our current connection is below:
sip phone< --->asterisk< ---->alcatel PBX< ---->PSTN
asterisk and alcatel PBX is connected via E1 isdn-pri.
when I use sip phone to dial outside PSTN world:
1. with 1.4 it is fine.
2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip
phone can not hear the ring and the beginning of the PSTN voice.
3. with 188.8.131.52, I can not hear ring and the beginning of the PSTN
voice. I try to play options with “prematuremedia” and
“progressinband”. but I can not find working settings.
I don’t know what other options I can try.
thank a lot for information!!