We are using Asterisk 15.4, sipML5, with bandwidth.com as our PSTN sip trunk provider.
For outbound calls we are not able to hear the initial 0.5-1 seconds audio (Like : Hi, this is, Welcome to, ) these things are cutting.
Can this be a Asterisk delay happening while connecting the two channels on bridge. Or it might be our SIP turnk provider.
Has anyone faced this problems before. Please let me known if anyone has solutions for this.