Add dial plan entries like this in extensions.conf [trunks-ld] ; long distance exten => _1NXXNXXXXXX,1,Dial(SIP/[EMAIL PROTECTED],20,Tr) exten => _1NXXNXXXXXX,2,Congestion
add a sip entry like this in sip.conf [mt-1204] type=peer host=172.20.16.7 mask=255.255.255.255 dtmfmode=inband context=default qualify=yes canreinvite=no > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Alastair Maw > Sent: Tuesday, September 09, 2003 5:27 AM > To: * Users List > Subject: [Asterisk-Users] Dynamic SIP outbound usernames? > > > Hi, > > I have * set up as a PSTN->VoIP gateway (with an E1 with multiple > numbers pointing to it). > > I'd really like to be able to dial out to a SIP server like so: > > exten => _X.,1,Dial(SIP/[EMAIL PROTECTED]) > > I.e. the remote SIP server receives a SIP INVITE with a "To:" header > containing the dialed number (e.g. [EMAIL PROTECTED]). > > This is equivalent to having a hostname extension in sip.conf with a > dynamic username of ${DNID}. > > How does one achieve this? > > Likewise, it would be nice to be able to use gnophone to simulate calls > into the system, by pointing it at the * box and getting the dialed > number on that to route things in the same way. > > Any ideas? > > -- > Alastair Maw <[EMAIL PROTECTED]> > MX Telecom - Systems Analyst > http://www.mxtelecom.com > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
