[OpenSIPS-Users] No record-route on reply's

[Răzvan Crainea]

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:255755813256 at 10.1.69.1:5079
> <http://sip:255755813256@10.1.69.1:5079> 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:200214 at 10.1.69.1
> <mailto:sip%3A200214 at 10.1.69.1>>;tag=HgcSt10Xa854e.
> To: <sip:255755813256 at 10.1.69.1:5079
> <http://sip:255755813256@10.1.69.1:5079>>.
> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
> CSeq: 53458861 INVITE.
> Contact: <sip:gw+os at 10.1.69.1:5069;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:200214 at 10.1.69.1
> <mailto:sip%3A200214 at 10.1.69.1>>;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:255755813256 at 10.1.69.1:5079
> <http://sip:255755813256@10.1.69.1:5079>>;tag=12ab34cd.
> From: "200214" <sip:200214 at 10.1.69.1
> <mailto:sip%3A200214 at 10.1.69.1>>;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:255755813256 at 10.1.69.1:5031;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 ?
>
>
>
>
>
>
>
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