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.