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" <anfec...@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. > > > _______________________________________________ > 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
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