Reading around the internet I found a lot of stuff about squeezing a bit more throughput from your drive.

I did a couple of benchmarks in Ubuntu 11.04:

• Adding noatime,discard to /etc/fstab

Result:  No change.  Not better or worse, but in theory the “discard” will help in the long term

Decision:  Switch this on.

• Setting the scheduler to noop in /etc/rc.local

Result: Average read rate increased by 2MB/s but Average access time went from 0.2ms to 0.3 ms.  In itself that’s not a real problem, but the graphs show a different picture.

Without NOOP

Note the scale on the right hand side.

With NOOP

The access speed is all over the place.

Decision:  Switch this off

I’d be interested in hearing if your experience differs, but it seems to me that “Doing nothing” is a valid choice.  Ubuntu, out of the box, doesn’t really require any fettling in order to get the best from your SSD.

So to make things a bit easier for me after a full re-scan, I’ve put together a list of the channels I don’t watch and with a tiny bit of SQL I can trim them from my channel list.

To make things a bit easier for you here is a SQL dump of my “unwatched channels” list:

unwatched_channels

And here is the SQL to trim these from your channel list:

update channel set visible=0,useonairguide=0 where name in (select name from unwatched_channels)

You’ll probably want to edit that list yourself to remove and add the channels as you prefer. Generally speaking, my list trims:

• Regional variations
• Specialist interest
• Shopping
• Games and other text based services

I’ll update this list occasionally, this page will always have my most up to date information.

• UPDATE: 6 Sept 11.  Refreshed channel list
• UPDATE: 8 Oct 11. Refreshed channel list
• UPDATE: 14 Dec 2011.  Refreshed channel list

I also couldn’t access voicemail from the SIPGate trunk either, but I could get it to work with a phone connected to an ATA.

I had a look in the logs:

channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)

Since slin is, I think, the Asterisk native format this is quite bad.  My assumption is that SIPGate sends through G.729 encoded audio regardless of whether you want it to or not (as configured in your allow/disallow lines for the SIP trunk PEER).

G729 from SIPGate works to the ATA because the ATA supports G729 and G729 to G729 pass-through works without any extra requirements from Asterisk.

So – in order to get SIPGate to Asterisk apps working again I installed the G729 codec binaries from here:

http://asterisk.hosting.lv/

and restarted Asterisk.  Now a:

         core show translations

shows I can convert between G729 and loads of other codecs.

They don’t appear to have advanced functions like switching off the load independently of the supply which means you can’t power down the servers and then have the UPS switch them back on again once the power is restored and the battery has had a chance to charge, but I can live without that sort of thing.

I assumed that something somewhere would support these, and for the most part NUT does a bang up job.  It’s a bit advanced for my needs but most of the work has already been done.  In NUT the LiebertPSP is sort-of-supported by the Belkin USB driver with a few notable exceptions:

• The Online/On Battery indication doesn’t work
• The numeric values don’t work for things like Output Voltage

The numbers I can live without but the Online / On Battery is really rather important.

So I’ve cobbled together a new sub-driver using the tools supplied with Nut.  It’s far from complete but it does fix most of the annoying problems.

http://www.whizzy.org/wp-content/uploads/2010/10/liebertpsp.tar.gz

You’ll need to download the Nut source and put the .c and .h files in the above tarball in to the drivers directory, then you’ll need to apply this patch in the drivers directory to get the new driver included in the build and to stop the native Belkin driver from claiming the ID of the LiebertPSP:

http://www.whizzy.org/wp-content/uploads/2010/10/liebertpsp.diff

Then recompile Nut.

If everything works you should see a bit more info from your UPS:

battery.type: PbAc

device.mfr: Emerson Network Power

device.model: LiebertPSP

device.serial:

device.type: ups

driver.name: usbhid-ups

driver.parameter.bus: 004

driver.parameter.pollfreq: 30

driver.parameter.pollinterval: 2

driver.parameter.port: auto

driver.version: 2.4.3-2519M

driver.version.data: LiebertPSP HID 0.1

driver.version.internal: 0.35

ups.input.frequency: 50.0

ups.input.voltage: 243

ups.mfr: Emerson Network Power

ups.model: LiebertPSP

ups.output.percentload: 54

ups.output.voltage: 240

ups.powersummary.capacitygranularity1: 1

ups.powersummary.capacitygranularity2: 1

ups.powersummary.capacitymode: 2

ups.powersummary.configvoltage: 12.0

ups.powersummary.designcapacity: 100

ups.powersummary.fullchargecapacity: 100

ups.powersummary.imanufacturer: 19

ups.powersummary.ioeminformation: 19

ups.powersummary.iproduct: 1

ups.powersummary.presentstatus.batterypresent: 1

ups.powersummary.presentstatus.belowremainingcapacitylimit: 0

ups.powersummary.presentstatus.charging: 1

ups.powersummary.presentstatus.discharging: 0

ups.powersummary.presentstatus.needreplacement: 0

ups.powersummary.presentstatus.shutdownimminent: 0

ups.powersummary.rechargeable: 1

ups.powersummary.remainingcapacity: 100

ups.powersummary.remainingcapacitylimit: 38

ups.powersummary.voltage: 14

ups.powersummary.warningcapacitylimit: 38

ups.productid: 0001

ups.serial:

ups.status: OL

ups.vendorid: 10af

Search hints:

Vendor ID: 0x10af

Product ID: 0×0001

I’d installed smartmontools but it just doesn’t seem to be working.

Finally sussed it. Have a look at /etc/default/smartmontools and enable deamon

But, it would appear there is a defect in these boards, or at least a strange design, which means that they won’t lock on to some frequencies which require the 22kHz tone sending to the LNB with new drivers because there’s no link between the flange and dolphin-points.  There’s plenty to read about here:

https://bugzilla.kernel.org/show_bug.cgi?id=9476

And there’s a patch which fixes the problem by controlling the tone generator directly but it’ll never get in to the main kernel.  For your convenience here is a link to a binary driver built for Ubuntu Lucid kernel version 2.6.32-23-generic:

http://www.whizzy.org/wp-content/uploads/2010/08/isl6421.ko

Replace the current isl6421.ko from /lib/modules/2.6.32-23-generic/kernel/drivers/media/dvb/frontends/isl6421.ko with this one.  It might also work for newer kernel versions, or not.  Who knows?  Not me.

I’ve also got a Hauppauge S2 HD and this patched driver doesn’t seem to effect it.

Search hints:

Hauppauge Nova S plus linux won’t lock horizontal 22khz tone can’t pick up some channels

Why do I always get myself in to these situations?

Many years ago I knocked up a set of scripts to record TV programs from a DVB-T card.  Getting the TV card working was really hard work.  I had to hack about with the driver source code, compile custom kernels, build endless versions of the driver, and so on and so on and so on.  It was a pain in the arse, which was compounded by the fact that at the time very few other people (end users) were trying to get these cards working, so there wasn’t much community support.  My scripts were great and everything and I could record TV, taxes were low and life was, on the whole, pretty good.

Then Stuart put me on to MythTV.  It quickly became evident that although my scripts were functional they were nothing compared to Myth.  So I moved over.  Again with the pain.

The basics were there but DVB support was in it’s infancy and it was a lot of hard work getting it working.  I persevered and learnt a lot about MythTV in the process.  Yay me.  At the beginning MythTV was a foreign country.  By the time I’d got it working the way I wanted it was my home town.

And so it will be with Asterisk, I hope.

Things I have learnt about Asterisk

1. The cheap FXO cards they sell on eBay are more trouble than they are worth.
2. The documentation on the voip-info site is often out of date.
3. UK Caller ID works with the TDM400′s out of the box.
4. Sending & receiving SMS’s does work, but it’s a bitch, and unreliable.
5. The logic in Asterisks’ dial plan (extensions.conf et al) is illogical to me
6. FreePBX is both a blessing and a curse.
7. I love it.

In a bit more detail then:

1.  The FXO cards you can get from eBay for about 20 quid work perfectly well.  For making calls.  But, there are a few drawbacks.  They don’t support polarity reversal which is the method by which incoming calls and Caller ID are announced to Asterisk.  You’ll still get ringing indications and be able to make calls, but you won’t get Caller ID.  Or rather, you can get Caller ID but you need to patch the Zaptel drivers and the Asterisk source.  The patches are old and don’t apply perfectly.  They are also seemingly unsupported now, in that no one cares if you are having problems with them.  I got the patches applied but Asterisk wouldn’t compile and I don’t know/care enough to fix it.  I was in the market for an ATA anyway so I put that 40 quid towards the cost of the TDM400p11 from NovaVox and if I hadn’t bought the AX-100 then the initial outlay for a fully supported card isn’t so bad.  It also works out cheaper to add another FXS port the to TDM400 than it does to buy another ATA.  The AX-100 also has a US spec. “hybrid” in it which means that the unbalanced UK spec phone lines and the AX-100 are not perfectly matched which can cause wicked echo.  The TDM400 has a tunable hybrid, and that’s a good thing.  All that said – for 20quid I still think it’s worth getting one to play with.  Ask me about doing a swap, if you’ve got any cool toys you don’t want  (I’ve got a DVB-T card going spare as well!).

2.  The VOIP-info.org site is a really good source of data.  The website might be ugly, but there is a lot of Asterisk data on there.  Unfortunately there is little in the way of information, and some of the data is way out of date.  It acts as a good reference point but isn’t for the beginner because most pages assume a lot of prior knowledge.  Sure – I could go in and edit the Wiki, but I don’t know if what I’m doing is the best way of doing it, or even the correct way to do it.  So for now that’s what this blog post is for.

3.  BT send the Caller ID information before the first ring.  The process goes something like this:

Line polarity reversed -> Caller ID sent as FSK (sort of modem tones) -> Phone starts to ring -> Call Connected -> Charlie Brown's Teacher -> Call Hung Up -> Line polarity reversed

You can see the importance of being able to detect polarity reversal in the call set-up and tear-down.  The AX100 simply doesn’t do it.  The TDM400 does, and it’s fully supported out of the box, as is UK style Caller ID, where the ID is sent before the first ring.  The hack for the AX100 keeps a buffer running all the time and then once a call comes in it looks in this buffer for the caller ID FSK and decodes it.  Shonky, I think you’ll agree.  My advice is that if you want to do caller ID in the UK buy a TDM400.  If you are a masochist then feel free to try with the AX100, I’d love to read your HowTo once you’ve got it working.

4.  SMS.  What a bitch.  I recently discovered my subscription with BT gives me 200 free SMS text messages a month.  So to make sure I was getting best value from the phone subscription I went out and spent 100quid on a PCI card and then spent hours and hours and hours fiddling about trying to get it working.  I don’t know why, I probably send about 3 text’s a month from my mobile, the whole “Because it’s there” thing I guess.  A pox on you, Mallory.

Last night I finally got it working semi-reliably for incoming and outgoing texts.  I can send and receive from my mobile, but Stuart doesn’t receive my messages.  Ho hum.

Firstly, receiving SMS’s. Now, I don’t really understand the extensions.conf language.  It’s all a bit, well, wrong, to my eyes.  All I can tell you is that this is how I got it working.  NOTE:  This WILL NOT work for you if you just copy and paste. Sorry about that.  Perhaps someone can help me re-write it so that it does?

I’m using FreePBX as a GUI and it rewrites some of the Asterisk config files for you.  I’m not sure if the context [from-pstn] is one that I created, or a standard one.  What you need to find out is what route incoming calls from the PSTN take i.e. which contexts they pass through before they start ringing on your internal phones.  You can do this my running the Asterisk console and turning verbosity up to about 6 (core set verbose 6) and then calling in from outside.  You should see a trace of what contexts get called as the call comes in.

For example, my call flow goes something like this:

[from-zaptel] -> [from-pstn] -> [macro-user-callerid] -> 600@[ext-group]

Where 600 is an extensions number I set up with FreePBX to ring all internal phones.

I chose to test for SMS’s in the [from-pstn] context. You can visualise what the [from-pstn] context does when a call comes in by using the Asterisk CLI.  Use the command “dialplan show s@from-pstn” to see the instructions that would be followed when [from-pstn] gets called at position “s”, or start.

Originally, mine went something like this:

[ Included context 'ext-did-0001' created by 'pbx_config' ]
's' =>            1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
2. Gosub(cidlookup|cidlookup_1|1)             [pbx_config]
3. ExecIf($[ "${CALLERID(name)}" = "" ] |Set|CALLERID(name)=${CALLERID(num)}) [pbx_config]
4. Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}}) [pbx_config]
5. SetCallerPres(allowed_not_screened)        [pbx_config]
6. Goto(ext-group|600|1)                      [pbx_config]

You can follow the call progression through this quite easily.  We jump in at position (or priority) 1, we set a variable, then we jump to some cidlookup sub-routine, then if the CALLERID(name) is blank we set it to be the same as the caller id number, then something about CALLINGPRES, not sure what that does, ditto the next line, and then we go to ext-group|600|1.  Ahhaaa!  I know that 600 is the group that I call to ring all the phones, so that must be where the call is handed off to other contexts or functions that let you do talking to people.  Since an incoming SMS doesn’t need to get as far as ringing on a phone, it makes sense to me to interrupt the call flow during [from-pstn].

So I added [from-pstn-custom] to /etc/asterisk/extensions_custom.conf like this:

[from-pstn-custom]
exten => s,3,GotoIf($["${CALLERID(num)}" = "08005875290"]?will-sms,s,1)
exten => s,4,Verbose("Not an incoming SMS")
exten => s,5,Gosub(cidlookup,cidlookup_1,1)
exten => s,n,ExecIf($[ "${CALLERID(name)}" = "" ] ,Set,CALLERID(name)=${CALLERID(num)})
exten => s,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}})
exten => s,n,SetCallerPres(allowed_not_screened)
exten => s,n,Goto(ext-group,600,1)

[will-sms]
exten => s,1,Verbose(=============Entered Will SMS)
exten => s,n,Answer()
exten => s,n,Wait(2)
exten => s,n,SMS(default|a)
exten => s,n,Verbose(=============Done with Will SMS)
exten => s,n,Hangup(16)

How does it work?  The [from-pstn-custom] overrides the [from-pstn] from the main extensions.conf file provided by FreePBX and adds a line that branches to a context called [will-sms] which then uses the SMS application to receive the SMS and then hangup the phone.  (Actually, I think SMS hangs up for you.  But, you know, whatever)

In [from-pstn] in extensions.conf is a line that “includes” the config from “from-pstn-custom”.  By virtue of being imported, the instructions in [from-pstn-custom] take precedence over the other config, because it is imported first.  The config files are read line by line in the order they appear in the file.  But, the imported instructions do not completely replace the config that is already there.  They are sort of merged. That’s why I’ve copied a load of bits from the original call flow in to my new [from-pstn-custom].  If I left them out, then some bits of my new context would be applied and some wouldn’t.  The way I got it to work was to replicate the call flow to the point of it branching off to [ext-group] in my new context.  If you do a “dialplan show s@from-pstn” now, you see this:

[ Included context 'from-pstn-custom' created by 'pbx_config' ]
's' =>            3. GotoIf($["${CALLERID(num)}" = "08005875290"]?will-sms|s|1) [pbx_config]
4. Verbose("Not an incoming SMS")             [pbx_config]
6. Gosub(cidlookup|cidlookup_1|1)             [pbx_config]
7. ExecIf($[ "${CALLERID(name)}" = "" ] |Set|CALLERID(name)=${CALLERID(num)}) [pbx_config]
[pbx_config]
9. Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}}) [pbx_config]
10. SetCallerPres(allowed_not_screened)       [pbx_config]
11. Goto(ext-group|600|1)                     [pbx_config]

[ Included context 'ext-did-0001' created by 'pbx_config' ]
's' =>            1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
2. Gosub(cidlookup|cidlookup_1|1)             [pbx_config]
3. ExecIf($[ "${CALLERID(name)}" = "" ] |Set|CALLERID(name)=${CALLERID(num)}) [pbx_config]
4. Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}}) [pbx_config]
5. SetCallerPres(allowed_not_screened)        [pbx_config]
6. Goto(ext-group|600|1)                      [pbx_config]

[ Included context 'ext-did-catchall' created by 'pbx_config' ]
'_.' =>           1. Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.) [pbx_config]
2. Goto(ext-did|s|1)                          [pbx_config]

You can see my new [from-pstn-custom] context at the start, along with the copied commands from ext-did-001 to ensure that they run in the way I expected.  The upshot of this is that I now branch to my new context [will-sms] when the incoming caller ID is the BT 0800 number from where SMS’s originate.  You should note that the 0800 number can change if you have more than one incoming SMS box registered, so don’t do that.  Just register one incoming address  (more on this later).

The [will-sms] is pretty straight forward.  Answer the phone, wait 2 seconds, and then start the SMS application which does all the clever noises.  The key here is Wait.  Without it SMS reception is intermittent at best.  The SMS(default|a) tells the SMS app to receive in to the default queue (this works, don’t understand fully why) and to do an “a” for answer.  I had a load of problems with the example from VOIP-info where it passes in the extension number, which is “s” for start.  If the SMS application sees an “s” it thinks it means “send”.  That won’t work!

If you manage to bodge all this in to your dial plan then you should find your incoming SMS’s in /var/spool/asterisk/sms/mtrx

Make sure that /var/spool/asterisk is writeable by the same user as Asterisk is running as.

Sending SMS’s. Was a whole bunch of no fun as well.  Especially if you don’t understand Asterisk properly.

The command line I’m using to send an SMS is:

smsq --motx-channel=Zap/2/17094009 -d <phone number>  -m "<message>"

That works because zap channel 2 is my PSTN line.  You might need to change this.  17094009 is the SMSC centre number for BT to accept SMS messages for delivery to other networks.

When I was trying to get sending working, which I actually did first, I couldn’t reliably get Asterisk to talk to the SMSC.  Before I had the receive working properly the BT system would ring me up and I’d answer the phone, then not hearing a carrier the BT system would hang up again.  This gave me an idea, if I could emulate the carrier then I could hear what the BT system was sending down the line.  I sent an SMS to the home phone, it rang in on one of the extensions and I whistled at it.  Because I am just so 1337 I hit the right tone and I could hear FSK noises coming down the line, but they were really really quiet.  I tweaked the gain up a little bit in the /etc/asterisk/zapata.conf file:

rxgain=2
txgain=0

and tried again.  It was louder!  Sending a test message to “00000″ with the text of “test”  causes the BT system to send a message back to you  (you need to send “register” to 00000 before BT will send you messages as text instead of reading them to you by the way – but if it can’t successfully deliver the message back to you to say it has received your “register” message it will ignore the request and you’ll keep getting the messages read to you by a friendly robot – hence the need to get receiving working first).  It sent!  I saw some hex coming from the SMS application in the Asterisk log.  I sent a text to my mobile, and it arrived.  Yay!  problem solved.

5.  Asterisk’s dial plan.  Bonkers.  I still can’t make sense of them.  Oh well, I’ll get there in the end.

6.  FreePBX is brilliant if you want to get Asterisk configured with some very complex applications (think voicemail) in a matter of minutes.  But, if it makes editing the config files by hand a bit more complicated because you have to use xxxx_custom files.  Thus, all the replication of code when as above when you don’t understand how it all works.

7.  Check out my hold music hacked together from various online sources, and no doubt subject to various copyright restrictions.  I’ve compressed the hell out of it, if you want a better quality version for your own amusement let me know.

Hold Music

I crack myself up, I really do.

Also – think of all the cool things you could do if you could phone or text your computer and get it to do things for you.

Now I’m off to write myself an gmail <-> SMS gateway doodad.

Ok – with Asterisk 1.6 the above is slightly less relevant.  Try this in your dialplan instead:

[will-sms]

exten => s,1,Verbose(=============Entered Will SMS)

exten => s,n,Answer()

exten => s,n,SMS(default,ap(1500))

exten => s,n,Verbose(=============Done with Will SMS)

exten => s,n,Hangup(16)

The SMS application has the option to add a pause which I've set to 1500 ms.  Seems to work.

I warned you, I warned you.  But would you listen?  Oh no.  You knew it all.

Spotify is on the ropes.

http://stuff.tv/blogs/music/archive/2009/07/13/why-spotify-might-die.aspx

Sign up to the paid for service if you can.  Now – if they had a proper native Linux client I would.

If you’re reading this then I will assume you already know about MythTV and you’ve searched Google to find out if the Aspire Revo box will make a decent MythTV frontend. In short, yes – it works fantastically well.

I bought the 8GB SSD Linux version from Play for about 150 quid (get a Play credit card and you can knock off about another 15 quid in vouchers and get 9 months interest free credit, for what it’s worth), and I also bought 2 x 2GB SODIMM.

The first thing I did was take the lid off to have a look inside.  This wasn’t as easy as I thought.  There’s one little screw under a sticker that says something about a warranty and then you just have to prise the lid off. It’s pretty stiff, to the extent that I was convinced there was another screw somewhere, but it comes off in the end.   I removed the WiFi card since I won’t use it and it might reduce the heat/power.   The RAM swap presented no surprises, but the appearance of a 160GB HDD did.

I had sort of decided that the SSD was the better option for me for two reasons; less heat and less noise. But, seeing as I’ve been gifted 160GB of disk space and under use the HDD makes no noise, I’m very happy!

I decided on XUbuntu over the normal version partly because of the reduced overheads and software bloat, I really don’t need Open Office and The Gimp installed on this box and I can’t be bothered with manually selecting packages at install, so I downloaded the XUbuntu ISO and stuck it on a USB pen drive with Unetbootin.  I had a few problems booting from the pen drive, it kept complaining that the initrd was corrupt, so in the end I had to use the alternative version and run through the install on the command line.   I blew away all the partitions on the disk, I won’t need any of the Acer software – there’s nothing much of use on there anyway.

Once XUbuntu was installed I downloaded and compiled the latest NVIDIA drivers:

http://www.nvidia.com/object/linux_display_ia32_185.18.14.html

(this is the latest version at I write this, check if there is a newer one)

and then I downloaded the SVN version of Myth and enabled VDPAU. If you’re looking for help setting up Myth and VDPAU check the MythTV wiki, there’s more information there than I can recreate here. Read the “Installing SVN on Breezy” document to give you a hand getting all the bits and bobs installed to allow you to compile Myth. Note: some libraries have changed name, e.g. liblame is now libmp3lame.

Then it’s a case of enabling VDPAU at compile time using the configure script and then creating a playback profile to use VDPAU from within the frontend.

My old frontend had a dual core 2GHz processor in and would sit at about 80 to 90 percent usage on both cores while watching 1080p video and sucked somewhere in the region of 80 to 100 watts. The Revo’s Atom processor sits at about 10% usage (obviously the graphics card is doing the work) and sucks less than 20watts, while also being nearly silent.

Sound was a bit of a faff – the Revo has an HDMI connector with the audio path built in. To hear anything I needed to enable and unmute the IEC958 (spdif) channel and then tell Myth to use ALSA:hdmi for audio.   Ubuntu detected the sound card perfectly well, so trust me when I say you do not need to download a later version of ALSA, you just need to get the settings right.   You might also need to tell your telly to pick up digital audio, not analogue.   I haven’t got system sounds going down the HDMI just yet , but I don’t think this will be a problem.

Summary then; very very good choice for a small, quiet and cheap frontend that can also double as an Internet browser.

I think I might buy another one.

The box under the telly at the moment is a dual-core Pentium 4 somethingorother running at about 2Gz in a nice looking “media” case.   It’s great and everything, and it decodes HD perfectly well but it has fans on the processor, fans on the video card and fans in the PSU which all add up to a noisy box.

So it’s going to get re-purposed as a general computer and the Revo will go in it’s place.   Actually it will get bolted on to the back of the telly to keep it out of the reach of children who like to press buttons.  I’ll report back on the building of a frontend on a Revo once I wrestle the package from the hands of the worlds most apathetic delivery company “Home Delivery Network”.  (I think they use the word “delivery” quite incorrectly)

Anyway, as part of this upgrade I need to add a DVB-T tuner to the Revo but with no PCI slots the only option is to use a USB tuner.   My father-in-law put me on to Kenable who have a lot of bits and bobs at good prices, specifically the PEAK DVB-T USB tuner for 15quid:

USB Peak DIGITAL DVB-T Freeview TV Card XP MCE Vista Dongle
http://www.kenable.co.uk/product_info.php/cPath/171/products_id/1435

I wasn’t able to find much about about it before I bought it, but I thought I’d take a punt and get one anyway and worry about getting it working under Linux later.

It arrived the other day and I popped the lid off to find an AF9015 demodulator and a QT1010 tuner inside.   I checked on the Linux TV wiki and things looked pretty good on the whole.  When I plugged it in to an Ubuntu 9.04 box nothing happened which was a shame but a bit more digging made me think that I should download the latest Linux TV drivers and have another go.

I followed the instructions from here:

http://www.linuxtv.org/wiki/index.php/How_to_Obtain,_Build_and_Install_V4L-DVB_Device_Drivers

and downloaded, compiled and installed the new drivers.  You’ll also need some different firmware to that which is supplied with Ubuntu, download 4.95.0 from here:

http://www.otit.fi/~crope/v4l-dvb/af9015/af9015_firmware_cutter/firmware_files/4.95.0/

and place the .fw file in /lib/firmware (not as most people say /lib/firmware/<kernel version>).  You’ll need to overwrite the other version, or rename it or whatever.

Reboot and you’re golden. The stick is detected and the firmware loads and I have been watching live TV on my Aspire One netbook courtesy of Mplayer.

You probably need to bear in mind that if you apply any future kernel/driver updates from Ubuntu then your drivers might get over written.  There’s a change that the new version of the Ubuntu drivers will include the required AF9015 driver, but it might not.  Also, you probably don’t need to compile all the drivers, just the ones for the AF9015 and QT1010 modules.  I’ll look in to this, and if you promise to be good I’ll provide a nice little package with everything in.

UPDATE:  So here we are two and a bit years later.  I’ve just found this same tuner in the bottom of a box and plugged it in to my 11.04 Ubuntu machine.  When I plugged it in Ubuntu automatically suggested I download the firmware.  Awesome.  It now “just works”.  Well pleased.