Hello,

sr-dev mailing list is for discussions about writing code/developing Kamailio. For question on how to use kamailio and how to configure it for various scenarios, send your messages to sr-users@lists.kamailio.org. That list has more members and the developers are also there, so the chance to get better answer is higher.

Cheers,
Daniel

On 09.01.19 08:14, Prashant Gupta wrote:

Hi,

I have the following architecture - SIP provider <-> Kamailio <-> Asterisk servers

Currently I have everything setup and incoming calls from Sip are routed to my asterisk server. The issue is however that when I answer the call, there is no media in the call. I have tried connecting with a normal local extension(not SIP,eg 1001) and there is a normal flow of media.

When i try to sniff my connection via Wireshark on the asterisk server, there is an outflow of RTP packets but the same RTP traffic does not appear on the Wireshark of my Kamailio server connection. 

I am not sure if this is an RTP engine issue and how to resolve this.

I have -

modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038")

this in my kamailio cfg but I don;t know which port to use here.

Any suggestions?

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