I've got some aastras (mostly 6731i's) that are connected to an Asterisk 1.2 server. That asterisk 1.2 server has a trunk (tried both SIP and IAX) to another Asterisk 1.4 server that connects to PSTN (it connects via SIP, not sure this is relevant)
When i place an outbound call out, the part of the call going from Ast1.4->SIP->PSTN connects the call, and passes back early media RTP ringing. I am able to identify that (using tcpdump) the early media RTP does infact make its way from the PSTN to the Asterisk 1.4 box, and as well from the Asterisk 1.4 to Asterisk 1.2 box, but then - it doesnt traverse back to the Aastra 6731i!
Presumably this is an issue with Asterisk 1.2 doing this, but I'm curious if you guys have any experience with this issue?
Connecting the 6731i to an Asterisk 1.4 box seems to make the problem go away - The easy solution would be to upgrade the 1.2 box, but there are so many customers on it, its a little difficult.
I had tried in the past adding the 'r' option to Dial (on the Ast 1.4 leg) and it fixed the problem, although it caused its own set of problems as well that are unrelated to aastra (more to do with the SIP PSTN provider)
Just hoping someone else has come across this and has a workaround/suggestion?
Member Since:
2007-11-20