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

Another area of redundancy that I need to address is connectivity. Some might assume this to imply network / Internet connectivity. But that’s only one possible connection methodology.

The simplest and probably most common approach here would be to keep one or two analog phones lines (POTS, for “Plain Old Telephone Service”) and connect them to FXO interfaces on the Asterisk server. These are often called analog trunk lines. When so configured, calls could proceed via traditional wire-line service if IP connectivity was lost.

This is not an approach I choose to deploy for three reasons. First, analog FXO interfaces have, in my experience, been fiddly and unreliable, to be avoided if at all possible. Admittedly this opinion was something that I arrived at some time ago, before the most current crop of Sangoma, Pika and OpenVox hardware was available. Even so, the extra hardware would add to the cost of the implementation and would force us to use a larger host platform that can handle a PCI card.

Secondly, the monthly carrying cost of POTS lines is not inconsiderable, especially if you add a few “extra calling features” such as caller ID. One of the many sorry aspects of the 1996 Telecom Act was the deregulation of fees for ancillary services like caller id, directory service, listing services, etc.

Finally, the only provider of analog lines in our area is AT&T, a company that I have decided I’d rather not give my business to for personal reasons. So it was that some time ago we decided to drop our analog lines and move to a 100% VoIP phone system, whatever that might entail.

Redundant IP Connectivity

On the surface, redundant IP connectivity would seem to make sense. But one fact of VOIP life is that certain things about networking can be more difficult than they should be.

For example, the SIP protocol can make firewall and NAT traversal troublesome. Making an Asterisk server load-balance across two ISPs is even more complex. At the root of this is the fact that the end-point source and destination IP addresses are passed in the VoIP traffic. The Asterisk server itself is expected to have only one IP address, not two.

Voice-over-broadband providers themselves go to significant effort to load balance traffic. Like clustering Asterisk servers, this takes more effort than makes sense for a home office. So on my case, even though I have both DSL and cable modem service at my location, it appears that my VoIP traffic will use only one of these connection, at least for now.

There is an often overlooked third connectivity mechanism that we can leverage—wireless via a cellular carrier. This can effectively create a digital cellular trunk into Asterisk and potentially solves a number of problems. This is the thrust of the rest of this article.

VoIP Service Providers

Most people who decide to move to VOIP based home phone service do so via one of the various VOIP-over-broadband service providers offering what has come to be known generically as “Digital Phone” services.

There are now literally hundreds of companies both large and small offering VoIP-over-broadband service. The leading independent provider of this sort is Vonage. The major cable companies and telcos also have some form of residential VOIP service offering. We might refer to these as the retail providers. Table 1 lists the major U.S. companies.

Company URL
AT&T CallVantage
Verizon VoiceWing
Comcast Digital Voice
Time-Warner Digital Phone
Cox Digital Telephone
Charter Telephone
Table 1: Major US Retail Voice-Over-Broadband Services Providing 911 Service

This group, by definition, provides their customers with access to 911/E911 and 411 services. The FCC, which regulates communications across the U.S., mandates that 911 service be provided by voice-over-broadband companies in the retail residential marketplace.

The most common pricing model among the large retail providers tends to involve xxx minute/month/line domestic long distance calling for a flat monthly rate. Some offer “unlimited” calling for a fixed monthly fee. Often “unlimited” really means some unstated but arbitrarily high number of minutes beyond which they consider that you are abusing the account.

There is yet another group of providers that target the small business, small office, home office and hobbyist marketplace. Their service offering may be wrapped in an application context as a “Hosted IP-PBX” or offered on a raw form as outbound call termination and inbound calling (aka DIDs)—all delivered via IP.

This tier of VoIP service providers is a wholly different class of companies. Not quite wholesale carriers, but not quite retail either. The generic term for these companies is “Internet Telephony Service Provider”, aka ITSP.

They run the gamut from very small players operating on an offer of super-cheap service, to larger companies in the broader networking space and providing SIP trunking to SMBs. Table 2 lists some of the U.S. based players.

Company URL
Voicepulse Connect
Junction Networks
Table 2: Sampling Of US Based Internet Telephony Service Providers

In this space, some players, VOIPJet for example, only offer one direction of service. That is, they may accept outgoing calls for termination afar, but do not provide any way to receive incoming calls. Many offer a diverse range of services at prices that make them extremely attractive to cost-conscious small business owners.

There are various pricing models among these providers. Some provide service based solely on cost per minute of calls placed with fixed no monthly fee. Some charge per minute for all calls, even local calls. Some charge fixed monthly fees per end-point (phone), while others charge per user.

One of the most significant differences between large retail players and smaller ITSPs is the ability to deliver 911 emergency calling service and 411 directory services.

In migrating my home and home office to a 100% VoIP system, we had to weigh many considerations, availability of 911 service key among them. If it was worth installing a UPS to keep the DSL, network and IP phones running during a power outage, then it follows that 911 service should be provided if possible. Yet none of the ITSPs we prefer to use provide 911 or 411 service.

Since we have several cell phones in the house, it, at first, didn’t appear that 911 service on the home phone line was an absolute necessity. Our cell phones could be used if the need arose. However, what about those times when a guest was at our home and without a cell phone? Further, would our household insurance provider look kindly upon not having 911 service?

Ultimately, my wife decided that a pure VoIP solution without 911 service was simply out of the question. What can I say? This is the Spousal Approval Factor that I have to live with.

Even 411 directory service can be important. My wife uses it with startling regularity and tends to find it very annoying if it’s not available. We have tried the free directory services such as TellMe, but found them wanting. They are interesting technology presentations, but don’t generally pass our internal spousal approval test.

Finally, we live in Houston, where we also have 311 service. This is a direct access number that reaches a call center run by the city government. It’s a non-emergency access line that can be used to report street lights out of order, fallen trees, leaking fire hydrants, etc. Sadly, access to 311 service is not provided by cellular providers in our area.

So a wireless cellular trunk sounds interesting. The next step is to examine the common cellular networks and devise a strategy for interface between them and the SIP domain.

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.

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