I am Patrick Perdue! I am bad for you!

Well, what do ya know? I'm actually writing in the blog again, properly this time, rather than in the guise of a delayed, semi-automated RSS feed. How's that for something?
Not impressed? Oh well, it's all good. I'm not either.

As you've probably noticed, no archive was posted for last week's show, which, by the way, did actually exist. While some fun things did happen, I wasn't really happy with the over-all result. At the time, I was being plagued by a migraine from hell. I did manage to do a few fun things, such as putting a well-known and quite horrible song through a midi controlled Granulab, which was rather amusing (at least to me), showed the last couple of minutes of one of the local TV station's analog feed as it went off the air forever, and complained about various things.
For some inexplicable reason, I extended the show for almost an entire hour just to talk to Alexander Nelson about keyboards and pointless things, and to play around a bit with the Ensoniq TS-12.
Yeah, it's all quite boring. Maybe I'll post a highly edited version of that archive at some point, but I really just wasn't in the mood to deal with it after the show, nor did I feel like it this week either, apparently.

Speaking of not feeling like doing things this week, that's the position I found myself in yesterday. Since Derek was busy, Arfy was... er... somewhere, and the Clower brothers were away visiting relatives, I decided that being the only live person on a Saturday was quite boring. Besides, it was really hot down there, as it hit 99 degrees Fahrenheit yesterday, and I didn't have fans or air conditioning running in the studio all day. These combined issues yielded the end result of my staying upstairs and being lazy. Pretty sad when you don't even feel like doing a self-appointed task from which enjoyment is usually gleamed, isn't it? But, such is life. Oh well.
Derek and I have both agreed to do our respective missed shows at some point on Wednesday, so stay tuned for that. I don't think I've ever done Things and Stuff on a Wednesday. I'm usually pretty inflexible about my time slot, since I like it just the way it is. I don't want three hours, or different hours, thanks for asking. On this TBRN fake station thingy, the only exceptions have been when I've missed shows completely due either to laziness or special events, or stupid things like the broadcastathons.

Enough of that... I know you're probably about as bored reading all this internal whining as I am of writing it, so no further effort will be exerted on my behalf, at least on that topic.

Now, for some stupid news about my recent life, or lack thereof:

I won third place in the NokiaMailtone contest for my entry made with sign and triangle waves and Sound Forge's tone generator, all without getting out of bed. Apparently, the third place prise is a new Nokia E63, which is a lot like the E71 I have. It's stripped in some ways, I.E. no internal GPS or HSDPA (although it's still 3g), the camera isn't as good (2.0 vs. 3.2 MP I think), and it's plastic instead of metal, which, ironically, makes it less resistant to fingerprints, although it might not look as shiny at first. However, it does have a real 3.5mm headphone jack instead of the 2.5mm of the E71, which is really one of my very few complaints about that particular phone. While I have converters and adapters for everything, I don't like putting extra weight on the jack, which is done a bit even if using a 2.5mm to 3.5mm jumper cable. Now that I'm actually starting to use it for downloading and listening to podcasts, reading books in text format, etc. it would be nice to have a real native jack.
So, I guess they'll send that to me at some point.
I haven't actually gotten a firm confirmation on that, so yeah, in either case, it's something to do.
freakyfwoof also submitted an entry, but sadly was not even put into the top five. Shame, really, because I honestly thought his submission was better than mine.

I also managed to break my Iriver H320's hard drive by dropping the player while it was recording, and the drive was caching from memory. I could easily get a new drive and a new battery, since it needs both, and I probably will still do that at some point, or more likely attempt to revive one of these busted H120's, but for now, I bought a used Iriver H10 on Ebay. No line-in recording, but I really just wanted a good solid player for the trip to Orlando. I rockboxed it, copied things to it, and played around with it a bit. Seems to be all right so far.
In place of the up/down buttons found on all the previous Iriver units I've owned, there is a touch strip, kind of like a really narrow rectangular laptop touchpad with only two contact points, one on either end, corresponding, of course, to up and down. Honestly, I like it better than the buttons, which I realize shouldn't be the case, but I do, so there. The only real issue I have at the moment is that the current Rockbox daily build seems to have issues with the H10's fm radio, mainly being that it doesn't actually turn the thing on. You can fake scan for presets, and go up and down the fake spectrum, but it doesn't really do much. Oh, I think I got it to go "click" a couple of times, but that's about it. Maybe I'm missing something really obvious, but the radios in either my 120 or 320 never did these things when controlled by Rockbox. Oh well, it's not really important anyway. It's just terrestrial commercial badness, though it's still good to have around sometimes.

In other news, I bought a mystery microphone on Ebay for $5 the other day. It's a dynamic mic of some kind, and the guy who sold it doesn't know what it is, since the switch plate containing the make and model has gone missing. It's rather vintage looking, and could be something from Electro Voice, which generally doesn't suck, although nobody seems to really know what it is. So, it could be good, or a pile of crap. In either case, it was $5, and I'm always looking for different microphones for different situations. I still wouldn't mind another Heil PR20, so I can use one on the snare drum in place of the Behringer XM8500 I have there now, which is still better than the Audix snare mic I was using originally. On the other hand, perhaps I could acquire a super cardioid dynamic. I currently don't have any of those, just standard cardioids, and the nice, old, slightly noisy Shure SM85 from 1983, which is an electret condenser that takes phantom power. It's probably got the narrowest axis of any of the mics I have, which makes it incredibly useful for stuff like singing while at the drum kit, or, in my case, being stupid while banging badly on drums. I want something with that kind of directionality, but not a condenser, and with maybe a bit more on the bottom end of the spectrum. It probably won't happen though. Even the Heil PR20's pick-up pattern is a bit wider than the SM85. I may have to eventually get something like a Shure SM7b, EV RE20, or god forbid... a pr40? Nah, wait, scrap those last two mics, and probably the first one as well. I've yet to spend $350 on a microphone, though I've come almost close, and I'm not really doing enough to warrant the expense. Oh well, hasn't stopped me before.

Sleep, however, has stopped me from doing many things, like continuing to type in this virtual box of issues. It is, in fact, telling me to do so now, so who am I to refuse? A sleepy person, that's who!

As most of you will know, I've been playing with VoIP stuff over the past few months. I've come to the conclusion, given how things are, that a good, solid sip softphone would be the best over-all solution for making/taking phone calls on live shows and such, although I wouldn't use a softphone for everyday phone use. That's just a bit restricting.
Last June, I purchased a JK Audio AutoHybrid to interface between my board and the Vonage line. After using several fake phone patch solutions over the years, this was the first true patch I had that was actually useful.
This is a dual transformer, passive patch, with no ICs or active electronics of any kind, so naturally it's not the best solution for those using pots lines unless you want to do lots of ducking on the output. If you don't, you'll get a very filtered instance of your dry signal when the line isn't active. This kind of thing works better on VoIP systems, because you don't have the loop length problem, plus input and output are rather hot compared to PSTN (unless it's attenuated by the ATA), giving a better ratio of signal-to-null by default.

Because this box is passive and has no DSP, and you're only using two wires, you'll get a bit of residual return from your input signal, which has this tendency to break things, particularly stereo images, when you have mics panned out and routed to the phone patch itself. Knowing this, and not wanting to spend $400 on a digital patch (and that was the cheapest one around), plus the fact that this particular one would have needed some serious attenuation to not suck on my line, I decided to use a side-chain on my Phonic PCL3200 compressor in conjunction with two aux sends to duck the signal down when the mic hit the side-chain. This worked OK, with the drawback of only partial full duplex. So, sometimes, things would get missed as a result of my saying things, which was slightly annoying. I say partial full duplex, because it was a bit less limited than half duplex, but not quite full. So maybe 3/4 duplex?

I was relatively happy with this setup, given it's weird issues and limitations, until I got my Sipura SPA-2000 ATA and Broadvoice. I could never get the phone patch to sound quite as nice with the new ATA, plus I was having some weird loopback issues. So, with Broadvoice's open equipment policy, I decided to try some softphones, thinking that would be the end to all those annoying and strange issues, plus I could cut it back down to one aux send instead of using two (one for side-chain and the other for phone input).
Well, this was great in theory, but there are still several annoying shortcomings that I'm still dealing with now.

The first softphone I tried was ExpressTalk. It's very accessible, sounds great, and works... most of the time, anyway.
The problem with this one, however, is that it comes packaged with a lot of software that you most likely won't need if you just want a softphone, such as IMS Telephone on hold player, Axon PBX, VRS recording system, and IVM answering attendant, which, I admit, is pretty cool in it's own right. However, unsuspecting users who install ExpressTalk with default settings might be slightly annoyed at the hold music and messages coming across the sound card, a phone recording system being launched and binding itself to the default recording interface, and separate auto answering attendant and PBX modules that you probably don't want or need being automatically launched and configured in the background.
For the record, I'm currently using Axon until I get a Trixbox system running, which I'm planning on doing shortly. It's decent if you want a basic PBX that will rout calls, but it delegates other things that Asterisk has built in, such as music on hold, answering attendant, etc. to NCH SwiftSound's other modules. The entire package of software can get quite expensive, and is not as flexible as the Asterisk-based stuff. The IMS system is pretty neat, and IVM can do rather a lot, especially if you don't feel like writing out conf files or playing with command line thingies.

So, all the extra software aside, which is fun cleaning up, by the way, there are some other issues with ExpressTalk. For one thing, it likes to play with your incoming and outgoing volumes, taking them both incredibly high by default. There are ways around this, but I really hate it when audio apps take over my stuff like that! If I have levels set where they are, it's for a reason!!!
Then there's the fun default behavior it exhibits when you put people on hold. Now, most other hard and softphones I've seen will simply be nice, and put people on hold like they're supposed to, and not be obtrusive about it, passing things off to the PBX and letting it do whatever it's supposed to do when phones are on hold.
ExpressTalk, however, has two options:
1. Play a really, really annoying wave file over and over again (ask beeping_becky if you don't believe me), or
2. connect to an IMS server.
Well, if you're using ExpressTalk strictly as a phone, you'll probably not be using an IMS server. Plus, the default thingies are bad, and it's one more thing you'd have to change.
Oh yeah, then there's this issue wherein sometimes, only one half of the conversation comes back when you take people off hold.
Conferencing, transfer and the intercom buttons (which I think are just quickdials) require the full version, which you can use for thirty days before lots of rather annoying reminders pop up telling you that your thirty day trial has expired, after which point you are reverted to the features of the free product. This wouldn't be so bad, although call transfer and especially conferencing are really basic features that can be found in a lot of free stuff.
I've also had it try to randomly connect to a VRS server which I don't run, regardless of the fact that I've told it that no VRS server exists.
And, as I found out last night, it will sometimes not forward incoming audio properly. For example: I have an extension on TBRN's PBX (very much in development at this time), which is registered on my own PBX. Calls in or out to TBRN mostly don't work with ExpressTalk, although they work perfectly fine with any other softphone or my ATA. I've seen it work before, and I know it's an ExpressTalk issue, but I can't seem to make it work or not work 100% of the time.

All this aside, it's the most accessible softphone I've used thus far, and it sounds great when it's not being stupid.

The next phone I tried was SJPhone which is completely free, and showed some potential. For one thing, call transfer and conferencing didn't cost extra, and unlike ExpressTalk, it has auto answer and auto conference features, which are really neat. I, however, found several problems with SJPhone as well.
For one thing, the defaults make it look a lot less accessible to screen readers than it actually is. This is actually one of the lesser problems, as you can simply turn off any active skin, and you get a basic looking program with some obvious buttons.
The problems show up when you try to answer calls, if you're not using auto-answer, that is...
With version 1.60 (stable at the time I tried it), the incoming call dialogue would sometimes forget to exist, meaning the phone could ring forever, but you'd have no way to answer it, as they thoughtfully didn't provide a global hotkey for answer and hangup, although there is a nice, dedicated hotkeys tab in the options dialogue.
Also, while it has an auto conference feature, the version I trried did not compensate for added amplitude from additional calls. As a result, all conferenced callers would come in at the same level, and clip everything out. You're supposed to devide the total amplitude with this sort of thing, so that doesn't happen.
By default, and I never found a way to change this, making an outgoing call was torture, since it would continuously play a 2-second loop of a U.S. ring, with no gaps at all, while the call was being placed. Imagine one long, continuous "riiiiiiiiiiiiiiiing" while making calls... Very bad!
Another annoying thing about SJPhone is it's tendency to flash the call status, including name, duration, and codec being used, on top of other windows.
Also, you can't assign sound events other than incoming call, which means no more "boing" at the end of a call. That's not the end of the world or anything, and it certainly makes it no less useful, but it does make it less fun.
DirectX audio support is very broken as the audio is being under-buffered, so don't use it. Fortunately, you can turn the option off, and it does let you change buffers manually on the new stable version (more on that in a second).
Hold functionality, however, works just like it's supposed to, and doesn't break things, like ExpressTalk does.
Supposedly, the latest stable build of SJPhone fixes some of these issues, but the sip functionality is either very broken, or I have something very badly configured.
Example; a call comes in. My computer hangs for about 10 seconds. I get a short burst of ring, then the call is droppped. Uh, yeah, really useful there.

Again, it's one of those things that sounds really nice when it works. Actually, the audio quality might be a bit more solid than ExpressTalk when properly configured, and I would use it over ExpressTalk if some of the issues would go away.

Now, we have X Lite,a free, semi-transparent, sort of accessible softphone.
When I first loaded it, I was able to see my Miranda, and through that my desktop, but not X Lite's call logs. This, of course, was rather bad. I did, however, manage to get some graphics labeled, and get it reading well enough to use it. Like SJPhone, it has a conference thingy, and auto-answer. Unlike SJPhone, it conferences properly, compensating for amplitude the way it's supposed to. However, it doesn't auto conference incoming calls, which may not be a terribly bad thing. It's another one of those things that's fun to play with, though.
Again, there are no global hotkeys for taking calls and hanging up; this all needs to be done from the X Lite window, or from the system tray. Raaaaaaaaah! This is a disturbing trend!
At least there are hotkeys for answering, hanging up, placing calls on hold and taking second calls while in the X Lite window, but how 'bout some global ones that will work in any window?
Despite your preferences, when a second call shows up, it will beep at you over your phone output device rather than the ringing device, which, in my opinion, it really shouldn't do. As a result, if you're streaming, everyone will no, by the very annoying "beep beep", that another call is coming in. This doesn't sound cool, and the other softphones I've seen have ways around this. Even setting the ringing device to "none" doesn't fix this behavior.
Also, I can't find a way to read the program and missed call logs reliably, although I can see parts of it sometimes.
The audio system on this phone isn't as good as ExpressTalk or SJPhone, as it breaks up sometimes for no particular reason, presumably another DirectX under-buffering issue which the end-user can't really do anything about, although it's not as bad as the next one I'll write about.
If the audio issues were fixed, and if the call waiting beeps could be redirected properly, I'd use it, even given it's slight lack of true readability, as it has a pretty small footprint.

Then, there's this other cute little thing called Phoner Lite. This has to be the most non-standard looking thing I've seen in a long time. It has a bunch of tabs in multiple dialogues, with each dialogue having it's own tab control. However, all the tab controls have only one tab on them. Why?
This phone is small and cute, requiring no installer. However, it's audio system is horribly screwed up, you have to go through the entire setup wizard in the help menu to change your sound devices (not such an obvious thing to do), and there are loads of graphical buttons that some people had issues finding. I found them, but... well... it wasn't really worth it in the end.

So, getting desperate, I even tried a sip plugin for Miranda, which really tried to work. However, it, like everything else, had some really odd issues.
Putting people on hold with this one is a really bad idea, as when you go off hold, instead of the callee hearing you, they get a nice, clean loopback of themselves. And I do mean a bloody clean one!
Incoming calls show up like authorization requests, and if you have missed calls, you'll get a long list of fake old calls that you supposedly still have coming in, which you have to manually deny, if you want them to go away.
It doesn't want to create multiple sip sessions, even though it has a conferencing feature. Of course, when you get an error every time a second call comes in when you try to answer the new call, or while making a second outbound call, that's sort of a moot point, now isn't it?
Plus, it has this bulky sort of session manager thing that really gets in the way while calls are active. So, while that would have been kind of nice in theory, that sort of broke.
Oh yeah, and of course the audio side of things just had to go and work perfectly... Just to annoy me a bit further, ya know?

I've yet to try Diax, which is a softphone that only works with the Asterisk IAX2 protocol, but it's last update was about two years ago. So if there are any issues with it, they probably won't get fixed any time soon.

I've tried Gizmo's sip support. Let's just not talk about that, shall we? Saves me a lot of typing.

Of course, Skype is out for two main reasons; it's huge, and it's not sip compliant. There are lots of sub-reasons, but most of you will already know my viewpoint on Skype, so I won't bother stating those again.

Conclusion; there is no decent softphone that actually does all the things it's supposed to, and does them properly, that I know of, anyway. If anyone knows something I don't, please feel free to share.
To be perfectly honest, I'd be happy if someone made a hardware sip phone with just a quarter inch (or balanced XLR) input and output, an RJ-45 with a web interface, a keypad for dialing, the ability to transfer calls, and conference both incoming and outgoing calls. Oh, and of course there would be residual from the input signal. Why should there be one at all when you're not using copper wires anymore. Sure, put a mix minus as well as mix+dry in there in case someone needs that sort of thing.
If such a device existed and it were cheap enough, I'd say screw all this softphone crap and use something like that.
This would be such a specialized thing that nobody would want to produce something like this, I'm sure.

Oh, one more thing I forgot; let's not have any filtering or other weird processing, such as autogain (unless it's optional, of course).
These hardware phone patch manufacturers have this thing for filtering the incoming and outgoing audio severely. For example, dgl1984s patch has DSP, and is limited from about 220 to 2.7 KHZ or something like that, when the line itself, even a PSTN, can pass way more than that, especially on the low-end. They say they're doing this to comply with telecom regulations... Right. So why then, do a lot of even really cheap phones have a better frequency response than something you're paying loads and lots of money for? And don't even get me started on the $2000+ Comrex and telos patches and such!

As the subject states, things have broken, and I have stuff to play with. All-in-all, there are too many things, and far too much stuff around, but that's neither here nor there.

As for the broke things; Mom's laptop has decided, intermittantly, to no longer have a hard drive. It's an old IBM 366 MHZ machine that Mary Ann gave me during the summer of 2005. When I got it, it had a very dead hard drive, a missing cover for the mouse pointing stick thingy, and a battery that didn't hold much of a charge. I replaced the battery and pointing stick cover, thereby making the mouse useable again, installed a new, old hard drive, and added more ram as well. And I have to say, for being so slow, when it works, it's stable. When it doesn't work though, it makes a lot of noise about it... litterally...
It's also got a bent pointing stick that slowly glides the mouse to the right of the screen when you're not using it, and half of the keyboard is caved in on the right side, as well as having a missing backspace key.
I've been telling them to get a new laptop since September, and now it's looking pretty bad for them. Due to space constraints and all that sort of thing, a desktop wouldn't work so well, which, of course, would be a lot more convenient to either build from scratch, find somewhere for really cheap, or deal with when it broke, but oh well.
I wonder how long it will take before they start asking to use my laptop? Hmm...

Also, my engenius technologies (or possibly not) access point decided to start dropping connections at random yesterday, so I rebooted it, reset to defaults, and upgraded the firmware. That was the last straw, apparently, as after what looked to be a successful upgrade, it has now become a little plastic paperweight. Ah well, I didn't particularly like that access point much anyway, although it was semi-expensive. I've bought a 3com one to replace it, so hopefully that won't suck?
Now watch it have an RSMA antenna connector instead of RTNC. I'll have to get a new cable for my 12 DbI antenna! Oh no!

Last week, I bought an ATA from an ebay store, which claimed it was a generic 1-port Sipura rip-off, but when I got it yesterday, I found that it is actually an unlocked Sipura spa-2000 with two ports. I don't know if it was supposed to be one of those or not, as the documentation that came with it seems to have been written for another, crappier ATA, but whatever.

As it's unlocked, I setup two test lines -- one on voipbuster, and another on fonosip, as they're both free, and can give me a pretty good idea of what to expect with this ATA. I also got one of those stupid free US phone numbers from IPKall, which is far from reliable and has no tech support, but what do you expect for free? When it works, it works fine, but most of the time it simply forgets to ring, which is sort of a bad thing. I actually don't know if this is the fault of Fonosip (another get-what-you-pay-for service) or IPKall that's breaking it. I'll probably try voxbone to see if the same issues occur with Fonosip and FreeWorldDialup for incoming calls. By the way, in case you haven't figured it out, my goal with this ATA is to be as cheap as possible without having things that don't work entirely. I'll probably setup a custom calling plan that allows me to call out to another service other than FWD on one line, while taking incoming calls from FWD on the same line. And now I've got to find a way, while the ATA is in the basement and not on the internal house wiring due to it's close proximity to my DSL modem, to have access to both the vonage, and at least one of the other two fake lines in my bedroom. Running cables everywhere... great fun.
If I get all this to work, I can then drop the virtual number that sits on the Vonage line and save them a bit of money.
Wouldn't that be nice, or something?

After playing with the sipura, I decided that it reminds me startlingly of my parent's old Linksys RT31P2, which I believe is a PAP2 packaged in a router, and forums seem to suggest that the SPA-2000 and the PAP2 are identical other than the outward appearance. The nice thing about this one over the RT31P2 I had, however, is that it doesn't have the highly annoying noise that came through the line, which I think was coming from the CPU or something.

There is one slightly annoying thing about this ATA, which I'm sure can be corrected, if I only knew what I was doing, or even what to look for.
The parents have a Linksys RT-P300 on the Vonage line, which has a texas instruments chipset, and probably sounds a little nicer than the SPA-2000 does. If unlocking policies weren't so blah, I'd steal that and use it instead of the Sipura just for the sound quality.
However, one nice thing about the Linksys over the Sipura's current configuration is that, when the caller drops the connection, it cuts power to the line for about 750 ms or so -- about the same length as a flash. This is a pretty standard thing to do if you happen to be an American PSTN line, but I'm not sure if other countries do that. I know the UK doesn't.
Anyway, when using my JKAudio auto-hybrid phone patch, this is especially convenient, since it doesn't attempt to regain access to the line again after it's been dropped in this way. With this ATA, I will have to manually kill the connection, which reminds me of the bad old days when I had a fake phone patch, comprised of a chordless phone with a 2.5 mm jack, some resistance on the cable to keep the line-level aux send on the board from over-loading the phone, and almost no null/separation. Of course, people might say that I'm lazy -- it's just one button, but when you're doing live stuff, it's fun to not worry about that sort of thing. Of course, this also breaks the potential use of the auto-answer feature when you're not around, since it won't die-on-demand like it's supposed to, keeping the line tied up instead.

Esentially, what I'm trying to say is that I'd like my Sipura to have this functionality, and I'm pretty sure an allowance can be made, but there's nothing obvious that suggests that it can be done, or how to do it. I've posted about it in a few forums, but so far, nothing has been returned.
I'd expect TFTP and config editing is in order, but, to be honest, I don't know what I'm doing, so now I will cry.
*cry*
There, that's better.
Well, not really, but I can pretend, I suppose.

So, I've been writing all this while in the downstairs bathroom, which is pretty cold, by the way, so I should probably send this off and go away to get some ibuprofen for this rather annoying headache that has decided, rather at random, to assault me. So, with that, I say 73's to you all?

Here's something that will probably interest any of you odd people who like strange old phone networks and related things.

This is a group of people who are using ancient technology, such as step switches, sintrex, crossbars, etc. and using Asterisk as outgoing trunks and tandums for all this old electro-mechanical switching equipment going back as far as the thirties, with old phones behind the switches to match. They have created cnet, which is a private network of this old stuff interconnected by VoIP. There are three American PSTN gateways that I know of (all in New York) into cnet, and some test numbers that look like they would be interesting. Only thing is, I can't get any of them to work to see if this is, indeed, the case. The 516 PSTN gives me an error that any given office code hasn't been assigned, 212 simply drops the call after having dialed after the PBX-side dial-tone, and the 914 PSTN sort of sits there and does nothing at all for a while after dialing any extension, then eventually drops. Maybe someone else will have better luck with these, or perhaps they're having issues?
There's also one in the UK, but I haven't tried that one yet.

The way it's supposed to work is; you dial one of the PSTN's, which takes you to a PBX, at which point you dial country code+seven-digit number. I guess there's no point in deploying area codes on a private network.
Here is an article relating to cnet.

I want to play with it, Mommy!