[sr-dev] kamailio SIP and RTP proxy

[Patrick Leybag]

Hi, Can someone help me?
I self host a kamailio using my raspberry pi as a load balancer for my two
asterisk servers and get a did number. when I call to my DID number it
points to my kamailio and kamailio will distribute to asterisk server but
the call has no audio. I tried port forwarding ports 5060 for SIP and
10000-20000 for RTP but it still does not work.

Any help is much appreciated. Thank you in advance
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