[Asterisk-Users] gsm playback garbled over sip

[Glen Gray]

I'm new to all this so I don't know where to look, some tips would be
most appreciated. 
I've enabled sip debugging and everything looks fine on the client and
server side. Using Linphone on the client side. 

GSM playback from the server console is fine. I've used Linphone to
connect to a vegastream VoIP system so I know if that installed and
working. I'm basically just trying to get the sample configs working,
dialing in to 500 at ipaddress. It connects and playback of the demo
ensues, but at the client end it's unrecognisable garbage. 

Any hints ?
-- 
Glen Gray <glen at lincor.com>                             17 Dame Court
Senior Software Engineer                            Dublin 2, Ireland
Lincor Solutions Ltd.                          Ph: +353 (0) 1 6746413