Basic Asterisk Configuration

I really like the stats package included with WordPress, it provides an excellent source of ideas for what I should write about next. Many of you reach me through a post titled Asterisk and Fedora Core 7 looking for more specifics on how to configure Asterisk. If you need to install Asterisk on FC7, read the Asterisk and Fedora Core 7 post first.

Alright, so you’ve got Asterisk installed but its not configured or has the default Asterisk sample configuration files.

The Asterisk configuration files are found in /etc/asterisk. If you are using Asterisk without telephony hardware, you really need to be concerned with 2 or 3 files. Of importance are the following files

1. sip.conf: this file contains everything to do with the SIP protocol, settings and authentication for Asterisk.
2. extensions.conf: At the most basic level, this file contains the call-plan; what happens on in-bound calls and how outgoing calls are to be treated.

You’re also going to need something to test Asterisk with. Either a soft-phone such as X-Lite or a handset device such as a Polycom 430, or a ATA device, such as the LinkSys PAP-2.

For the sake of this sample, we’re going to configure Asterisk to handle the SIP registration of 3 IP devices.

1. X-Lite Softphone [extension 203]
2. a Polycom 301 [extension 200]
3. a Linksys PAP-2 (Vonage un-locked) [extensions 201 and 202]

sip.conf is easy enough to get going; some advanced features found on some handsets may require additional settings but to keep things simple lets start with the softphone alone.

Backup your existing /etc/asterisk/sip.conf and go with something like this

[general]
port=5060
bindaddr=0.0.0.0
context=home
tos=0x18
nat=yes
externip=YOUR PUBLIC IP
disallow=all
allow=g729
allow=gsm
allow=ulaw

[203]
type=friend
host=dynamic
context=home
secret=agoodsippasswordgoeshere
callerid=CIA FBI ATF <1-555-555-5555>
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw

Feel free to have some fun with the “callerid” parameter, you can have some great fun with your friends and family.

Make a backup of /etc/asterisk/extensions.conf and replace it with this

[general]

static=yes
writeprotect=no

[home]
exten => 55,1,Playback(demo-echotest) ; Let them know what's going on
exten => 55,2,Echo ; Do the echo test
exten => 55,3,Playback(demo-echodone) ; Let them know it's over

This bare bones configuration sets up extension 55 in the “home” context to be an echo server.

Get Asterisk to re-read the sip.conf and extensions.conf files (if already running, to start Asterisk, just run “asterisk” as root).

[root@athlon asterisk]# asterisk -r
Asterisk 1.2.20, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.20 currently running on athlon (pid = 3193)
Verbosity is at least 42
athlon*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
athlon*CLI> extensions reload
  == Parsing '/etc/asterisk/extensions.conf': Found
    -- Registered extension context 'home'
    -- Added extension '55' priority 1 to home
    -- Added extension '55' priority 2 to home
    -- Added extension '55' priority 3 to home
athlon*CLI>

Configure the X-Lite phone by adding a SIP account with the following details:

Save the settings and X-Lite should register with Asterisk. If watching the Asterisk CLI, you should see the following

Connected to Asterisk 1.2.20 currently running on athlon (pid = 3193)
Verbosity is at least 42
    -- Remote UNIX connection
    -- Registered SIP '203' at 10.10.50.198 port 14126 expires 3600
    -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 203
athlon*CLI>

At this point you should be able to dial the echo server. Go ahead and dial “55″ and hit the green send button. You’ll connect with the “Echo Test”, basically just echoes whatever you say to it. It’s simple but a great way to get the basics right without having to deal with the large “sample” configuration files.

If your call to the echo server worked, the rest of the setup is basically the same as before, editing both the sip.conf and extensions.conf files.

Moving on, lets add the other extensions to the mix. We’ll have our own little PBX by the end of this!

Add the rest of the SIP devices to the sip.conf file

[general]
port=5060
bindaddr=0.0.0.0
context=home
tos=0x18
nat=yes
externip=YOUR PUBLIC IP
disallow=all
allow=g729
allow=gsm
allow=ulaw

[200]
type=peer
host=dynamic
context=home
secret=agoodsippasswordgoeshere
callerid= CITY MORGUE
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw

[201]
type=friend
host=dynamic
context=home
secret=agoodsippasswordgoeshere
callerid= CITY MORGUE
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw

[202]
type=friend
host=dynamic
context=home
secret=agoodsippasswordgoeshere
callerid= CITY MORGUE
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw

[203]
type=friend
host=dynamic
context=home
secret=agoodsippasswordgoeshere
callerid=CIA FBI ATF
dtmfmode=rfc2833
nat=yes
mailbox=200@home
disallow=all
allow=ulaw

Now lets add the sip registrations to the extensions.conf file so that calls can be placed amongst the devices.

[general]

static=yes
writeprotect=no

[home]
exten => 55,1,Playback(demo-echotest) ; Let them know what's going on
exten => 55,2,Echo ; Do the echo test
exten => 55,3,Playback(demo-echodone) ; Let them know it's over

exten => 200,1,Dial(SIP/200,20)
exten => 201,1,Dial(SIP/201,60)
exten => 202,1,Dial(SIP/202,60)
exten => 203,1,Dial(SIP/203,60)

Reload sip.conf and extensions.conf and you should now be able to place calls between the registered devices and each device should be able to dial the echo server.

This is a very basic Asterisk configuration that should allow you to further explore other Asterisk options. In the next parts of this post, we’ll explore adding “trunking” with a VoIP provider, ZapTel hardware and voice mail with email notification (In no particular order).

Asterisk and Fedora Core 7

Looking over my blog stats, most of you reach me via a Google search for FC7 and Asterisk. With that, this post will detail installing Asterisk on FC6-7 or any modern Linux distribution for that matter as nothing here is Fedora specific anyways.

Getting Asterisk compiled and installed is quite trivial, so I thought it would be wise to spend a few moments examining the hardware you intend to use with Asterisk. If you are using a PSTN interface card through ZapTel such as Digium TDM or something else, you need to be aware that these cards generate a tremendous number of interrupts during their operation and the target system must be able to handle these interrupts quickly.

I’ve deployed Asterisk at the office where it provides call queues, automatic attendant, voice-mail and telephony services for 30+ people. This particular setup runs on a Dell 2850 with 1GB of RAM at 3.0ghz with a Digium TDM2400 card to analog PSTN lines. This setup’s load average rarely peeks over 0.01 during the day and the only complaint is occasional echo on the PSTN lines. We will soon be ditching the PSTN lines in favour of SIP channels from Primus providing both a huge cost savings and improved call quality.

At home, I’ve run Asterisk on a PIII-800 with 256MB of RAM using SIP only (no PSTN hardware) and have never had a problem with load related glitching or call drops.

The point in all this is to ensure that you select hardware appropriate for the application; don’t expect that old surplus P-III550 to serve as a corporate PBX for very long; but for home purposes that’ll only ever have 1 or 2 concurrent calls, it’ll probably be fine.

If you go with PSTN hardware, make sure to run the “zaptest” (or other for non-Zaptel hardware) program to test the interrupt timings.

Getting on with the installation, you should probably start by making sure any distribution specific Asterisk packages or Asterisk look-a-like clones are removed before you start. On FC7, OpenPBX is an available package, but in my opinion isn’t worth wasting the download. Go and get Asterisk from Digium directly and build your own.

This assumes you have a working build environment; generally the development packages for your particular distribution should have installed the basics like “make” and “gcc”… You’re really on your own here.

We’re going to be compiling Asterisk 1.2 at this time. Although 1.4 is out and in reasonable shape, I only have 1.2 experience and I’m quite happy with it so far.

Download the following source packages:

Asterisk 1.2.23 The Asterisk Code

Add-Ons 1.2.7 Asterisk Add-ons

Asterisk Sounds Pre-Recorded PBX sounds.

If you are going to be using ZapTel hardware, you’re going to have to build a kernel module. Download ZapTel as well.

To make things really easy, I’m going to include the commands to get this done. Lets start with Asterisk. Get, untar and build.

[skin@borg ~]$ mkdir asterisk
[skin@borg ~]$ cd asterisk/
[skin@borg asterisk]$ ls
[skin@borg asterisk]$ wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.23.tar.gz
--21:22:42--  http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.23.tar.gz
Resolving ftp.digium.com... 216.27.40.102
Connecting to ftp.digium.com|216.27.40.102|:80... connected.
...downloading...
[skin@borg asterisk]$ tar -xzvf asterisk-1.2.23.tar.gz
... lots of files...
[skin@borg asterisk]$ cd  asterisk-1.2.23
[skin@borg asterisk-1.2.23]$ make
...compiling...
 +--------- Asterisk Build Complete ---------+
 + Asterisk has successfully been built, but +
 + cannot be run before being installed by   +
 + running:                                  +
 +                                           +
 +               make install                +
 +-------------------------------------------+
[skin@borg asterisk-1.2.23]$ su root -c "make install"
[skin@borg asterisk-1.2.23]$ su root -c "make samples"

If everything worked out, you are the proud owner of a new Asterisk installation. There are a couple more things that need to be done though.

Lets do the sounds next.

[skin@borg asterisk-1.2.23]$ cd ..
[skin@borg asterisk]$ wget http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz
[skin@borg asterisk]$ tar -xzvf asterisk-sounds-1.2.1.tar.gz
[skin@borg asterisk]$ cd asterisk-sounds-1.2.1
[skin@borg asterisk-sounds-1.2.1]$ su root -c "make install"

Lets do the add-ons now. If you want to use MySQL for CDR support, you need to make sure the MySQL-devel packages are installed for your distro before you run this. It going to look for mysql.h, errmsg.h and mysql_version.h.

[skin@borg asterisk]$ cd ..
[skin@borg asterisk]$ wget http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.7.tar.gz
[skin@borg asterisk]$ tar -xzvf asterisk-addons-1.2.7.tar.gz
[skin@borg asterisk]$ cd asterisk-addons-1.2.7
[skin@borg asterisk-addons-1.2.7]$  make
[skin@borg asterisk-addons-1.2.7]$  su root -c "make install"

At this point, you should have a fully installed, ready for configuration Asterisk PBX. The installation gets a little more complicated if you need ZapTel for some PSTN hardware. If you’re planning to use the Conference Room feature, you’re going to need a ZapTel timer even if you don’t have hardware. It probably a good idea to build ZapTel anyways.

Like above, Get, untar, and build ZapTel.

[skin@borg asterisk-addons-1.2.7]$ cd ..
[skin@borg asterisk]$ wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.19.tar.gz
[skin@borg asterisk]$ tar -xzvf zaptel-1.2.19.tar.gz
[skin@borg asterisk]$ cd zaptel-1.2.19
[skin@borg asterisk]$ make
[skin@borg asterisk]$ su root -c "make install"
Building /etc/modprobe.d/zaptel...
***
*** WARNING:
*** If you had custom settings in /etc/modprobe.d/zaptel,
*** they have been moved to /etc/modprobe.d/zaptel.bak.
***
*** In the future, do not edit /etc/modprobe.d/zaptel, but
*** instead put your changes in another file
*** in the same directory so that they will not
*** be overwritten by future Zaptel updates.
***

Now we can try the ZapTel basics. Become root and try:

[root@borg ~]# modprobe zaptel
[root@borg ~]# lsmod
Module                  Size  Used by
wctdm                  37056  0
zaptel                183972  1 wctdm
crc_ccitt               6337  1 zaptel
.
.
.

The lsmod should show ZapTel loaded. This doesn’t imply that your hardware is configured, just that the ZapTel basics are in place. You’ll have to reference the specific module and configuration for your hardware, but the basics should be in place to get you going.

Blog Moved to the hosted WordPress…

Why the heck not? Its one less thing to worry about. I had installed the WordPress that came with Fedora core 7 but have since had some trouble with the i2o disk controllers. I basically lost everything on a RAID-5 array. Nothing aside from the few posts I made were really at risk of being lost, everything was just backed up prior to the system shuffling anyways.

The cause of this bizarre failure still escapes me.

I’d been tinkering with some new 100mm fans that I picked up at the local automotive/everything surplus store. The plan was to stuff as any of these as possible throughout both my desktop PC case and the server. 5 fans at $2.49ea…. great deal. I started with my desktop PC; in there is an eVGA 8800 GTS that idled in the mid 60C and would peak into the 80’s under load. 2 intake fans were added to it as well as an intake fan directly over the video card mounted on the case door. This did a great job of bringing the idle temps down to the mid-50’s and does a great job dissipating the heat under load.

Anyways, this is where it started to go wrong. After completing the fan install on my Desktop PC, I shelled into the FC7 box and issued a ‘halt’. Usually, the system crunches around, kills off all the processes and preps the system for shutdown. The usually finishes with a message like “System Halted. Power down”. 15 minutes have gone by and the shutdown process seems to have stalled; its not doing anything. I think this sort of thing happens on all Linux installs occasionally; perhaps due to some process that was killed manually and the PID file stays behind perhaps making the init script wait for a process that has already ended… It really could be anything and there is no way of knowing after-the-fact what it was.

So I gave it the “Big Red Button”. Powered-off, unplugged it and installed a few case fans. The fan went onto the PSU lead with the other fans using a full 12volt line. (There will be a rant about Antec’s “Fan Only” PSU lead in a future post). Checked my work, closed up the case and reconnected everything. Flicked the power switch…

It passes POST as usual but this time, instead of booting off the Adaptec 2400, it fell through and tried to boot off the CD, then tried to PXE off of each NIC. “No biggie” I thought; on rare occasions CMOS data can get reset when tampering inside a case… it does happen. So, power on again, mash the [DEL] key. Immediately I recognize that the settings are fine, as I left them.

Try to boot again… same deal.

Boot again and access the Adaptec hardware configuration, no problem, the array is still there. The tool only lets you manage the array, nothing configurable about the card such as boot interrupts or memory addresses.

Alrighty…. I don’t quite have the “Oh No!” feeling yet, so I boot off the FC7 DVD to see if I can determine if the card is still working on the Linux side; I can get my data off or even repair over-top. So I boot the Recover option from the DVD. The I2o drivers come up, the card is seen, capacity of the drives listed… but NO PARTITION TABLE! WtF? I break out of the installer to the first available console and try ‘fdisk’ for myself…. yup, its a big old empty partition table.

The “Oh No!” feeling sets in.

Fu-diddly-ck.

Having let FC7 configure the partition table when first installed, I didn’t note the specific partition sizes and disk layout. I’m no expert with the guts of the FC installer so manually invoking the partition tool escapes me. So I began to evaluate the old “diminishing returns” formula… RAID controllers tend to re-initialize when partition tables are written, so either way it looks like this installation is up-the-creek.

FC only takes a few minutes to re-install and I’d only be “out” 2 things, my WordPress installation/posts (a bit of searching later yields an XML file. w00t) and my latest configuration of Asterisk, Samba and such. So rather than messing around for hours on end, I quickly re-installed and got things going again.

At this point, I’m entertaining wild theories and ideas about what might have happened here and what I could have done to recover from this. Something mangled the partition table, I know that much. I’d suggest something about the RAID superblocks… but nothing hinted that the array was ever broken, its like something just blew away the partition table.

I suppose the lesson is, no matter how young and installation is, its never going to be immune from a good backup regime. Back-it-up or else!!

Fedora Core 7: Distro Review

My first ever WordPress blog entry… goodbye cherry!

Starting off with a few notes and observations of Fedora Core 7 seems appropriate as it (used to, I moved to WordPress.com) powers this site and some supporting elements for my home network.

Someone looking in might think I’m crazy, but my job and interests keep my stables quite full of computers. My wife and I each have our desktops, as well as one older HP NetServer 2U server as a firewall/ LAMP install and finally a PIII-800 with about 200GB of disk space that serves as my file dumpster and SAMBA NT domain controller.

Eco-concerns being what they are, it makes sense on a few fronts to combine the best of left over parts and build something to replace the HP and the PIII clone.

The target machine is my old desktop: (only the hydro company would reboot it)

• an AMD XP Athlon 3200+,
• 512 MB RAM and
• 140GB of IDE disks on an Adaptec 2400 RAID controller
• 8x DVD Burner
• Nvidia GeForce2 TI 64MB AGP
• Intel E1000 NIC

The obvious challenge in this configuration is an install kernel thats aware of the Adaptec 2400. Trustix 3.1.x supports this and if it was a lesser machine I’d certainly have gone with it; however I found it wasteful to dedicate such hardware to the menial task of packet forwarding. This naturally lead me down the path of an X based distro and my previous experiences with Fedora Core 6 being quite good I opted to try FC version 7.

The ISO was downloaded from ibiblio in record time and committed it to DVD.

The installation was seamless. It detected all the hardware! A Web Server configuration with a GNOME desktop was selected and the install was invoked. The format took a little while, but the disk array had not fully synced; not that it hurts anything, it just slows things down. The installer went about its business and rebooted.

The GUI came up in an optimal 1600×1200 resolution and I logged in as root.

Strangely, eth0 (e1000) would not get an IP address from my current DHCP server. After reviewing the message log, I discovered that the onboard 1394 controller was being aliased as eth0 during boot up. You’d have to own a 1394 device in order to need a 1394 adapter, so I took the coward’s way out and disabled it in the BIOS. Surprise, surprise, eth0 came right up and got an IP.

Once the YUM update manager did its thing and if memory serves me correctly, there were a 104 updates. The server was left to update the software and the installation continued the next day.

Next article we’ll look at FC7’s OpenPBX package and why it should be avoided.

All in all, this is a good distribution.  It hasn’t given me much grief and I’ve been able to accomplish everything I needed it to do.  Nothing new really seems to jump up at you, and everything is where you’d expect it to be found.

Fedora Core 7: OpenPBX

Following the theme of my previous post on Fedora Core 7, I really wanted to touch on my experience with OpenPBX.

Just for a bit of background, we’ve opted out of traditional telephone in favour of a SIP based VoIP, provided by a Montreal area re-seller named BabyTel. I had figured out and configured BabyTel to run under Asterisk 1.2 on my HP NetServ machine where it was happily doing its job answering the phone and taking messages etc… However, with all this swapping and upgrading, the HP NetServ is being retired.

So, looking through the FC7 package manager I see a listing for OpenPBX; I dig a bit deeper. OpenPBX is built on Asterisk and is somewhat different. Here’s what the authors claim make it different.

• OpenPBX.org has built-in STUN support for SIP NAT traversal
• OpenPBX.org uses SpanDSP which means better codecs and full T.38 fax over IP support
• OpenPBX.org uses Sqlite instead of Berkeley DB as its internal database
• OpenPBX.org has a universal jitterbuffer for use with any channel type
• OpenPBX.org uses POSIX timers which means there are no Zaptel timing dependencies
• OpenPBX.org has much faster and more efficient dialplan execution because it uses hashing
• OpenPBX.org evaluates correctness and integrity of configuration data (work in progress)
• OpenPBX.org runs well under a virtual machine such as Xen or VMware

Looking at this list, one assumes that the basics work and the folks at OpenPBX are diligently working on icing and candy. Sounds pretty good right? Prepackaged solution, easy to install, standard Asterisk config? What could go wrong?

So I embarked on the journey; OpenPBX installed via the FC7 Package Manager and then configuration began…

Having working extension.conf, sip.conf and voicemail.conf files, the theory is you should be able to drop these files in and basically start OpenPBX and life should go on. So, I grabbed the files off the old Asterisk server and copied them to /etc/openpbx.org/ on the new rig.

The copy was immediately followed by a service openpbx start.

Lo and behold, OpenPBX did start and did register the SIP channel with BabyTel. So I proceeded with a test call to the local talking clock (613-745-1576); great success! The call was answered and I was told the time.

Next and finally I moved onto voicemail. My callplan has the * key linked with VoiceMailMain; so to access any voicemail that was left, one would simply dial * and follow the prompts.

As soon as I tried the * key from a desktop phone; “click” the call was dropped. Looking through the logs the problem seems obvious; OpenPBX doesn’t ship with sound files.

Right around here is where OpenPBX begins to resemble a jet engine, in that it concurrently sucks and blows. I begin to search of a tarball of sound files for OpenPBX. OpenPBX seems to be a product without a real identity; searching for OpenPBX yields a sparse wiki and something much worse called CallWeaver. It would seem that OpenPBX is now something called CallWeaver. Take your pick, documentation for both packages is just plain terrible.

After searching and following countless 404’s the sound files were eventually fished out of svn://svn.openpbx.org/openpbx-sounds/trunk/sounds/ using svn. These files included a Make that ran each WAV through SOX for formatting presumably. Once complete the files were copied to the sound path specified in OpenPBX’s config files.

So, I try again. Dial voicemail using “*”. Click… hangup.

In the log again, this time its bemoaning the fact that it cannot find a codec to handle the translation of ulaw to g729. Which I suppose is fine as there is no g729 codec installed under OpenPBX. I deal with this by removing all references to g729 from SIP.conf to see if that helps.

Reloaded OpenPBX and tried again… A similar error this time; Unable to find a codec translation path.

mmmmkay. So now its bitching about codec paths that aren’t even configured. I figured a “reload” was enough, so I stopped and started OpenPBX. During the subsequent start-up it complained about the g729 codec. Loader.c: /usr/lib/openpbx.org/modules/codec_g729.so: undefined symbol: ast_log.

Assuming that an Asterisk compatible module would work in OpenPBX was a mistake. Based on the error one can only assume that in the process of “making” OpenPBX the authors did a wholesale “Find and Replace” on certain “asterisk” like words that resulted in breaking binary compatibility all over the place.

With a sour mouthful of OpenPBX tainting my palette, enoughs enough. YUM was called in to remove OpenPBX with extreme prejudice. I visited the Asterisk home page, downloaded the latest version of Asterisk and the supporting files and rolled-my-own build of Asterisk. I copied my configuration and g729 codec over and rigged up an init.d script based on the stock proftpd script.

First try, the talking clock worked.

Second try, the voicemail worked.

Given the 5 minutes of effort it took to download, compile, install and configure it took to get Asterisk going, I feel I’m way ahead of the 2 hours I’ve been tooling around with OpenPBX.

Why I can’t recommend OpenPBX to anyone:

1. Its a package looking for an identity. Not very clear what it is… or who is in charge of it.
2. The name change from OpenPBX to Callweaver further skews this identity.
3. Documentation is piss-poor. You might as well look up Asterisk how-to docs or use voip.info.org .
4. Binary compatibility with Asterisk is broken
5. Doesn’t seem to persist SIP registration between shutdowns. All the devices need to re-register. There may be a setting for this… but Asterisk does this out-of-the-box
6. Uses SQLite… but it doesn’t work nor is there any documentation to help.
7. Installation does not work out-of-the-box. There are a number of obscure downloads and builds required.

What does it have going for it?

1. Its a FC7 package. Maybe the worst one, but its easy to install but easy stops there!

Recommendation? Stay away. Get the real-deal from Asterisk. It works and has a future.