Hi, I have a SIP phone connecting to my asterisk server, using dtmfmode=rfc2833. When calling from the SIP phone to internal asterisk services, such as voicemail, it works fine.
But when I call out to the PSTN, from the SIP phone, via my X100P, the call will be connected fine. After that, though, any numbers I dial on the SIP phone are lost. I hear them on the phone, but I don't hear them on the remote end of the PSTN connection. I know that rfc2833 is correct for the SIP phone since it is working fine with internal asterisk services. I have tried the wiki, searching the list, and google. No luck. Ideas would be welcome! -- John _______________________________________________ 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
