[SR-Users] Parallel Forking with Different Call-Id

SamyGo govoiper at gmail.com
Wed Jul 6 16:44:24 CEST 2016


Thanks Daniel,
Both are  good ideas and I will try and see how it goes. I just wanted to
confirm that there is no way we can modify CallID of branches via script.
This is fine too.
Thanks again,

Best Regards,
Sammy
On Jul 6, 2016 06:11, "Daniel-Constantin Mierla" <miconda at gmail.com> wrote:

> Hello,
>
> maybe you can loop through a 2nd instance of kamailio (can be same server,
> different port) that has topoh enabled.
>
> Otherwise, if the gateway is matching on full r-uri, you can try to add
> some extra uri params, which are not relevant for target number.
>
> Cheers,
> Daniel
>
> On 05/07/16 21:54, SamyGo wrote:
>
> Hi,
>
> I've a very strange scenario to work on which requires me to parallel fork
> the call to the same Destination provider. The only problem here is that
> they think that the second INVITE with different branch tag is a
> re-transmission and hence only take one call forward.
>
> I do not have to modify any R-URI or any headers, hence just
> 'append_branch() before t_relay() is in the code.
>
> ...
> $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port);
> append_branch();
> route(RELAY);
> ...
>
> *Question:* Is there anyway possible I can change the CallID of the
> forked INVITEs ?
>
> I have tried using Topoh module, but it still puts the same CallID before
> sending out. Different from the A-leg but the Sent out INVITEs have same
> Call-ID value.
>
> In other weird scenario, I've also tried branching, and looping call
> within Kamailio before sending out to carrier hoping that Kamailio would
> treat the two different calls and Topoh would change the Call-ID on both
> INVITEs before sending out..
>
> ...
> if(is_present_hf("X-FORKED")) {
>     $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port);
>     route(RELAY);
> }else {
>     append_hf("X-FORKED: 1\r\n");
>     append_branch();
>     route(RELAY);
> }
> ...
>
> Still I get same Call-ID on outgoing branched call.
>
> I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk
> but again I don't want to have that component bottleneck the throughput.
> That could be my very last option.
>
>
> Looking for some ideas.
>
> Regards,
> Sammy
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
> _______________________________________________
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