[Asterisk-Users] SIP call doesn't execute the 's'-extension

[Rich Adamson]

> i have just started to configure access to the * over SIP-Phones. 
> Therefore I have defined this SIP-Phone in sip.conf:
> 
> [tobias]
> type=friend
> username=tobias
> secret=tobias
> auth=md5
> host=dynamic
> reinvite=no
> dtmfmode=inband
> callerid="Tobias" <1087006>
> allow=all
> context=javaAgi
> dtmfmode=rfc2833
> 
> 
> As you can see i am directing calls from this user to the context 
> [javaAgi] which is defined here in extension.conf:
> 
> [javaAgi]
> exten => s,1,Answer()
> exten => s,2,Playback(code1000)
> exten => s,3,Hangup()
> exten => 1,1,Answer()
> exten => 1,2,Playback(code1000)
> exten => 1,3,Hangup()
> 
> If i dial 1 on my SIP Phone everything works as suspected, the call is 
> answered and the gsm-file is played. My understanding of the 
> 's'-extension is, that it is executed then a call comes in an there is 
> no extension wich matches the called number. But if i dial a random 
> number i get an "404 Not found" error.

The "s" extension matches only when "no" digits are dialed. Dialing a "1"
is a digit, so no match.

Try playing around with exten=>_XXXX.,1,Answer() and understand what
the differences are.