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HOWTO: SIP Trunking between Avaya IP Office R5 and Flowroute
Kyle L Holladay, Sr
In this example we will configure a SIP trunk between the Avaya IP Office and Flowroute using registration on LAN1 behind a firewall/NAT. Alternative
configurations would include a static public IP or static IP behind a NAT/Firewall which will not be covered in this document.

NOTE: Flowroute claims T.38 version 0 support as of May 2009 however I have not been able to successfully negotiate any T.38 traffic with a Colorado DID at
this point using version 0 at 9600bps and IP Office EI version 5.0 due to a requirement by Flowroute of information in the INVITE that the IP Office does not
currently support. Thus far Flowroute's technical support has been responsive and helpful. The call quality has been fantastic, far beyond what I have come to
expect from other ITSP (over the Internet SIP providers).

First we need to obtain the IP address of the Flowroute 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, you can find the SIP Proxy name by clicking on Interconnection on the Flowroute Manager web site. From your
computer enter ping sip.flowroute.com from the command line (CMD) and make note of the IP address that is returned. From this point on we will use
70.167.153.130 to refer to the Flowroute SIP server, if you resolve a different IP address you should use that instead.

The IP Office will need a number of ports opened on your firewall in order to properly communicate with the Flowroute 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.

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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 Flowroute.

We will skip most of the basic signup options for Flowroute however you will want to start by visiting http://www.flowroute.com and selecting the Sign Up
Today option create an account once done they will give you a $0.25 credit which will allow you to test their service for outbound calls. Purchasing a DID or
placing inbound calls will require a minimum purchase of $35 and at least a single DID.

While Flowroute does support routing to a public IP their defult is registration which is what we will cover in this document. Go back to the Flowroute Manager
web site and once again click on the Interconnecton tab.
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

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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.

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 sip.flowroute.com (70.167.153.130 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 or Automatic Select 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 the Username
you obtained from the Flowroute's Interconnection tab on the Flowroute Manager web site as your "Primary Authentication 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:

With IP Office 5.0 you may wildcard your URI on the SIP Line tab to allow for ambiguous validation of the URI on INVITE without the need to build a
specific list of URI for validation. With IP Office R4 this was not possible and a specific URI list had to be created for inbound validation of the INVITE.
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 wildcard this entry by adding a * for each of the three
drop down fields.
You will also want to set your Outgoing Group to some number not associated with any ARS table so that your outbound caller ID can be controlled by the "Use
User Data" URI entry created in the next step.

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Now repeat the step above but this time we will 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 which will be referenced in your associated ARS tables.

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 "@flowroute7.net" so that the entry looks as such:

Next we must add our URI information for 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.

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.

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

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