Thanks Daniel.

Here is the use case.

An INVITE received from an upstream Proxy to Kamailio.  Kamailio needs to save the Record-Route header from the INVITE before proxying it to an Asterisk server and  when  the Asterisk server sends a BYE for that INVITE back thru Kamailio,  Kamailio is required to attach the Record-Route as a Route header in the BYE message to the upstream Proxy.

This way the upstream Proxy can route the BYE.

Hope this makes it a bit clearer.

Thanks, --Jignesh

________________________________ From: Daniel-Constantin Mierla miconda@gmail.com To: jignesh gandhi jhgandhi@yahoo.com; Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Sent: Monday, April 28, 2014 4:32 AM Subject: Re: [SR-Users] Record-Route header

Hello,

$hdr(Record-Route) should return the record-route header value. You can store it in htable using callid as key.

However, if you explain the issues you try to solve, there might be other alternatives.

Cheers, Daniel

On 25/04/14 22:00, jignesh gandhi wrote:

Hello All,

I have a need to save the Record-Route header from the INVITE and  use it as a route header in requests like BYE.

PROXY --> Kamailio(as PROXY) --> Asterisk

I am using rr module to add a record_route when passing the INVITE to Asterisk.

Thanks, Jignesh Gandhi

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Hello,

yes -- if you use t_relay() to send out the SIP requests, then kamailio acts as a stateful proxy.

If you use record_route() and loose_route() in kamailio.cfg (the default kamailio.cfg uses them, so you can look at it if you started something from scratch), then requests within dialog end to proper asterisk.

Cheers, Daniel

On 28/03/16 06:26, jignesh gandhi wrote:

Hello All,

Is it possible to use kamailio to be a stateful outbound proxy ?

The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them.

After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ?

I know on the inbound side I can hash over the call-id to ensure that all requests within a dialogue are routed to same Asterisk instance.

Thanks, --Jignesh

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users