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: > <sip:[email protected]>;tag=f254695ad980185f5ba46cc313375d56.4b85. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > Server: OpenSIPS (1.6.2-notls (x86_64/freebsd)). > Content-Length: 0. > . > > > U 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060 > SIP/2.0 302 Moved Temporarily. > Via: SIP/2.0/UDP > 69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>;tag=1235203116. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 INVITE. > Content-Type: application/csv. > Contact: sip:rn=6024810000;npdi;[email protected]. > User-Agent: eXosip/3.1.0. > Content-Length: 0. > . > > > U 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060 > ACK sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: > <sip:[email protected]>;tag=f254695ad980185f5ba46cc313375d56.4b85. > Contact: <sip:[email protected]>. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 ACK. > User-Agent: None. > Max-Forwards: 70. > Remote-Party-ID: "Test" <sip:[email protected]>;privacy=off;screen=no. > Content-Length: 0. > . > > > U 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060 > ACK sip:[email protected] SIP/2.0. > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport. > From: "Test" <sip:[email protected]>;tag=as4f36ab60. > To: <sip:[email protected]>;tag=1235203116. > Contact: <sip:[email protected]>. > Call-ID: [email protected]. > <mailto:[email protected].> > CSeq: 102 ACK. > User-Agent: None. > Max-Forwards: 70. > Remote-Party-ID: "Test" <sip:[email protected]>;privacy=off;screen=no. > Content-Length: 0. > > On Tue, 2010-08-24 at 10:57 +0300, Bogdan-Andrei Iancu wrote: >> Hi Brad, >> >> Maybe I do not fully understand your case, but opensips is not sending a >> 302 after 200 OK...Maybe you can post the call flow (a SIP trace) from >> the SIP server showing the entire scenario. >> >> Regards, >> Bogdan >> >> Brad Bendy wrote: >> > Hi, >> > >> > Im having a heck of a time figuring this out: >> > >> > INVITE comes to our switch, we send a INVITE to another proxy that >> > responds with a 302, we parse that 302 in failure route then use a >> > route() command to go to another route block which does some other >> > processing (will send out more INVITE's, do certain things on failure, >> > etc), if the original call does get canceled or completes successfully >> > with a 200 OK the originating proxy receives the original 302 request >> > plus what ever our final failure response code we want to send. >> > >> > The behavior does seem correct as openSIPs is just forwarding the >> > 302, but in this case I want it to send only the final response code >> > back to the originating client. >> > >> > The initital route block which sends the INVITE to get the 302 is very >> > simple, we just write the rU and rd and send via t_relay, >> > onreply_route does a little parsing then failure_route sends to a new >> > block. >> > >> > Any help on this would be great, I think it's my logic in the switch >> > that is wrong somewhere. >> > >> > Thanks! >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > Users mailing list >> > [email protected] <mailto:[email protected]> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> >> >> > -- > Brad Bendy > Chief Technical Officer > [email protected] <mailto:[email protected]> > > Benga Networks, LLC. > 10115 E. Bell Rd, Ste. 107-451 > Scottsdale, AZ 85260-2189 > > Toll Free: 877-44-BENGA > Local: 480-970-5200 > Cell: 602-550-4004 > Fax: 866-852-4468 > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
