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 = [email protected] 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: [email protected] [mailto:[email protected]] 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 -- 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 _______________________________________________ -- 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
