Using LiveCD or a running system

• Press Super (Window) key, and open Terminal
• Type sudo modprobe -r acer_wmi

That’s all. You should be able to connect to the wireless till next reboot.

Making the change permanent

Edit the file /etc/modprobe.d/blacklist.conf and in the end type

blacklist acer_wmi

Reboot and you’re done. If you’re doing it entirely from the GUI, then editing the file could be done by pressing ALT+F2 (this would open a “Run” window) and typing there gksu gedit /etc/modprobe.d/blacklist.conf

Please note: This applies to wireless card having rtl8188ce chipset. You can view that by typing lspci in a terminal and searching for the entries respective to Network Controller.

I recently got a Lenovo E420 laptop, and I just went ahead and installed Ubuntu 11.04 on it. Few are some of the things that may help those who’re facing issues with it. I got 4 GB RAM on Core i5. It’s 64bit processor, so I preferred 64bit Ubuntu instead of 32bit pae kernel.

Wireless

The machine has rtl8188ce chipset, and the Realtek drivers available from Realtek’s website don’t work well. You just need to blacklist acer_wmi module

Edit /etc/modprobe.d/blacklist.conf and add

blacklist acer_wmi

Save and reboot. This won’t work if you’ve compiled Realtek drivers and installed them. It’ll work only in the default install. Here’s more detailed link -  http://askubuntu.com/questions/53625/wireless-on-thinkpad-edge-e420s

Random screen lockups

I am not sure why it occurred – due to the kernel or due to the compiz settings. I did both. Installed Kernel 2.6.39 from http://kernel.ubuntu.com/~kernel-ppa/mainline/v2.6.39-rc4-natty/

Also, I installed CompizConfig Settings Manager (ccsm) from Ubuntu Software Center, and then disabled this option

General -> OpenGL -> Sync to VBlank (uncheck this)

Reboot. The system may not hang. http://www.preshweb.co.uk/2011/05/ubuntu-11-04-x-freezing-after-screensaver-active-laptop-lid-closed/

Update 26 July 2011: There are still instances of system hanging Unable to identify the cause as yet. However, this time the system just hangs with no mouse movement at all. This behavior I experienced in 32bit-pae kernel too and usually happens when system has gone through more than a few standbys. Will diagnose and update further.

Update 2 – 30 July 2011 : I updated the BIOS to v1.16 (earlier it was 1.10), but it didn’t do any good. The system still got hung. It is not due to standby also as I disabled the “Suspend” option from Power Management section and rebooted the computer and used it for around 3 hours after which I got a Kernel Panic. Realtek has put a new version of their drivers for rtl8188ce chipset – I’ve just installed those and rebooted and it seems to be working. Will update later if it has cured the hanging problem.

Update 3 – 31 July 2011 : After installing the realtek rtl8188ce drivers as mentioned in the post above, I’ve not rebooted the machine since more than 24 hours. Had suspended it more than a few times, and did various experiments which earlier eventually led to a freeze. But I’m happy to report that till now all has been good and it has not yet hung. I’ll keep working on it for next few days and see if there has been a proper resolution.

Indicator icons for unsupported apps

I installed gnome-do and I was unable to see the indicator icon for that. Installed dconf-editor from Ubuntu Software Center.

Got the name of the application from .xsession-errors file (TrayChild). Opened dconf-editor and added ‘Do’ in there. Logged out and logged back in and all is well.

http://askubuntu.com/questions/36898/how-can-i-see-a-list-of-all-the-systray-icons-that-are-not-whitelisted/39750#39750

Don’t set systray-whitelist to “all” as that may lead to the icons in indicator being unclickable.

To be honest, it was horrendous to use Ubuntu 11.04 due to severe productivity issues for the first week. I installed 32bit pae kernel so that the 4GB RAM could be supported. But I couldn’t upgrade it to 2.6.39 as no pae kernel was available for 2.6.39. I rather installed 64bit Ubuntu, and updated it immediately and rest did what I mentioned above. No more hangs till now, and life has been happy.

• Use “Nokia Messaging” which means you receive your emails via Nokia servers (similar to blackberry) and it’d not be your phone but nokia servers which would connect to your IMAP server and push mails to your phone. This works better, but then you’re giving your details to Nokia, and could be a privacy issue for some.
• Use the in-built IMAP client. This would mean you don’t have a ‘push’ email, but a check interval of 5 minutes. Not awesome.

Following is one more way to set to use IMAP via the Nokia Mail for Exchange client, and you don’t need to have any Microsoft Exchange setup.

Concept

1. You need to be running an IMAP server.
2. There’s an open source PHP based software called “z-push” which allows activesync connectivity and has IMAP backend, which means you can use it to ‘sync’ your phones using the Nokia Mail for Exchange client.
3. You need Apache web server with php5 and php5-imap support where z-push would be setup.
4. You download and configure z-push.
5. You configure your phone settings, and then use Nokia Mfe client to connect to your IMAP server and thus have “push” functionality.

Assumptions

1. An IMAP server is running with SSL support with self-signed certificates (I used dovecot), which has usernames as ‘user@domain.com‘ format.
2. You have Apache with PHP5 and php-imap support. On ubuntu/debian you can install it using apt-get install php5-imap.
3. Nokia Mail For Exchange supports only a single account on a phone. So if you have multiple email accounts to be synced, bad luck.
4. It was tested on a Nokia E6 running Symbian Anna. Your setup may vary or could be different.

Server side setup

1. Download z-push from Source Forge http://z-push.sf.net ; I used version z-push-1.5.4RC-705.tar.gz.
2. Extract it in a suitable location, for example, /var/www/z-push
3. You can refer to the INSTALL document too for more information inside the z-push extracted files.
4. Make the ‘state’ directory web server user writable. So do chown www-data.www-data state or chmod 777 state.
5. Configure z-push. We will modify the following

   $BACKEND_PROVIDER = "BackendIMAP";
   
   // ************************
   //  BackendIMAP settings
   // ************************
   
   define('IMAP_SERVER', 'localhost');
   define('IMAP_PORT', 993);
   define('IMAP_OPTIONS', '/ssl/novalidate-cert');
   define('IMAP_DEFAULTFROM', 'domain');
   define('IMAP_SENTFOLDER', 'Sent');
   define('IMAP_USE_IMAPMAIL', false);

6. So that the “Sender Name” when sending out email works well, there’s a IMAP_DEFAULTFROM setting, which didn’t work too well in my case, so I modified the backend file to display the “Sender’s Name” correctly in the format Firstname Lastname <user@domain.com>. The following changes were done in backend/imap.php

   # Added the following lines in the section below
   # $v = "$this->_domain <$this->_username>";
   # $envelopefrom = "-f$this->_username";
   
               if ($k == "from") {
                   if (trim($v)) {
                       $changedfrom = true;
                       $v = "$this->_domain <$this->_username>";
                       $envelopefrom = "-f$this->_username";
                   } elseif (! trim($v) && IMAP_DEFAULTFROM) {

7. In Apache configuration, I have SSL setup. So in the configuration file for the SSL, I specified an alias as under

           Alias /Microsoft-Server-ActiveSync "/var/www/z-push/index.php"

8. Restart Apache and your z-push is ready.

Client side/phone setup

1. Now that the server side setup is ready, we’ll configure the phone.
2. You can enter the settings as per the requirements while configuring the Mail For Exchange client.
3. The “domain” would be your full name (as you wish to display when you send someone an email).
4. That’s all. Sync it and you’re ready to go!

Troubleshooting

1. To enable debug logs, create a file debug.txt in the z-push directory and assign it web server user writeable privileges. For example:

   cd /var/www/z-push
   touch debug.txt
   chown www-data.www-data debug.txt # or; chmod 777 debug.txt

2. You may want to experiment with the IMAP_OPTIONS by referring to http://php.net/imap_open
3. Your ‘Sent’ items folder could be different. You may want to keep it blank (the default).

It didn’t work well for

• Linksys WUSB54GC v3 USB adapter. Till Ubuntu 10.04, there were no good drivers for use with hostapd (supporting mac80211 / nl80211). In Ubuntu 10.10, though the drivers were working, but the connection was highly unstable – in fact unusable. The drivers that I tried were rt2800usb (which allowed the adapter to come into Master mode), and rt2870sta which didn’t allow the adapter to come into “Master” mode. It always remained “Auto” or “Managed” (checked via iwconfig). Need to test in 11.04 when it arrives.

What worked pretty well

• 02:00.0 Ethernet controller: Atheros Communications Inc. AR5001 Wireless Network Adapter (rev 01), using ath5k drivers. hostapd didn’t complain at all and all was smooth. This was in Ubuntu 10.04.

Concept

1. There’s an application called hostapd which allows converting a wifi adapter into an access point and provide privileges such as WPA authentication and ssid name definition etc. I used it and it worked well.
2. When a client connects to the access point, apart from authentication it’ll require IP addresses to be assigned. For that a DHCP server is used.
3. You need to have 2 interfaces, one which accesses the net (e.g. eth0), and other which provides the access point services (e.g. wlan0).
4. You start the wlan interface, assign it an IP address, start the dhcp server, setup firewall/nat and start hostapd. That’s all to it. Your devices would be able to use the wifi adapter as the access point.

Procedure

apt-get install dhcp3-server hostapd

Modify /etc/hostapd/hostapd.conf and put the following

interface=wlan0
driver=nl80211
ssid=MyAP
hw_mode=g
channel=11
wpa=1
wpa_passphrase=MyPasswordHere
wpa_key_mgmt=WPA-PSK
wpa_pairwise=TKIP CCMP
wpa_ptk_rekey=600

The dhcpd.conf section in /etc/dhcp3/dhcpd.conf would have something like the following

subnet 10.10.0.0 netmask 255.255.255.0 {
        range 10.10.0.25 10.10.0.50;
        option domain-name-servers 8.8.4.4, 208.67.222.222;
        option routers 10.10.0.1;
}

Modify /etc/default/dhcp3-server

INTERFACES="wlan0"

Check what name your adapter got via iwconfig. You can change the name also and make it persistent via /etc/udev/rules.d/70-persistent-net.rules so that it always gets a single type of name. In our example we’re assuming wlan0, but it could be changed. Make necessary changes in your configuration too.

Configure the new interface

ifconfig wlan0 10.10.0.1

The above could also be done in a better way via the /etc/network/interfaces file, but didn’t try it out. In any case if you shutdown hostapd, the network interface (wlan0) loses its address, so need to put a script which assigns it again before hostapd is started. An example could be

iface wlan0 inet static
 address 10.10.0.1
 netmask 255.255.255.0

Restart the dhcp3-server. It should now be ready to serve addresses and is also bound to the network interface too.

Allow ip masquerading

echo "1" > /proc/sys/net/ipv4/ip_forward
iptables -t nat -A POSTROUTING -o eth0 -j MASQUERADE

Now start hostapd and see the messages that it shows

hostapd -dd /etc/hostapd/hostapd.conf

Wait for a few seconds, it should show some probes being done by other wifi devices. If it shows, then probably you’re in good luck.

Now try to connect via your device to this access point. It should work.

To make this work on boot, can put the relevant config in /etc/default/hostapd

RUN_DAEMON="yes"
DAEMON_CONF="/etc/hostapd/hostapd.conf"
DAEMON_OPTS="-dd"

and also put the firewall rules in /etc/rc.local (make sure its executable).

Remember – if you shutdown your hostapd, the network card would lose its address. So you have to assign that again before starting hostapd. The usual practice would be

• Stop hostapd
• Stop dhcp server
• Restart network (or rather ifup wlan0 / ifconfig wlan0 10.10.0.1 would do)
• Restart dhcp server
• Start hostapd

Miscellaneous

1. You can check the wifi interfaces via

   iwconfig

2. To set a wifi adapter into master mode, try the following. If it doesn’t work and shows an error that it’s not possible or something, fret not – use hostapd as that’ll do that in any case.

   iwconfig wlan0 mode Master

3. Network Manager could create issues, though in my test environment – instead of using an ethernet interface, I used two wlan interfaces, one being controlled by Network Manager for internet access, and other for making it an access point.
4. modprobe -r ath5k / modprobe -r rt2800usb etc. is to be used for unloading the modules.
5. If you wish to proceed without using authentication so that you can test it easy, then put the following in /etc/hostapd/hostapd.conf

   interface=wlan0
   driver=nl80211
   ssid=MyAP
   hw_mode=g
   channel=11

Credit goes to

• Michael Montey from http://groups.google.com/group/noki-sms/browse_thread/thread/1012e376016ca2ba
• petrusek – who created NbuExplorer (http://nbuexplorer.sourceforge.net)

Here’s how I retrieved all the Contacts. It assumes you have a Backup.arc with you. If you’ve taken backup on the memory card and view the contents of the memory card on computer, you’ll see a Backup/ folder in which the Backup.arc would be there. If not, tough luck.

1. It works only on Windows. I used Windows XP.
2. Install NbuExplorer from http://nbuexplorer.sourceforge.net – it requires .Net Framework 2.0 which I downloaded from Microsoft website, so install the framework prior to opening NbuExplorer.
3. Install Java 1.4.2 and Symbian SDK. As of now I’m not too sure that Java 1.4.2 would be necessary at all. For Symbian SDK, you need to be a member of Nokia Forum – the version that I downloaded was S60-SDK-0548-3.0-f.3.215f.zip
4. Now open NbuExplorer and point it to the Backup.arc file. It should show the contents in it including pictures, sounds etc.
5. You need to locate the file having the name Contacts and ending with the extension .cdb. In my case the name was DBS_100065FF_Contacts.cdb and it was under C: -> private -> 100012a5 within the NbuExplorer interface. Once you locate it, right-click on the file name and select ‘Export selected file(s)’. It’ll ask you for a location – you may select “Desktop”. Keep that file safe – it has all your contacts.
6. Now close NbuExplorer and start Symbian Emulator. It would open a Symbian interface – you need to add a new contact in there. You can add anything. We just want it to build a contact database of its own which we’ll eventually replace with our own contact database. Once done, close the Emulator.
7. Next step is to replace the contact database created by the emulator by our actual database which has our contacts. In my case I went to C:\Symbian\9.1\S60_3rd\Epoc32\winscw\c\private\100012a5. The file DBS_100065FF_Contacts.cdb was already present. I replaced it with my actual file which I’d retrieved via NbuExplorer.
8. We’ll add a memory card to our Symbian Emulator. For that edit the file epoc.iniwhich is located in C:\Symbian\9.1\S60_3rd\Epoc32\Data and modify the following values so that they look like the following
   _EPOC_DRIVE_E \epoc32\winscw\e
   _EPOC_LocDrv_1 E: FAT
9. Let’s start the Emulator now. If you go to contacts, if all goes well you should be seeing all your contact listing. Hurray! Step 1 done.
10. Now we’ll need to transfer the contacts to the memory card. Mark all contacts and then select “Copy to memory card” from within the emulator interface. It should say that it has copied them to the memory card.
11. Now go to the “e” drive created via epoc.ini, that is, in C:\Symbian\9.1\S60_3rd\Epoc32\winscw\e. There would be a folder called “Others” inside which there’d be “Contacts”. If all goes well, you should see a lot of .vcf entries inside it. Those are your contacts. Copy them to an actual memory card of the phone – in the same directory (Others\Contacts). Now put the memory card in your phone.
12. In your phone, open “Contacts”, and then select the option Copy -> From memory card. The contacts would be copied. For me it retrieved all information with precision.

I had evaluated Nokisoft.com’s Noki Explorer application. There were two things I wasn’t sure of, (a) it retrieved 30 contacts in the trial version, but the data that was retrieved was not complete; and (b) it seemed to be a tad expensive for one time use. Nokisoft’s explorer may be easier to retrieve contacts – that you can decide.

Prepared by : Vivek Kapoor http://exain.com
Prepared on : 01 July 2010
PDF Version : Asterisk_Basics_Load_Balancing_DUNDi_Vivek_Kapoor

Purpose of this document

Asterisk is an open source PBX. Which means, you can setup your own little telephone exchange in your home/office environment. On a much larger usage, you may look forward to setting up a redundant phone setup wherein multiple boxes are interconnected with each other and provide some sort of failover capability and easier management, which is possible through something called DUNDi. This document aims to provide such a solution with a possibility of horizontal scaling instead of vertical scaling. So, this may mean that by using low-end commodity hardware, you can setup a strong telephone infrastructure which can cater to thousands of users.

Read the document carefully. Read it fully before you start implementing it. It is intentionally detailed so that you know what you’re doing.

Acknowledgements

All credit of this document goes to JR Richardson. I’m afraid I’ve not been able to find a suitable link for him, but his whitepapers and presentations are the ones which helped me move forward. And of course, Mark Spencer – the guy behind Asterisk. Without him this document wouldn’t have existed.

The writings by JR Richardson

• http://www.voip-info.org/storage/users/813/47813/images/1654/DUNDi_So_Easy.pdf
• http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf

Disclaimer

Though I have significant experience doing Linux and related systems, but I’m more of a novice in Asterisk. The following may or may not work for you. I don’t provide any warranty whatsoever of it being successful, or rather, consistently successful, and providing an actual load balanced solution. I’ve yet to implement it in a production setup, and hence all you do is at your own risk. The below procedure SHOULD work, but I CANNOT say that it WILL work.

Before we begin

A little about the nomenclature

• Asterisk : http://asterisk.org It is a PBX (Private Branch Exchange) software which runs on Linux platform. It allows you to have VOIP facility without incurring cost on proprietary hardware, and can run off a home desktop system.
• DUNDi : http://dundi.com DUNDi is a peer-to-peer system for locating Internet gateways to telephony services. Unlike traditional centralized services (such as the remarkably simple and concise ENUM standard), DUNDi is fully-distributed with no centralized authority whatsoever. We will be using it to interconnect multiple asterisk boxes and setting up a load balanced environment which will also act as a failover.
• SIP Softphone : A softphone running from your desktop which allows you to register to asterisk and enable VOIP communication. I, for this document, use Zoiper – http://zoiper.com
• RealTime : http://www.voip-info.org/wiki/view/Asterisk+RealTime It will allow us to connect to MySQL and have the users registered in there instead of being hard coded in sip configuration. This way it will be possible to have users register to any registration server and not just one on which they’re configured.

You need to have some experience related to Linux systems. You should be comfortable in compiling applications from scratch. The method mentioned in this email should work as is, but if there are issues, then you would be resolving them on your own. Also, it is highly recommended to have some working knowledge about Asterisk. You should have setup some Asterisk based environment – if you’ve not done that yet, then I think you should first concentrate on that. You’ll be delighted to see what all is possible through Asterisk.

Quick look at the steps

To make it work with the minimum configuration, we’ll proceed step by step so that we don’t miss a thing. This means we’ll not jump straight-away to the fully functional DUNDi setup, but do it slowly, understanding each step as we do it. Here’s how we’ll move forward

1. Installation of Asterisk on a Debian/Ubuntu systems (2 systems initially, later 3)
2. Basic SIP User Creation & Testing
3. Making it work with DUNDi – 2 Server Setup
4. More DUNDi – 3 Server Setup
5. Adding RealTime to it
6. What’s next

Asterisk Installation

1. You need to have atleast TWO working Debian/Ubuntu machine. I will be using THREE machines eventually, but will start with two. Though the following would work for any Linux distribution available, the steps related to installation and initialization of packages could vary – so you may refer to the documentation of your distribution.
2. I am using Virtual Machines running via KVM for this exercise. You may use Virtualbox, Xen, or any of your preferred solutions. Here’s the basic structure
   • Server A, IP: 192.168.1.5, MAC Address: 00:16:36:99:54:62
   • Server B, IP: 192.168.1.6, MAC Address: 00:16:36:2f:11:9c
   • Server C, IP: 192.168.1.10, MAC Address: 00:16:36:6e:d7:dc
3. Three packages are required from the asterisk website, the asterisk package itself, dahdi & asterisk-addons for mysql support. Get them from http://asterisk.org – I chose the following versions which were the latest at the time of writing this article
   • asterisk-1.6.2.8.tar.gz
   • dahdi-linux-complete-2.3.0.1+2.3.0.tar.gz
   • asterisk-addons-1.6.2.1.tar.gz
4. The installation steps are as under. The asterisk installation documents are readily available on the internet, so you may try out a more detailed ones. Here’s what will get you started
5. Install the pre-requisites first

   apt-get install linux-headers-$(uname -r) build-essential automake autoconf bison flex libtool libncurses5-dev libssl-dev libgsm1 libgsm1-dev libgsmme-dev libgsm-tools libxml2-dev libmysqlclient-dev
   

6. Untar all the downloaded packages (asterisk, dahdi and asterisk-addons) into /usr/local/src and then one by one run the following

   # Dahdi First
   cd dahdi-linux-complete-2.3.0.1+2.3.0
   make clean
   make
   make install
   make config
   
   # Now for Asterisk
   cd asterisk-1.6.2.8
   make clean
   ./configure
   make install
   make samples
   make config
   
   # Last, the asterisk add-ons (important for MySQL support)
   cd asterisk-addons-1.6.2.1
   make clean
   ./configure
   make menuselect # Optional - I usually disable h323 support
   make
   make install
   make samples
   

7. DAHDI may require some downloading from the Internet, so make sure you have the machine connected to the Internet, or use a proxy by using

   export http_proxy=http://your.proxy.ip:port
   

8. That’s all. Asterisk is configured. You may run the following to verify.

   /etc/init.d/asterisk start
   asterisk -r
   /etc/init.d/asterisk stop
   

Basic SIP User Creation & Testing

For each file listed below, certain changes are done. You can name as per your convenience, but it should match in all the relevant sections further in the document.

Server A,B & C : /etc/asterisk/asterisk.conf

Uncomment the following

verbose = 14

Server A : /etc/asterisk/sip.conf

[general]
srvlookup=no
NAT=yes
bindport=5060
allowguest=no
canreinvite=no

[1001]
type=friend
context=internal
host=dynamic
disallow=all
allow=gsm
canreinvite=no
qualify=yes

You can change bindport as per the requirements. If you happen to change the bindport to, say, 5070, then while configuring Zoiper, in the domain section, write something like 192.168.1.5:5070

Server A : /etc/asterisk/extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[internal]
exten => 123,1,Playback(demo-congrats)
exten => 123,n,Hangup()

Server B : /etc/asterisk/sip.conf

[general]
srvlookup=no
NAT=yes
bindport=5060
allowguest=no
canreinvite=no

[1002]
type=friend
context=internal
host=dynamic
disallow=all
allow=gsm
canreinvite=no
qualify=yes

Server B : /etc/asterisk/extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[internal]
exten => 456,1,Playback(demo-congrats)
exten => 456,n,Hangup()

Testing the setup

What we see above is that there are two servers, Server A and Server B. Server A has SIP user 1001, and Server B has 1002. Both the users belong to a context called “internal”, which probably means that when they dial in any extension, the first context that they’ll hit is “internal”, and from there onwards they’d be redirected to other contexts as per the dial plan.

The context “internal” is defined in “extensions.conf” file. There’s not much in that context except one single extension, 123 in Server A and 456 in Server B. When that extension is called, a congratulations message is played. So to test it, I have used an application called Zoiper which allows two lines to be connected. You can download Zoiper from http://www.zoiper.com/softphone/ – I used Zoiper Classic for Linux.

Let’s start asterisk first on Server A & Server B by using

/etc/init.d/asterisk start

and then connect to an asterisk session by using

asterisk -r

I created two SIP accounts in Zoiper, for Server A I entered 192.168.1.5 as domain, and 1001 as username, and for Server B I entered 192.168.1.6 as domain and 1002 as username. Rest of the details can be added as per your preference. I deselected the “Register on startup” option under advanced settings. Then on the Zoiper interface I registered to both the servers.

On Server A, something like the following should appear when you list the sip peers

CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1001/1001                  xx.xx.xx.xx     D   N       5060     OK (116 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]
CLI>

Same goes for Server B. It should show something like the following

CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
1002/1002                  xx.xx.xx.xx     D   N       5060     OK (134 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]
CLI>

Now select in Zoiper, the Server A Account and dial 123 – it should playback the congratulations message. If you try to dial 456, it’ll not work. But if you select Server B in the Account, and then dial 456, it should work as expected.

A sample screenshot of Zoiper is below

Also check the messages that appear in the console that you’d opened earlier (by using asterisk -r). A sample of the same on Server A is as under

CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [123@internal:1] Playback("SIP/1001-00000001", "demo-congrats") in new stack
    -- <SIP/1001-00000001> Playing 'demo-congrats.gsm' (language 'en')
    -- Executing [123@internal:2] Hangup("SIP/1001-00000001", "") in new stack
  == Spawn extension (internal, 123, 2) exited non-zero on 'SIP/1001-00000001'
CLI>

So, our basic configuration is working fine. If you add more SIP users in sip.conf file, and add relevant extensions in extensions.conf file, then you can have a single server PBX running perfectly fine. It is important to have the above working so that we can be sure that our setup is ready for DUNDi.

Making it work with DUNDi – 2 Server Setup

Before we delve further into DUNDi, please remember that the most important thing here is “contexts”. If you specify them incorrectly, it wouldn’t work as expected and you’ll be left wondering what’s happening.

Let’s begin now in a two server configuration. The purpose here is that extension 1001 would be registered on Server A, and 1002 on Server B. But, via DUNDi, Server A would know that extension 1002 is on Server B, and would effectively be able to call it too.

Server A : /etc/asterisk/sip.conf

Add a new context (regcontext to be precise) called “dundiextens” in the [general] section. It’s a pretty simple file

[general]
regcontext=dundiextens
srvlookup=no
NAT=yes
bindport=5060
allowguest=no
canreinvite=no

[1001]
type=friend
context=internal
host=dynamic
disallow=all
allow=gsm
canreinvite=no
qualify=yes

The regcontext parameter is very important here. It auto-creates a context as soon as a SIP user logs in (in our case the context will be dundiextens). This is used and searched by DUNDi to get to the desired extension. Later we’ll see how to view the users who belong to “dundiextens” context.

Server B : /etc/asterisk/sip.conf

Similar to Server A, the new context has to be added. The following would be the final sip.conf for Server B.

[general]
regcontext=dundiextens
srvlookup=no
NAT=yes
bindport=5060
allowguest=no
canreinvite=no

[1002]
type=friend
context=internal
host=dynamic
disallow=all
allow=gsm
canreinvite=no
qualify=yes

Server A & B : /etc/asterisk/iax.conf

We will be using IAX2 trunk to interconnect the DUNDi boxes. IAX2 doesn’t work too well in virtual machines as it requires a timing device/timing source. For me it didn’t work well as far as Playback() was concerned, but when doing a voice call everything seemed to be just fine. I’ve not been able to figure out if it’s a timing device/virtual machine issue or it’s just some setting in iax.conf that needs to be done. If I run dahdi_test in a virtual machine, the results are always above 99.6%. The same is with a dedicated machine, so I guess the timing source seems to okay.

Tried using SIP Trunk also, but it causes a number of issues when doing RealTime. When using RealTime the users were registered multiple times on Asterisk box, so 1001 which was initially registered on Server A got registered on Server B as well. Also, I wasn’t able to figure out a way to send a call from Server A to a particular context on Server B. It also caused a bit of scalability issues as well since if using dedicated trunks then I’d have to provide each server’s details in sip.conf of all servers. So when a new server is added, the sip.conf would need to be updated on all the servers, which isn’t that elegant (because then sip reload would also be called).

Here’s the content of the “iax.conf” file where [iaxuser] is added. It’s the same on each server

[general]
jitterbuffer=no
nochecksums=no
disallow=all
allow=gsm

[iaxuser]
type=friend
dbsecret=dundi/secret
context=incomingdundi

So, as we see, there are three contexts we’ve encountered till now. Context “internal”, which is associated with the SIP user being registered, context “dundiextens” which is auto-created as soon as the first user registers on a particular server, and lastly context “incomingdundi” – this is used when a call arrives via the IAX trunk. All these have to be mentioned in extensions.conf of each server.

If you notice, there’s a parameter “dbsecret” in the file. I guess it allows one server to authenticate to another by using a username/password combination. The advantage is that it’s just a username/password authentication and not any server information hard coded. Which has the best advantage that if any new server is added to the cluster, it’ll automatically be authenticated. Also, I had read somewhere, probably in the links below, that “dundi/secret” value for “dbsecret” is to be taken literally – you shouldn’t be changing it.

If you want to read about using IAX and SIP trunks in DUNDi, you can refer to the following

• http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+IAX
• http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP (this doesn’t work yet)
• http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords

DUNDi key generation

Before we proceed further, we will need to create a couple of keys on one server, and COPY it to another server. I haven’t tried creating different keys, so cannot confirm otherwise.

On Server A, go to /var/lib/asterisk/keys directory, and execute the following

astgenkey -n dundi

and when it asks for pressing ENTER, just do that. A couple of files would be created in that directory – dundi.key & dundi.pub. Copy both the files to Server B in the same directory – /var/lib/asterisk/keys. This essentially means that dundi.key and dundi.pub are identical on both the servers, and would remain identical as new servers join the cluster.

Server A : /etc/asterisk/dundi.conf

dundi.conf has three primary sections, [general], [mapping] and peer information to which it can place queries. Here’s what needs to be in dundi.conf

; Part 1
[general]
department=Your Department
organization=Your Company, Inc.
locality=Your City
stateprov=ST
country=US
email=your@email.com
phone=+12565551212

bindaddr=192.168.1.5
port=4520
entityid=00:16:36:99:54:62
cachetime=5
ttl=2
autokill=yes

; Part 2
[mappings]
priv => dundiextens,0,IAX2,iaxuser:${SECRET}@192.168.1.5/${NUMBER},nopartial

; Part 3
[00:16:36:2f:11:9c] ; MAC Address of Server B
model = symmetric
host = 192.168.1.6 ;(this is the IP address of the DUNDi Server B)
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

Let’s move to each item step by step

Part 1 – The [general] section is self explanatory. Following points should be looked into

• bindaddr=192.168.1.5 <- The IP Address of your existing machine
• entityid=00:16:36:99:54:62 <- The MAC address of the interface for 192.168.1.5. However, it could be any mac address that you own, that is, it should be something unique.
• cachetime=5 <- Better to have it at 5 seconds than 3600 seconds if it’s a closed group network. If there’s a failure of one server, the re-requests shall be redirected to other servers after 5 seconds have elapsed. May not be good if you’re connecting to outside servers not in your domain as it has to potential for frequent queries if there’s heavy traffic.
• ttl=2 <- The number of DUNDi lookup requests made. So, if request goes to Server B from Server A, then that’s one request. Now Server B can make a maximum of 1 more request to other servers it has paired with.

Part 2 – The [mapping] section defines what you will provide as the contact address when another server gets in touch with you. So, if a SIP extension is registered with you, you will provide the server asking for that extension a complete path to your machine. If we break the following statement,

priv => dundiextens,0,IAX2,iaxuser:${SECRET}@192.168.1.5/${NUMBER},nopartial

As per the documentation, here “priv” is the DUNDi context and will be used by DUNDi. We’ll see later how it’s used in extensions.conf file. Then comes the local context which it’ll search. In our case it’s “dundiextens” which we defined in sip.conf under the regcontext parameter. Then is the weight, the technology being used, the destination that is to be sent to the server requesting information, and then further options. You can review DUNDi documentation at http://www.voip-info.org/wiki/view/DUNDi for more details on each of the parameters.

Part 3 – These are the peers that you’ll register with for looking up / doing DUNDi searching. So, if you have multiple servers, then you’ll need to have multiple peer definitions similar to the one mentioned earlier. Since we’re using only one Server (Server B) as the peer, we’ve entered that information in dundi.conf. I think the MAC addresses are more for consistency than any technical verification, so you should be able to connect machines outside your network too – across the globe maybe.

Server B : /etc/asterisk/dundi.conf

In continuation to the concept above, we’ll add the dundi.conf for Server B. Here’s an example

; Part 1
[general]
department=Your Department
organization=Your Company, Inc.
locality=Your City
stateprov=ST
country=US
email=your@email.com
phone=+12565551212

bindaddr=192.168.1.6
port=4520
entityid=00:16:36:2f:11:9c
cachetime=5
ttl=2
autokill=yes

; Part 2
[mappings]
priv => dundiextens,0,IAX2,iaxuser:${SECRET}@192.168.1.6/${NUMBER},nopartial

; Part 3
[00:16:36:99:54:62] ; MAC Address of Server A
model = symmetric
host = 192.168.1.5 ;(this is the IP address of the DUNDi Server A)
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

Our DUNDi configuration is complete. Now Server A has Server B as DUNDi peer, and vice versa. This means if Server A queries Server B, then Server B would respond favorably and provide the location to the SIP extension if it exists on that server (and of course vice versa).

Server A : /etc/asterisk/extensions.conf

Now, for DUNDi we’ll need to work up a bit more. Following is the extensions.conf for each of the contexts mentioned above, that is, internal, dundiextens, and incomingdundi

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[lookupdundi]
switch => DUNDi/priv

[internal]
exten => 123,1,Playback(demo-congrats)
exten => 123,n,Hangup()

include => lookupdundi

exten => _XXXX,2,Playback(demo-congrats)
exten => _XXXX,n,Hangup()

[incomingdundi]
exten => _XXXX,1,Goto(internal,${EXTEN},2)

We’ve created a new context called [lookupdundi]. This performs the actual lookup, and if you notice, it does search the “priv” context mentioned in dundi.conf file.

The [internal] context still has the extension 123 on which demo-congrats is being played. However, we have included “lookupdundi” context in there, and further to that created a pattern of 4 digits starting from Priority 2. Priority 2 is very important here. Why? Because when a SIP Phone is registered, due to the regcontext being specified in sip.conf, a Priority 1 with NoOp is already created, which you can see by typing “dialplan show dundiextens” on the asterisk prompt after the SIP phone is registered. So, when the phone will reach the “internal” context, it’ll look for Priority 2, and that’s what we need to provide.

The last context, [incomingdundi] is called via the IAX Trunk that we created (called “iaxuser” in sip.conf). This transfers the request to “internal” context on Priority 2.

So, it signifies that, primarily the “internal” context is the one where in the primary dialplan would be written related to what will happen to a call once it arrives. At this stage, we are not doing anything except playing a congratulations message.

Server B : /etc/asterisk/extensions.conf

There’s not much change for Server B, except having a different extension 456 instead of 123.

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[lookupdundi]
switch => DUNDi/priv

[internal]
exten => 456,1,Playback(demo-congrats)
exten => 456,n,Hangup()

include => lookupdundi

exten => _XXXX,2,Playback(demo-congrats)
exten => _XXXX,n,Hangup()

[incomingdundi]
exten => _XXXX,1,Goto(internal,${EXTEN},2)

Testing the setup

So here we are, our DUNDi configuration with two servers is complete, and I just hope it works without issues for you. Here is how we’ll test it.

1. Restart Asterisk and start the console on both the servers

   /etc/init.d/asterisk restart
   asterisk -r
   

2. Use Zoiper and register to both Server A and Server B. You should see appropriate registration messages in the asterisk console of both the servers.
3. Now we will see if on both servers the “dundiextens” has been created and if the users 1001 and 1002 are part of it.

   # On Server A
   CLI> dialplan show dundiextens
   [ Context 'dundiextens' created by 'SIP' ]
     '1001' =>         1. Noop(1001)                                 [SIP]
   
   -= 1 extension (1 priority) in 1 context. =-
   CLI>
   
   # On Server B
   CLI> dialplan show dundiextens
   [ Context 'dundiextens' created by 'SIP' ]
     '1002' =>         1. Noop(1002)                                 [SIP]
   
   -= 1 extension (1 priority) in 1 context. =-
   CLI>
   

4. All cool. Both our phones are registered and as you see the first priority of NoOp is created in there. Next is to see if the peers are displayed correctly. Server A would show Server B as its peer, and vice versa.

   # Server A : Listing DUNDi peers
   CLI> dundi show peers
   EID                  Host                Model          AvgTime  Status
   00:16:36:2f:11:9c    192.168.1.6         (S) Symmetric  Unavail  OK (1 ms)
   1 dundi peers [1 online, 0 offline, 0 unmonitored]
   CLI>
   
   # Server B : Listing DUNDi peers
   CLI> dundi show peers
   EID                  Host                Model          AvgTime  Status
   00:16:36:99:54:62    192.168.1.5         (S) Symmetric  Unavail  OK (1 ms)
   1 dundi peers [1 online, 0 offline, 0 unmonitored]
   CLI>
   

5. Next we will see if what happens if a lookup request is made by DUNDi from Server A to find 1001 (which is on Server A), and 1002 (which is on Server B). As you will see, Server A will not give a result of the extensions registered on its own instance, but only those which are registered on other servers. So, 1001 when searched doesn’t retrieve any info, but 1002 does.

   CLI> dundi lookup 1001@priv
   DUNDi lookup returned no results.
   DUNDi lookup completed in 7 ms
   CLI> dundi lookup 1002@priv
     1.     0 IAX2/iaxuser:5FGlTXaaJQ+YDTZ0xmbNdQ==@192.168.1.6/1002 (EXISTS)
        from 00:16:36:2f:11:9c, expires in 5 s
   DUNDi lookup completed in 8 ms
   CLI>
   

6. Doing the above on Server B gives the results as expected. 1002 returns no results as it is on the same machine, but 1001 does.

   CLI> dundi lookup 1001@priv
     1.     0 IAX2/iaxuser:tgz3cZ1U22WtapJcibkpFA==@192.168.1.5/1001 (EXISTS)
        from 00:16:36:99:54:62, expires in 5 s
   DUNDi lookup completed in 6 ms
   CLI> dundi lookup 1002@priv
   DUNDi lookup returned no results.
   DUNDi lookup completed in 8 ms
   CLI>
   

7. Now let’s place a call from Server A selected in Zoiper to 1001. It shouldn’t play anything. Then place a call to 1002 – it should play a congratulations message, and there should be some information displayed on Server B’s asterisk console.

If it works for you, then it means that the DUNDi configuration is working well. If it doesn’t, then it would show some messages at the console of what didn’t work. You can try enabling sip debugging through

sip set debug on

More DUNDi – 3 Server Setup

We’ll now add Server C to our existing cluster. This will be our DUNDi lookup server and no user would register to it.

JR Richardson mentioned in his document of having a separate DUNDi lookup server which will have all information about the peers. The advantages that it would have is

• There would be only one peer entered in all registration servers. So if a new registration server is added to the cluster, the other registration servers needn’t have the DUNDi entry in them.
• It will be easier to manage since the new registration server would be added into the DUNDi configuration of Server C only, and all other servers would know of it instantly.

Let’s move to the configuration part on each server. I am assuming that you’ve done the basic installation of Asterisk on Server C. There’s no need for setting up sip.conf, iax.conf or extensions.conf on Server C, only “dundi.conf” is needed.

Do the following before proceeding so that you don’t encounter weird behavior

• Unregister the SIP connections in Zoiper
• Stop Asterisk on all three servers via

  /etc/init.d/asterisk stop
  

Server A : /etc/asterisk/dundi.conf

We’ll remove Server B’s entry from dundi.conf and add Server C’s entry. Here’s the final dundi.conf on Server A

; Part 1
[general]
department=Your Department
organization=Your Company, Inc.
locality=Your City
stateprov=ST
country=US
email=your@email.com
phone=+12565551212

bindaddr=192.168.1.5
port=4520
entityid=00:16:36:99:54:62
cachetime=5
ttl=2
autokill=yes

; Part 2
[mappings]
priv => dundiextens,0,IAX2,iaxuser:${SECRET}@192.168.1.5/${NUMBER},nopartial

; Part 3
[00:16:36:6e:d7:dc] ; MAC Address of Server C
model = symmetric
host = 192.168.1.10 ;(this is the IP address of the DUNDi Server C)
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

Only the Peer’s MAC Address and host are changed in Part 3 of the configuration. So we have added Server C as a peer having IP Address of 192.168.1.10.

Server B : /etc/asterisk/dundi.conf

The configuration is almost similar to Server A except of course the bindaddress and entityid.

; Part 1
[general]
department=Your Department
organization=Your Company, Inc.
locality=Your City
stateprov=ST
country=US
email=your@email.com
phone=+12565551212

bindaddr=192.168.1.6
port=4520
entityid=00:16:36:2f:11:9c
cachetime=5
ttl=2
autokill=yes

; Part 2
[mappings]
priv => dundiextens,0,IAX2,iaxuser:${SECRET}@192.168.1.6/${NUMBER},nopartial

; Part 3
[00:16:36:6e:d7:dc] ; MAC Address of Server C
model = symmetric
host = 192.168.1.10 ;(this is the IP address of the DUNDi Server C)
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

Server C : /etc/asterisk/dundi.conf

Remember, you should have copied dundi.key and dundi.pub from either Server A or Server B to Server C’s /var/lib/asterisk/keys. The two said files should be identical on all the machines. The DUNDi configuration will have information on the peers to which it will connect. Since it will connect to all asterisk registration servers, hence the peers would be Server A and Server B.

Here’s the configuration

; Part 1
[general]
department=Your Department
organization=Your Company, Inc.
locality=Your City
stateprov=ST
country=US
email=your@email.com
phone=+12565551212

bindaddr=192.168.1.10
port=4520
entityid=00:16:36:6e:d7:dc
cachetime=5
ttl=1 ; Notice the TTL. You wouldn't want this server to loop, but only ask its peers ONCE
autokill=yes

; Part 2
[mappings]
priv => 

; Part 3
[00:16:36:99:54:62] ; Peer 1 : MAC Address of Server A
model = symmetric
host = 192.168.1.5 ;(this is the IP address of the Server A)
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

[00:16:36:2f:11:9c] ; Peer 2 : MAC Address of Server B
model = symmetric
host = 192.168.1.6 ;(this is the IP address of the Server B)
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary

A few details on above. The “ttl” is 1 instead of 2. Since our lookup server (Server C) is peering with all the registration servers, it should ask them only once and the registration servers shouldn’t forward the request to other servers. ttl=1 ensures that. If it had been ttl=2, then registration servers (Server A & B would query their peers – and their peer is only 1, Server C – and thus it would get into an unwanted loop.

There’s an empty “priv => ” parameter. It is there because [mapping] somehow requires atleast one parameter, even if it’s empty.

Peer 1 and Peer 2 are added. If you add another server to it, then you just need to add an entry in this file only and make it Peer 3 and so on.

That’s all to it. Let’s test this out.

Testing the 3 Server Setup

In our test we have two servers, Server A and Server B on which the SIP phones would register. Server C would act just as a lookup server and answer requests.

1. Start Asterisk on all the three servers and connect to their consoles.

   /etc/init.d/asterisk start
   asterisk -r
   

2. Now use the command “dundi show peers”. It should show only 1 peer on Server A & B, and 2 peers on Server C. This is just fine.
3. Register your SIP Phone on Server A & B, so both extensions 1001 & 1002 should be registered.
4. Now you can either do “dundi lookup 1001@priv” or “dundi lookup 1002@priv” from Server C and it’ll show you both. Doing “1002@priv” from Server A or “1001@priv” from Server B would also give you results which are retrieved from the peer (Server C).
5. Make a call now from Server A selected in Zoiper to extension 1002 (on Server B).

If you’ve been able to do a 2 server setup, this type of setup should be trivial for you. Remember, dundi.key and dundi.pub should be identical on all three servers, so make sure you do that.

Adding RealTime to it

Yeah, having users hard coded in sip.conf file is so so so uncool! Let’s have them in database. Of course, for this to work you should have asterisk-addons installed on Server A and Server B. Server C is just the lookup server, so we don’t need anything there as far as asterisk is concerned.

Setting up MySQL Server

You need to have some understanding of MySQL to continue. On a Debian/Ubuntu machine you can install MySQL via

apt-get install mysql-server

For our setup, we will install it on Server C as it is sitting idle in any case, doing just the lookups. You can have a dedicated machine for it depending on your requirements. Also, as JR Richardson had mentioned in one of the documents, asterisk realtime does a lot of database activity, so it is rather good to have it on other server and it should go via a separate network interface which is not being used for VOIP. Since usually all servers have atleast two ethernet cards, it should be trivial. Think about it and implement as per your requirements.

After installation, comment out “bind-address” in /etc/mysql/my.cnf file and then restart mysql through /etc/init.d/mysql restart. This is what I did on the mysql server after connecting to it

mysql> create database asterisk;
mysql> grant all privileges on asterisk.* to userA@192.168.1.5 identified by 'secretA';
mysql> grant all privileges on asterisk.* to userB@192.168.1.6 identified by 'secretB';

I separated the user so that if in future I need to identify which user is doing what, then it’s possible. Otherwise you can have a single user, it’s much more convenient that way, specially when you add more servers. You can also use ‘%’ (quotes intended) instead of IP Address so that anyone can connect to it.

Next import the database structure in which you’ll store the users. You can copy paste the following directly into mysql prompt, or save it as a file and import it later. This is copied directly from asterisk source and is in the contrib/scripts/sip-friends.sql file. A minor change you’d need is that “fullcontact” column should have a varchar(128), otherwise asterisk would complain.

CREATE TABLE `sipfriends` (
  `name` varchar(40) NOT NULL default '',
  `type` varchar(10) NOT NULL default '',
  `defaultuser` varchar(40),
  `fromuser` varchar(40),
  `fromdomain` varchar(40),
  `secret` varchar(40),
  `md5secret` varchar(40),
  `auth` varchar(10),
  `mailbox` varchar(20),
  `subscribemwi` varchar(10), -- yes/no
  `vmexten` varchar(20),
  `callerid` varchar(40),
  `cid_number` varchar(40),
  `callingpres` varchar(20),
  `usereqphone` varchar(10),
  `language` varchar(10),
  `incominglimit` varchar(10),
  `context` varchar(40) NOT NULL default '',
  `subscribecontext` varchar(40),
  `amaflags` varchar(20),
  `accountcode` varchar(20),
  `musicclass` varchar(20),
  `mohsuggest` varchar(20),
  `allowtransfer` varchar(20),
  `callgroup` varchar(20),
  `pickupgroup` varchar(20),
  `autoframing` varchar(10), -- yes/no
  `disallow` varchar(20) default 'all',
  `allow` varchar(20),
  `maxcallbitrate` varchar(15),
  `host` varchar(40) default 'dynamic',
  `outboundproxy` varchar(40),
  `ipaddr` varchar(20) NOT NULL default '',
  `defaultip` varchar(20),
  `port` int(6) NOT NULL default '0',
  `fullcontact` varchar(128),
  `insecure` varchar(20),
  `qualify` varchar(15),
  `regseconds` int(11) NOT NULL default '0',
  `regexten` varchar(20),
  `regserver` varchar(20),
  `rtptimeout` varchar(15),
  `rtpholdtimeout` varchar(15),
  `rtpkeepalive` varchar(15),
  `lastms` int(11) NOT NULL default '-1',
  `setvar` varchar(200),
  `useragent` varchar(50),
  PRIMARY KEY  (`name`),
  INDEX host (host, port),
  INDEX ipaddr (ipaddr, port)
) TYPE=MyISAM;

Last step – add two users into it, which will be our SIP users.

mysql> insert into sipfriends set name='1001',type='friend',context='internal',host='dynamic',disallow='all',allow='gsm',qualify='yes';
mysql> insert into sipfriends set name='1002',type='friend',context='internal',host='dynamic',disallow='all',allow='gsm',qualify='yes';

Let’s move on to other configurations.

Server A : /etc/asterisk/res_mysql.conf

The file contains the database details

[general]
dbhost = 192.168.1.10
dbname = asterisk
dbuser = userA
dbpass = secretA
dbport = 3306
requirements=warn

Server B : /etc/asterisk/res_mysql.conf

If you didn’t configure two separate users in MySQL, then Server A’s values could be used. But since in my setup each server has a separate user, therefore the following is needed

[general]
dbhost = 192.168.1.10
dbname = asterisk
dbuser = userB
dbpass = secretB
dbport = 3306
requirements=warn

Server A & B : /etc/asterisk/extconfig.conf

Here’s what should be there in the file on both the servers

[settings]
sipusers => mysql,general,sipfriends
sippeers => mysql,general,sipfriends

Now we’ll remove the users in sip.conf files on both the servers (1001 & 1002). You may want to comment them out by prefixing a semi-conon (;).

Server A & B : /etc/asterisk/sip.conf

We’ll just remove the sip user entries (of 1001 & 1002) in sip.conf file as it’s no longer needed. And we’ll add one parameter “rtcachefriends” in the general context so that users registered in real time are cached by sip and thus qualify=yes works for them. You may omit it as per your requirements.

[general]
regcontext=dundiextens
srvlookup=no
NAT=yes
bindport=5060
allowguest=no
canreinvite=no
rtcachefriends=yes

Testing RealTime Setup

1. Unregister the phones from Zoiper.
2. Restart asterisk and then open the consoles of Server A & Server B.

   /etc/init.d/asterisk restart
   asterisk -r
   

3. On Server C’s mysql prompt, if you enter “show processlist”, you will two persistent connections from both the servers. All’s good to go.
4. Now try registering the phones. It should register. Server A’s asterisk console should print out something similar to the following

   CLI> sip show peers
   Name/username              Host            Dyn Nat ACL Port     Status       Realtime
   1001/1001                  xx.xx.xx.xx     D   N       5060     OK (122 ms)  Cached RT
   priv                       192.168.1.6         N       5060     Unmonitored
   2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline]
   CLI>
   

5. Let’s try calling extension 1001 now from Server B selected in Zoiper.
6. It should playback the congratulations message that we’d defined in extensions.conf.

What’s next

Enabling actual calling out

Till now we had just used playback. If you intend to actually call out the users, which of course you’d want to do, then do the following.

/etc/asterisk/extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no

[lookupdundi]
switch => DUNDi/priv

[internal]
exten => 123,1,Playback(demo-congrats)
exten => 123,n,Hangup()

include => dundiextens

exten => _XXXX,2,NoOp(internal context called on Server A to ${EXTEN})
exten => _XXXX,n,Dial(SIP/${EXTEN},10)
exten => _XXXX,n,Hangup()

include => lookupdundi

[incomingdundi]
exten => _XXXX,1,NoOp(Landed on Server A ${EXTEN})
exten => _XXXX,n,Goto(internal,${EXTEN},2)

If you notice, I’ve included “dundiextens” in the “internal” context. This would allow you to call users who are registered on the same server as yours. If they’re not registered, then “lookupdundi” would be called. The SIP extension if found would ring for 10 seconds unless answered, and then hangup would be called. This is very very basic, but it should give you an idea where to start.

Using extensions in RealTime

If you plan to use extensions also in RealTime instead of hard coding them, such as for IVR, Voicemail etc. Then the following should work. I would prefer that you don’t put the user extension information in there, but rather use it as a replacement for extension 456 or extension 123 that we created above.

In database, the table structure would be

CREATE TABLE `extensions` (
  `id` int(11) NOT NULL AUTO_INCREMENT,
  `context` varchar(20) NOT NULL DEFAULT '',
  `exten` varchar(20) NOT NULL DEFAULT '',
  `priority` tinyint(4) NOT NULL DEFAULT '0',
  `app` varchar(20) NOT NULL DEFAULT '',
  `appdata` varchar(128) NOT NULL DEFAULT '',
  PRIMARY KEY (`context`,`exten`,`priority`),
  KEY `id` (`id`)
) ENGINE=MyISAM;

In /etc/asterisk/extconfig.conf you will add

extensions => mysql,general,extensions

In /etc/asterisk/extensions.conf, in “internal” context you will add

switch => Realtime

As an example, let’s insert an Echo() application as extension 100. Enter the following information in the database

mysql> insert into extensions set context='internal',exten='100',priority='1',app='Playback',appdata='demo-echotest';
mysql> insert into extensions set context='internal',exten='100',priority='2',app='Echo',appdata='';

Restart asterisk and then try dialing out extension 100 from either Server A or B. One thing to note is that we are restarting only because we’ve made significant changes in the configuration. The best part of RealTime is that you don’t need to restart any of the services if new SIP users are added, or new extensions are added.

Securing the setup

1. Secure the SIP users. In the database, put some value for “secret” column.
2. Put firewall on your servers. You need to open ports for SIP, for RTP and DUNDi. Here’s a sample configuration

   # For IAX
   $IPTABLES -A INPUT -p tcp --dport 4569 -j ACCEPT
   $IPTABLES -A INPUT -p udp --dport 4569 -j ACCEPT
   # For DUNDi
   $IPTABLES -A INPUT -p tcp --dport 4520 -j ACCEPT
   $IPTABLES -A INPUT -p udp --dport 4520 -j ACCEPT
   # For SIP
   $IPTABLES -A INPUT -p udp --dport 5060 -j ACCEPT
   $IPTABLES -A INPUT -p udp --dport 5070 -j ACCEPT # In case bindport is different
   # For SIP RTP
   $IPTABLES -A INPUT -p udp --dport 10000:20000 -m udp -j ACCEPT
   
   # In case bindport is different in sip.conf then you can do a bit of port forwarding
   $IPTABLES -A PREROUTING -t nat -p udp --dport 5060 -j DNAT --to xxx.xxx.xxx.xxx:5070
   

Do load balancing & fail over

• Round Robin DNS for poor man’s load balancing. This means having two (or more) “A” records for your domain. The softphone would connect randomly to any. No failure can be detected in this scenario, but still you can maintain uptime by using linux-ha (high availability)
• DNS SRV for load balancing as well as failover. Zoiper supports DNS SRV and it’s not that difficult to implement. All you need is a few SRV records in your DNS Server, and that’s pretty easy via GoDaddy’s DNS. You can assign weight to each server as well as port numbers, and the softphone would automatically connect to another server in case the one it’s connected with goes down.
• GeoDNS in case you are having users spread across the globe. Have multiple registration servers spread all over the world and use GeoDNS to make the user register to the ones closest to them.
• Have multiple dundi lookup servers in case dundi lookup server itself goes down. Recommended is a High Availability scenario (ha – http://linux-ha.org/)
• MySQL Cluster setup for database. So you’ll have high uptime.
• Master Slave/Replication configuration of MySQL for better performance. JR Richardson mentioned somewhere that we can have multiple MySQL servers running – the data is read from one database and written to another. This way we can have faster and localized reads in case the servers are geographically apart.

Hope this serves a good purpose for you. Any comments or questions are welcome.

The possibilities are endless, and I personally feel that this is the way future applications would be developed. What I thought Java Web Start could have achieved (had it been a bit faster and the plugin etc. didn’t involve huge download and cumbersome installation), now I see that in Adobe Flex.

Regardless, I spent last few months working on a Flex application, and it has now culminated into what I’ve named as Board800 (http://www.board800.com). It is an interactive multi-user whiteboard application, on the lines of Dabbleboard, though much simpler. You can use it to collaborate with other users, and also *purchase* it if you wish to deploy it on your own servers. Go check it out now! http://www.board800.com

http://thewikireader.com/

I have always been fond of encyclopedias – when I was a kid, I loved the World Books, then when I grew up and got familiar with computers, had heard of Microsoft Encarta – but I think since childhood I wasn’t a big fan of MS, so went ahead and bought IBM Worldbook CDs. It was amazing, and helped much in the school projects.

Now, at this date, I’ll admit I am getting a lot paranoid. That is more related paranoia associated with 2012, End of the World and what not. I have always believed in stuff which is WITH YOU and in your reach WHEN YOU WANT. Internet is something which scares me – a lot of stuff I “need”, which is there, but as soon as you get offline, all’s gone (well, this scare has prompted me to setup 3 different ISP connections at my home, apart from the mobile wireless internet). I don’t want that. I had been attempting to understand and planning that I”ll download the entire Wikipedia some day, and set up on my machine so that I can access it offline. Well, it would have been cumbersome and real painful. I couldn’t believe when I read about WikiReader, and I realized that’s the device I need (originally I’d been reading about Openmoko and FreeRunner). So, I found out the dealers in India (IDA Systems), and immediately placed the order.

I have been so happy with the purchase that now I can concentrate on the other survival tactics for EoW I don’t think anyone wouldn’t want knowledge at their fingertips, in such a convenient and cheap manner. If you don’t have WikiReader, then I think you’re missing out on a lot of stuff that you could’ve learnt.

MUST SEE! – check out the “catch” while raising your angle of vision