asterisk users - Jun 2005 - Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?

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@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@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@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

Steve wrote:
>
> I have read LOTS of docs and played quite a bit to get this far....
>
Good, keep playing!!!

(a lot of your typing time deleted)
>
> ----------------------------
> OK here's what messes it all up (and I admit I'm clueless here)
>
>
> register => 2135551212:2135551212@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...
>
What does it mean for you, that this "breaks" it. Did it work before?

What is your "jump in point" to the dial plan? You do not have a 
context= line. So you may jump into the context=default as usually 
mentioned in sip.conf
[general]
context=default            ; Default context for incoming calls

Do you have something in the dialplan like:
[from-sipproviderexample.com]               ; if you would use this as 
your context= in sip.conf, otherwise the next lines in the default 
context of your extensions.conf
exten => _X.,1,Dial(SIP/601,60,tr)       ; if any number calling from 
this provider should ring extension 601
or
exten => 2135551212 ,3,Dial(SIP/601,60,tr)   ; if only a dialed in 
number 2135551212 of your provider should dial extension 601

Keep playing with:

context=xxxxx  and    [xxxxx]

exten  =>  something with  and without  starting  _ 

Hope it helps, ...


bye

Ronald





> 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
>
>
>
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>
>

-- 
Ronald Wiplinger  (CEO of ELMIT)
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