[asterisk-users] probably an rtfm but... need to dial out to 2 PSTNlines from AMI

[Danny Nicholas]

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 = dummy at 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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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
-- 
JohnM


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