I have Asterisk running and 2 SIP phones (X-Lite) plus an iaxtel gateway set up. I used the configurations from the O'Reilly article and I haven't even set up voice mail (the only change was to add the iaxtel entry). My problem is the audio out from my SIP phone isn't reaching the destination phone (whether it is the other SIP phone or a PSTN/POTS phone that I am calling through the iaxtel gateway). The call makes it through OK and audio comes back from the remote end. The audio looks like it is leaving the SIP phone - the level meter is reacting to the speech - but it doesn't reach the other end. Also - I have tried turning off some of the codecs in the X-Lite phone to see if it was a codec compatibility problem with no success. Any ideas? Is this likely a SIP phone setup problem? Or a config setting? (something in the echo canceling maybe??)
TIA for any help.
Ben
(Once this is set up I might have to re-record all the prompts in the en-us-mn dialect as mentioned earlier) :)
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