• Dual FXOports
• Easy installation, no need to open up computer to install PCI/PCIe card
• Supports up to 2 simultaneous calls
• Compact plastic enclosure
• Low power consumption, takes power from USB bus
• USB 2.0 compliant (compatible with USB 1.1)

TORONTO, ONTARIO–(Marketwire – Feb. 4, 2009) – Sangoma(R) Technologies Corporation (TSX VENTURE:STC) today announced general availability of its USBfxo product in international markets.

USBfxo is designed for Asterisk(R) open source phone system software users who need hardware support for two non-VoIP analog lines for off-network calls. To add more lines, users can connect multiple Sangoma USBfxos in one system.

“The USBfxo worked like a charm,” said Gerard K.M. Lim, CEO of the GENME Group in Malaysia. “We have both voice and fax working fabulously with Asterisk 1.4 on our desktop class Intel workstation with Fedora 7.”

“It’s ideal for OEMs looking to offer low cost PSTN access in appliances that do not have PCI or PCI express interfaces, developers who need an external solution, and even SOHO end users who are new to Asterisk,” said Doug Vilim, vice president of sales and marketing for Sangoma. “It’s easy to install and configure, fully supports Asterisk on Linux systems, and includes Zaptel/DAHDI drivers.”

The Sangoma USBfxo, part of the B-Series product line enables cost-conscious customers to opt for Sangoma’s premium audio and engineering quality by choosing a product specifically designed for smaller-budget installations.

The product is available through authorized Empowered by Sangoma resellers.

Sangoma is exhibiting the USBfxo today at TMC’s ITEXPO in Booth 703.

About Sangoma Technologies Corporation

Sangoma is the premium provider of software-centric media and signal processing hardware. The company develops and manufactures the most scalable and reliable voice and Wide Area Network data cards in the industry, including the award-winning Advanced Flexible Telecommunications (AFT) product line. By offering the building blocks for all processing in any architecture, Sangoma gateways are tightly integrated and easily scalable. The result is a comprehensive toolset for creating cost-effective, powerful, and flexible solutions.

Founded in 1984, Sangoma Technologies Corporation is publicly traded on the TSX Venture Exchange (TSX VENTURE:STC). Additional information on Sangoma can be found at: www.sangoma.com.

While there are a ton of settings, there are really only about 7 settings that need to be setup to make this work.

In this article, we will show you the idiots guide to making this cool product work within a trixbox or FreePBX environment. We will walk you through the FreePBX setup of the SIP trunk configuration and then show you the few settings that you must set in order to make the SPA-3102 work. Before starting, go to http://linksys.com and get the latest firmware and update your device, then read the article and get your SPA-3102 working within just a few minutes.
Linksys SPA-3102 FreePBX Configuration Setup
Go into your FreePBX interface, click on the Setup tab, then go to the Trunk menu. From here, add a new SIP trunk. The settings are pretty simple here but do need some explaination:

General Settings

Outbound Caller ID: “Your Company”
Within quotes, put in your caller ID name information and within the greater-than/less-than brackets put in your outbound caller id phone number. An example of this is as follows:

Outbound Caller ID: “Acme Widgets” <9495551212>

Maximum Channels: 1
You need to set the Maximum channels to 1 since the device can only handle one concurrent call at a time.

Outgoing Settings
Trunk Name: SPA3102

PEER Details:
allow=ulaw
canreinvite=no
context=from-pstn
disallow=all
host=10.10.10.50
insecure=very
nat=yes
port=5060
qualify=yes
type=peer

Unless you are using G729, you will need to allow ULAW (G711) as that is the default Codec for the SPA-3102. The host address is the IP address of the SPA-3102, if this device is outside of your local network, then set nat=yes, otherwise if it is on the local network, use nat=no.

Incoming Settings
User Context: SPA3102_In

USER Details:
allow=ulaw
canreinvite=no
context=from-pstn
disallow=all
host=192.168.0.252
insecure=very
nat=yes
port=5060
type=user

In the incoming settings, the host address needs to be the IP address of your Asterisk/trixbox PBX system, again use nat=yes if the SPA-3102 is on another subnet or nat=no if the SPA-3102 is on the same subnet as the PBX.

Submit your changes and click the red apply bar, your trixbox/FreePBX system is ready to go, now you need to configure the SPA-3102.

SPA-3102 Configuration
On the rear of the SPA-3102 are four ports, you should plug the phone line into the line port, an analog phone into the phone port, and connect a cable from the Ethernet port to your computer. This will assign your computer an IP address, then you can access the web interface of the SPA-3102 at http://192.160.0.1.

If you want the device to sit on your network and not hand out DHCP addresses, plug your regular network into the Internet port, if you have a DHCP server it will assign an IP address to the device or you can set it manually. If you are putting it as a device on your network, be sure and turn on Enable WAN Web Interface otherwise the only way to configure it is to plug a cable from the device to a computer.

Here comes the fun
Now, to configure it as a SIP trunk (this article does explain how to use it as an ATA) you need to click on the Voice tab, and then on the PSTN Line tab, all of the settings are going to changed here.

Proxy and Registration
This section tells the SPA-3102 how to handle calls back and forth, since we are not using authentication, we need to disable those settings. First, set the Proxy setting to point to IP address of your PBX box, then change the settings in the section to match the list I will give you here, you should only have to adjust a few settings, but I will give you all of them to make sure you have everything setuip right:

Use Outbound Proxy: No
Register: No
Register Expires: 3600
Use DNS SRV: No
Proxy Fallback Intvl: 3600
Use OB Proxy in Dialog: Yes
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
DNS SRV Auto Prefix: No
Proxy Redundancy Method: Normal

Dial Plans
What we are doing in this section is telling the SPA-3102 how to route a call when one comes in. Basically the dialplan we use is going to say, “go to the IP address I specify and tell the system the phone number I am giving you”, this will also allow you to use the phone number you specificy as a DID for inbound route control.

Dial Plan 1: (S0<: 7145551212@10.10.10.50This e-mail address is being protected from spam bots, you need JavaScript enabled to view it :5060>)

The 7145551212 is YOUR phone number, change this to be the phone number people dial to reach you.
The 10.10.10.50 is the IP address of your PBX, change this to match your system configuration.

VOIP-To-PSTN Gateway
This section of the configuration tells the SPA-3102 how to handle calls that come from your PBX and go out to the phone line. The factory default settings should work properly but again, I will give you the working settings just in case something isnt set right:

VoIP-To-PSTN Gateway Enable: Yes
VoIP PIN Max Retry: 3
Line 1 VoIP Caller DP:3
Line 1 Fallback DP: 1
VoIP Caller Auth Method: None
One Stage Dialing: Yes
VoIP Caller Default DP: 1

PSTN-To-VOIP Gateway
This section tells the SPA-3102 how to handle calls from the phone line to the PBX. While the default settings will mostly work, there are two changes you will most likely want to make, again, here is the complete list:

PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line 1: No
PSTN CID For VoIP CID: Yes
PSTN Caller Default DP: 1
Line 1 Signal Hook Flash To PSTN: Disabled
PSTN Caller Auth Method: none
PSTN PIN Max Retry: 3
PSTN CID Number Prefix:
Off Hook While Calling VoIP: No
PSTN CID Name Prefix:

Save your settings and your SPA-3102 should be ready to go!