Situation: SIP phone A calls Asterisk. Asterisk forwards to another SIP agent B. The SIP agent B forwards A back to an Asterisk extension that is mapped to a TDM400P channel Once all of this has transpired, there is no audio channel between the SIP phone A and the TDM400P. If A is not registered with Asterisk performs the same sequence, a half- duplex connection results in which A can hear the TDM400P, but not the other way around. If the above procedure is repeated in reverse, that is: TDM400P dials the extension of SIP agent B. SIP agent B forwards TDM400P to the asterisk extension of SIP A. A full duplex connection results, and everything is normal. Any ideas? Elaborations needed? ===sip.conf==============[general] port = 5060 ; Port to bind to bindaddr = 192.168.0.104 ; Address to bind to context = from-sip-internal disallow=all allow=ulaw [201] username=201 type=friend secret=password host=dynamic dtmfmode=inband ============================== 201 is SIP phone A. -Dustin Mulcahey