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

[John A. Sullivan III]

Hello, all.  My apologies up front but I must be brain cramping on
something very simple.  I've tried to pare down my configuration to the
absolute minimum for SIP traffic just to understand how it works.  My
incoming calls are not finding the "s" extension in my dial-plan.  I am
assuming SIP calls can do this.  I am using Asterisk 1.6.1.1

sip.conf has nothing but:
[general]
context=incoming

extensions.conf has:
[globals]

[general]
autofallthrough=yes

[default]
;exten => s,1,Verbose(1,Unrouted call handler)
;exten => s,n,Answer()
;exten => s,n,Wait(1)
;exten => s,n,Playback(tt-weasels)
;exten => s,n,Hangup()

[incoming]
exten => s,1,Answer()
exten => s,n,Playback(hello-world)
exten => s,n,Hangup()

[internal]
;exten => 515,1,Verbose(1,Echo test application)
;exten => 515,1,Answer()
;exten => 515,n,Echo()
;exten => 515,n,Hangup()
;exten => 1000,1,Verbose(1,Extension 1000)
;exten => 1000,n,Dial(SIP/1000,30)
;exten => 1000,n,Hangup()
;exten => 1001,1,Verbose(1,Extension 1001)
;exten => 1001,n,Dial(SIP/1001,30)
;exten => 1001,n,Hangup()

[phones]
include => internal

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.

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
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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