[asterisk-users] Incoming SIP and the 's' extension

[John A. Sullivan III]

On Thu, 2009-06-18 at 03:50 +0000, Joseph L. Casale wrote:
> >I then fire up twinkle on my desktop and dial sip:36 at pbx.mycompany.com.
> >The Asterisk console shows:
> >[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
> >Call from '' to extension '36' rejected because extension not found.
> >
> >If I use the same extensions.conf but change "s" to 36", it works.  I
> >would have expected the SIP channel to see that it had nothing which
> >matched my name or IP address and sent processing to the [incoming]
> >context where it would encounter "s" and process accordingly.
> 
> http://www.voip-info.org/wiki/view/Asterisk+s+extension
> http://voip-info.ehd.com.br/wiki/view/Asterisk+standard+extensions.html
> 
> >What concept am I missing? Does SIP always have a FROM and TO and thus
> >never uses "s"? I'm obviously misunderstanding a fundamental concept.
> >Thanks - John
> 
> You have a known #, your explicitly calling 36 from your soft phone.
> 
> What you want is a pattern match for your sip phones, and the "s" for
> a dahdi line for example...
<snip>
Ah, ok.  Thanks very much. That's what I thought might be happening but
didn't trust my instincts over my ignorance and over the tutorials I was
following which did not point that out when describing a minimal
dialplan. It makes perfect sense.  Thanks again - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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