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
Eric Wieling aka ManxPower
2005-Aug-18 14:29 UTC
[Asterisk-Users] SIP message 183 and in band info
Tom?? Kom?rek 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?