[Asterisk-Users] Configure DID

[Adam Moffett]

In the most basic case you create a SIP user and create extensions that 
point to those SIP users.
in sip.conf:
[sipuser1]
username=sipuser1
secret=123456
type=friend
host=dynamic
disallow=all
allow=ulaw  (<-- put your most preferred codec here)
allow=gsm   (<-- other codecs you will support on subsequent lines)

in extensions.conf:
exten => 6071234567,1,Dial(SIP/sipuser1|60)        (<--- replace with 
your actual DID)

I also suggest:
exten => 16071234567,1,GoTo(6071234567|1)
exten => 1234567,1,GoTo(6071234567|1)              (<--  these lines 
allow for the number to be dialed in different ways and still get to the 
SIP user)


You could also create arbitrary extensions for your internal use:
exten => 101,1,GoTo(6071234567|1)
exten => 102,1,GoTo(some other extension)


>Hi All,
>
>I am a newbie to Asterisk and I was able to install Asterisk and call out.
>Recently I purchased two DID's, can someone please tell me or point to some
>links showing how to configure these DID's for SIP based softphones like
>Express talk?
>
>Thanks,
>Manoj.
>
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