Hello,

you have to provide the sip trace taken on kamailio server, capturing the traffic on both sides. What you provided is missing any description and ips of sender and receiver.

As a guess, if you don't get bye on kamailio, very likely you didn't do record_route() for invite.

Cheers,
Daniel

On 25/09/14 12:22, balu wrote:

Hi

I am using kamailio with rtp proxy module. I have 2 questions /issues .

1. When caller or callee ends the call the other end call is not disocnnecting .

UA is pjsip based and behind NAT router. Present call flow is

pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP proxy----> ThridParty SIP Server

UA local ip : 192.168.2.11

UA public IP : 89.78.92.23

Kamailio Public ip: 94.50.203.32

Third party Sip server : 76.42.89.25

Here When I disconnect call from either side , it is not disconnecting other side .

2. My second requirement is , how can I define port of third party server .

for example if have 3 or 4 sip servers with different sip registration ports other tahn 5060

How can I route registration requests coming from UAs to different ports of third party servers.

Please bear my ignorance I am new to kamailio .Hope some experts will help me here .

Attached kamailio config and SIP trace taken from kamailio server

Thank you
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