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:[EMAIL PROTECTED]

[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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