Trying to understand the default asterisknow extensions.conf

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Trying to understand the default asterisknow extensions.conf

Postby frodefi » Mon Apr 16, 2007 6:02 am

As the title gives away, I am trying to understand extensions.conf that asterisknow is creating.

I am using a SIP-trunk and have installed two SIP-phones (extension 602 and 603), and I also have a call queue 600. I have also read the appropriate chapters in the online book Asterisk: The Future of Telephony. But I cannot understand how the incoming call is connected to the SIP-phone, I have only one incoming rule, all calls for _XXXXXX02 must go to 602. I though that there had to be something like:
exten => 602,1,Dial(....) or something similar, but...

When I call XXXXXX02 from my mobile phone, the SIP/602 is ringing correctly, and the conversation is correctly established when I answer the call.

So where is the sentence that actually connects the inncoming call and SIP/602? Thanks in advance! Here is (what I believe to be) the insteresting parts from extensions.conf that asterisknow has automatically produced:

[default]
exten => 850,1,VoiceMailMain
exten => 8500,n,Hangup

; Lots of comments-lines removed....

include => voicemenu-custom-1
exten = 600,1,Queue(${EXTEN})

[voicemenu-custom-1]
comment = mainmenu
exten = s,1,Answer
exten = s,2,Background(thank-you-for-calling)
exten = s,3,Background(if-u-know-ext-dial)
exten = s,4,Background(otherwise)
exten = s,5,Background(pls-hold-while-try)
exten = s,6,Background(to-reach-operator)
include = default

[DID_trunk_1]
include = default
exten = _XXXXXX02,1,Goto(default|602|1)
frodefi
Newsterisk
 
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