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Asterisk Configuration Guide

Here are sample asterisk configs for Voipfone:

<<== start of sip.conf==>>
[general]
register=> : USERID:PASSWORD@sip.voipfone.net/USERID

[voipfone]
type=friend
secret=PASSWORD
username=USERID
fromuser=USERID
fromdomain=sip.voipfone.net
host=sip.voipfone.net
insecure=invite,port
dtmfmode=rfc2833
context=from-pstn ;inbound calls falls in this context of dialplan
<<== end of sip.conf==>>

Peer Details

authuser=USERID context=from-pstn dtmfmode=rfc2833 fromdomain=sip.voipfone.net fromuser=USERID host=sip.voipfone.net insecure=very qualify=yes secret= PASSWORD type=peer username= USERID

In the Incoming Settings section all entries should be blank

In the Registration Section enter your userid and password and select Submit Changes

USERID:PASSWORD@sip.voipfone.net/USERID

Setting up an Inbound Route

Log onto the FreePBX web interface
Select Connectivity
Select Inbound Routes
Select the add incoming route

Add the DID number (USERID)

Finally you need to set the destination for all incoming calls. Depending on your configuration you can send the calls to a Digital Receptionist (IVR menu), a single extension, voicemail, a ring group, a queue etc.

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