[SR-Users] Hi Daniel.

[graham]

I am no Daniel, but here is a stab at it ...

The pseudo variable ³oP² maybe something you can do a quick if/else on?

Taken from http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.5.x :

        Transport protocol of SIP request original URI
       $oP - reference to transport protocol of original R-URI

There is also dP and rP as well.

Hope this helps,

-graham

On 3/16/12 12:11 PM, "Andres Collazos" <anfecora at gmail.com> wrote:

> thanks for all the support for all this years.
> 
> Can you please help me to know  if there is any way to route sip calls based
> on transport protocol, for example a call incoming on tcp i will assign a
> route and if a call comes in udp i will assign a different route.
> 
> my scenario is calls coming from different devices registering to kamailio and
> then kamailio send those calls to asterisk.
> unfortunately i have to create a different peer set for each device. for this
> scenario i have two types of UAs and they need completely different
> configuration on the asterisk switch.
> 
> i am planing on segregate the traffic and build a media server for each type
> of device means 2 asterisk, teh only way that i can identify those incoming
> registrations is by the use of the port one client connects trough udp the
> other trough tcp.
> 
> I appreciated any input in this matter.
> 
> and again thank a lot to all for the great support.
> 
> Andres Collazos.
> 
> 
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