For the first time I felt that this is more than a phone, for the first time calender integration works, for the first time I can use the builtin GPS, for the first time accessing the web with a phone works, for the first time downloaded software really works.

My favorite apps so far is:

• Nagroid, to be able to view my op5 Monitor status
• FONMaps, find hotspots for LaFoneras
• Car Cast, listen and download podcasts
• MapDroid, to use preloaded OpenStreetmaps and GPS without using any bandwidth. Perfect when abroad because of the crazy price of data roaming outside Sweden.
• HTC Facebook, read and post on facebook
• FON Access, automatically connect to FON hotspots when traveling.
• GPS Logger, logg tracks in gpx format that almost any GPS software understand
• Sipdroid, to connect to my Asterisk PBX using 3G or WiFi

Of  cource the device is not perfect, I miss:

• The phone must be ‘rooted’ to be real useful, why? Open the phone so the community and others can develop apps that are real useful. Vendor lock-in always sucks.
• IPSec VPN so I can connect to my IPSec based OpenBSD firewall. The IPSec implementation in the phone sucks.
• OpenVPN, there exists OpenVPN apps but to use the the phone must be ‘rooted’
• Bluetooth modem, it is not possible to use the phone as a modem using bluetooth
• Screenshots, the phone must be rooted or using the SDK to take screenshoots. Why?

I really hope that Google and/or HTC understand and use the power of the community to make the Android even more succesfull by open it even more.

I have read the book Asterisk now by Nir Simionovich, published in March 2008. The book was a big disappointment, the reason is that the book do not cover the software used in AsteriskNOW today. The book cover the Asterisk GUI but AsteriskNOW is using FreePBX instead which is totally different. One main reason to use AsteriskNow is to avoid the sometimes cumbersome task to install Linux or a similair operating system, download, compile and configure Asterisk using cryptic text files. So an accurate description of the GUI used is essential for a book like this and unfortunatly the book is to old. I do not intend to install an old version of AsteriskNOW just for a bookreview so I cannot tell how accurate the book is. So my recommendation is to wait for an updated version of AsteriskNow book.

Links

• A sample chapter Chapter-7-For-Annoyance-Press-1-Voice-Menus-and-IVR
• Link to the book AsteriskNow

I registered on their homepage and after a couple of days I got a snail mail with my credentials.

I configured my FreePBX and calling in worked directly, but not outgoing. After 2 hours of troubleshooting I started to google "phonzo asterisk" and found several people that has the same experience. The reason is that Phonzo does not accepted "Asterisk PBX" as useragent and that is default in Asterisk.

After changing sip.conf

[general]

...

useragent=it-slav PBX

....

It worked!

I do not like unlogical stupidity so I sent an email to the support and the following bizare mail conversation occoured:

I will update this post when new info arrives.

I have finished reading the great book "FreePBX 2.5 Powerful Telephony Solutions", by Alex Robar and it is 277 pages.

The intended target audience for this book are system administrators who want to get started with Asterisk and FreePBX.  It is perfect for administrators who want to reduce costs by replacing a proprietary PBX with a PBX that runs on open source packages. The pre required knowledge is basic knowledge of Linux and telephony, though neither is strictly required.

I find the targeted audience and the pre required knowledge correct, however I think the Linux knowledge is an absolute demand or at least have someone to ask. A basic knowledge of Telephony is also more or less required.

The book cover installation on CentOS and Ubuntu Server, configuration, adding handset, trunk setup, call routing, voicemail, digital receptionist, music-on-hold, call recording, maintenance, backups and much more.

I find the book very valuable and I really like the concept of FreePBX. When starting with Asterisk the myriads of parameters are overwhelming. FreePBX makes the approach to Asterisk easier and makes the startup much faster then go to the Asterisk config via textfiles. FreePBX comes with many modules that should be enough for most demands and if needed it is still possible to use configuration using config files.

The book describes the most important modules in FreePBX in a structured and pedagogic way with many screenshoots and figures. I really likes the examples in the end of the book with trunk configuration to some big SIP providers. Even if none of them is valid in Sweden it gives you an idea what parameters to start with when configure trunks.

When it comes to trouble shooting I think the book should have contained a chapter about that. In many cases the way of trouble shoot is to use the Asterisk CLI, read logs and in some cases use a network analyser like WireShark. I think the book should have contained an introduction to trouble shooting and the different tools available.

Some of the script examples contain some typos but with basic scripting knowledge it is easy to fix.

I reccomend this book for anyone interested in FreePBX, FreePBX admins and Asterisk newbies. This book makes the start with FreePBX much faster compared to  trying to collect the information from the web and other sources. The FreePBX website is a great as a reference, however this book makes it easy and fast to get started with FreePBX and Asterisk.

Links:

• More info about  the book "FreePBX 2.5 Powerful Telephony Solutions"
• Example chapter 8 Recording calls
• FreePBX homepage
• Asterisk homepage

I have read the excellent book Asterisk 1.6, Build feature-rich telephony systems with Asterisk by David Merel, Barrie Dempster and David Gomillion.

The book is inteended to anyone interested in bulding a telephony system using Asterisk and are 224 pages. The book claims that no preknowledge about opensource, Linux and Asterisk is required. I think it is a little bit hard to start without any of these knowledge. I would recommend to have basic Linux/Unix knowledge to be able to go through the installation part.

It starts from the begining with introduction to Asterisk, brief telephony system introduction, installing Asterisk,configure Asterisk, creating dialplans. It continues with call logging, different pre-made Asterisk alternatives like trixbox/FreePBX, asterCRM and Case studies and ends with some hints that makes the life of an Asterisk admin easier.

I relly liked the book. When starting with no Asterisk knowledge the number of config files and parameter is overvelming and this book help the reader to penetrate that. The book walk through basic configuration and help to get your a Asterisk installation up and running. I like that it gives an overview of different things to consider when to start an Asterisk implementation and builds up a basic knowledge about the subject.

I appreciate the case studies where three different scenarios are described: Small office/home office, Small business and a hosted PBX.

The drawback of the book is that it just gives an introduction to the different topics. I would like to have a little bit deeper knowledge in some of the topics, for an example dialplans.

I would recommend the book to newbies to Asterisk, the book will help them to get started and boost their knowledge. It is possible to gain all the knowledge in the book by visiting different Asterisk forums, read documentation and so on but that will take much effort compared to read the book.

• To buy the book: Asterisk 1.6, Build feature-rich telephony systems with Asterisk

I have been running an asterisk system to handle my VoIP calles the last years and I have been using it for a number of things during the time.

I also use firewalls to secure incoming and outgoing IP traffic in my home network, so I thought the asterisk system could be used to do handle a “blacklist” of telephone number on my PSTN.

I would also like the asterisk system to kick in before my normal telephone gets the call and starts ringing.

If it detects a telemarketing caller, I would like it to answer the call and give a prerecorded message saying something like “You are calling from a unauthorised source. Please do not call again, this telephone line is being monitored.” and then just drop the line.

I found a list of known Swedish CallerID numbers on the Internet that I can use to “jumpstart” my blacklist. I only needed to have a dialplan and to configure my Sipura SPA-3000 ATA device to send my PSTN calls to my asterisk system.

This is how I configured it:

I started with configuring my SPA-3000 box to pick up the CallerID from my PSTN company.

The important configuration parameters are on the PSTN Tab in the Sipura web gui.

Line Enable: yes

Proxy: <my asterisk server IP>

User ID: pstn

Password: pstn

PSTN-To-VoIP Gateway Enable: yes

PSTN Ring Thru Line 1: no

PSTN Caller Default DP: 1

Line 1 Signal Hook Flash to PSTN: Disabled

Dial Plan 1: S0<:1010@gw1>

The Sipura dialplan is (as usual) pretty complicated (and powerful) and this dialplan sends the call to extension “1010” at the “gw1” interface in the SPA box.

I created a new context for the incoming PSTN, called [blacklist].

I also needed to create a sip device configuration for the Sipura box, so that the call would enter the correct context in the dialplan:

[pstn]

type=friend

host=dynamic ;the extension can register from a dynamic IP

context=pstn ;incoming calls to to this extension

secret=pstn

dtmfmode=rfc2833

disallow=all ;disallow all codecs

allow=ulaw ;allow only the ulaw codec

deny=0.0.0.0/0.0.0.0 ; block all registrations from other networks

permit=10.0.0.0/255.255.255.0 ;only allow registrations from my LAN

qualify=600 ;monitor the ping delay

If you are using another PSTN-To-VoIP gateway adapter, the box needs to be configured to send the incoming CallerID to the asterisk box before answering the call.

The list of telephone numbers I downloaded needed to be inserted into a database in the asterisk system. Because there is about 1600 telephone numbers in the list, I had a script to load it into the asterisk database from a unix shell.

The list can be updated in realtime by using the commands:

ast> database put blacklist <number> 1 (for blacklisting)

ast> database del blacklist <number> 0 (for removing blacklisting)

You can run these commands on the asterisk console from the shell using the form:

# asterisk -r “databse put blacklist <number> 1″

To check if the list is populated, you can use the command:

ast> database list

It will give you the list of blacklisted number.

To make the blacklist matching work, I neede to insert a few lines of code into my /etc/asterisk/extensions.conf file:

[pstn]

exten => 1010,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1);

[blacklisted]

exten => s,1,Answer

exten => s,n,Wait(2)

exten => s,n,Playback(ss-noservice)

exten => s,n,Wait(1)

exten => s,n,Hangup

So, if a call is detected in the SPA3K box, it is sent to the “1010” extension in the pstn context. The CallerID is compared to the blacklisted entries in the asterisk database.

If the CallerID number is found in the databse, the call is sent to the blacklist extension and the “s” extension kicks in and handles the call.

The call is of course logged in the CDR database so I can monitor the number of telemarketing callers by day and week. But that is a different story.

Now, my family is happy to not answering those stupid calls every day!

Theory

In my implementation of Asterisk I have a couple of softphones, 2 hardphones and connections to two SIP providers. I want to monitor the following:

• Possibility for a phone to be able to register at the Asterisk server
• The registration at the SIP providers are OK
• The Operating system is not overloaded
• The server where Asterisk is running is up

Implementation

Sip check

I found the following after Googling: http://bashton.com/osprojects/nagiosplugins/ Define it in commands.cfg

# command 'check_sip'
define command{
    command_name                   check_sip
    command_line                   $USER1$/custom/nagios-check_sip-1.2/check_sip -u "$ARG1$"
    }

Define the sip check in services.cfg, I also created a service group called ip_telephony

# service 'Asterisk Check SIP'
define service{
    use                            default-service
    host_name                      dull
    service_description            Asterisk Check SIP
    check_command                  check_sip!sip:XXXXX@dull.mynet
    servicegroups                  ip_telephony
    contact_groups                 it-slav_msn,it-slav_mail,call_it-slav
    }

Monitor the Peers

First I tried to find a suitable plugin at Nagiosexchange, Google and other sites if there was anybody that has created and published a Nagios plugin to monitor the sip peers. I could not find any. With the asterisk command "sip show peers", information about the connected sip peers can be found:

[root@dull custom]# asterisk -rx "sip show peers"
Name/username              Host            Dyn Nat ACL Port     Status
pulver                     69.90.155.70                5060     OK (154 ms)
digisip/XXXXX              82.209.165.194              5060     OK (44 ms)
6016                       (Unspecified)    D          0        UNKNOWN
6005                       (Unspecified)    D          0        UNKNOWN
6004/6004                  10.1.1.168       D          5060     OK (139 ms)
6003/6003                  10.1.1.168       D          5060     OK (136 ms)
6002/6002                  10.1.1.152       D          5060     OK (8 ms)
6011                       (Unspecified)    D          0        UNKNOWN
6010                       (Unspecified)    D          0        UNKNOWN
6000                       (Unspecified)    D          0        UNKNOWN
6001                       (Unspecified)    D          0        UNKNOWN
11 sip peers [Monitored: 5 online, 6 offline Unmonitored: 0 online, 0 offline]

I discussed this issue with Marcus Rejås when we meet and he told me that he has written a Nagios plugin to monitor the sip peers. I got the script and modified to fit my needs, i.e. get performance data for graphing: /opt/plugins/custom/check_asterisk_sip_peers.sh

#!/bin/bash
#
# Simple Asterisk Peer Check.
# Copyright (C) 2008 Marcus Rejås / Rejås Datakonsult
#
# Modified with perfdata by Peter Andersson
# http://www.it-slav.net/blogs/?p=123
# peter@it-slav.net
#
# This program is free software: you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation, either version 3 of the License, or
# (at your option) any later version.
#
# This program is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
# GNU General Public License for more details.
#
#
# Very simple plugin that checks if a peer is ok. The peers needs "qualify=yes"
# in its configuration.
#
# A peer that is not registered or non-existent will result in error. If the
# peer is OK a short statusline (from Asterisk) is written. There is timing
# information suitable for graphing as well.
#
# You should have received a copy of the GNU General Public License
# along with this program.  If not, see .
#
# Example use of this script:
#
# sip:~# ./sip_check_peer mysecretary-100
# mysecretary-100/461762501 62.80.200.53 5060 OK (10 ms)
# sip:~#
#
#

if [ $# == 0 -o "$1" == "-h" -o  $# -gt 1 ]; then
	echo "Usage: $0"
	exit 3
fi

LINE=`asterisk -r -x "sip show peers" | grep $1 | grep "OK ("`

#
# This is a uggly. Just to check that the expression above does not match more
# then one line.
#
HITS=`asterisk -r -x "sip show peers" | grep $1 | grep "OK (" | wc -l`

if [ $HITS -gt 1 ]; then
	echo "ERROR: Multiple match, tweak your arguments or fix $0  "
	exit 3
fi

if [ "$LINE" ]; then
	echo -n "OK: "
	echo -n $LINE
	#Create perdata
	echo -n "|time="
	echo $LINE | awk '{gsub(/\(/,"")};{gsub(/\)/,"")};{print $(NF-1)$NF}'
	exit 0
elif [ -z "$LINE" ]; then
	echo "CRITICAL: Something is wrong with $1";
	exit 2
else
	echo $LINE
	exit 2
fi

The command run by hand looks like this:

[root@dull /]# /opt/plugins/custom/check_asterisk_sip_peers.sh pulver
OK: pulver 69.90.155.70 5060 OK (166 ms)|time=166ms

The plugin is started by nrpe at the asterisk server and configured in /etc/nrpe.d/mycommands.cfg

command[asterisk_peer_digisip]=sudo /opt/plugins/custom/check_asterisk_sip_peers.sh digisip
command[asterisk_peer_pulver]=sudo /opt/plugins/custom/check_asterisk_sip_peers.sh pulver
command[asterisk_peer_6002]=sudo /opt/plugins/custom/check_asterisk_sip_peers.sh 6002
command[asterisk_peer_6003]=sudo /opt/plugins/custom/check_asterisk_sip_peers.sh 6003
command[asterisk_peer_6004]=sudo /opt/plugins/custom/check_asterisk_sip_peers.sh 6004

/opt/plugins/custom/check_asterisk_sip_peers.sh must run as a high privileged user so I’m using sudo, modify /etc/sudoers with visudo:

# visudo -f /etc/sudoers

--snip--
nobody ALL= (root) NOPASSWD: /opt/plugins/custom/check_asterisk_sip_peers.sh
--snip--
Turn of requiretty, because it will run without a console
--snip--
#Defaults    requiretty
--snip--

The service checks are defined in services.cfg and also put into the same servicegroup

# service 'Asterisk Peer Pulver'
define service{
    use                            default-service
    host_name                      dull
    service_description            Asterisk Peer Pulver
    check_command                  check_nrpe!asterisk_peer_pulver
    servicegroups                  ip_telephony
    contact_groups                 it-slav_sms,it-slav_mail,call_it-slav
    }

I also defined the other peers, i.e. digisip, 6002, 6003, 6004

The servicegroup is defined in servicegroups.cfg

# servicegroup 'ip_telephony'
define servicegroup{
    servicegroup_name              ip_telephony
    alias                          IP - Telephony
    }

This is the result, a screenshot from my op5 Monitor system:

I leave to the reader to define the standard checks for load, ping and total processes. If you use op5 Monitor or pnp4nagios the graphs are automatically created:

Links

• Nagios
• op5 Monitor
• pnp4nagios
• Asterisk
• Marcus Rejås

I’ve configured op5 Monitor to call me when it detects a problem, to listen to an example click on link below:

11880larm1

The text below is op5 Monitor centric. op5 Monitor is based on Nagios so the principle is the same the only difference in this case is the possibility to test it from the op5 Monitor webconfig GUI.

Here is how I did it.

1. Have a working asterisk with connected phones
2. Create a notify script on the op5 Monitor Server or Nagios server

Add the following to misccommands.cfg
# command ‘host-notify-call’
define command{
command_name host-notify-call
command_line sudo -u monitor /opt/monitor/op5/notify/call_asterisk_message.sh ‘A\ message\ from\ Op5\ monitor.\ Host\ “$HOSTNAME$”\ is\ “$HOSTSTATE$”‘
}
and
# command ’service-notify-call’
define command{
command_name service-notify-call
command_line sudo -u monitor /opt/monitor/op5/notify/call_asterisk_message.sh ‘A\ message\ from\ Op5\ monitor.\ “$SERVICESTATE$”.\ Service\ “$SERVICEDESC$”
\ on\ host\ “$HOSTNAME$”\ is\ $SERVICESTATE$’
}

Comment: sudo to monitor is probably unnecessary but I had som problems with ‘test this service’ in op5 Monitor webconfig GUI because it runs as apache and when the script is run when a problem occour it runs as monitor.

3. call_asterisk_message.sh

ssh dull sudo /root/scripts/make_call3.sh $1

i.e. run make_call3.sh with first argument, dull is my asterisk server.

4. On dull /root/scripts/make_call3.sh

#!/bin/sh
#Creates a phone call via asterisk to a certian number and reads a message

#By Peter Andersson, peter@it-slav.net

echo “$1″ | /usr/bin/text2wave -scale 1.5 -F 8000 -o /tmp/$$larm.wav
echo “$1″ >/tmp/$$larm.txt
cat <<EOF > /tmp/$$monitorcall.call
Channel: SIP/6000
Callerid: Op5 Monitor
MaxRetries: 5
RetryTime: 60
WaitTime: 60
Application: Playback
Data: /tmp/$$larm
EOF
mv /tmp/$$monitorcall.call /var/spool/asterisk/outgoing
#rm /tmp/$$larm.wav
echo “rm /tmp/$$larm.*” |at now + 1 day #Removes the wav and txt file tomorrow

Comment: 6000 is the phonenumber to call, in my case a softphone.

And now it should work.