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?
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