*scottstuff*

CNet just posted an interview with Mark Spencer, the guy behind Asterisk. There aren’t a huge number of details, except for this one:

[Digium] has something that eluded many a Silicon Valley wannabee during the bubble: real revenues. The company pulls in about $10 million a year

Zing! Way to go Digium.

Since I’ve been having problems with my Verizon POTS line, I decided that it’s finally time to look into porting my main home phone number to a VoIP service. Right now, I’m using call forwarding on the line, so all incoming calls come in via VoIP anyway; using local number portability (LNP) to move the number directly onto a VoIP service won’t really make things any less reliable, but it will save me $15 or $20 per month.

I’ve been using NuFone for most of my VoIP traffic for over a year, but they only provide local numbers in Michigan; I’m in Washington, so that’s not very useful to me. I did a bit of research, and Teliax comes highly recommended on the Asterisk-Users mailing list and they’re able to do LNP ports in my rate center. Teliax seems to be sort of a cross between full-service VoIP providers like Vonage and pre-paid wholesale providers like NuFone–they have both prepaid by-the-minute plans and monthly minutes-included plans. Even the prepaid plans include the option of having them provide voicemail; it’s not very useful to me when I have my own Asterisk server, but I like the idea of having a provider that I use that I can recommend to non-Asterisk users.

Since Teliax can provide the service that I need, I went ahead and signed up. It only took a couple minutes to enter all of my data on their website, and my new incoming number was active immediately. Their support website includes Asterisk configuration snippets, dynamically-generated with my username and password, so all I had to do was paste them into my Asterisk config files and everything worked.

The one thing that I noticed immediately was that Teliax calls sounded a lot better then NuFone calls. By default Teliax uses the GSM codec, while I had NuFone set up for ILBC. By most accounts, ILBC should sound slightly better then GSM, so I’m not really sure what’s up. I changed my NuFone settings to use GSM, and suddenly the slightly swimmy sound that I’d been hearing on NuFone calls for the past few weeks went away. I suspect that my Asterisk build has a slightly-broken ILBC codec, but I wouldn’t have noticed this if I hadn’t added a Teliax account. Since this isn’t the first time that I’ve had ILBC problems, I’m going to drop it and stick with GSM for now. If it starts bothering me too much, then I’ll consider paying $10 and licensing G.729 from Digium, but I doubt there’s any purpose in doing that.

So far, I’ve only used Teliax for about 12 hours, and I haven’t ran more then a handful of calls through them, but so far them seem great. I have Asterisk set to reject them and log a message if their ping time rises over 400ms, but it didn’t trigger overnight. If it can make it a couple days without 400ms worth of network problems, then I’ll start the process of porting my home phone number to Teliax. For now, I’m going to change the forwarding on my Verizon POTS line to point to Teliax instead of NuFone.

Verizon’s doing weird things with my home phone again. At the moment, I’m paying for call forwarding on my home line, so any time someone calls my home number, the call is forwarded to my NuFone VoIP account. This was working just fine, but I’ve received busy signals when calling home twice in the last week. Since I’m able to call home via NuFone directly without any problems, I’m assuming that the problem is on Verizon’s end.

It’s taken a bit of poking around, but I think I’ve figured out what’s happening–Verizon will only forward one call at a time. If a second caller calls while a forwarded call is in progress, then the second caller will receive a busy signal. That’s a really nasty behavior–I’ve effectively lost the ability to do call waiting and to take voicemail messasges while I’m on the phone.

Since this is clearly happening on Verizon’s end, I called their local repair line and talked to someone. In the past, I’ve had much better luck with their repair number then their sales number–the repair people have access to the switch–but this time the agent was terminally confused by this–she got mixed up and thought that I was forwarding my calls to my cell phone, and concluded that it was the cell provider’s problem. Once she got into that state, I couldn’t figure out how to un-confuse her, so I guess I’ll have to wait until tomorrow and try to get a new repair tech.

I guess I should see this as an incentive to find someplace that do a LNP port of my home number to a cheap-ish VoIP service. It looks like Teliax might be able to do it, and most people on the Asterisk-Users list seem to think highly of them.

This isn’t exactly new news, but Cisco bought Sipura yesterday. Sipura makes a number of VoIP products, including the SPA-841 phone that I’ve been using for the past few weeks. They’re generally considered to have the best SIP implementation of any of the cheap vendors, and they make good, solid products for low prices. It’s a nice combination. Cisco has been licensing Sipura’s technology and using it in Linksys’s cheap VoIP hardware for around nine months now. Linksys has had to jump through a number of hoops to keep Sipura happy recently; apparently Sipura didn’t like customers buying the unlocked Linksys PAP2-NA instead of the more expensive Sipura SPA-2000. Now that Cisco owns both companies, I suspect that they’ll work out their differences.

Hopefully Cisco won’t gut Sipura to keep them from competing with Cisco’s more expensive products. The jury is still out on Cisco’s Linksys acquisition–they haven’t released many exciting new products since Cisco bought them, but they haven’t killed off any of their interesting product lines or tried to stop the flood of alternate Linux firmware distributions for the WRT54G family either.

One thing that’s interesting about this acquisition is that Sipura was formed by a bunch of ex-Cisco people. After Cisco bought Komodo in 2000, a bunch of the Komodo people left Cisco to go form Sipura. Now they’re back at Cisco again. This seems to be how Cisco does R&D these days–it spins employees off to work on their own products and then acquires them if they accomplish anything interesting. I’m not convinced that it’s a bad way to deal with R&D risk in a huge company–it shields Cisco from the cost of failure and promotes risk-taking by R&D engineers, but it doesn’t do anything to help unify Cisco’s massively fractured product lineup.

From looking back over my phone records, it looks like today marks the beginning of my second year of VoIP. I’m not sure which day I actually set up my Asterisk server, but I signed up with NuFone on March 23, 2004, one year ago today. At the time, I paid them $30 for pre-paid long-distance service, and have been chiseling away at that ever since. This morning, I still had $3.423 left of the original $30.

Since my monthly long-distance bill was around $20 before I started sending long-distance calls out over VoIP, I’m feeling pretty good about this. Of course, it’s probably best to ignore the $600 or so in VoIP hardware that I’ve purchased over the past year–it sort of screws up the cost-benefit equation.

Asterisk has grown a lot in the last 12 months–it’s reached version 1.0, and is rapidly approaching v1.1. Asterisk itself is very usable as a corporate PBX, and with tools like the Asterisk Management Portal and Asterisk@Home, it’s starting to be usable by people without a deep understanding of Linux, networking, and phone systems.

On the other hand, some problems still haven’t went away–it’s still really hard to find decent providers that sell local numbers across the country at decent rates. A year ago, good DID providers were almost completely nonexistent. Now there are a half-dozen or so companies that sell DIDs for Asterisk, but most of them are flaky in one way or another–they have bad support, or their terms of service are bizarre, or they don’t actually have any numbers in my local calling area. Hopefully a handful of solid providers will pop up this year, and I’ll actually be able to recommend them to people without any disclaimers.

I ordered a Sipura SPA-841 SIP phone from VoIPSupply.com last week, and it arrived last night. I haven’t had enough time with it yet to write a really comprehensive review, but I’d like to share a couple first impressions.

First–the SPA-841 is a lot smaller then I’d expected. It’s under half the volume of my Cisco 7940. It fit into a 2” tall FedEx mailing box, which I didn’t expect at all. Even though the base is small, it’s not very light–it feels like a real office phone, even if it’s a lot smaller then most of the office phones that I’ve used. It doesn’t seem to slide around too much on my desk.

Once I plugged it in, it booted very quickly. The Cisco phone takes around 30 seconds to boot, while the Sipura is ready for use in under 10 seconds.

The SPA-841 comes in a box with no documentation. Once you plug it in, you can configure it via HTTP using a web interface that the phone provides. Supposedly it’s also possible to feed it a configuration file, but Sipura only gives out the configuration file documentation and tools to VoIP service providers, not end-users. Personally, I’d rather edit text configuration files on a server and upload them to the phone then fiddle with the hundreds of settings that Sipura provides on their web interface, but if I’m only dealing with one phone, it isn’t a big difference. If I end up buying another couple SPA-841s for around the house, I’ll probably start agitating for open provisioning tools.

Even though there isn’t a whole lot of documentation, the phone isn’t too hard to configure. I spent about 15 minutes with it and had it accepting incoming calls, dialing out, and handling voice mail. The voice mail light (Message Waiting Indicator, or MWI) is just a dinky red LED sitting in the middle of the phone; I really like Cisco’s MWI a lot better. The Sipura also provides a MWI stutter dial tone, and it’s hard to miss that, even if you don’t see the tiny LED shining at you.

At this point, it seems to work, but I’m not completely happy with the way it’s configured. Once I’ve finished tweaking the config, I’ll write up a full review with pictures comparing it with the Cisco phone and provide a few configuration recommendations.

Update: I haven’t had time to finish the review yet, but I wanted to add a couple quick notes:

• The phone does come with a getting-started flyer, a glossy 8.5x11 mini-booklet with directions for plugging it in, connecting it to the network, and configuring it to talk to a few different SIP providers. It doesn’t come with anything more substantial. Sipura’s website has had a 71-page PDF Users’ Guide for a while, and just recently added a 79-page PDF Admin Guide. I haven’t had time to read the admin guide yet.

• The audio quality seems perfect. I’ve only spent a half-hour or so on the phone, but I haven’t noticed any dropouts. The handset is pleasantly loud.

• The latest firmware release, 3.1.1 (the update from last week’s 0.9.5–nice version number jump) includes support for “SIP-B,” which is apparently a standard being pushed by a few phone and softswitch vendors that make it easier to add PBX-like features to SIP phones. This includes bridged line appearances, shared missed-call DBs, called-number ID (the opposite of caller ID–it shows the name that goes with the number that you dialed), standardized call park/pickup support, and a few other useful features. Unfortunately, the SIP-B spec doesn’t appear to be public right now, even though the vendors involved have made some attempts at running pieces of it through the IETF’s standardization process. I suspect that SIP-B is really just a blanket name that covers a bunch of small, independent SIP enhancements that will be pushed through the I-D/RFC process one at a time, but for now there’s no real documentation available. Hopefully that will change soon so Asterisk can better support SIP-B hardware. (Micro-update: Sylantro has sent me a pile of documentation on SIP-B. I’m not sure that it’s complete, but there’s quite a bit of it, and they’re getting ready to put it on their website. So I’m mentally adding them to the “good guys” list when it comes to standards compliance and promotion)

• Several people have mentioned that they’ve had problems with the rubbery phone buttons on the SPA-841 sticking. I suspect that they’ve fixed this with more recent phones, as mine has been perfect. I wouldn’t say that the buttons are as nice as Cisco’s, but I don’t have any complaints.

I guess that’s a good summary of the phone–it’s not as nice as Cisco’s phones, but I have no complaints about it, either. It seems to work well enough, it has a decent feature set, and it’s cheap. I’d love to see them add PoE support, a ‘SPA-842’ model with a built-in Ethernet switch, a backlight for the LCD and buttons, and some way of supporting external dialing directories, but none of these are really critical–as it is, the phone works quite nicely, and I’ll probably order 2-3 more SPA-841s over the next few months.

How to get decent-quality sound recordings into Asterisk from a Mac without a ton of work:

1. Record with GarageBand. Use a real microphone, not the one built into the Mac. If your Mac doesn’t have a microphone jack, consider buying a Griffin iMic.
2. Export to iTunes. With GarageBand 1.0, this seems to be the only export option available.
3. Find the track in iTunes and convert it to an MP3. This shouldn’t be necessary (or really even a good idea), but my copy of sox (below) couldn’t handle the AIFF file that GarageBand produced.
4. Run sox with these options: sox recording.mp3 -r 8000 -w -s -c 1 recording.wav resample -ql
5. Verify that the WAV file sounds okay.
6. Copy the WAV file into /var/lib/asterisk/sounds. You can now use it with Asterisk’s Playback application.

The WAV file produced is sampled at 8 kHz, with 1 channel of 16-bit signed linear audio. This seems to be the best format for Asterisk, assuming that you don’t mind using around 16 KB/sec for audio files.

It looks like the FCC has ruled that VoIP providers can get phone numbers directly from the North American Numbering Plan Administration, without having to go through state regulators in each state first. While this FCC ruling was specifically for SBC, hordes of smaller VoIP companies will almost certainly do the same thing.

As anyone who has tried to get local phone numbers from smaller providers can attest, it’s really hard to find companies with local numbers across the entire country. I’ve been looking for almost a year, and I didn’t find numbers in my rate center until last week, when I discovered that iax.cc has numbers for only a $1.49/month plus $0.0137/minute. Before I found them, my best choice would have been paying someone like Vonage $25/month for services that I don’t really want.

Hopefully this FCC decision will lower the price of numbers substantially over the next few months.

Somehow I missed this when it came out yesterday–Level3, the US’s largest VoIP wholesaler, is now offering enhanced 911 services to its customers. Since their customers include Speakeasy, Net2phone, and a bunch of other big-name VoIP retailers, that means that 911 service is suddenly a lot easier to provide over VoIP. Also, it means that Level3’s competitors will probably be racing to match them.

The lack of reliable 911 service is generally considered to be residential VoIP’s single largest shortcoming right now; this should go a long ways towards fixing the problem.

Of course, it’s only going to work right when you’re at home, with your registered address. Travellers who take their VoIP adapter with them and then dial 911 will end up talking to their 911 operators at home, which isn’t going to be very useful to them. Until we have a way to send e911 location information as part of the SIP call setup, there won’t be a good way around this. Still, Level3’s new service is a big step forward.

I keep telling myself that sooner or later, I’ll install MythTV at home to replace one of my ailing TiVos, but I’ve never really got around to it. I like the idea of an open, networked PVR that I can easily fix when it breaks.

During today’s semi-monthly visit to the MythTV web site, I noticed a new feature: MythPhone. It’s a SIP-based videophone for MythTV. It’s new, and probably doesn’t work right, but it’d be a nice addition to my home Asterisk setup.