skip to Main Content

How To Add a Cellular Trunk to Your VoIP System: Part 1

Originally published May 9, 2008 at

Our Asterisk based home/office phone system provides tremendous flexibility in handling our phone calls. It gave us the opportunity to migrate away from using analog phone lines from a traditional carrier. We now send and receive all calls via IP over our DSL. Of course, the monthly cost of our calling is a lot less. However, it’s not a prefect system – yet.

From the outset, we have worked to make the system more robust. This we have done in many ways, including providing various redundancies in hardware and configuration. Most recently, we have added a cellular trunk to ensure calling capability should our DSL service fail. The process involved in arriving at this decision has proven interesting.

VOIP systems are more than just a phone line coming into the office/home, so there is more potential for problems. To prevent issues from occurring, it is best to address the potential problems before they happen. In our system, our VOIP has a number of redundancies built in to increase its reliability.

To begin with, the phones themselves, a combination of Polycom and Snom SIP phones, are registered both to the local Asterisk server and a remote service. The phones automatically failover, so if the local server is unresponsive, we can still make outgoing calls.

In addition, our phone numbers (aka DIDs, for Direct Inward Dialing numbers) provided through an online service provider also provide a failover system. My preferred service providers offer failover to a normal phone number, which I generally point to my cell phone.

Our Asterisk server provides it’s own form of redundancy as well. It is registered with three commercial service providers and a couple of free VOIP services including Free World Dialup. If any service provider is not available, it will cascade to another, and then a third. This is really only useful, however, when making outgoing calls.

Redundant Asterisk servers are, I think, a little beyond the scope of a typical home or home office environment. The sort of Linux clustering skills required to set this up are certainly beyond my present skill set. Also, redundant servers are more than a little inconsistent with my use of appliance-like Asterisk embedded systems.

This Post Has 11 Comments
  1. Digium will be shipping a Skype connector module (commercial product) in Q1 of 2009, so you could connect any channel type that you can link into Asterisk through to your Skype account, bi-directionally.

    I read through the docs on the MV-370, and it doesn’t seem like there is a way to receive SMS messages via any interface other than the built-in web server, even though there seems to be a way to send them via a telnet and “AT” command set. Have you found anything differently? Has anyone created any Perl routines to automated the transmission of SMS’s (not that it’s a big deal to do it oneself.) What happens when you log into the device’s AT command interface, set a debug mode, and send the unit an SMS? It seems that two-way SMS would be a pretty snappy feature even if it just relayed to a fixed SMTP address – XMPP would be asking too much. 🙂


  2. John,

    I’ve not really pushed the SMS issue, as until recently I didn’t make much use of it even on my cell phone.The MV-370 unit that I have appears to have bidirectional SMS capabilities by way of AT commands over telnet. I know nothing of the command set.

    This capability was implemented in firmware after I had the unit installed. I probably should upgrade, but it’s been doing what I need. The latest firmware with SMS is dated 10/13/2008.

    The MT-350S specifically offers SMS service. It’s cheaper as its voice connectivity is by way of FXS/FXO.

    I’m not much of code jockey so playing with AT commands is a little beyond my scope. However, there is a manual specifically on using the AT command set to send/receive SMS.


  3. TellMe’s directory service (800-555-TELL) is actually pretty spartan. I’m not sure where they get their business directory from, but it’s pretty antiquated, and seems to be missing a LOT of businesses.

    Free411 (800-free-411) has a MUCH better search capability, and includes municipal government numbers as well as business and residential, but at the expense of listening to an advert at the beginning of the call and before the call is connected.

    GOOG-411 (800-GOOG-411 — Google’s Service) has a nice search feature that will rank top listings, and works for Canadian businesses as well, but it’s only business listings. It does not currently have adverts in it, but, being Google, one can assume those will appear eventually.

    Incidentally, it’s rumoured that in the UK, this sort of cellular bridging is illegal to provide as a service. I’m not sure if it’s illegal in the SoHo arena. Again, this is just what I’ve heard. We’ve not done any real extensive research into that sort of issue, since we’re not in the UK. Any UKers out there care to comment on this?

  4. Yes, I have heard that it’s illegal to provide such a gateway as a commercial service in the UK. Using it for private use is allowed AFAIK.

  5. It is perfectly legal for any end user (business or consumer) to use GSM gateways, it is however illegal to use GSM gateways in order to offer a communications service.

    Taken from

  6. Hmmm… I tried using the tags but it looks like it hasn’t worked. Please see below for the information I intended to post:

    “Ofcom has recently clarified that it is entirely legal under UK law for end-users (whether businesses or ordinary consumers) to buy, install and use GSM gateways for their own use. However it is currently illegal under UK law for anyone to use GSM gateway equipment to provide a communications service by way of business to another person or organisation, irrespective of where the gateway equipment is located, or how many or few end-users are connected to each gateway. This prohibition on ‘commercial’ use applies equally to the mobile network operators (MNOs) as to other organisations, since the MNOs’ licences do not currently extend to the installation and use of GSM gateways.”

    Taken from

  7. I’m currently running two MV-370’s on a OnSip PBX from Junction Networks. One gateway is in US and the other is in Poland. This setup allows me to call toll free between my cell phones from US to Poland and back.I have one problem. I’m experiencing a delay (kind of an echo) pretty much every time a make a phone call between the gateways (sometimes more sometimes less). It’s usually better when one of the parties is on a regular sip phone and only one is talking via gateway. Have you expirienced such a delay? Do you know what can cause this?


    1. No, this has not been my experience. But then again, I only run one gateway. I would expect that the actual media would be negotiated to flow directly from one gateway to the other, not through OnSIP. That should be a very direct path.

      You could try using another PBX to see if the situation changes. You could get a free account from IdeaSIP and give it a whirl. Or even set the gateways up to direct dial via IP, presuming that you have fixed IPs.

Comments are closed.

Back To Top