[Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?

[Steve]

Over the past 2 weeks I have been able to compile and get an asterisk 
system up
& running on a debian Linux box.

I have setup 5 internal sip clients on the lan and all works great!

I can also call from outside (PSTN) into the system and reach extensions 
and
services no problem.

All is up & running behind a nat firewall with proper ports forwarded and
locked down on each device to work properly.

For outbound calls to the PSTN I'm using a single SIP provider account...


I have read LOTS of docs and played quite a bit to get this far....

The only thing I have not been able to figure out is how to set it up so 
the
internal extensions can actually make an outbound call via our SIP 
provider
account.


I've very confused by the docs I have read and some of them pertaining to 
this
matter  are not perfectly clear (to me) on if what they are doing is what 
I
think they are doing.

I'm probably totally off here and doing the wrong thing because whenever I 
try
it My system no longer even works with the sip provider at all and won't
register...



--------------
Here's what works:

register => 2135551212:2135551212 at sipproviderexample.com
--------------

Just having that single line in place allows incoming calls to everything 
and
works flawlessly

but please forgive me for asking.....

register => 2135551212:2135551212 at sipproviderexample.com/2200

ALso works perfectly and rings in to extension 2200

How in the world to you make it ring into more than one extension? :-)



----------------------------
OK here's what messes it all up (and I admit I'm clueless here)


register => 2135551212:2135551212 at sipproviderexample.com

[sipproviderexample.com]
type=peer
host=10.77.77.133
fromuser=2135551212
secret=2135551212
fromdomain=sipproviderexample.com

adding this secttion breaks it and I really do not understand what it's 
even
for...

does it work with the register line somehow? or is it totally seperate?
what is it for?


All the docs I have looked at seem to suggest adding this extra section 
but do
not really seem to explain it or what exactly it does.
I'm not sure what it's for or if it has anything to do with making 
outbound sip
calls from the internal extensions.

when I add it my sip provider account stops working and I get registration
retries and timeouts without any successful registrations after that.

I'm just looking for a good pointer in where to go for an example of how 
to use
my provider account for outbound connections...

I understand the dialplans themselves  but do not know how to associate 
them
with the actualy sip provider account for an outbound call.


Thanks....

I'll keep reading until I figure these things out but any pointers to
specific documentation that answers any of these questions would be very 
much
appreciated....

I *think* I am familiar with just about all of the standard asterisk
documentation I have been able to find...

More than likely I am just missing some key points that I have read but 
have
misinterpreted!

Take care!

Steve