[asterisk-users] Incoming call

[michel freiha]

Dera All,

I have the following scenario,
A customer dial a DID number...The call is routed to a PSTN GW that send the
call to asterisk...
On asterisk I created an AGI Script that send the call to an extension
registered on OpenSIPS server...
The extension is ringing successfully, but as soon as I accept the call on
OpenSIPS side the call is hangd up...
I checked rhe SIP debug and it seems that I have a Codec issue as you can
see on http://pastebin.com/m767a2172

Need some help please
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