search for: siptm_connect

...y tries to negotiate it, but eventually disconnects. It HAS, > however, twice, successfully connected the call for a short time, but no > browsing was possible. I've done some debugging output on the Adtran whic= h > seems to indicate that an RTP BYE command is received: TM.T01 01 > SipTM_Connected =A0 =A0 =A0rcvd SIP call-leg request: BYE ... This is the= first > difference between a debug output where the call connected and one that d= oes > not work. This is the one that doesn't work. > > TL;DR - Is it possible to do dial-up through Asterisk? Or is it like t38 > faxin...