At first, thanks for an answer.
Well the problem starts at the side of a cell phone operator. In the IAM
this operator sends ALERTING, eventhough the cell phone is switched
off. It means, that the gateway sends to the asterisk 180 with SDP and
also with the RTP stream, that contains the operator's announcement:
"the phone is not available at the moment". Asterisk receives this 180
message and cuts off the SDP part and sends to the extension the 180
without SDP. And also the outgoing RTP stream contains ringing, that
Asterisk generates.
So, my question is: Is it possible to sent 180 with SDP part? I've
changed a little bit a source code, so Asterisk sends such message.
Is there any way, how to force Asterisk to send the RTP stream, that it
receives?
Because if it receives 183 Session Progress, the outgoing RTP stream is
the same as the incomming.
Thanks for an answer
Tomas
Eric Wieling aka ManxPower napsal(a):
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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