Any Ideas?
I’ve just upgraded our asterisk from 1.2.13 to 1.2.22, then suddenly when calling our toll free numbers, users cannot hear the calling party (one way sound). This only happens with the incoming. When i revert back to 1.2.13 sounds went back to normal.
Tried to tail -f /var/log/asterisk/messages.
channel.c: Translation to slin failed, dropping frame for spies
Is this error have any connection to the one way sound?