[asterisk-users] 1.8 and prematuremedia problem

d tbsky tbskyd at gmail.com
Mon May 9 20:57:56 CDT 2011


hi:
    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 1.8.3.2, 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!!



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