Here are sample asterisk configs for Voipfone:

 

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

[voipfone]
type=friend
secret=
username=
fromuser=
fromdomain=sip.voipfone.net
host=sip.voipfone.net
insecure=very
dtmfmode=rfc2833
context=fromvoipfone ;inbound calls falls in this context of dialplan
deny=0.0.0.0/0.0.0.0
permit=195.189.173.27/255.255.255.255
<<== end of sip.conf ==>>

This is example of minimal extensions.conf dialplan. It simply forwards
all incoming calls to user 123 over SIP. <<== extensions.conf ==>>
[voipfone]
exten => _.,1,Dial(SIP/123)

Setting Up TrixBox

Below you will find the configuration details for TrixBox which is an open source PBX that you can build yourself:

 

You can find descriptions, handbooks and guides here:

http://asteriskathome.sourceforge.net/index.html

http://geekgazette.com/index.php?option=com_content&task=view&id=2&Itemid=26

 

Log onto the Asterisk@home Management Portal (AMP)

Select Setup

Select Tunks

Select the add SIP Trunk

 

In the General Settings section. Enter your Outbound Caller ID

 

In the Outgoing Settings section

 

Trunk Name VOIPFONE-SIP

Peer Details

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

 

In the Incoming Settings section. All entries should be blank

 

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

 

: @sip.voipfone.net/

 

Setting up a DID route

Log onto the Asterisk@home Management Portal (AMP)

Select Setup

Select inbound routing

Select the add incomming 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 Recepitonist (IVR menu), a single extension, voicemail, a ring group, a queue, a custom application or as per your Incoming Calls settings