It should become immediately clear to you that this material is rather old. The number of visits this area gets is astronomical.

I have caught the VoIP bug and have been building VoIP systems lately. A recent opportunity presented itself which involves Voice Over IP (VoIP) technologies using open-source software. This section collects my notes on the subject.


Several years ago I pulled Category-5 cable around the house (along with other media) and installed a BBS Telecom IPS Hybrid-Key telephone system with three display phones and four regular analog phone extensions that served the entire house. This system worked well for a couple of years but had several problems that could not be easily overcome:
  • Caller-ID only works on the $100 display phones, not the regular phones or even the fancy cordless phone system.
  • No phones ring until after the second incoming ring if the Caller-ID support is turned on. (This was to support Caller-ID decoding for the Display Phones; I have no idea why they don't ring while waiting for the CID data--probably an oversight).
  • Incoming calls ring forever. (This is apparently a hardware failure. I attribute this to the dust from recent basement renovation work. There were no settings changed and no further changes have fixed this problem.)
I have made the decision to rip out the IPS and replace it with a SIP-based phone system.

VoIP for the Home

Outside Lines

Vonage subscribers have a problem when they want to keep their existing phone service--how to support more than one outside line without having two phones or a special phone system? Currently these are the only answer, but in the future there may be a new alternative, already implemented by the GrandStream HandyTone 486, which supports incoming and outgoing calls on a backup phone port which works in concert with Vonage. This does not support simultaneous use of lines inside the house but at least you can use both your Vonage and your Verizon lines. I like to build systems, and building phone systems is particularly satisfying to me, so I will be replacing my existing analog IPS phone system with a SIP-based phone system that also supports the analog phones in the house.

For outside lines we have one Verizon and two Vonage lines. I have been a Vonage subscriber for a couple of years and have been impressed with the service except for the fact that the cable modem doesn't always stay online. A recent day-long dig in the yard has resolved the cable modem issues to our satisfaction and we finally feel happy with the whole broadband situation.

Inside Lines

As I said earlier, I originally installed the IPS telephone system to support three outside lines. One is Verizon and the other two are Vonage. It works beautifully except for the chief complaint of the lack of caller-ID on the regular house phones, including the fancy cordless phone system that always say "NO DATA" for caller-ID. With the new VoIP system and the current generation of SIP-based analog phone adapters we will have caller-ID on all phones in the house and will have access to all three outside lines. I also purchased a couple of SIP-based desk phones to replace the IPS display phones.


I have not built the PBX server yet, but I have over 15 working servers in varying capacities at my beck and call. For my setup with three outside lines I need:
  • Three FXO interfaces to outside lines
    These are called "FXO ports" and refer to outside lines. FXO stands for "Foreign eXchange Office" and merely refers to a gateway that connects to your analog outside line on one side and your phone system via SIP on the other. These are the ports that allow you to place and take calls via Verizon and Vonage. Notice that I am connecting with Vonage via FXO ports so I will be double-transcoding. Vonage does provide direct SIP from your phone system but it is sold in blocks of minutes that keep the cost out of our reach. Digium sells cost-effective PCI cards but they are strictly Asterisk-only and I have problems with their marketing philosophy (see below). I want SIP-based commodity hardware ONLY.
  • Two FXS interfaces to inside lines
    FXS stands for "Foriegn eXchange Subscriber" and refers to inside lines. This allows the regular analog phones (the fancy cordless phone system) to connect to my SIP phone system. One side is SIP and the other side is a regular analog phone jack. These units generate ring current, caller-ID, voice mail stutter-tone and FSK signal, and also have the ability to transcode analog to digital and back. You may simply hook all your inside analog phones to one FXS port but be sure you don't exceed more than 5 non-powered phones or they don't ring. (You can use an unlimited number of phones that don't use AC power, like cordless phones and fax machines because they don't draw power from the phone jack). I will install two FXS only because I don't want to overload the dual-purpose FXO/FXS Grandstream 488 unit with both FXO and FXS traffic. I will reserve that for a seldom-used analog phone like the garage phone.
  • Three SIP phones for inside lines
    If you're going to go through the trouble of building any phone system you are cheating yourself if you don't have at least one display phone per floor in the house. I have one Sipura SPA-841 with an excellent dot-matrix display. I also have a cheap Grandstream Budgetone 100 with a single-line CID display. I am contemplating which vendor to use for the third but it will probably be a software phone like X-lite or a higher-end phone.


I intend to start with Asterisk because, like Linux, it gets the most users, and because it has the most users Asterisk has the most documentation and support options. After Asterisk I intend to move to another solution such as sipX. Asterisk has a small limitation where it transcodes all data to a private protocol instead of keeping it as SIP. My intention is to use SIP-based commodity hardware so Asterisk and the Asterisk-only hardware are not going to be used. In addition, while Asterisk is free, it is no secret that the Asterisk software is used as a vehicle to sell the Digium line of phone hardware. As a result of this marketing ploy the SIP support is inevitably a second-class citizen. In addition the politics behind Asterisk is troublesome--the original Asterisk FXO card cost $135 and was a re-branded Intel Winmodem with the chip part numbers scratched off. Worse, that Intel Winmodem costs only $15 at retail. Digium has discontinued this unit and replaced it with a fully proprietary PCI card. I think this situation says more about Digium and Asterisk than anything else and is why I intend to use another system.


  1. Get FXS adapters [done]
  2. Get FXO adapters [waiting for Grandstream 488]
  3. Get SIP phones [done]
  4. Test phones with FreeWorldDialup [done]
  5. Test FXS adapters with FreeWorldDialup [done]
  6. Build the Asterisk server with its SIP proxy thinger and register SIP FXO and SIP FXS with it [lost interest]
  7. Register SIP phones with Asterisk via SIP [...]
  8. Register FXO adapter(s) with Asterisk via SIP [...]
  9. Register FXS adapters with Asterisk via SIP [...]
  10. Replace, supplant, or peer Asterisk with flavor-of-the-month open-source VoIP system [...]
  11. Replace those snooty-sounding Asterisk voice prompts with someone better (the Cheyenne Bitstream voice prompts or even the synthesized AT&T Natural Voice would be better... even Microsoft Sam would be more tolerable). [I did get several voice synthesizer demos from several vendors; Cepstral and Loquendo are excellent, Nuance is frighteningly bad, and AT&T is disappointing.]

Random Notes


The Grandstream HandyTone 488 is on back-order. It is a new unit that just went into production and provides a true FXO and a true FXS port. I originally purchased a Handytone 486 but its FXO port is not a true FXO port--it does not gateway to SIP but merely provides emergency backup when the SIP service is out. Hopefully the Vonages of the world will provide this excellent solution for those who need to have a hard line for the alarm system and 911 service but still want to use VoIP for regular calling. For me this box does not provide outside line support for the phone system so I'm waiting for the 488 which should be arrive before the end of March 2005. [I got it after losing interest]

SIP Desk Phones

The Sipura SPA-841 is a very high-quality unit with superb sound and two lines (four if you pay for a $30 software upgrade). The Grandstream Bugetone 100 uses embarassingly cheap-looking plastic but the sound quality is excellent. I will probably get a Cisco or the higher-end Grandstream but this Sipura looks better than any of them and at $80 you can't really argue with the price. [Cisco/Linksys has since acquired Sipura]

Voice Prompts

I have it on good authority that the makers of Asterisk paid a locally-known radio voice talent to produce the voice prompts used in Asterisk. They should get their money back. Even Lernout & Hauspie voice synthesis is more polite to callers! Asterisk sounds like a female used car salesman with a stuffy nose and lisp. [This comment was unfair to the radio announcer, as Nuance/L&H voices are terrible]

Voice Recognition

In days past I was to work on a research project that, in part, was supposed to merge voice recognition, voice synthesis, telephony, and smart conversation agents in order to provide automated data-retrieval services for users who only have access to telephones. Since I was already working on phones at home it was a natural choice. I am continuing to work on the VoIP system to replace the worn-out phone system in my house. I will be exploring the voice parts and data-retrieval system as time and motivation allow, but they're not as fun as building cutting-edge phone systems. [No, I won't, I'm totally turned off by Voice Recognition]

Free VoIP Services

FreeWorldDialup provides free VoIP services. Just register your SIP phone with FWD and you're ready to go. It has gateways to many providers, including Vonage, but I can't get the Vonage gateway to work. I suspect it's no longer available. This is a great way to test your SIP phones and SIP FXS devices so you know if the problem is in your Asterisk server or if it's in your SIP device. The echo and tone generator functions are good ways to understand how VoIP and how each codec works. It appears to consist of a group of Asterisk servers (that snooty voice prompt is everywhere) with some custom services on the extensions. Phone numbers are six digits. Calling FWD numbers is usually an international toll call from other services (at least it is from Vonage) but calling to other FWD numbers are always free. More info at

VoIP Centrex

This is the natural evolution of Centrex, the phone company's virtual PBX for PBX-less offices. This is where you have a dedicated telephone line for each extension in your office and the actual PBX is located at the phone company. In Centrex, all you do is pay for each line. It costs more but you don't have to set up a PBX in your office and the phone company is responsible for the upkeep. Traditional Centrex means that you usually just get a trunk line (bundle of lines like T1, ISDN PRI, or similar) that is broken out into individual lines for each extension. The more expensive Centrex systems use ISDN for each extension (I have personally used such systems where the desk phones start at $400 per unit). ISDN Centrex gives you multiple line appearances (which are multiple lines with the same phone number). Your voice travels digitally from the phone to the PBX, much like VoIP, but it's ISDN (again, this is known as several brand names like "5ESS" or whatever). The ISDN version is intended to provide you modem-less 56kbps connections, but with pervasive high-speed ethernet in every office it never lived up to its promise or made any sense.

The natural evolution of Centrex and ISDN Centrex is VoIP Centrex. The VoIP version of this solution is based on the same principle but it sends data over an internet connection. The better providers will sell you a dedicated internet connection that connects directly to their "soft switch". You can always use your own internet connection to connect to the "soft switch" but you risk problems like jitter and delay and dropped calls due to network congestion. I have called several places that use this solution and the delay is annoying--in conversations it seems like the person on the other end is pointing at their phone thinking "what an idiot." Many companies are starting to sell VoIP Centrex under lots of different names (I think Centrex is a trademark). You can buy phone packages with VoIP Centrex at places like and Some of these bundles can come with a pallet full of those cheap Budgetone phones (I have seen at least one office with these phones). The more well-known brands are using Cisco or Nortel phones. If you're going to use VoIP Centrex, use a dedicated internet connection and also keep your phone network on its own private ethernet network or you risk jitter, delay, and (heaven forbid!) dropped calls. I can spot budget-foolish VoIP office phone systems a mile away. [And I still can.]

Wi-Fi VoIP

Some publications are talking up WiMAX and Wi-Fi VoIP phones. I'm not sure this fits into the scope of my discussion but it sounds interesting except for the pathetic cell size and the cell hand-off issue. If the CDMA, TDMA, and GSM networks can replace the application layer with some form of IP, then it makes perfect sense--one protocol (IP) for everything. It's too bad they spent all their money and effort on 3G which never actually paid out. The problems here are obvious--CDMA, TDMA, and GSM are optimized for voice, spectrum sharing, spectrum hopping, and cell hand-off. Does any of this exist in 802.11? I don't remember seeing anything about that in 802.11, so I'm not sure if WiMAX and Wi-Fi VoIP phones can possibly work the same way we expect cell phones to work--seamless hand-off while driving on the highway or walking several blocks downtown. [Filed under "didn't happen, won't happen." There is not enough bandwidth unless the entire phone system is IP, which LTE would be, but that is not WiFi or WiMax.]