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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