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

Wed Jun 1 20:51:48 MST 2005

Ok I'm still playing and the way it's supposed to work is making much more 
sense now.

However it is still 'not working' as soon as I add this:

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

to sip.conf

I also learned that I needed to replace the internal IP host= with 
sipproviderexample.com

for whatever reason the example I had been working off of was showing
an internal address for that

What is breaking is that the asterisk box stops accepting ANY inbound 
calls via sipprovider.com as soon as I add the extension noted above.

I have also tried many variations of that...
and using another context besides default.
No change.

It flat out 'stops working' no inbound calls (from PTSN via sip provider) 
as soon as I add that  to  sip.conf and associate it with any ANY context.

even associating it with an empty context (no dialplan lines) in 
extensions.conf it behaves the same way.

If I comment it out inbound calls work perfectly well all day long

I'm baffled and it's very frusterating!

if I attempt an inbound call in this non-working state from PSTN this is 
what I see with sip debug:

Sip read:
ACK sip:s at y.y.y.y SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKdf72.63b2d483.1
From: "unknown" 
<sip:2135551122 at x.x.x.x:5060;user=phone>;tag=e2d90014-1cffac42-8cd690d8
Call-ID: acaf9453-2c29-1cffac42 at x.x.x.x
To: <sip:2135551122 at x.x.y.y:5060;user=phone>;tag=as51e3a2ab
CSeq: 1131072 ACK
User-Agent: SIPProxy
Content-Length: 0

---------> About 5 seconds later this comes back

8 headers, 0 lines
Destroying call 'acaf9453-2c29-1cffac42 at x.x.x.x'

remove those few lines, reload and incoming works just fine again.

I also tried this with a different provider (stanaphone) and that behaves 
the same way.

Anything else to try?

Thanks for all the help and this is EXTREMELY cool! I am playing and 
having a blast wit hit....

Sure would like to be able to make some outbound calls! :-)

Take care!

Steve

On Thu, 2 Jun 2005, Ronald Wiplinger wrote:

> 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 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...
>> 
> 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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