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

[Joseph L. Casale]

>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...

jlc