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By WelshPaul
#138
Setting Up Asterisk
Here are my asterisk configs for my VoIP provider Voipfone:
Code: Select all
<<== 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.
Code: Select all
<<== 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:
  1. Log onto the [email protected] Management Portal (AMP)
  2. Select Setup
  3. Select Tunks
  4. Select the add SIP Trunk
  5. In the General Settings section enter your Outbound Caller ID
  6. In the Outgoing Settings section enter the Trunk Name: VOIPFONE-SIP
Peer Details:
Code: Select all
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:
  1. Log onto the [email protected] Management Portal (AMP)
  2. Select Setup
  3. Select inbound routing
  4. Select the add incoming route
  5. 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.

Hello Paul, Thank you for your response. Much app…

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