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,Busy
And 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 in Asterisk | Tags ael, asterisk, rubyonrails, voip | 6 comments
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 in Asterisk | Tags asterisk, rubyonrails, voip | 2 comments
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 in Asterisk | Tags asterisk, asterisk1.2, voip | no comments
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:
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 in Phones, Asterisk | Tags asterisk, gsm, mvno, phone, sip, voip | 2 comments
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 in Asterisk | Tags asterisk, extensions.conf, iax.conf, sip.conf, voip | 8 comments
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:
- Set up the local equivalent of a CLEC in some third-world country.
- Publish exorbitant rates for a block of numbers (say, $5/minute).
- Sign up with a US-based VoIP provider using a freshly-stolen credit-card number.
- Verify that the VoIP provider doesn’t know about your rates, and only charges a small fee to dial the country in question (say, $0.10/minute).
- Have a SIP auto-dialer call your numbers via the VoIP provider until the CC is declined or the account’s balance limit is hit.
- Repeat.
- At some point in the future, the VoIP provider will receive a massive bill from their upstream provider, while the Telco in whatever third-world country you’re using will hand you a big check.
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 in Computer Security, Asterisk | Tags asterisk, fraud, nufone, scam, voip | 3 comments
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 in Asterisk | Tags asterisk, digium, markspencer, voip | no comments
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 in Asterisk | Tags asterisk, gsm, ilbc, lnp, nufone, teliax, voip | 2 comments
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 in Asterisk | Tags callforwarding, nufone, teliax, verizon, voip | 1 comment
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 in Phones, Asterisk, Computer Networking | Tags cisco, sipura, voip | no comments
Posted by Scott Laird
Mon, 11 Apr 2005 19:49:03 GMT
Yikes, Digium has announced a DS3 card for Asterisk. So, the next time you need 672 channels of voice data all on a single PC, you know where to go.
It’ll be interesting to see what they’ve done with this–their current T1 cards have compatibility problems with some motherboards and have extremely tight timing requirements. In general, you don’t want to try running more then one of their 4-port T1 cards on a single PC. Since a DS3 carries 28 T1s worth of traffic, they must have done something to clean up their designs, or there’s no way it’ll be able to keep up with the load.
Posted in Asterisk | Tags asterisk, digium, ds3, ds3000p | no comments
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.
Posted in Asterisk | Tags asterisk, nufone, voip | no comments
Posted by Scott Laird
Tue, 15 Mar 2005 20:06:59 GMT
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.
Posted in Phones, Asterisk | Tags asterisk, phone, reviews, sipura, spa841, voip | 4 comments
Posted by Scott Laird
Sun, 06 Feb 2005 00:38:02 GMT
How to get decent-quality sound recordings into Asterisk from a Mac without a ton of work:
- 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.
- Export to iTunes. With GarageBand 1.0, this seems to be the only export option available.
- 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.
- Run
sox with these options: sox recording.mp3 -r 8000 -w -s -c 1 recording.wav resample -ql
- Verify that the WAV file sounds okay.
- 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.
Posted in Asterisk | Tags asterisk, garageband, mac, voip | 1 comment
Posted by Scott Laird
Sat, 05 Feb 2005 01:44:18 GMT
It all started on Wednesday night, when I discovered that my Asterisk VoIP server couldn’t make long-distance phone calls any more. I could dial, and the phone would ring, but as soon as the other end answered, the line went dead and the logs started filling up with error messages:
Feb 2 20:24:03 WARNING[-1250968656]: Huh? An ilbc frame that isn't a multiple of 50 bytes long from IAX2 (16)?
That’s kind of weird looking. Google wasn’t particularly helpful, so I decided that this was probably a VoIP provider problem and sent mail to NuFone’s support people. Historically, NuFone’s support department has a reputation of being a bit spotty–either they handle your problem right off the bat, or you never hear from them again, but that was probably a combination of broken email servers and staffing shortages due to rapid growth. In this case, I got a response back from NuFone’s president first thing in the morning. We’ve exchanged a bit of mail in the past, and in this case he agreed that it was a weird error, and the first thing to try was upgrading. I was running Asterisk 1.0.2, while the current version is 1.0.5. He suggested that I might be happier with checking a development version of out CVS–they’ve been more stable for him, anyway, and there are a ton of new features and bug fixes that aren’t in 1.0.5.
What the heck, I said. Sounds like fun.
So, I checked a copy of Asterisk and several of its supporting libraries and drivers out of CVS, built them, installed them, and tried restarting Asterisk. First problem: I’d built my kernel without support for module unloading, somehow. So, I had to reboot to upgrade the drivers for my two PCI VoIP cards. Grumble, grumble. Oh well, I’d been meaning to shut it down and add more RAM anyway; the RAM was sitting right there, and it shouldn’t take long to slide the server out from under its desk and install the RAM. Five minutes later, the box was booting back up with the new drivers and twice as much memory.
Problem 2: As soon as I started Asterisk, I discovered that the X101P PCI card that handles my regular POTS line was reporting a ‘red alarm’ on the line, suggesting that it wasn’t connected right. I crawled under the desk, unplugged the phone line, wiggled connectors, and tested it with a $10 phone that I keep on hand for situations like this. I got a dial tone just fine with the phone, so I plugged the line back into the PC and crawled back out from under the desk to discover…
Problem 3: the server locked up while I was fiddling with the phone line. So I rebooted it, only to discover…
Problem 4: it couldn’t load the drivers for the PCI card anymore. The driver spit out a number of weird errors:
kernel: wcfxo: Out of space to write register 06 with e0
kernel: wcfxo: Out of space to write register 0f with 00
That’s ugly. I developed a new theory: the card wasn’t getting reset right on reboot. So, I powered the box completely down, pulled the power cord for a minute, and then tried again. It worked this time; everything came up right, and I was able to verify that everything was working correctly. Incoming calls worked right, and outgoing calls worked, via both POTS and VoIP.
So fast-forward a few hours. I try to call out, and nothing happens–I just get a dead line. Checking Asterisk’s console, it looks like it dialed out on the POTS line, but I wasn’t hearing ringing or anything. The driver for the X101P PCI card had apparently choked again. I tried stopping and restarting Asterisk, but it didn’t help. I didn’t worry too much, though–incoming calls on my POTS line get forwarded to VoIP after about 15 seconds worth of ringing. Somewhere in the middle of this, a couple calls came in from family members, but I wasn’t close enough to a phone to answer them, so I let them go to voice mail. Except neither call left a message. That seemed strange (my family is big on leaving messages), so I called back, and they said that the line had went dead right after it answered–they never heard the “leave a message” message.
Problem 5: incoming VoIP calls weren’t working, either. The connection died in the middle of a Playback instruction. I tried changing codecs, restarting Asterisk, and even rebooting, all without success. My frustration level was rising. Quickly. At this point, almost nothing was working right–POTS was dead, and incoming VoIP calls only worked right if I answered the phone. I took a break, put the kids to bed, and resisted the urge to scream. I mean, I debug problems like this for a living, which means that I really hate to do it at home.
So I spent a minute thinking–it was really weird that rebooting didn’t fix the VoIP Voicemail problem. I’d tested it earlier in the day, and it had worked after the upgrade to CVS Asterisk. And I hadn’t changed any other software since then. I rebooted again, and this time the X101P PCI VoIP card disappeared completely. Asterisk wouldn’t even start now–my config files are sprinkled with references to this card, and Asterisk was aborting with “can’t find card” errors. So I thought a bit more and came up with a new theory–when I’d pulled the box out to install more RAM, maybe I’d pulled on the phone cable that goes into the X101P a bit too hard, and the card had popped partway out of the slot. It’s easy to test, so I shut the server down again, slid it out from under the desk, popped the PCI card out, checked it for obvious problems, and put it back in firmly. I then plugged everything back in, rebooted, and discovered that everything was working perfectly. POTS worked, incoming voicemail worked via POTS or VoIP, outgoing VoIP calls worked.
I checked again this morning, and everything’s still working fine.
So, did I really pull the card halfway out, or was yesterday’s Asterisk CVS tree just really unstable? And what sort of person goes through this just to save $15 per month on phone calls? And speaking of that–if I’d had a second VoIP provider set up, then I could have shunted calls to them when outgoing calls to NuFone failed, and I would have been able to avoid part of the mess. Hey, yeah, and I still need to finish the Asterisk-caller-ID-on-MythTV project, so I can see who’s calling on the TV while I’m watching movies. Oohh, and my shiny new internal PCI ADSL card is supposed to ship soon. And I need to mod an Xbox so I can run MythTV on it downstairs…
I guess that should answer the “what sort of person” question. A person with incurable gadget lust. I’m getting better. Really.
Posted in Asterisk | Tags asterisk, broken, ilbc, nufone, upgrade | 3 comments
