[SR-Users] Kamailio With RTP proxy BYE not receiving

[Daniel-Constantin Mierla]

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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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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