thanks for all the support for all this years.<div><br></div><div>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.</div>
<div><br></div><div>my scenario is calls coming from different devices registering to kamailio and then kamailio send those calls to asterisk.</div><div>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.</div>
<div><br></div><div>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.</div>
<div><br></div><div>I appreciated any input in this matter.</div><div><br></div><div>and again thank a lot to all for the great support.</div><div><br></div><div>Andres Collazos.</div>