> I use mediatrix gateways running SIP protocol. > > I have installed hisax compatible passive adapter on my asterisk box > (HFC-S PCI Active chip). > > Problem is following; when I dial through my ISDN adapter and run echo > test I got excellent response (clear sound, no breaks), when I connect > my SIP gateways between each other users hear each other perfectly, no > jitters, errors or breaks. > > But; when I try to call from ISDN to SIP gateway I can hear perfectly > what is said to me from SIP side, but my voice "recorded" on isdn > adapters appears jittered or broken to the other party, and if I speak > to loud it is cut completely.
There is an option in the Mediatrix called Voice Detection (or something like that) that is set to Auto. Turn that "off". The problem relates to asterisk needs a constant flow of rtp traffic (not just traffic when you are speaking), and with the voice detection feature turned on, asterisk does not get that constant flow. Rich _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
