I’ve installed and configured an asterisk environment as follows:
gtalk <–jingle–> asterisk #1 (as gateway) <–sip–> astersik #2 as pbx
I can fire up the gtalk client, dial the asterisk #1 gtalk account which is configured to Dial(firstname.lastname@example.org).
This all works fine! Congrats! (I’ve also been playing with freeswitch (which seems to be a little more hacky at the moment in it’s code, and it doesn’t work for me yet). So Kudos to the Asterisk team.
Now, I’ve discovered not so much a problem, but an incompatibility with the way GTalk montiors the RTP stream. I’ve set up the pbx SIP URL to be answered by an extension with voice mail attached to it. This all works, and I get the message “Extension is on the phone, press 1 to leave a message” or whatever.
The problem is the silence. My guess is that for silence, Asterisk (as PBX) stops sending RTP data while a prompt is not playing. Normally this is good as it saves bandwidth. However, GTalk thinks that the RTP stream has died and ends the call.
My question is this: Is it possible to configure asterisk (the MIDDLE box) when doing a bridged RTP stream, to “fill in” the silence so that GTalk will remain happy? I suspect this may be useful for other bridging applications as well.