Hello Razvan, thank you for your help, i check about this function before, i will try that and i let you know if solve , thank you and happy hollidays
2013/12/24 Răzvan Crainea <[email protected]> > Hi, Mike! > > Have you tried matching the dialogs using the match_dialog() function[1]? > Also, for sequential requests, you should try using the fix_route_dialog() > function[2]. > > [1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144 > [2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295287 > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > On 12/21/2013 06:25 PM, Mike Tesliuk wrote: > >> Hello Guys, >> >> >> Im getting a strange situation here that i dont know how to deal >> >> i have an enviroment where i have freeswitch receiving a call to billing >> and opensips doing the load_balance to the gateways. >> >> When i send the call to the gateway i receive the reply without the >> record-route header, i try to put an asterisk server as gateway and >> this not happen in this scenario . >> >> Below the invite that i send to the gateway >> >> U 10.1.69.1:5079 <http://10.1.69.1:5079> -> 10.255.2.31:5031 >> <http://10.255.2.31:5031> >> INVITE sip:[email protected]:5079 >> <http://sip:[email protected]:5079> SIP/2.0. >> >> Record-Route: <sip:10.1.69.1:5079;lr;ftag=HgcSt10Xa854e;did=9d2. >> 723c6252>. >> Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0. >> Via: SIP/2.0/UDP >> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage. >> Max-Forwards: 68. >> From: "200214" <sip:[email protected] >> <mailto:sip%[email protected]>>;tag=HgcSt10Xa854e. >> To: <sip:[email protected]:5079 >> <http://sip:[email protected]:5079>>. >> >> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4. >> CSeq: 53458861 INVITE. >> Contact: <sip:[email protected]:5069;transport=udp;gw=os>. >> User-Agent: vBilling. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, NOTIFY. >> Supported: precondition, path, replaces. >> Allow-Events: talk, hold, conference, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 195. >> X-FS-Support: update_display,send_info. >> Remote-Party-ID: "200214" <sip:[email protected] >> <mailto:sip%[email protected]>>;party=calling;screen=yes;privacy=off. >> >> >> >> and below the 200 ok that i receive >> >> U 10.255.2.31:5031 <http://10.255.2.31:5031> -> 10.1.69.1:5079 >> <http://10.1.69.1:5079> >> >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP >> 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0;received=10.1.69.1. >> Via: SIP/2.0/UDP >> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage. >> To: <sip:[email protected]:5079 >> <http://sip:[email protected]:5079>>;tag=12ab34cd. >> From: "200214" <sip:[email protected] >> <mailto:sip%[email protected]>>;tag=HgcSt10Xa854e. >> >> CSeq: 53458861 INVITE. >> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4. >> Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER. >> Supported:. >> Allow-Events: telephone-event. >> Contact: <sip:[email protected]:5031;transport=udp>. >> Content-Type: application/sdp. >> Content-Length: 196. >> >> when i send the call to this gateway the loose route did not execute, >> and i get error's on dialog because the dialog is not matched >> >> >> how should i deal with a situation like this ? >> >> >> >> >> >> >> >> _______________________________________________ >> 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 >
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