[Asterisk-Users] SIP message 183 and in band info

[Tomáš Komárek]

Hello, I have such a problem. I have an * configured as a peer connected 
to the gateway to PSTN.

While calling to the switched off cell phone, the gateway sends to the * 
the SIP message 180 with the SDP part, and also a lot of rtp packets 
containing the operator's in band info.

But * forwards the 180 to the UAC without the sdp part and also without 
the rtp stream.

Is there any way, how to setup the * dialplan to translate all incoming 
180 SIP messages to 183 with the SDP part and also to forward the rtp 
stream to the UAC??

Thanks for advices...

Tomas