How to configure trixbox CE with CogentVoip Service

Step 1 – NAT Settings

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The first stage is to ensure that the NAT (Network Address Translation) configuration is

correct in your sip_nat.conf.

Login as maint and the password you have set, find the Asterisk link and hover over it.  You should see “Config Edit”. Select and look for “sip_nat.conf”, enter the following lines in iof which are in quotes;

“nat=yes”

-This tells the Asterisk box that some adjustments will need to be made while using SIP protocol over a WAN link.

For fixed IP addresses:

“externip=XXX.XXX.XXX.XXX “

-This sets the IP address in the SIP packets to your outside IP address. Use the external IP address you have from your ISP.

“localnet=192.168.X.X/255.255.255.0”

 -This enables the trixbox to know who is registered from the inside LAN and allows the SIP messages to be generated with the correct IP addresses. Use your local subnet settings.

Also, under the Security Settings tab, set "Allow Anonymous SIP Calls" to "Yes".

Step 2 – Trunk Settings

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Trunk Name : cogentvoip.com

Peer details:

dtmf=rfc2833

dtmfmode=rfc2833

fromuser= local user name [can be used for CLI generation at user level]

host=cogentvoip.com

insecure=very

nat=yes

qualify=no

secret= [CogentVoip issued proxy password. This is not the web login password]

type=peer

username=[CogentVoip issued proxy user name. This is not the web login user name]

User Context=[local user name]

User Details

context=from-trunk

fromuser= local user name [can be used for CLI generation at user level]

host=your External IP address

insecure=very

nat=yes

qualify=no

secret=[CogentVoip issued proxy password. This is not the web login password]

type=user

username=[CogentVoip issued proxy user name. This is not the web login user name]

Register String: Username:Password@cogentvoip.com

Create your inbound routes for inbound numbers, e.g. 7035790200 routes to phone A,

7035790201 routes to phone B and that should cover your CogentVoip trunk setup.

Firewall/Router

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You will need to open port 5060 on your firewall to enable the SIP signaling to traverse.

You will also need to open the RTP or audio ports. This is different for each customer

premise device. Please reference trixbox for this detail.