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

[Eric Wieling aka ManxPower]

Tomá¹ Komárek wrote:
> 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??

That would be a function of a SIP Proxy, which Asterisk is not.

What is the specific PROBLEM you are experiencing?