>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... jlc _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
