On Thu, 2009-06-18 at 03:50 +0000, Joseph L. Casale wrote: > >I then fire up twinkle on my desktop and dial sip:3...@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 jsulli...@opensourcedevel.com
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