[Asterisk-Users] First seconds of outgoing SIP call are cut-off

[Hielke Christian Braun]

I have a * setup with Grandstream SIP phones dialing out through
Nikotel via SIP. When i dial out and the other side picks up, the
Grandstream keeps ringing for another seconds and two and the sound
coming from the other side is lost. After these two seconds the call
is connected find and works flawless. 

Calling the same path the other way to the Grandstream phone works
fine without the cut-off. 

I read about a similar problem with a cut-off at the beginning when 
calling in to the voicemail service. Don't know about if that is
maybe related. 

Is there something i can configure to get it working correctly? 

Thanks in advance, 
 Christian