asterisk users - May 2009 - probably an rtfm but... need to dial out to 2 PSTN lines 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

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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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 = 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
Thanks for the info Danny.

I also found while doing more reading that I can use
Action: Originate
Channel: local/1 at 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