This guide is to assist you in setting up SIP.US as a SIP Trunk provider on Avaya IP Office Manager version 8.0 and above with Digest Authentication. This particular configuration was done on an Avaya IP Office 500v2 with a VCM 32 card. This Avaya System was configured via open internet and was not behind any firewall. The following configuration represents a minimal configuration to provide inbound and outbound calling with a single DID and channel with SIP.US. It includes utilization of both gateways for failover capabilities as a best practice.
IMPORTANT: Refer to Interconnecting with SIP.US for the full list of gateways along with whitelisting recommendations.
1. Log in and load your configuration in Avaya IP Office Manager.
2. Go to "System" and then select your IP Office System.
3. Select the "LAN 1" tab.
4. Select the "VoIP" tab and ensure that "SIP Trunks Enable" is checked.
5. Select the "Network Topology" tab and fill in the settings to match your topology.
NOTE: This device was not behind NAT, but you may use the SIP.US STUN server (stun.sip.us) if necessary. The "Binding Refresh Time" is a setting that controls the frequency of SIP options messages. We chose 60 seconds as it provided a good result in testing.
6. Go to "Lines" and then right-click and select "New" > "SIP Line".
7. Under "ITSP Domain Name" input "gw4.sip.us". For the Call Routing Method, select "Request URI". For the Association Method, use "By Source IP Address". All other settings can be left as default or adjusted as necessary.
8. Under the "Transport" tab, input the IP address for gw4.sip.us (104.219.162.40) in the "ITSP Proxy Address" field.
9. Ensure that you have input valid DNS settings, network configuration information, and that "Calls Route via Registrar" is checked.
10. Select the "SIP Credentials" tab and add an entry with your trunk credentials as shown below:
11. Select the "SIP URI" tab and add an entry. The applicable settings from below are for the "Incoming" and "Outgoing" groups in addition to the "Registration" selection. These need to match the line number that you just created and the "SIP Credentials" entry entered in step 9. All other settings were left at the default.
NOTE: If you are having trouble getting calls to route, use an asterisk (*) in the "Local URI" section. However, this will cause your outbound caller ID to show up as the extension number that is sending the call. To fix this new issue, set the "PAI" field to "Use Internal Data" and then go to the "SIP Trunking" tab in the SIP.US customer portal and click "Modify Trunk" for the relevant SIP trunk. Set the "Outbound CID Header" field to "PAI". Your caller ID will then be based on the User in your Avaya associated with the extension that is dialing out.
12. Select the "VoIP" tab, and then click "Advanced".
13. Ensure that only G.711 U-LAW and G.729 are selected.
14. Check the box for "Re-invite Supported" and ensure that RFC 2833 is selected for DTMF mode.
15. Create a second "SIP Line" and duplicate all settings except for "ITSP Proxy Address" under the "Transport" tab. This is for gw5.sip.us (104.219.163.117) and will provide redundancy for inbound and outbound calling when configuration is complete.
17. Navigate to the "User" tree and select "No User".
18. Select the "Source Numbers" tab and add "SIP_OPTIONS_PERIOD=1" as an entry. This is an additional time-based control on the frequency of SIP Options messages.
19. Select the User who will be dialing and receiving calls on the SIP Line and go to the "SIP" tab. Enter the DID that will be transmitted as their caller ID under "SIP Name" and "Contact."
NOTE: SIP.US requires 11 digits in the To header for outbound calls. If your requirement is to only dial 10 digits within the United States, create the ARS rule that follows:
- Code: XXXXXXXXXXN
- Telephone Number: 1N
- Feature: Matching
- Line Group ID: Matching
NOTE: Refer to these articles on Setting Up Caller ID and Setting Up Outbound Calling for more information on configuring each with SIP.US.
20. Navigate to the "Short Codes" tree and create a new short code. This code will be used to direct calls to the SIP Line we created.
21. Ensure that your "Telephone Number" field follows the N@"gw4.sip.us" format where N=Any Number, or you may specify a specifically dialed number if you wish.
22. We chose to use an ARS entry in the "Line Group" field which will be discussed in upcoming steps as it allows us to provide failover capabilities in case gw4.sip.us is unavailable. You could simply skip the ARS steps and use the Line Group you previously created.
23. Navigate to the "Incoming Route" tree and create a new route.
24. Under the "Standard" tab, ensure that the "Line Group" matches what you created previously and that you list the DID that will be delivered with a "+" symbol in front.
23. Under the "Destinations" tab, ensure that you have set an appropriate destination for your inbound route to deliver the call to.
THE FOLLOWING STEPS ARE OPTIONAL BUT RECOMMENDED
24. Navigate to the "ARS" tree and create a new entry.
25. Give the route a distinctive name and mark it as "In-Service". Add a code for calls to be routed through. The code pictured below transmits any number dialed through the Line Group that was previously created.
26. Note that the out of service route is directed to a second ARS group that was created using the same process outlined here, except it uses the secondary Line Group previously created. The second ARS group is also used as a lower priority "Alternate" route in this ARS group routing.
27. Save the Avaya configuration and load it into the system.
28. Once the system reboots, you should be able to make and receive calls through SIP.US.
If you experience problems after following this guide, please refer to our Troubleshooting Guides. If you are still unable to resolve the issue, please open a ticket to contact the Support team at support@sip.us.