Hi, is there any way to increse the buffer or something to make SIP connections sound better? When I make the calls with Asterisk as a SIP client (through sip.voipbuster.com) the sound quality is poor - constantly breaking (there are few occasional seconds when the sound is OK)- but with any other SIP client on the same network and through sip.voipbuster.com sound is allways OK. Matic