Danny Nicholas wrote: > users.conf > [108] > username = 108 > transfer = yes > mailbox = 108 > call-limit = 100 > fullname = General Messages > registersip = no > host = dynamic > callgroup = 1 > context = DLPN_DialPlan1 > cid_number = 108 > hasvoicemail = yes > vmsecret = 1234 > email = du...@dummy.com > threewaycalling = no > hasdirectory = no > callwaiting = no > hasmanager = no > hasagent = no > hassip = yes > hasiax = no > secret = > nat = yes > canreinvite = no > dtmfmode = rfc2833 > insecure = no > pickupgroup = 1 > disallow = all > allow = ulaw,gsm > autoprov = no > label = > macaddress = > linenumber = 1 > > no entry in sip.conf > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Millican > Sent: Thursday, May 28, 2009 1:17 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] probably an rtfm but... need to dial out to 2 > PSTNlines from AMI > > Hello all, > I have a need to be able to use the originate AMI command to dial out to > the PSTN, have that person answer and then have the second PSTN > connection dialed out. > I have tried to use: > Action: Originate > Channel: sip/<number>@<provider> > Context: default > Exten: <othernumber> > Priority: 1 > Timeout: 30000 > > This does not dial the number through the provider, actually, it seems > that the number never gets passed to the provider. > I suppose I could create a dummy sip exten but it would have to be one > that had no device attached and I am unclear on how to do that. > Any Sugestion on either method? > > TIA Thanks for the info Danny.
I also found while doing more reading that I can use Action: Originate Channel: local/1...@mynewcontext Context: default Exten: <othernumber> Priority: 1 Timeout: 30000 and then setup a context in the dial plan that dial out to the needed number. I new as soon as I sent the question something rtfm ish would hit me Thanks again -- JohnM _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users