I have 1.0.6 installed from the ports collection on FreeBSD 5.3 and have it registering to SIP account @ sipgate.co.uk. If I call my sipgate number from a landline I get the test voice prompts however when I press “2” I do not get re-routed. I have tried a few things but nothing seems to work. I’m in the UK but doubt that would make any difference. Any ideas???
Howdy,
SIPGate uses RFC2833 for DTMF? There was a substantial rewrite of the RFC2833 support in Asterisk between 1.0/1.2 and 1.4 Asterisk. You’ll probably want to upgrade to a newer version.
Cheers.