Asterisk @ Home is a marvellously easy to set up, except when it doesn't work.
Setting up with Voipuser causes a lot of people some grief, so here we offer a configuration that works.
These settings are known to work for AAH up to 1.7
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From the AMP setup screen select Trunks and Add SIP Trunk
Go to Outgoing Settings
For the Trunk Name enter '
voipuserout'
In the panel below enter:
| Quote: |
canreinvite=no
context=from-pstn
fromdomain=voipuser.org
fromuser=Username
host=sip.voipuser.org
insecure=very
qualify=no
secret=Voipuser password
type=peer
username=username |
Leave the Inbound entries blank
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Enter DID Route and select Add DID
The DID number to enter here is 44(Voipuser number without the zero e.g. 8444321321)
Select the destination as you wish
Save the entry.
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Go to Outbound Routing and Add Route
Give the route a name and create a dial pattern.
Something like this is good for dialing national and international, while using a 9 for selecting out bound calls
Finally select the trunk
Voipuserout
And save the route
You can now click the red bar at the top of the screen to load the configuration.
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Now we can create the Inbound Context for the account
Go to the AMP Maintenance panel and select Config Edit (it opens a new window)
Scroll down the list on the left until you find
sip.conf and click
In the text window scroll to the bottom and enter:
| Quote: |
[voipuser]
type=friend
context=from-pstn
username=username
secret=password
host=sip.voipuser.org
fromuser=username
fromdomain=voipuser.org
insecure=very
qualify=no |
Update this and click on the link to /etc/asterisk at the top of the now blank screen
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Scroll down until you find the entry sip_nat.conf
Here you can enter you network details which will usually include:
| Quote: |
nat=yes
externip=Connection IP
localnet=Internal network/Subnet |
and add your register string at the bottom:
| Quote: |
| register=>username:password@voipuser.org/Your DID number as used above |
Update these changes and select Re-Read Configs
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Subject to the membership delay, firewall and port forwarding settings, you should have a working connection.
You can check that you have registered with VoipUser by selecting Asterisk Info from the AMP Maintenance screen. You should see something like this
| Quote: |
| voipuser.org:5060 rgower 105 Registered |
under SIP Peers
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Edit 2/11/05
If there are problems receiving calls, try replacing host=sip.voipuser.org with voipuser.org in the [voipuser] context in sip.conf
Any faults, better solutions, or requests for simillar VoipUser based guides, please drop me a PM