Set Up SIP trunk with Les.net on IP Office

Setting up a SIP trunk between the IP Office and Les.net (Compliments of Kyle L Holladay Sr.,  R.I.P.)

In this example we will configure a SIP trunk between the Avaya IP Office and LES.net using a static IP address assigned to LAN1 behind a firewall/NAT. Alternative configurations would include registration which is supported by LES.net for users with a dynamic IP addresses or you could assign a public IP to the IP Office on LAN2 outside of your firewall/NAT.

 

First we need to obtain the IP address of the LES.net SIP server. Because the IP Office does not use DNS to resolve the name of the SIP server you will want to use PING on your PC to resolve this name. From your computer enter ping did.voip.les.net from the command line (CMD) and make note of the IP address that is returned. From this point on we will use 64.34.181.47 to refer to the LES.net SIP server.

 

The IP Office will need a number of ports opened on your firewall in order to properly communicate with the LES.net SIP server. You will need to insure that the following ports are open for both inbound and outbound traffice, without restriction.
UDP 5060 (SIP Signaling)
UDP 49152 – 53246 (RTP Traffic)
NOTE: The RTP range can be modified under System>LAN1>Gateway if requested by your provider.

 

Ensure that you have a default route created in your IP Office under IP Routes that points to your LAN1 interface.

 

Once you have your NAT created and all your ports opened in your firewall (refer to the documentation provided with your firewall/router for directions on how to do this if you are unsure) you can verify that you are ready to move forward by running STUN in the IP Office. Open Manager and click on System>LAN1>Network Topology and click on the [Run STUN] button. This process will take a while so just wait for the results which will be indicated by a flashing  next to several fields. You are looking for “Open Internet” or “Full Cone NAT” in the Firewall/NAT Type field. Any other entry indicates that you have not properly setup your firewall/NAT. A proper result should look similar to that below (we will use 71.39.255.255 as the public IP of the IP Office for this example):

 

Once you have properly configured your firewall/NAT and your IP Office has properly recognized your public IP address via STUN and sees all necessary ports open you can proceed with configuring your peer/trunk with LES.net.

 

We will skip most of the basic signup options for Les.net however you will want to start by visiting http://www.les.net and selecting the Sign Up option on the left to create an account and create your inital Trunk/Peer and select your DID(s).

After you have created your Trunk/Peer you will need to configure it to use your static IP address. Login to your LES.net account and click on the Peers / Trunks option on the left and click the “Edit” option for your Peer Name.Set the options to match the picture to the right making sure to select “Public IP” as your “Peer Type” and enter the IP address that you obtained via STUN in the step above as your “Peer Address”. You can enable G.711 as well, if you wish, however G.729 is typically the best option for voice conversations over the Internet.

 

********** ALTERNATIVELY ********** 
If using registration (such as with a dynamic public IP address).
Set the options to match the picture to the right making sure to select “Registration” as your “Peer Type” ignoring the “Peer Address” and enter any password you like in the “Password” filed. You must also make note of the full “Peer Name” at the top of this form as you will need to enter that later.REGISTRATION EXAMPLE

 

Now we will create a SIP line in the IP Office. In Manager right click on Lines and select New>SIP Line. In the pane on the right enter did.voip.les.net as the ITSP Domain Name and the IP address obtained from your ping to did.voip.les.net (64.34.181.47 in our example) as the ITSP IP Address. Check the In Service and Use Offerer’s Codec check boxes and select G.729(a) 8k as the Compression Mode from the drop down. Select UDP as your Layer 4 protocol and LAN 1 as your Network Topology. Leave 5060 as your Send Port. Generally your screen should look similar to this:

 

 

********** ALTERNATIVELY ********** 
Now we will create a SIP line in the IP Office. In Manager right click on Lines and select New>SIP Line. In the pane on the right enter did.voip.les.net as the ITSP Domain Name and the IP address obtained from your ping to did.voip.les.net (64.34.181.47 in our example) as the ITSP IP Address. Check the In Service and Use Offerer’s Codec check boxes and select G.729(a) 8k as the Compression Mode from the drop down. Select UDP as your Layer 4 protocol and LAN 1 as your Network Topology. Leave 5060 as your Send Port. You will need to select “Registration Required” and enter teh “Peer Name” you obtained from the LES.net web site as your “Primary Authentication Name”. Note that this is NOT the same as your LES.net web site loging name! You will also need to enter the password that you selected in the step above as the “Primary Authentication Password”. Generally your screen should look similar to this:

Now here is where we get a bit unique. There are a number of ways to handle your SIP URI and none of them are wrong. When an inbound SIP call is sent to the IP Office the destination/called party is checked against the list of URI in the system to insure that the IP Office should handle the call. The system doesn’t use this information to actually route the call, it simply checks to see if the called party belongs to the IP Office. Every DID you own must be entered in the IP Office somewhere as a URI or the call will be rejected by the system. There are two places to enter a URI. You can enter the information under the SIP line itself, which requires a reboot to modify. Alternatively you can enter URI information under a user’s SIP tab which can be modified and saved as a merge. The same URI information can also exist in multiple places. In order to avoid reboots we will use phantom users to enter URI information.

Under your SIP Line click on the SIP URI tab and then click on the [Add] button. Typically you would enter your DID in the three drop down fields and repeat this step over and over for each DID that would not be associated with a specific user. Instead we will select select “Use User Data” for each of the three drop down fields.

You will also want to assign an appropriate incoming and outgoing group ID for this line.

Now you must modify your existing ARS table or create a new one to accomodate the domain information needed to make an outbound call over the SIP trunk. For each entry in your ARS table add “@did.voip.les.net” so that the entry looks as such:

 

Next we must add our URI information to allow for incoming calls. Right click on Users and select new. In the Name field enter URI followed by the DID number (I start with URI as you can not have a username that begins with a numeric digit). I also enter a description for the DID under the full name field although this is completely optional. You may leave the extension number blank as it is not needed.

 

Now select the SIP tab for this user. This tab was previously hidden and is only available once a SIP line has been created and a SIP URI exists with the “Use User Data” option. Remove all but the numeric DID information from all three fields.

 

And now your individual users. For each actual user you will also want to set a URI. Either a unique DID for that user or a main telephone number that may be the same for multiple users. This will be used to display the outbound caller ID when that user places a telephone call. Again this information is NOT used for inbound call routing.

 

Finally we need to create our incoming call routes. This is no different than the incoming call route you would create for an ISDN trunk. Enter your line group ID, number and destination. No special domain or SIP information is needed.

 

NOW SAVE! Save your configuration and reboot your IP Office and you’re ready to test!

Unfortunately at this point it probably will not work. Out of the last several system’s I have setup with LES.net I have had the results when making my first outbound call. A recording saying something like “one zero zero, your peer is not configured correctly, one zero zero”. But do not fear! We have a solution!

Log back into the LES.net web site and click on the Support Ticket link on the left site to start a new support ticket and enter the following:

 

Once they have configured your username to be your peername in their database all should be good.