Posted by Scott Laird Thu, 17 Nov 2005 14:08:49 GMT
It looks like Asterisk 1.2 is out. From the release announcement:
This release of Asterisk contains over 3,000 improvements on version 1.0, including hundreds of new features and applications.
I’m planning on upgrading my home Asterisk server soon, possibly today, and then I’ll look at migrating pieces of my dial plan to RAGI and/or AEL.
Posted by Scott Laird Mon, 10 Oct 2005 19:28:54 GMT
While I really like all of the things that Asterisk allows me to do with my phone system, I’m really not very fond of its configuration language. The language provided in Asterisk 1.0 is slightly better then sendmail.cf, but it’s still a lousy language. They made a few small improvements early in the Asterisk 1.2 development process that helped a bit, but I assumed that we’d have to wait for another year or two before someone broke down and wrote a decent language for Asterisk.
It looks like I may have been overly pessimestic. I discovered the Asterisk extension language on a list of new features for Asterisk 1.2 today. Somehow I’d missed this when it first went into Asterisk.
Here’s a chunk out of my old config file. It’s not perfect, but it’s what you get when you’re stuck dealing with line numbers:
[macro-diallocal]
exten => s,1,AbsoluteTimeout(7200)
exten => s,n,SetAMAFlags(default)
exten => s,n(analog),Dial(${TRUNK}/${ARG1})
exten => s,n,Congestion
exten => s,analog+101,Macro(condsetcid)
exten => s,n,SetCIDName(LAIRD SCOTT)
exten => s,n,SetAMAFlags(billing)
exten => s,n(nufone),Dial(${NUFONE}/1${ARG1})
exten => s,n,Congestion
exten => s,nufone+101,BusyAnd here’s the equivalent using the new configuration language:
macro diallocal( number ) {
AbsoluteTimeout(7200);
SetAMAFlags(default);
Dial(${TRUNK}/${number});
if(${DIALSTATUS} = "CONGESTION" || ${DIALSTATUS} = "CHANUNAVAILABLE") {
&condsetcid();
SetCIDName("LAIRD SCOTT");
SetAMAFlags(billing);
Dial(${NUFONE}/1${number});
switch(${DIALSTATUS}) {
case BUSY:
Busy;
default:
Congestion;
}
}
}It’s still not ideal–${VAR} is ugly–but it’s vastly better then the old syntax. This plus a bit of Rails integration should give us a really nice phone environment. I’m rapidly approaching the “tear it down and rebuild it” point with my Asterisk system–there’s a lot of stuff that I’d like to add that just doesn’t fit in with my current configuration files–so I’ll have a set of articles on integrating Asterisk, AEL, and Rails in a few weeks.
Posted by Scott Laird Sat, 08 Oct 2005 00:54:18 GMT
One side effect of running my own business is that I’ve been spending a lot of time on the phone. Unfortunately, the $10 analog phone that I was using was hard to use for more then 15 minutes at a time, and it wasn’t always very easy to hear what the other party was saying. That’s not a great mix for a business phone.
So, I’ve been looking around to find a cheap way to get a good phone with a headset on my desk. None of the analog headsets phones that I’ve looked at have been very appealing, so I’ve been looking at cheap VoIP phones with 2.5mm phone jacks. Amazingly enough, VoIPSupply.com decided to clear out a bunch of old phones via eBay at just the right time, so I was able to pick up a Pingtel Xpressa for a song.
Pingtel Xpressa
The Xpressa was a first-generation VoIP phone, and it was orphaned by Pingtel over a year ago. However, in many ways it’s still the highest-end SIP phone on the market. It comes with a Palm-like 160x160 grayscale display, supports (pre-standard) Power-over-Ethernet, and runs Java applets natively on the phone. The SDK is still available from Pingtel, but you have to hunt for it a bit. I paid about the same for the Xpressa as I did for my Sipura/Linksys SPA-841, but the Pingtel is clearly better in nearly every area–it’s easier to use, it’s more solid, it’s more attractive, it has more features, and sounds better. It’s lacking a few NAT features and I can’t find a way to use different ringtones for different lines, but other then that it does everything that I need. It comes with a standard 2.5mm headphone jack, so I picked up a cheap Plantronics headset; I’ve spent nearly two hours on the phone so far today, and everything has been perfect.
Er, well, mostly everything. I actually ordered the mango colored model from VoIPSupply, but somehow ended up with a charcoal-colored phone instead. Hopefully they’ll be able to fix that soon. Also, since Pingtel discontinued the phone (and actually sold it off to an unnamed vendor, alleged to be 3com), they’ve pulled all of the add-on software packages off of their website; that means that I haven’t been able to find their LDAP-based phonebook. I’ve been fishing around and I’m sure that I can find a copy somewhere. For now, I’ve been using Jon’s Phone Tool to dial from Quicksilver. JPT is swimming in features, 95% of which are useless to me, but it seems to do a decent job dialing my phone for me, so I’ll probably pay the $12 fee for it if I can’t get the Pingtel Phonebook to work.
Update: As expected, VoIPSupply is fixing the mango-colored problem. I should have a mango phone on the way shortly.
Posted by Scott Laird Tue, 13 Sep 2005 17:31:16 GMT
So pretty much nothing worked yesterday.
It started with an 8:00 phone call over VoIP that only had one-way audio. Then my 11:00 phone call gave a potential client an “all circuits busy” instead of ringing through to my VoIP phone. I’ve been using Asterisk for almost 18 months, and this is the first time that I’ve ever seen either of these, so I spent a while trying to reproduce the problems and sending support email off to two different VoIP providers.
After that, I finally started in on The Evil Thing–I bought a copy of XP and I was planning on installing it on a spare PC so I could test Typo with IE 6. How hard could it be, right?
I gave up on it at 8:00 PM.
Here’s a short list of things that went wrong:
I couldn’t find an old Windows CD to use to make the upgrade test on my XP CD happy. I should have CDs for ‘98 and 2000 sitting here somewhere, but I couldn’t find either. I had to borrow one to make XP’s installer happy. It would have been easier to download a cracked copy then to use the legitimate version and fight with its copy and licensing protection.
Once I got past the upgrade test, the installer refused to format my hard drive. No matter which options I picked (full disk or small partition, NTFS or FAT, quick format or full format), it would always die out within 5 seconds with a “Setup was unable to format the partition” error. The error suggests that I check the power on my external SCSI drive. Since I’m installing onto a completely standard 80 GB internal IDE drive, the error isn’t very helpful. Digging around a bit, bad IDE cables and bad CD drives seem to be the most common causes for this error. Since this is an old box that I put together from spare parts, the system is using old 40-pin IDE cables; I need to swing by a store and pick up a couple 80-pin IDE cables. Maybe that will help.
For the fun of it, I tried booting my borrowed XP disk (the one that I was using to pass the upgrade test), and *it* partitioned the drive without any problems. Unfortunately, it refused to take my license key. The nice hologrammed one that came directly from Microsoft. Apparently my key is just good for XP Pro Upgrade CDs that come with SP2 pre-installed or something. Rebooting with my CD put me right back into formatting limbo.
I swear, I should have just downloaded and installed a cracked version–I would have been done early yesterday afternoon.
Posted by Scott Laird Mon, 12 Sep 2005 14:11:20 GMT
Oooh. Just when it looks like I’m going to have to stoop to programming in Perl to do a bit of Asterisk integration, someone comes out with a way to use Rails with Asterisk’s AGI scripting interface.
This has the potential to be really interesting, because Asterisk+Rails should be about a decade more advanced then anything that the IVR people are used to seeing.
I have a couple small Asterisk scripting jobs that I’ve promised to various family memebers; if my schedule allows, I’ll give RAGI a spin and share how it goes later this week.
Posted by Scott Laird Tue, 26 Jul 2005 18:41:11 GMT
I’ve fallen a bit behind on Asterisk with all of the time that I’ve spent working on Typo lately, but I found this announcement sitting in my inbox today:
As previously mentioned on the lists by Olle Johannson, we are actively trying to get Asterisk in shape for a 1.2 release within the next 60 days.
…
I will produce the first release candidate on August 20th, with followup versions produced every week until we deem the release ready for public consumption. I expect it will require at least three -RC releases for us to get things in shape, so that means that 1.2 itself may be ready by September 15th.
Asterisk has desperately needed a new stable release for months now, and it looks like version 1.2 will be out within the next two months.
Posted by Scott Laird Fri, 08 Jul 2005 23:50:28 GMT
Darn it, I think I’ve fallen in love with another unreleased phone.
This week, it’s the Nokia N91. It was announced a couple months ago, and isn’t supposed to ship until late in 2005. It’s going to be marketed as a “music phone,” but I think the specs more or less speak for themselves:
You can get more details from Nokia’s own flashcrapular site.
It’s not a small phone by any stretch. It’s very slightly smaller then a Treo 650–4mm narrower, 1mm thinner, 16g lighter.
The thing that I find so fascinating about the N91 is that it can replace practically every device that I’ve been cramming into my pockets:
The other thing that fascinates me about the N91 is its SIP support. The specs list support for JSR-180, which is SIP for J2ME apps. There are rumors online that the demo N-series phones have native SIP support in the phone UI. That’d let me use the N91 as a cordless phone when I’m at home or at work, which is just one more thing to like about it.
Of course, there have to be downsides–the camera doesn’t look as good as the one that comes with the N90 (but the N90 doesn’t have 802.11 or the hard drive). It doesn’t have the N90’s video-conferencing camera, either (that’d be cool with SIP). It’s kind of big. It doesn’t have a keyboard (although external bluetooth ones will work). It won’t ship until the very end of 2005. The specs don’t list 850 MHz support, although they’re clearly marketing this to the US, so presumably there will be a US model with 850/1900 MHz support. Finally, the price: at least $700 US before subsidies, possibly closer to $900. So, frankly, it’s probably too expensive for me to buy, but I’m going to be really tempted. Since Palm is rumored to be saving the next Treo for Spring of 2006, the N91 may not even have any competition for “Cool Phone of the Year” in my mind.
Posted by Scott Laird Fri, 08 Jul 2005 18:35:26 GMT
I want to see a combination VoIP/MVNO double-play. That’s one company that sells cellular service (using someone else’s network), sells VoIP service, and integrates the two services.
There are two specific scenarios that seem obvious:
They sell the customer mobile phone service and VoIP service via an ATA or SIP hardphone. This would be great for people who have turned off their home phone service while continuing to pay for some form of Internet access. One number would ring both devices, the first one to pick up wins. Outgoing calls would have the same caller ID from either device. Alternate products would be family plans with multiple phones, each with their own number, then a shared number that will ring all devices; and centrex plans for small businesses, where the company provides both VoIP desk phones and mobile phones.
They sell the customer mobile phone service and act like a SIP client. This way the customer can integrate their mobile phones directly into their existing phone system. Ideally, the MVNO’s SIP gateway will register and unregister with the SIP PBX as the phone gains and loses service; this will let the PBX do the Right Thing with voicemail, and also enable a number of other services.
Personally, I’d love to buy services like this. I’d prefer GSM phones, simply because the most interesting phones are almost always GSM-only. Accoring to Cellular News, there are at least three GSM MVNOs in the US right now, using both Cingular and T-Mobile’s networks, so this is certainly possible.
On the hardware front, several companies seem to be providing GSM/SIP gateway equipment:
Bridgeport Networks seems to be marketing these exact ideas to carriers.
Blueslice concentrates more on WiFi/GSM roaming, but the basic technology seems similar.
It’s possible that Earthlink will be rolling something like this out soon–they’re building a MVNO, and they share at least one board member with Bridgeport Networks. They seem to be concentrating more on EV-DO then anything GSM-related, though.
There seems to be a huge push in the cellular industry to integrate SIP into their networks, so something like this will be possible sooner or later. My current contract with Cingular is up in 6 months; it’d be nice if someone has something on the market that I can buy by then.
Posted by Scott Laird Thu, 07 Jul 2005 16:12:00 GMT
I’ve had several people ask me for a more recent version of my Asterisk configuration files, and it’s been about a year since I last posted them, so here they are.
Read more...Posted by Scott Laird Tue, 28 Jun 2005 02:43:19 GMT
It’s amazing the ways that people can find to steal money. The latest scam goes like this:
This is just a twist on the old “modem dialer” scam, but it’s been costing VoIP providers big money. NuFone has apparently lost $400,000 recently, and other providers are reporting huge fraud rates–according to Teliax, roughly 1/3 of their new customer signups are fraudulent.
There’s a long thread following this on Digium’s list server.
It looks like this has helped kill at least one VoIP provider: LiveVoIP has declared bankruptcy, although they had a spotty reputation to start with.
Posted by Scott Laird Thu, 09 Jun 2005 17:53:30 GMT
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.
Posted by Scott Laird Thu, 12 May 2005 14:11:30 GMT
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.
Posted by Scott Laird Tue, 10 May 2005 23:58:37 GMT
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.
Posted by Scott Laird Wed, 27 Apr 2005 18:56:11 GMT
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.
Posted by Scott Laird Wed, 23 Mar 2005 18:25:15 GMT
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.
