Just received a Mediatrix 1204 fxo sip gateway and playing with the initial config's, etc. It's working, but have a ways to go before it could be considered usable. The box was not designed to "register" like sip phones do. The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm using canreinvite=no to forcably keep * in the middle for now.
Questions: 1. The 1204 answers incoming pstn calls correctly, cycles through the invite/ trying/ringing (I have * config'ed to simply ring an internal sip phone for testing purposes), and I answer the call just fine from the sip phone. When I hang up the sip phone, * sends a Bye and the 1204 says OK. The 1204 then sends "one" more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? 2. The 1204 seems to be set to a 30 millisecond sampling rate while all other sip phones, etc, are set to 20. Anyone have any thoughts as to whether that would cause a problem later, or should I change that to 20 milliseconds for consistency? 3. Has anyone played with this box and found any unusual problems, weird config's, etc? The box is essentially in a test/eval mode, anticipating using it to replace a couple of x100p's. 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