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ANO
Analog Gateway User Manual
2 &y
www.allo.com
Version 2.0
21
By Default: Based on Username
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Host Name
Configure the IP address or URL for the VOIP provider’s server of the
trunk.
Host Port
Enter the port number given by the VOIP provider. 5060 is the default SIP
port. If configuring any roaming extension or if connecting gateway to
public network then consider changing the SIP port for better security.
Once port changes are done, for registering IP phones/sip phone use
<domain name ><sip port number>.
Username
Enter the username to register to the trunk from the provider.
Password
Enter the password to register to the trunk from the provider.
Qualify
If you enable Qualify, asterisk will send the SIP Options command
regularly to check that the device is still online
DID Routing
DID Routing Routes calls to a single, specific extension
.
Outbound Proxy
Configure Outbound proxy to send Outbound signaling to that proxy.
4.3.4 Editing / Deleting an Existing VOIP Account
On the right side of the page, you can see the list of VOIP trunk you have setup. To edit, or
delete any of them, simply click the appropriate icon provided to the right of each trunk. Once
you click on the edit button of a VOIP trunk then it will display the information of that particular
VOIP trunk, here you can change the required details and click on the Save button and then click
on the apply changes tab to save the changes made.
To ensure successful registration of your VOIP Trunk, you must click the “System Status” tab on
the top navigation menu (see Status section for more info)