>From the log, it showed lookup(): 'account name' Not found in usrloc. I think it is the NAT problem, so I use stun for the case below.
Telephone (A) | PSTN | G/W | openser | NAT -- IP phone(B) -- STUN + ----- IP phone(C) ------+ Below is the result: A to B/C is ok B/C to A is ok However, there is no sound when B to C or vice versa. What reason will cause no sound between B and C? Is the the reason from the NAT/STUN? On 12/12/05, Klaus Darilion <[EMAIL PROTECTED]> wrote: > Use ngrep to watch for incoming SIP requests on the SIP proxy. > > Take a look at the logfiles on the gateway. > > klaus > > unplug wrote: > > Below is the common configuration of the network. > > > > Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger) > > > > I can make call from softphone to Telephone. However, it is failed to > > make call from Telephone to softphone. I wonder why it happened and > > any reference to trace the problem. Anyone have such experience? > > > > _______________________________________________ > > Users mailing list > > [email protected] > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ Users mailing list [email protected] http://openser.org/cgi-bin/mailman/listinfo/users
