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Avaya IP office

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Avaya's IP Office IP500 V2 Control Unit is a communication solution that can scale up to 272 telephones and 8 T1/E1 trunks. This system supports 8 expansion modules providing a combination of up to 272 analog, digital, and IP telephones, with capacity for 16 analog trunks or 8 digital trunks (192 T1/PRI channels or 240 E1 channels). Optionally, SIP trunks are also supported.

This guide was kindly provided by our customers, it was tested and currently working for them, however, you may need to make modifications according to your own needs.


Configuring an Avaya IP Office IP500v2 (Version 9.0)

This guide applies specifically to the Avaya IP Office platform running on an IP500v2 running the software release 9.0. Settings on different versions and editions of the Avaya IP Office platform will be similar but may difer slightly.


Configuring the network

Under the LAN that is connected to the Internet, on the VoIP page, "SIP Trunks Enable" must be checked.

Further down the page, take note of the incoming RTP port range.

In your firewall, open port 5060 UDP and the port range from the RTP settings, 10000-20000 UDP by default, and forward them to your system's Internet facing LAN IP.

Configuring LAN- click to enlarge

If you experience an issue with the RTP or NAT ports, you could set the Avaya's default ports (Minimum: 46750 and Maximum: 50750)

On the Network Topology page, set a public STUN server, "stun.stunprotocol.org" for example, the click "Run STUN". The detection process will take several minutes. Be patient. Once detection has complete, the "Firewall/NAT Type", the "Public IP Address", and the "Public Port" will automatically be filed in.

Set a "Binding Refresh Time" to something appropriate, such as 60 seconds.

Check "Run STUN on startup" to ensure that the system detects any changes to the public IP address if using a the system with a dynamic public IP.

LAN network topology - click to enlarge

Once the network configuration settings are applied and the firewall is properly configured, the system is now ready to accept the voip.ms SIP trunk.

Adding the voip.ms SIP trunk

In the "Line" section of the system, add a new "SIP Line".

Choose a "Line Number", in this case we chose this to be Line 1.

Enter the voip.ms server associated with your DID under the "ITSP Domain Name". In this case, montreal4.voip.ms (You can choose any of our multiple VoIP.ms servers).

Choose a "Prefix", usually 9.

Set the "Association Method" to use "VIA" in order to match against the resolved IP address from the ITSP domain name.

SIP line - click to enlarge

Add your voip.ms account credentials under the "SIP Credentials" page using the "User name" and "Password" sections.

User ID: 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 100000 or 100000_sub)
Password: Your VoIP.ms SIP Password

Set an "Expiry" to something low such as 5 minutes, and check "Registration required".

SIP credentials - click to enlarge

After adding your account credentials, select them on the "SIP URI" page in the "Registration" dropdown.

To allow incoming calls, set the "Local URI", "Contact", and "Display Name" to a star ("*").

Set the Incoming Group and Outgoing Group as appropriate for your configuration.

SIP URI - click to enlarge

For a more efficient interchange with voip.ms, set the "Codec Selection" to custom and choose only the codecs supported by voip.ms and check "Re-invite Supported".

SIP - click to enlarge

If your networking and firewall were correctly setup, and all the previous settings applied, you should now have access to both outgoing as well as incoming voip.ms calls. Enjoy!

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