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

[Mike Tesliuk]

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 <razvan at opensips.org>

> 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 ?
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20131225/afd853b6/attachment.htm>