For a variety of ranty reasons, we are moving our phone services to voip, with the free Asterisk software based PBX server. This covers the equipment/software/providers being used, and what we’ve done with the system.
- Cisco SPA-3102 (formerly Linksys)
- Siemens Gigaset A580 IP + bundled handset; plus three extra S76H handsets.
- Mac Mini (not dedicated - this is on my desktop currently, and fairly low CPU).
- Asterisk PBX server
Siemens Gigaset A580 IP
This is a wireless phone system (DECT6 standard) that handles both landline and voip. This system can handle 2 voip calls and one landline phone at the same time, with up to 6 handsets, using a common base. In our configuration we are only using voip; all calls are routed through to the phone system.
Call quality is quite good; and very few audio dropouts due to the wireless. I’ve been able to use this phone next to 2.4ghz systems without interference. We have noticed that when calling out, there are a few seconds where the phone is syncing up with the voip provider and no audio is heard. I’ve compensated by having our pbx delay outbound connections by 3 seconds.
The user interface on this phone is a bit awkward. For the most part, if you treat it like a cell phone, it will do the right thing. However, if you have a second call come in, you have to use the cumbersome menuing system to answer the call or to switch calls. The flash button is useless, at least when using Asterisk as your PBX. One neat feature is if you do have a second call, whether inbound or outbound, you can join it to your first.
Ring tones: The stock A58H handset has a nice pleasant “ring” tone. The other 3 handsets (S67H) we purchased, are either loud and annoying and shrill, or are soft muted music tones. Not thrilled here, but it isn’t a deal breaker. We chose the S67H handsets for the wired headset capability - we depend on this feature.
Speaking of ring tones and UI: You can choose which SIP providers ring which extensions. You can have one SIP phone number ring all of your phones, or just one. However, there is a catch: Since these are not hardline phones, you can’t just pick up a random extension. And the phone only lets you answer a call if the phone is ringing. Handsets that are set to not ring for a given phone number, are useless when you want to answer your spouse’s line. Furthermore: No distinctive ring; the ringer on the phone applies to all numbers, and not just “yours’.
Battery life: Darn good so far. Even better: AAA NiMH batteries are replaceable, no special battery packs.
Opinion after 1 week of use: Decent system but a handful of quirks. None of them deal breakers yet; and even if they were, I don’t have any other multiple handset wireless system that would be suitable. Especially in the consumer price point category.
This is what is known as an “ATA”. This will translate an analog phone line from the phone company, and allow incoming calls to be routed into your voip based PBX. Similiarly, your PBX can use this line for outgoing calls. Additionally this device has a port for your analog phone. This device is flexible - you can use it to convert your regular phone line to voip; or, you can use it to convert a voip account from a SIP based provider, into something your regular phone can use.
Ultimately, playing with voip can be a lot cheaper _without_ this box (see “Voipo” further down). However, I am keeping this box primarily for 911 purposes. We will keep one analog phone line active in this house. I have more faith in a landline based 911 call than I do with any voip based one. We will also use this as the line of last resort, in case the voip providers are not accessible (ie, internet is down).
Here are the things I dislike about this box:
For incoming calls, you need to let ring a good solid ring and then some (To get the caller id signal from the telephone company). Your callers don’t know that you missed this ring; you’ve effectively lost time to answer the phone.
For both incoming and outgoing, it is real easy to have calls that have a lot of echo on the line. There’s a lot written up on this; and suffice it to say, dealing with the echo is hard. However it is only there if you have this analog/digital conversion thing happening on your landline; voip based calls don’t have it.
I’m using my desktop mac - nothing special here. Not even dedicated.
On my todo list: Set up a linux virtual machine, move the whole setup into that. Sync it nightly to another Mac and make it trivial for the wife to start it on that other host, just in case this rig dies when I’m unavailable to fix it.
Asterisk - Open Source PBX server
This is the heart of the beast. An overkill heart, at that, considering that our phone system could have just as easily spoken directly to our voip providers. The main reason we’re running it is for flexibility, and will be the subject of a followup article about our home pbx. The short gist of it is, we’re using it to filter incoming calls.
I first tried FreePBX, which is Asterisk with a web based GUI on top. Unfortunately, the UI is too confusing; the documentation too lacking. This really would make sense for some folks - once they knew how Asterisk worked underneath.
Now I’m using Asterisk 1.6 built from source, without a gui. Just plain old config files for everything.
The programming language for this is quite arcane. However, it is mostly functional. More importantly, you can have it call an outside script, one you can build in whatever language you want. Your script can issue commands, watch for responses, and otherwise intelligently do whatever you want with it. If you’re a hacker/tinkerer, this is quite handy. You could, if you so wished, actually handle all dialing features from within your own scripts, and just use the primitives that Asterisks provides.
Asterisk comes loaded with professionally created sound files - you can do almost anything you want with it, at least for typical business use (and then some). If you find you need something created, that matches the system’s voice, you can hire that out to Digium (the makes of Asterisk). Short sound files are \$12 for up to 15 words. I’ve been quite happy with the service :-).
FlowRoute - voip provider
FlowRoute’s main usefulness to us is as a very cheap pay-as-you-go voip provider, with no monthly minimums. They are prepay only (using Amazon’s payment system). So far, voice quality has been pretty good - better than I expected, actually.
Outbound calls are a penny a minute. Additionally, they let me set caller ID to whatever I see fit. In my case, calls that are forwarded retain their original caller ID; and any calls outbound, I can advertise my prefered phone numbers. International rates are pretty cheap (5.5c to India, 1.3c to China; 1c to 25c for London, depending on landline or mobile).
Inbound calls require you to purchase one or more incoming phone numbers. For my line, I’m paying 1.2c/minute with no monthly commitment. For the wife, a \$7/month with unlimited incoming calls.
Depending on how much my wife actually does use the phone, we may switch to a different provider (Voipo); but we won’t do that until we are ready to cancel her land-line and can “port” the phone number over to the voip provider. (Voipo does not let us set caller-id to whatever we wish).
This was the first provider I picked up; this is another pay-as-you go provider. This provider does only outbound calls. I originally planed to have the Cisco SPA-3102 handle analog calls for incoming; but based on voip quality (And lack thereof, on the cisco box) things changed.
VoipVoip does not allow setting caller id (all calls appear to be from “unknown caller”). The only thing I’ll probably continue using them for, is to call India (3c a minute to Bangalore).