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

[Karl Fife]

Your ITSP is giving you the DNIS digits.  You have to match them in your 
dialplan.

What if your ITSP routed calls from FIFTY different numbers to your switch? 
How would you differetntiate between them if they all just routed to the S 
extension?  That's why the paradigm is based on passing and matching digits. 
Otherwise you'd need fifty contexts/registrations.  Number X routes to 
Jimmy, Y to Johnny, Z to customer service queue.  Outbound calls starting 
with *67 block the outgoing caller ID.
See what I mean?
-Karl



----- Original Message ----- 
From: "John A. Sullivan III" <jsullivan at opensourcedevel.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, June 17, 2009 10:06 PM
Subject: [asterisk-users] Incoming SIP and the 's' extension


> 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
>
> http://www.spiritualoutreach.com
> Making Christianity intelligible to secular society
>
>
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