Convincing MythTV to tune in to DVB-T2 MUXes

November 25th, 2013 — 7:09pm — By Will Cooke

So that I remember for next time, and so I can start writing down some of the issues I’ve had to sort out since re-building my servers, here is what you have to do to tune to a DVB-T2 mux in MythTV (0.27 fixes).

This is the frequency for the PSB3 mux on Sandy Heath, I expect the encoding scheme will work pretty much where ever. At the very least it should get you started:

 

Frequency:  474200000
Bandwidth:  8 MHz
Inversion:  Auto
Constellation:  QAM 256
LP Coderate: 2/3
HO Coderate: Auto
Trans. Mode: 8K
Guard Interval: 1/32
Hierarchy:  None

If that doesn’t work, and you know it should (as in you are on the same transmitter) try setting the tuning timeouts higher. Also, just blindly re-trying because “WHY DOESN’T IT WORK” seemed to help.

Comment » | linux, tv

And we’re back…

November 22nd, 2013 — 7:51pm — By Will Cooke

I’ve moved to another server, and in the process I’ve had a whole lot of problems.  For anyone that’s read any of my blog before that won’t come as a surprise.

Once I’ve got all the kinks worked out, I’ll tell you how I fixed it.  For now, this is just a test…

1 comment » | Uncategorized

Asterisk UK Caller ID & SMS redux

February 12th, 2013 — 12:41pm — By Will Cooke

Update 28 Dec 2013:  As far as caller ID goes, I might have been barking up completely the wrong tree here.  Have a look here: http://forums.digium.com/viewtopic.php?f=1&t=85028.  I’ve re-implemented that patch for Dahdi and I’m testing it now.  Get in touch if you want to test it as well.  Once I’m happy it’s actually working, I’ll post it here.

 

tl;dr: Scroll down to “How to fix Caller ID in the UK on Asterisk 11 and Dahdi 2.6.1″

I’ve been having some problems with Asterisk 1.6 for a while now, in that when I’m dialled in to conferences via SIP, or when I call someone via the work Asterisk server from my Asterisk server at home I get cut off every 15 minutes. My colleagues have all kind of accepted this now, it was starting to get annoying. It was a known problem in Asterisk 1.6, and so I was going to have to upgrade.

The Ubuntu Asterisk packages are fairly up-to-date but FreePBX isn’t packaged, and it’s a pain in the backside to get set up with the correct permissions, especially if you install Asterisk from packages. I had a spare machine anyway so I decided that I’d give FreePBX Distro a go and once I’d burnt the ISO to a CD (ya rly) I found the set up process to be simple and quick.

FreePBX Distro is a stripped down CentOS (2.6.32 kernel) which boots quick, has minimal footprint and bundles in loads and loads of FreePBX “apps” like conferences, voicemail, blacklisting and makes setting up extensions a breeze. It also includes things like kernel source for I sell settlement so building extra modules from source is pretty easy. Which is a good job, because it was about to get messy.

If you’ve found this page via Google then this will probably sound pretty familiar:

  • You got a TDM400p four port FXO/FXS interface card, or a clone from someone like OpenVox (which work perfectly by the way – as do those cheap daughter cards off eBay).
  • You’ve upgraded to Asterisk 11, or Asterisk 1.8 perhaps, or changed to the Dahdi 2.6.1 drivers and all of a sudden your ”UK Caller ID” has “stopped working” or at the very least become “intermittent“.
  • It was working fine before, so it’s not a hardware issue and your phone provided have NOT suddenly stopped sending caller ID information, regardless of what people of forums tell you. No, in fact this is clearly a bug which has crept in somewhere along the line.
  • You’ve searched and searched and tried things like setting rxgain=x.x and txgain=x.x or whatever. (Side note: setting txgain and rxgain in chan_dahdi_channels_custom.conf has no effect, the only place is seems to work is in chan_dahdi_groups.conf)
  • You might have even stumbled across a post hinting that cid_rxgain=x.x will solve your problems, only it didn’t

If that sounds like you, dear reader, then read on. I am your salvation.

  1. This page: http://downloads.openvox.cn/pub/drivers/callerid_patches/ (I can only attribute this discovery to divine intervention)
  2. Find a new wctdm.c driver here: http://downloads.openvox.cn/pub/drivers/callerid_patches/2.6.1-wctdm.c
  3. Download the 2.6.1 Dahdi source from here: http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.6.1+2.6.1.tar.gz
  4. Replace dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctdm.c with the version downloaded from openvox
  5. Note it doesn’t compile, add the missing semicolon to line 333 (or thereabouts), successfully build and install the new driver
  6. You might need to change /etc/modprobe.d/dahdi.conf and add:
options wctdm opermode=UK fwringdetect=1 battthresh=4
  1. restart Dahdi and Asterisk and caller ID should now be fixed! Yay!

How to fix Caller ID in the UK on Asterisk 11 and Dahdi 2.6.1

Here is a patch against wctdm.c from stock Dahdi 2.6.1 with the fixes in: wctdm.c.patch

OR: Replace dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/wctdm.c with this one: wctdm.c

OR: If you’re running FreePBX distro 64 bit “BETA-3.211.63-5 Release Date-01-24-13″ (or close) here is a binary driver: wctdm.ko  Move it to /lib/modules/2.6.32-279.11.1.el6.x86_64/dahdi/wctdm.ko

It works for me. I have not had to increase the rxgain, or txgain or anything else. It just works.

Once that’s working, we can move on to…

How to send & receive SMS in the UK on FreePBX Distro Beta 3.211

I’ve talked about receiving and sending SMSes before, but that was quite a long time ago, and things have moved on a bit since then. I’ve also learnt a bit more about the Asterisk dial plan, and have a slightly cleaner way of doing it now.

You need to have working caller ID. Hint: see above.
You need to install smsq.
smsq doesn’t come built in FreePBX Distro so you have to build it yourself if you want to send SMS. This is pretty easy:

  1. Download the Asterisk 11.2.1 source from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.2.1.tar.gz
  2. Install popt: yum install popt-devel.i686 popt.i686 popt.x86_64 popt-devel.x86_64 (prerequisite for smsq)
  3. In the Asterisk source directory do “./configure”
  4. run “make menuselect”
  5. Scroll down to “Utilities”
  6. In the right hand pane, check “smsq”
  7. Save & Exit
  8. run “make”
  9. DO NOT run “make install” as this will trash your FreePBX config
  10. Once that’s done you should find an executable smsq in the utils directory

Just to be sure, create /var/spool/asterisk/sms/motx and /var/spool/asterisk/sms/mtrx
You can find more details on building Asterisk here: http://blogs.digium.com/2012/11/05/how-to-install-asterisk-11-on-centos-6/

You need to be able to send SMS in order to register with the BT SMS system, otherwise when you receive an SMS you will get a phone call from the BT robots who will read the text to you, badly.

To register with BT try this:

<path to smsq>/smsq –motx-channel=DAHDI/2/17094009 -d 00000 -m “test”

where DAHDI/2 is the channel number of your outgoing line.

If you look in the Asterisk logs you should see the message going out with some TX and RX hex dumps. If it works, a short while later you will receive a phone call from either the SMS system trying to talk modem at you, or the robot. Either way, this means your text went out. Great success!

Now we need to set up receive, which is pretty straight forward.

Edit /etc/asterisk/extensions_custom.conf and add the below. Note – I’ve added a lot of comments to explain what’s going on, it might improve readability if you take them out.

[from-internal-additional-custom]
;; This makes sure our new 'app' gets included in the FreePBX dial plan
include => app-rx-sms
[app-rx-sms]
;; This is the app to receive an SMS which will be called when we match
;; the caller ID of an incoming call to the BT computer
exten => s,1,Answer()
;; This wait seems to help
exten => s,n,Wait(0.5)
;; We simply use the built in SMS function
exten => s,n,SMS(default,a)
;; Hangup with code 16, normal termination.
exten => s,n,Hangup(16)
;; Done
;; We also extend the from-pstn context with a branch to the SMS app
;; if the caller ID matches, hence why it's necessary to get caller
;; id working first
[from-pstn-custom]
;; we add ourselves in to the end of the current dialplan
;; priority 7 gets us there...
exten => s,7,GotoIf($["${CALLERID(num)}" = "08005875290"]?app-rx-sms,s,1)
;; pretty simple - if the caller ID matches, go to the sms app
exten => s,n(dest-ext),Goto(ext-group,1100,1)
;; Note: 1100 is my "ring all" group. You will need to change 1100
;; to what ever you use. You should see this at the end of the
;; from-pstn context in extensions_additional.conf

Save that, and in asterisk do a “dialplan reload”. From smsq send an SMS to 00000 saying “register”· If everything works you should get a text file in /var/spool/asterisk/sms/mtrx.

Comment » | asterisk

Device control over HDMI via CEC. libcec FTW.

November 26th, 2012 — 3:28pm — By Will Cooke

Blimey, it’s been a while.  I’ve been a bit busy, and let’s be honest; writing up blog posts always sounds like a good idea, but when you get in to it – it’s really hard work.

Anyway – I finally got round to buying a Pulse-Eight CEC to USB adapter:  http://www.pulse-eight.com/store/products/104-usb-hdmi-cec-adapter.aspx

This awesome little box of tricks makes up for the lack of CEC control in the vast majority of HDMI-Out equipped graphics cards.  It’s the final piece in the jigsaw of a Linux based home entertainment device.  It allows you to talk to the other devices in your HDMI network; your surround sound amplifier and your big screen TV being the best examples (assuming they support CEC of course).

CEC has been around for a long time but for some reason it doesn’t seem to be widely used, or very well implemented, in most consumer electronics devices.  It’s a published standard but with OEM manufacturers wanting to differentiate their products and introduce a bit of vendor lock-in, they all call it something different.  The fact is that your Toshiba Regza Link TV will talk to your Sony Bravia Link Amp just fine, for the most part.  There might be a couple of proprietary things which don’t work, but on, off, volume up, volume down etc will all just work.

Pulse Eight’s USB to CEC adapter lets your computer get in on the act too, and opens up a whole realm of automatic switching, which really cuts down on the number of remote controls you need and the number of buttons you have to remember the purpose of.

The good folk at Pulse Eight have also made libCEC, an open source library to allow pretty much any software to take advantage of the USB adapter.  http://libcec.pulse-eight.com/

It comes with C++, C and .NET interfaces, and a CLI utility called cec-client.  XBMC & MythTV already support libCEC and have some neat features baked right in.  It’s good, but it’s not exactly what I was looking for – I want a bit more control.

Now, I don’t know anything about C++ or C or .NET, so until someone writes some Python bindings, my ticket to this party lies solely with the CLI utility cec-client.  It can do most things on the “transmit” side, so you can send commands to your other CEC devices fairly easily.  Acting on a request, or listening to the CEC traffic is a bit more complex – but not beyond the realms of possibility.

This weekend I wrote (and rewrote and rewrote) a couple of Bash scripts to:

  • Let me control the system volume (i.e. the real hardware, not the mixer on the computer) on the TV & Amp from the PCs remote control
  • Let me switch the TV & Amp on and off
  • Activate proper muting, again not the mixer on the computer – the hardware itself
  • Switch the amp & TV to my MythTV PC
  • Power off all the hardware when the PC suspends, and then switching it all back on again
  • Shut the whole lot down when the screensaver on the PC kicks in.

I make these scripts available for your amusement:

cecsimple is a client of the “server” which itself is a client of cec-client.  Fire up the server and then issue it commands down the FIFO either directly or via the abstraction layer which is cecsimple.sh.

 

I hooked up the volume control and amp power via “irexec” and lirc.  I tell irexec to execute, for example, “cecsimple.sh volup” or “cecsimple.sh ampon”.  If the server component is already running then these commands are sent very quickly and you don’t really notice the lag.

To switch the TV off when the PC goes in to suspend mode I added a script in /etc/pm/sleep.d which calls “cecsimple.sh tvoff” and then “cecsimple.sh tvon” when it resumes.  In theory if the TV is using the Amp to output surround sound audio then the TV will tell the Amp to turn off, it it’s not – it wont.

To switch things off when the screensaver kicks in, I simply “sudo pm-suspend” from an “xscreensaver-command -watch” script.

 

The practical upshot is that I can now control 99% of my media centre from a single remote control.  I’ve opted to use the remote connected to the PC as I found it to be the least laggy – using the TV remote to send up/down/left/right etc to the PC was sluggish.

 

I think it should be possible to parse the log output from cec-client and write a “listener” component too, but it’s probably a better idea to learn some rudimentary C and do it properly.  Or some Python bindings.  Oh yeah, and you know what would be really cool, a hook in to MythTV so that when I’m watching something in surround sound the amp turns on automatically. That would be cool.

 

UPDATE 3 Dec 2013:  When someone leaves the amp’s HDMI switch set on the PS3 and you switch the MythTV box on from the remote, the amp doesn’t automatically switch to MythTV.  This has been annoying me for a while now, so I fixed it.

In the scripts linked to above the “active source” command does this:

send_command "tx 45 82 11 00"

4 (the MythTV device) to 5 (the amp) – 82 (switch active source) to 1.1.0.0 – but my Sony amp just ignores this request.

I spent some time trying out a few alternatives with cec-client and I’ve found one which works, and it kinda makes sense why:

send_command "tx 45 70 11 00"

4 (MythTV) to 5 (Sony amp) 70 (System Audio Mode) 1.1.0.0 (the input where MythTV is connected)

My assumption is that amp only speaks “system audio” – what with it being an amp.  I’ve changed the “activesrc” with the 45:70:11:00 code and now it works!  (It also switches the amp on, whether I like it or not – so it’s not perfect).

 

 

Useful links:

 

Comment » | linux, Making the world a better place, tv

Fixing “warning: Please check that your locale settings”

August 16th, 2012 — 5:01pm — By Will Cooke

I took an Amazon AWS t1.micro instance for a spin the other day. A free server is not to be sniffed at. Of course I installed Ubuntu 12.04 on it.

I was getting a lot of locale errors, things like this:

perl: warning: Setting locale failed.
 perl: warning: Please check that your locale settings:
 LANGUAGE = (unset),
 LC_ALL = (unset),
 LC_MESSAGES = "en_GB.UTF-8",
 LC_COLLATE = "en_GB.UTF-8",
 LC_CTYPE = "en_GB.UTF-8",
 LANG = "en_US.UTF-8"
 are supported and installed on your system.

I thought this would just go away by itself, but it didn’t – so I had to fix it. Note: I’m in the UK, so I’m using en_GB as my locale, change yours to en_US or whatever.

Type:

export LANGUAGE=en_GB.UTF-8
sudo locale-gen en_GB.UTF-8
sudo dpkg-reconfigure locales

And you should be all set.

Comment » | Uncategorized

Monetizing my feeds

August 4th, 2012 — 3:32pm — By Will Cooke

I decided I wanted to add adverts to my RSS feed. Hardly anyone subscribes to this blog, a dozen people or so (Hello! I bet I know exactly who you all are), but I do get a few hundred hits a month on my Beginners Guide To Google Maps tutorial. So adding adverts to the feed shouldn’t really hurt anyone, and as a bonus the bandwidth used serving the RSS feed should reduce.

AdSense lets you easily monetize a feed by sucking in an RSS feed, parsing it and adding the advert code and then spitting it back out again via FeedBurner.  You can do this by logging in to your AdSense account, click “My Ads” and then expand the “Feeds” section on the left. Once you’ve filled in the boxes, it’ll provide you with a new URL for your feed complete with adverts.

All this is pretty easy, but then comes the question of how you get people to see the new monetized feed?

First things first, FeedBurner can generate a little bit of HTML for you to add to your WordPress Sidebar. It’s described here http://support.google.com/feedburner/answer/78487/?hl=en& and for the sake of completeness also described here:

  1. Log in to FeedBurner using the same credentials as you log in to AdSense
  2. Click on Publicize from the menus near the top of the page
  3. Click on Chicklet Chooser
  4. Choose the type of button you’d like to appear on you blog. Personally I think the smaller “Subscribe in a reader” is the best choice
  5. Then the Javascript at the button of the page is automatically updated, so copy it to the clipboard (ctrl-c)
  6. Go to the Admin page of your WordPress blog
  7. Choose Appearance
  8. Choose Widgets
  9. Drag “Text: Arbitrary text for HTML” to your sidebar
  10. Drop down the Sidebar widget and paste the text from above in to the big box
  11. Click Save and click close
  12. You’re done!

That was the easy bit. Now new subscribers to your blog feed can simply click the button and will be taken to the new monetized feed.

But what about existing subscribers? Or 3rd party links which point direct at your “/feed/” URL? There’s the rub.

What you need to do is redirect them to the new URL, but then if you put a global redirect from the old URL to the new URL, how is FeedBurner going to keep itself updated? (It’ll get redirected back to itself and so never see any of your new posts)

The trick is to use mod_rewrite and check for the HTTP_USER_AGENT of “FeedBurner”.

I’m going to assume that you’ve already got mod_rewrite enabled in your Apache config. If you’re running WordPress on your own server then you most likely have it enabled without even realising.

In your blog’s web directory you will find a “.htaccess” file, open this up and you’ll probably see something like this:

# BEGIN WordPress
 RewriteEngine On
 RewriteBase /
 RewriteRule ^index.php$ - [L]
 RewriteCond %{REQUEST_FILENAME} !-f
 RewriteCond %{REQUEST_FILENAME} !-d
 RewriteRule . /index.php [L]

What you need to do is add these lines between “RewriteBase /” and “RewriteRule ^index.php$ – [L]

RewriteCond %{REQUEST_URI} ^/feed/* [NC]
 RewriteCond %{HTTP_USER_AGENT} !FeedBurner
 RewriteRule ^feed/?.*$ http://feeds.feedburner.com/Whizzyorg [R,L]
 #Next rule

How does it work?

Mod_rewrite matches all the conditions in order, when it hits a Rule it decides if all the preceding conditions are met, and if so acts on the rule. If not, if moves on to the next line. The rules are the the separators between the conditions.

So the new logic says:

  • IF you’re trying to get to “/feed/” (the [NC] makes it case insensitive)
  • AND your USER_AGENT is NOT “FeedBurner” (the ! inverts the logic)
  • THEN re-direct people to the new URL

So, if you are FeedBurner you get access to the old standard WordPress RSS feed and keep yourself updated, if you’re anyone else you get redirected and see the adverts.

The adverts are fairly inconspicuous and are easily blocked by those in the know, so I don’t have a problem with it.

Neat!

2 comments » | Making the world a better place

Over engineering FTW

March 8th, 2012 — 8:26pm — By Will Cooke

Working from home has many advantages.  No commuting, relaxed approach to being dressed, loud music.

One of the drawbacks however, is that you are entirely reliant on your broadband connection.  You could, as many do, have a 3G modem/phone as a backup.  Fine for a few hours, but you’re soon going to run in to data caps, invisible proxies and traffic shapers (more so than a fixed line telco).

 

So what to do?  My solution has been to get a second line installed.  The big UK telcos are running a lot of offers at the moment to try and secure as many customers as they can in readiness for their Triple & Quadruple Play strategies.  This means you can get a reasonable ADSL line and line rental for about 15 to 20 quid a month.

When you’re as reliant on a connection as I am, that’s money well spent.  It also gives you a perfect opportunity to play with some new toys!

Now, you could simply plug in another wifi router to the new line, and when the primary connection fails move your connections over to the secondary one.  But where’s the fun in that?  What you really want to do is some load-balancing and multi-path routing over both the lines.  You won’t see a sudden doubling in download speeds in normal browsing, but certain activities which open a lot of connections to different IP address should benefit, and you get to play with routing tables.

Here’s what it looks like:

 

We don’t really get much diversity until we get south of the exchange, and even then we’re probably going to live in the same fibre all the way to The Internet, but we do work round the much more common problem of ISP configuration errors.

Here’s how it works:

Router 1 (R1) and Router 2 (R2) are two independent ADSL modem/routers  connected to a phone line each and the ethernet port connected to a NIC in a server.  These modem/routers also take care of NAT, so that any traffic coming from inside your network will be translated to appear as if it had come from the appropriate router on it’s way to the rest of the internet.  The server has three NICs in total, one each for the two routers and one back to the LAN for everyone else on the network to use as a gateway.  Let’s use the standard naming convention from http://lartc.org/howto/lartc.rpdb.multiple-links.html

$IF0 = eth0
$IF1 = eth1
$IF2 = eth2

The IP addresses associated with those interfaces are:

$IP0 = 192.168.3.100
$IP1 = 192.168.1.1
$IP2 = 192.168.2.1

The routers are configured with these LAN IPs:

$P1 = 192.168.1.254
$P2 = 192.168.2.254

As you can see on the diagram, the networks are all /24 (255.255.255.0) which gives us network numbers of:

$P0_NET = 192.168.3.0
$P1_NET = 192.168.1.0
$P2_NET = 192.168.2.0

Then we configure the Ubuntu box as a router & NATing gateway:

1.  Enable IP forwarding by editing /etc/sysctl.conf and uncommenting or adding a line like this:

net.ipv4.ip_forward=1

2.  Create two new routing tables.  Edit /etc/iproute2/rt_tables and add a line for each of your providers.  For example:

1 my_isp_name
2 my_other_isp_name

3. Create some routes in each of the routing tables so the server knows where to send data.  They take the general format of:

ip route add $P1_NET dev $IF1 src $IP1 table my_isp_name
ip route add default via $P1 table my_isp_name
ip route add $P2_NET dev $IF2 src $IP2 table my_other_isp_name
ip route add default via $P2 table my_other_isp_name

 

Using the real numbers and the template above gives us this actual set of commands:

ip route add 192.168.1.0 dev eth1 src 192.168.1.1 table my_isp_name
ip route add default via 192.168.1.254 via eth1 table my_isp_name
ip route add 192.168.2.0 dev eth2 src 192.168.2.1 table my_other_isp_name
ip route add default via 192.168.2.254 via eth1 table my_other_isp_name

 

In english, that means:  In order to send traffic to the 192.168.1.0 network or the 192.168.2.0 network send it out of eth1 or eth2 from IP address 192.168.1.1 or 192.168.2.1.  These routes are then stored in the relevent routing table; either one called my_isp_name or one called my_other_isp_name.

 

4. Create some additional routes for communicating between the routing tables and the local interfaces.  This is as much of an efficiency saving as anything, telling each routing table the quickest way to the other interfaces.

ip route add $P0_NET dev $IF0 table my_isp_name
ip route add $P2_NET dev $IF2 table my_isp_name
ip route add 127.0.0.0/8 dev lo table my_isp_name
ip route add $P0_NET dev $IF0 table my_other_isp_name
ip route add $P1_NET dev $IF1 table my_other_isp_name
ip route add 127.0.0.0/8 dev lo table my_other_isp_name

5.  Add the routes to the main routing table, so that traffic knows how to find the networks which have the gateways to the internet on:

ip route add $P1_NET dev $IF1 src $IP1
ip route add $P2_NET dev $IF2 src $IP2

6.  Apply the correct routing rules depending on where the traffic is coming from:

ip rule add from $IP1 table my_isp_name
ip rule add from $IP2 table my_other_isp_name

7.  The science bit.  Create a route which will send traffic down each of the internet connections in turn.  This is a global default route, so any traffic which isn’t bound for the local $P0_NET network will use this rule to get to the internet.

ip route add default scope global nexthop via $P1 dev $IF1 weight 2 nexthop via $P2 dev $IF2 weight 1

As you can see, this isn’t a 50/50 split.  We’ve applied some weighting, in this case because I get more GBs with my_isp_name.  We’re actually going to send twice as much traffic down $IF1 as we are down $IF2.

8.  By default, routes to other networks are cached for 10 mins.  This means that once a route is decided on, traffic will always follow the same route for at least the next 10 minutes.  The counter will be reset every time that particular route is used.  This is great for large downloads because it means that your IP address isn’t suddenly going to change half way through getting the file.  But, for something like IRC where it’s possible that no traffic will be sent for 10 minutes, then the IRC server could well see your IP address changing as traffic flips between the two routes.  IRC servers don’t like that.  So you need to fix some routes so that traffic will always follow the same path.

ip route add 174.143.119.91/32 via $P2

This sets the route to a specific Freenode IRC server to always go via my_other_isp_name.

9.  Last of all, we need to add an extra layer of NAT so that traffic can find its way back to the host on the LAN side.  This is necessary because our internet modem/routers are not really routers at all.  They simply take traffic from one side, rewrite the headers and push it out the other side, while maintaining a look up table of what came from where and doing the reverse when a reply arrives. If you can configure the routes on to your modem, then you don’t need to be reading this in the first place, so let’s assume you can’t.

iptables --table nat --append POSTROUTING --out-interface $IF1 -j MASQUERADE
iptables --table nat --append POSTROUTING --out-interface $IF2 -j MASQUERADE
iptables --append FORWARD --in-interface $IF0 -j ACCEPT

For things like port forwarding you do now have to deal with a bit of an odd NAT situation.  For example, let’s assume I want to run a web server on the same machine that runs the routing.  I decide to only make the server available on the WAN IP address of $R1.  I set up Router 1 to NAT port 80 on the WAN to port 80 on $IP1.  Then I need to configure the web server on the routing server to listen on that interface.  Not too bad really.

And there you have it.  Two internet connections, both available to all the clients on your network.  Make sure that your clients use the IP address of the server as their default gateway and everything should just work.  The additional load on the server is negligible.

If you reboot the server you’ll need to add all the routing information again, so create yourself a Bash and have it run from /etc/rc.local or something.

 

1 comment » | linux, Making the world a better place

Overriding Outbound Dial Command Options

September 5th, 2011 — 7:03pm — By Will Cooke

I’ve previously noted that you can apply time limits to calls going out of your Asterisk box:

http://www.whizzy.org/2011/02/more-asterisk-hints/

using the L(nnn,mmm,yyy) options for DIAL_TRUNK_OPTIONS. But, what if you don’t want to limit the length of calls for a specific trunk?

Well, FreePBX has a context called [macro-dialout-trunk-predial-hook] which lets you jump in at the very last moment and override any settings you like, which is perfect for this sort of thing.

I got the idea from here:

http://www.freepbx.org/book/export/html/5893

I’ve added this to extensions_custom.conf:

[macro-dialout-trunk-predial-hook]
 exten => s,1,NoOp(Trunk ${OUT_${DIAL_TRUNK}} selected)
 exten => s,n,Gotoif($["${OUT_${DIAL_TRUNK}}" != "SIP/TRUNK_NAME"]?skip)
 exten => s,n,NoOp(Setting DIAL_TRUNK_OPTIONS to Ww)
 exten => s,n,Set(DIAL_TRUNK_OPTIONS="Ww")
 exten => s,n(skip),MacroExit()

Which tests the name of the trunk which is to be used and explicitly sets DIAL_TRUNK_OPTIONS if it matches, otherwise nothing happens. This is a very powerful feature and my extremely crude hack doesn’t do it justice. I wanted to make a note of this before I forgot.

A better way might be to look at the dial prefix? I’ll investigate that option, but for now that will have to do.

Comment » | asterisk

NOOP scheduler – not worth it for SSD

August 16th, 2011 — 9:45am — By Will Cooke

I’ve got a new Thinkpad X220i with a 128GB SSD.

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.

 

 

Comment » | linux

Off to pastures new

July 22nd, 2011 — 3:33pm — By Will Cooke

Comment » | Uncategorized

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