Dec 152017

The Challenge

Finding a cost-effective SIP trunk provider in Canada can be one of the biggest challenges that a business may have when trying to adopt VoIP technology. This is also a common problem for VoIP PBX re-sellers, as it’s hard to find a good provider to refer.

Back in 2007, just a year in to running my own company, my telecommunication and voice requirements massively grew. I needed a phone system to handle multiple extensions, call forwarding, conference rooms, follow-me services, rings groups, and needed the ability for staff and contractors to have their phones (and extensions) in remote offices or home offices. Also, I was travelling quite frequently so I needed to be able to have an extension running on my smartphone (so it would appear as if I was at the office, and to save on international roaming and long distance costs).

Implementing a VoIP PBX phone system handled all of this, and was very easy to implement however finding an SIP trunk provider was not. Originally I was using FXO/FXS adapters to pipe analog lines in to my PBX, however I wasn’t happy with the quality or the complexity of a solution. I wanted a true 100% digital, and 100% Canadian hosted solution.

The Solution (The Review)

After spending months researching providers, I came across a company called Iristel. There were numerous great reviews on the internet, and most importantly they had a following of Trixbox (Asterisk) users, so I could verify they would work with my PBX. They were a Canadian company (important to me), who provided SIP trunks at a great cost. I signed up for service, and tech support was actually amazing at providing assistance for configuring the SIP trunks with my Asterisk PBX, their sales staff was pretty awesome as well!

Here’s where the review gets boring (which is a good thing), I’ve been using them for around 10 years now, and everything has always just worked! I think in 10 years, I may have experience a single 1-2 hours of downtime, and this was due to a compatibility issue with Asterisk and their SIP gateways caused by an update (SIP registration bug). In this one-off case, tech support was immediately available and made configuration changes to resolve this issue. Outstanding service to say the least!

Over the years, I’ve also re-configured and deployed new PBXs. I’m now using FreePBX, and Iristel is still working great! AND YES, Iristel supports T38 faxing!

I would definitely recommend Iristel as a your VoIP SIP provider for your business digital telephony needs!


Feel free to reach out (comment) if you have any questions about my review, or the quality of the services.

Jul 232012

Interesting story:

On the weekend, my Trixbox VoIP PBX (which runs Asterisk) failed. Unfortunately, during the restore process the hard drive also blew up. Temporarily I setup a ML350G5 as a temp VoIP PBX, however today I had the chance to setup an old Acer Aspire Netbook which I had sitting in a box as my new permanent VoIP PBX.

I used a USB CD-Rom to install Trixbox, as for some reason I couldn’t get it to load the kickstart file during a grub boot off a USB key, also couldn’t get it to mount the NFS install export (maybe the kernel didn’t have support for NFS?).


The netbook had decent specs:

Dual-Core 1.5Ghz Process

1GB Ram

Battery (this means I don’t have to put it on my UPS for all my server equipment)


Got it setup and it’s running great! 🙂

Mar 022012

Well, today I was cleaning up and found an old Linksys SPA-3102 device that I purchased 3-4 years ago. I originally purchased this device to connect my Trixbox (Asterisk) PBX to my land line at my house.

The SPA-3102 is a device manufactured by Linksys/Cisco that provides one FXS terminal, and one FXO terminal. This device can connect your PSTN phone line to your VoIP PBX, and it can also allow you to connect a standard phone to your VoIP PBX as an extension, all at the same time.

While I wasn’t to happy with performance of the solution, nonetheless I figured it out and got it running. I decided to write up a little blog post as a How-To get the SPA-3102 working with Trixbox. This solution is mostly just a bunch of config, so excuse the lack of How-To and the bulk of config dumps:

Update (2017): This also works on new versions of FreePBX Asterisk Linux Distro. I can confirm T.38 faxing works on the FXO line, but have not been able to get it working on the FXS PSTN line (fails to re-negotiate).


1) Configure the Asterisk extension (this configures the line you hook up to a phone on the SPA-3102):

Create a extension inside of Trixbox or FreePBX. Leave everything default except:

Display Name: Fax Machine (change this to whatever you want)

Extension: 199

secret: password (choose you password)

canreinvite: yes

host: dynamic

type: friend

nat: yes

qualify: yes

2) Configure the Asterisk Trunk for the SPA-3102

Go to the Trunk Menu inside of Trixbox or FreePBX PBX configuration. Add a new SIP Trunk. Leave settings default except:

Outbound Caller ID: 1234567890 (Change the number to your PSTN line, if the number doesn’t match, it could break things)

Trunk Name: spa3102

PEER Details:

secret=P4SSw0rdz (replace with your password)

3) Configure Outbound and Inbound Routes

The configuration for the Outbound route is normal and doesn’t require any special configuration other than the normal outbound route you’d normally create for a trunk. However, the Inbound route does require special attention. When creating the Inbound route, make sure that the DID Number value exactly matches the 10 digit number you configure for the PSTN. This is how it will recognize this and categorize the incoming call under that specific inbound route.

4) Now for the SPA-3102 Configuration

There’s no way I’m writing all the config out for the SPA-3102, so instead I took screenshots for each tab that requires configuration.



And Voila!

You now have your SPA-3102 configured to both act as an extension and a gateway to the PSTN. If anyone has any better configuration please write a comment, I’d love to update this article, and I’d like to get this working better than it currently is of possible. One additional note: When the SPA3102 is factory reset, it’s default settings are optimized for the North America region.

Jul 012010

Here’s a few oldies I found while going through the millions of pictures I’ve taken over the years…

Apr 112010

Some time ago, I needed to configure an SIP trunk between a Trixbox/FreePBX (Asterisk on Linux) PBX and a Cisco Call Manager PBX. It was pretty hard to find any relevant information on the internet, however eventually I figured out how to do it. Originally this article was written for Trixbox, however the same configuration applies to FreePBX (with minor differences in steps due to the UI differences).

Please note that the following configuration reflects a Trixbox/FreePBX PBX configured with phones with extensions of 1XX and the Cisco Unified Call Manager configured with extensions of 3XX.

If you are simply using CUCM for Cisco IP Phone handset connectivity, you don’t even need CUCM anymore, you can simply use the commercial “EndPoint Manager” on FreePBX to handle Cisco IP Phone connectivity to FreePBX (includes the Cisco 7961 phone’s I use).

Trixbox/FreePBX Configuration

Create an SIP Trunk (Leave settings default unless otherwise specified below)

Outgoing Settings

Trunk Name: CallManager

Peer Details:












Incoming Settings

USER Context: ip.address.of.CUCM

USER Details:





host= ip.address.of.CUCM

fromdomain= ip.address.of.CUCM






Create an Outbound Route to route calls made to 3XX to the Cisco Call Manager

Create outbound route “Cisco”. Check the “Intra Company Route”, and inside of the Dial Patterns type in 3XX. Under Trunk Sequence select “CallManager”.

This pretty much sums up the amount of configuration required on the Trixbox/FreePBX side of things. Now onto the Cisco stuff.

Cisco Unified Call Manager Configuration

Create an SIP Trunk

Device -> Trunk -> Add New

Trunk Type: SIP Trunk

Device Protocol: SIP

Device Name: TrixboxPBX

Call Classification: OnNet

Check the “Media Termination Point Required” checkbox (this is to handle transfers, hold music, etc…)

Check “Remote-Party-Id”

Check “Asserted-Identity”

SIP Information

Destination Address: IP.address.of.trixboxfreepbx

Uncheck “Destination Address is an SRV”

Destination Port: 5060

MTP Preferred Originating Code: 711ulaw

SIP Trunk Security Profile: Non-Secure SIP Trunk Profile

Change the “Non-Secure SIP Trunk Profile” security profile from TCP to UDP

System -> Security Profile -> SIP Trunk Security Profile

Hit the “Find” button

Select “Non Secure SIP Trunk Profile”

Incoming Transport Type: TCP+UDP

Outgoing Transport Type: UDP

Uncheck “Enable Digest Authentication”

Incoming Port: 5060

Out of the last 6 checkboxes, all should be checked except the First and Last.

Create a Route Pattern to route calls from the Cisco Call Manager to Trixbox

Call Routing -> Route/Hunt -> Route Pattern

Create New

Route Pattern: 1XX

Gateway/Route List: TrixboxPBX

Route Option: Route this pattern

Call Classification: OnNet

Enable Required Services on CUCM

I’m not too sure which ones are actually required, however the below configuration works great. To get to the CUCM services go to the “Cisco Unified Serviceability” section (Top right of web interface).

Enable Services

Tools -> Serviceability

Enable the following:

CM Services

Cisco CallManager

Cisco Tftp

Cisco Messaging Interface

Cisco Unified Mobile Voice Access Service

Cisco IP Voice Media Streaming App

CTI Services

Cisco CallManager Attendant Console Server

Cisco IP Manager Assistant

Cisco WebDialer Web Service

Select “Save”, afterwards select “Set to Default”. Please note that it may take some time to bring the services up.

It’s always a good idea to restart both the Trixbox/FreePBX PBX and the CUCM PBX.

After you have configured the above, configure phones in the 1XX range for the trixbox, configure phones on the CUCM for the 3XX range and they should be able to call each other. Please remember that if you have a PSTN line on your Trixbox or FreePBX you will need to create another route pattern for how to transfer 9XXXXXXXXXX from your CUCM -> Trixbox, then configure the applicable route in Trixbox -> PSTN.

Feedback is welcome, leave a comment!