Wednesday, 26 April 2017

What is VoIP (Voice Over Internet Protocol)?

Over the past few years, the acronym “VoIP” has become a household term. Anyone who
has made an international phone call has probably heard the term used at one point or
another. It has become a popular topic among “techies” discussing its numerous
applications in today’s society, as well as politicians discussing its growth and implications
on the world’s economy. Although most people have heard the term, very few actually
know what VoIP is and what the hype is all about.
VoIP stands for Voice Over Internet Protocol, and it simply means what it implies: sending
voice conversations through the internet. Rather than sending voice through the traditional
telephone networks owned by the telephone conglomerates, VoIP uses the internet as its
medium to send voice. Since voice is nothing more than data, any internet connection in
the world can be used as a VoIP terminal for making calls with the use of a VoIP gateway or
an IP Phone.
A great reference for gaining a better understanding of how Voice Over IP works as opposed
to the traditional telephone network can be found on the How Stuff Works website at the
following URL address:
What are the benefits of choosing VoIP?
Everyday more and more organizations switch their communication networks partially or
fully to VoIP. There are many benefits of choosing VoIP over the traditional telephone
network. The major benefits that these organizations are realizing include cost of
equipment and deployment, versatility, flexibility, interconnectivity, and time to market.
Cost Benefits of Choosing VoIP
· VoIP equipment costs are generally cheaper than traditional telephone switches.
· VoIP in a direct point-to-point environment has no toll charge.
· VoIP technicians are usually more readily available and more cost-effective than
standard telephone switch technicians.
· Billing & Call Management software for VoIP is generally less expensive than similar
software for legacy technology.
 Versatility

VoIP is allows your network to be extremely versatile. VoIP allows you to interconnect with
multiple carriers (routes) very easily and quickly. Within a matter of minutes you can add
additional carriers that will provide you with the lowest route to a particular destination.
There is added flexibility so that if one route is down, your network can automatically reroute
to another carrier.
Interconnectivity
VoIP makes it easy to interconnect to multiple & redundant carriers. With traditional PSTN
networks, interconnecting with multiple carriers required additional expensive hardware,
and an extensive amount of time to setup the connection, sometimes taking several
months. As well, there is a very high connection charge occurred per month to maintain the
connection. With VoIP, the interconnection is very simple, taking only a matter of minutes,
and without the need for additional hardware and no connection charges.
Time to Market
VoIP allows your company to bring its solutions into service quicker than standard telecom.
You can connect to different routes quickly and reliably. This allows you to bring new routes
to customers at cost-effective pricing in a timely manner. You no longer need to wait for
months to connect to new routes.
How does PEC fit into the VoIP Industry?
PEC is a developer of software applications that integrate seamlessly with VoIP technology.
PEC is also a proud partner and reseller of Quintum Technologies, developers of the popular
Tenor VoIP Gateways. Combining the power of Quintum VoIP hardware with the versatility
and robustness of our software, we have developed numerous products to take advantage
of the benefits that VoIP offers. PEC develops the following products for the VoIP industry:
CardSaver™ VoIP Billing and Call Management
· Prepaid Calling Card Industry
· Postpaid Calling Services
· Pinless Calling Services
· Call Shop Services
· Inter-office communications
· Wholesale Long Distance Termination
VoiceSaver™ Voicemail and Auto Attendant
· Voicemail Message Bureaus
· Auto-attendant systems for offices
Insight IVR (Interactive Voice Response)
· Allows customers to develop customized call flows
· Used in almost every industry
Calling Card and Pin-less Solutions
In the prepaid calling card or pin-less calling business, like most other businesses, more
startup companies fail than succeed. Our intention is not to deter you from starting your
own prepaid calling card business; it can be and has been for many people an extremely
lucrative business. Our intention is rather to explain to you the differences between a
successful card business and a failing one so that you may learn from their mistakes.
PEC has a great deal of experience designing and integrating with products developed for
the Prepaid Calling Card and pin-less calling industries. We have helped over 400 clients
deploy VoIP and calling card solutions around the world. We have platforms running on 6
continents and in over 70 countries around the world. Through our day to day
conversations with our clients, we learn a good deal about how the industry is changing and
about new trends that are appearing in the market. If you have any questions, please do
not hesitate to contact us.
Setting Up a Solid Infrastructure
First and foremost, you must develop a solid infrastructure for your prepaid calling card
system to run on. Without a solid infrastructure, you will be spending more time worrying
about your system’s integrity than marketing your product. The following is a list of
equipment and services that you will require to start the business. We have also included
recommendations that we feel will ensure a solid infrastructure.
1. Billing, Card, and Call Management Software
We list this item first not only because it is one of the products that we provide, but because
it is the most vital on the list. Choosing a good billing software will ensure that every call
made through your system is tracked and no call is left unbilled. CardSaver™ by PEC is not
only a robust, dependable software package, but it is also the simplest and most userfriendly
package in the industry. CardSaver enables you to view reports on traffic passing
in and out of your system. CardSaver also gives you complete control over your calling
cards, including PIN generation, usage fees, expiration dates, and much more. It also
allows you to manage your agents/distributors and effectively sell your products through
resellers. CardSaver is a vital component for developing a solid infrastructure.
2. VoIP Gateway
The VoIP gateway that handles all of the calls into and out of your system is the next vital
component. If your hardware fails, your customers’ calls will not go through. 98% of our
customers use either Cisco or Quintum products. Some higher-end customers also use
Nextone gateways. These two vendors have a large following and many online forums exist
where you can go to receive answers to any questions that you may have regarding
configuration or specifications.
3. Server(s)
Server technology is constantly changing, and unless you are in the business, it is a
challenge to stay abreast of the latest and greatest. For this reason, our recommendations
on the right server to choose are based on the current technology available.
Systems with 60 concurrent calls or less:
Rackmount Server
Intel Dual Core Processor
1 GB RAM
(2) 80 GB+ hard drives
10/100/1000 Ethernet Adapter
Systems with 60 to 240 concurrent calls:
Intel Quad-Core Xeon Processor
2 or 4 GB RAM
Dual 10K SCSI hard drives (120 GB or more)
Gigabit Ethernet Adapter
Systems with 240 to 750 concurrent calls:
(2) Intel Quad-Core Xeon Processor
8 GB RAM
Dual 15K SCSI hard drives (250 GB or more)
Gigabit Ethernet Adapter

If you are handling more than 750 calls – we recommend using multiple servers.


VOIP SERVICE PROVIDER FOR BUSINESS

VOIP SERVICE PROVIDER FOR BUSINESS
Established in 2008, ARN Server LLC is a Call Center and Hospitality industry leading VoIP service provider across many countries including United States, Canada, United Kingdom, India, Philippines, Australia and Dominican Republic.

Always aiming to provide top customer support and best telephone service ARN Server LLC offers the lowest worldwide calling rates for services like VoIP for Call Centers, Toll Free Numbers, SIP trunking for call centres and many more. Get a free USA or Canada virtual phone number when you signup for a new account.
CALL CENTRE / DIALER VOIP
ARN Server LLC is the choice VoIP service provider for call centers using predictive dialers, businesses using virtual hosted PBX systems and hotels looking to save money on their hosts calls. Our advanced dialer VoIP service can handle unlimited multiple calls at once without delays with a superb voice quality. Our billing is optimized to be best cost effective in a pay-as-you go manner.

ARN Server LLC call center customers usually call to countries like UK, USA, Canada, India and Australia for telemarketing campaigns with predictive dialers to automate the calls, ARN Server LLC serves call center VoIP service needs and support dialer calling. Contact us to get the best-in-market rates in call center VoIP services.
CARRIER VOIP / SIP TERMINATION
ARN Server LLC provides VoIP SIP service to call centers, hotels and businesses using IP PBX systems. We meet the telecommunication VoIP needs of worldwide call centers across Europe, America, Africa, Australia and Asia for local and long distance calls as well.

We aim to serve call center with the best VoIP quality service to our clients and the best customer support of 24×7.

We value your business and understand your situation, that is the reason why our VoIP services are contract free with no minimum call volume required, no setup fees and no cancellation penalty.

ARN Server LLC also provide services such as:

1-800, 1-866, 1-877 United States, Canada Toll Free Numbers
800 United Kingdom Toll Free Numbers

Free USA and Canada Virtual Phone Numbers on every new VoIP account.

Virtual Hosted Predictive Dialers and IP PBX's

Bulk SMS Messaging Service


SIP TRUNKING FOR CALL CENTERS
ARN Server LLC is a reliable VoIP service provider that offers inbound and outbound SIP trunking to the Call Centre industry since 2008.

What is SIP Trunking? SIPTrunking replaces analog traditional telephone service called Public Switched Telephone Network (PSTN) with a connection provided over the internet through an Internet Service Provider (ISP) normally called SIP provider (Session Initiation Protocol). In VoIP, SIP is the protocol used to send calls to the applications.

ARN Server LLC is includes a highly experienced team of technical engineers to provide the best VoIP service to its clients.
SIP / VOIP TOLL FREE NUMBERS
ARN Server LLC strives to provide the best SIP / VoIP - Toll Free Services to its clients in the call center industry. Our target is to provide the best and most reliable, cost effective VoIP service to all our call center clients to enhance their scalability and productivity. Hotels are also starting to use VoIP as an standard for their long distance calls .

ARN Server LLC Toll Free number services: Our customer calls are routed through a 1-800, 1-866 or 1-877 phone number using the ARN Server LLC switching platform to an IP, DID or a combination of end points. Our billing plan gives the opportunity to call centers to pay only for the minutes they use with no extra charges.

Currently we provide Toll Free Numbers from United States, Canada and United Kingdom. Our International Virtual Phone Number services are provided to our customers accounts so they receive calls from anywhere in the world.
INTERNATIONAL TOLL FREE / LOCAL PHONE NUMBERS
A large number of companies maintain their call center activities outside UK, USA and Canada. For companies with call center operations outside their homeland ARN Server LLC provides International DID/Virtual Phone Number Services, your clients will always get in touch with you by dialing a local number.

ARN Server LLC provides DID numbers and SIP / VoIP Toll Free Numbers across over 32 countries countries. Your customers will interact with your call center agents more productively by calling a local phone number instead of an international phone number.






Asterisk 11 + FreePBX on CentOS 6.5

Step by step instructions for installing software PBX Asterisk 11.x and control panel FreePBX on CentOS 6.5
Conditions:
CentOS installed and has access to the Internet.
Contents

Before install:

1.  # yum -y update
  1. Turn off SELinux:
# vi /etc/selinux/config
3.  # reboot
  1. Additional repos:
5.  # rpm -Uvh http://pkgs.repoforge.org/rpmforge-release/rpmforge-release-0.5.3-1.el6.rf.x86_64.rpm 
6.  # rpm -Uvh http://dl.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm
7.  # yum -y install mc make gcc kernel-devel gcc-c++ ncurses-devel httpd mysql-server mysql mysql-devel php iksemel iksemel-devel crypto-utils openssl-devel gmime gmime-devel spandsp spandsp-devel openldap openldap-devel popt-devel libresample libresample-devel speex speex-devel libcurl-devel speex-tools svn sqlite-devel mod_ssl php-mysql php-pear-MDB2-Driver-mysqli php-mbstring php-gd php-mcrypt libxml2-devel lame bison audiofile-devel sox php-pear-DB php-posix
  1.  
9.  # cd /root
10.# wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz 
11.# tar -zxf asterisk-11-current.tar.gz
12.# wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz
13.# tar -zxf libpri-1.4-current.tar.gzwget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
14.# tar -zxf dahdi-linux-complete-current.tar.gz

Installing DAHDI, libPRI and Asterisk

1.  # cd dahdi-linux-complete-2.9.0+2.9.0.1
2.  #make all
3.  #make install
4.  #make config
5.  # cd libpri-1.4.14
6.  # make
7.  # make install
  1.  
9.  # cd asterisk-11.7.0
10.# ./configure --prefix=/usr --exec-prefix=/usr --with-crypto --with-dahdi --with-iconv -with-libcurl --with-gmime --with-iksemel --with-mysqlclient --disable-xmldoc --with-pri --with-spandsp --with-ldap --with-libcurl --with-popt --with-resample --with-speex
11.# make menuselect
12.# contrib/scripts/get_mp3_source.sh
13.# make
14.# make install
15.# make samples
  1.  
17.# ln -s /usr/lib/libasteriskssl.so.1 /usr/lib64/libasteriskssl.so.1
18.# ln -s /usr/lib/libpri.so.1.4 /usr/lib64/libpri.so.1.4
19.# cat contrib/init.d/rc.redhat.asterisk | sed 's/__ASTERISK_SBIN_DIR__/\/usr\/sbin/' > /etc/init.d/asterisk
20.# chkconfig --add asterisk
21.# cp -rf configs/* /etc/asterisk/
  1.  
23.# service dahdi start
24.# service asterisk start
  1.  

Installing FreePBX

1.   # wget “http://downloads.sourceforge.net/project/phpmyadmin/phpMyAdmin/3.5.5/phpMyAdmin-3.5.5-all-languages.tar.gz?r=http%3A%2F%2Fsourceforge.net%2Fprojects%2Fphpmyadmin%2Ffiles%2FphpMyAdmin%2F3.5.5%2FphpMyAdmin-3.5.5-all-languages.tar.gz%2Fdownload&ts=1358931580&use_mirror=ignum”
2.  # tar -zxf phpMyAdmin-3.5.5-all-languages.tar.gz -C /var/www/
  1. in /etc/hosts:
4.  [IP PBX] pbx-mysql.example.com 
[IP PBX] pbx.example.com
  1. to  /etc/httpd/httpd.conf:
NameVirtualHost *:443
  1. to /etc/httpd/conf.d/ssl.conf:
7.  DocumentRoot "/var/www/phpmyadmin"
8.  ServerName pbx-mysql.example.com:443
9.  ErrorLog logs/ssl_phpmyadmin_error_log
10.TransferLog logs/ssl_phpmyadmin_access_log
11.LogLevel warn
12.SSLEngine on
13. 
14.SSLProtocol all -SSLv2
15.SSLCipherSuite ALL:!ADH:!EXPORT:!SSLv2:RC4+RSA:+HIGH:+MEDIUM:+LOW
16.SSLCertificateFile /etc/pki/tls/certs/localhost.crt
17.SSLCertificateKeyFile /etc/pki/tls/private/localhost.key
18. 
19.SSLOptions +StdEnvVars
20. 
21.<Directory "/var/www/cgi-bin">
22.SSLOptions +StdEnvVars
23. 
24.SetEnvIf User-Agent ".*MSIE.*" \
25.nokeepalive ssl-unclean-shutdown \
26.downgrade-1.0 force-response-1.0
27. 
28.CustomLog logs/ssl_phpmyadmin_request_log \
29."%t %h %{SSL_PROTOCOL}x %{SSL_CIPHER}x \"%r\" %b"
30. 
31.<Directory "/var/www/phpmyadmin">
32.Order allow,deny
33.Allow from [Admin IP]
34. 
35.DocumentRoot "/var/www/html"
36.ServerName pbx.example.com:443
37.ErrorLog logs/ssl_html_error_log
38.TransferLog logs/ssl_html_access_log
39.LogLevel warn
40.SSLEngine on
41. 
42.SSLProtocol all -SSLv2
43.SSLCipherSuite ALL:!ADH:!EXPORT:!SSLv2:RC4+RSA:+HIGH:+MEDIUM:+LOW
44.SSLCertificateFile /etc/pki/tls/certs/localhost.crt
45.SSLCertificateKeyFile /etc/pki/tls/private/localhost.key
46. 
47.SSLOptions +StdEnvVars
48. 
49.<Directory "/var/www/cgi-bin">
50.SSLOptions +StdEnvVars
51. 
52.SetEnvIf User-Agent ".*MSIE.*" \
53.nokeepalive ssl-unclean-shutdown \
54.downgrade-1.0 force-response-1.0
55. 
56.CustomLog logs/ssl_html_request_log \
57."%t %h %{SSL_PROTOCOL}x %{SSL_CIPHER}x \"%r\" %b"
58. 
59.<Directory "/var/www/html">
60.Order allow,deny
61.Allow from [Admin IP]
62.# cd /var/www/
63.# ln -s phpMyAdmin-3.5.5-all-languages phpmyadmin
64.# service httpd restart
65.# chkconfig --levels 345 mysqld on
66.# chkconfig --levels 345 httpd on
67.# service mysqld start
68.# /usr/bin/mysql_secure_installation
69.  Change root password for MySQL
70.# cd /root
71.# wget http://mirror.freepbx.org/freepbx-2.11.0.tar.gz
72.# tar -zxf freepbx-2.11.0.tar.gz -C /usr/src 
73.# cd /usr/src/freepbx-2.11.0
  1. Edit php.ini:
75.# vi /etc/php.ini (or /etc/php4/apache2/php.ini)
76.upload_max_filesize=20M
77.# mysqladmin create asteriskcdrdb -p
78.# mysql -p asteriskcdrdb < SQL/cdr_mysql_table.sql
79.# mysqladmin create asterisk -p
80.# mysql -p asterisk < SQL/newinstall.sql 
81.# mysql -p 
82.mysql> GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY '[your password]';
mysql> GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY '[your password]';
mysql> flush privileges;
83.# groupadd asterisk 
84.# useradd -c "asterisk PBX" -d /var/lib/asterisk -s /sbin/nologin -g asterisk asterisk
  1. Edit:
86.# vi /etc/httpd/conf/httpd.conf (or /etc/apache2/apache2.conf):
87.User asterisk
88.Group asterisk
89.# chown asterisk /var/lib/php/session
90.# ./install_amp
  1. In the database named asterisk for table named freepbx_settings change password for asterisk manager:
92.# mysql -p
mysql> UPDATE `asterisk`.`freepbx_settings` SET `value` = '[your password 2]' WHERE `freepbx_settings`.`keyword` = 'AMPMGRPASS';
93.# chown asterisk /etc/modprobe.d/dahdi.conf
94.# rm /etc/asterisk/ccss.conf /etc/asterisk/logger.conf /etc/asterisk/iax.conf /etc/asterisk/extensions.conf /etc/asterisk/sip_notify.conf /etc/asterisk/features.conf /etc/asterisk/sip.conf
  1. Change something in freepbx admin panel, click Apply (default username/password – admin/admin)
  2. Change default password

Additional installation:

1.   # yum -y install bind bind-utils mailx ntp
2.  # chkconfig --add named
3.  # chkconfig --add ntpdate
4.  # service ntpdate start
5.  # service named start
6.  # echo “nameserver 127.0.0.1” > /etc/resolv.conf
7.  # echo “nameserver 8.8.8.8” >> /etc/resolv.conf
8.  # yum -y install xinetd tftp-server 
9.  # vi /etc/xinetd.d/tftp
10.-s /tftp -c
11.# mkdir /tftp && 
12.# chown nobody /tftp 
13.# vi /etc/sysconfig/iptables 
14.# service xinetd start
  1.  
  2. Add to iptables config:
 -A INPUT -m udp -p udp -d [IP PBX]/32 --dport 5060 -m hashlimit --hashlimit 10/m --hashlimit-burst 10 --hashlimit-mode srcip,dstip,dstport --hashlimit-htable-expire 180 --hashlimit-name SIP -j ACCEPT


How to start a VOIP Business

*Rates shown do not include E-911 charges or government mandated taxes. Providers offering unlimited calling plans may have restrictions. Read providers terms and conditions carefully before buying.
The first thing to do is decide what part of VOIP marketplace you want to serve. Here are some possibilities:
  • VOIP Provider services
  • VOIP consulting
  • Independent Sales/Service Agent for existing VOIP service providers
  • Value Added services with VoIP
  • etc.

Some general suggestions:
  • Pick an area that plays to your strengths. For example, if your strength is sales and marketing, pick an area where you can leverage those abilities
  • Learn all you can about the maketplace
  • Attend industry tradeshows
  • Read industry magazines, blogs, forums, etc
  • Read books
  • Do market research - talk to your potential customers
  • Ask questions
  • Test the waters — to the extent possible try before you buy, test the waters before making large commitments of time or money

Value Added services
If you have experience with VoIP or already in VoIP business, you can get benifit / new customers by introducesing some value added services on VoIP. Few value added services are mentioned in following, Within each service there are many choices.
  • PBX sales and service
    • Hosted PBX
    • Virtual Numbers
    • Hosted IVR / Auto attendents
    • etc

  • Message broadcasting / Call Center Solutions

  • Prepaid Cards
    • Retail prepaid cards from existing wholesale providers
    • Start your own brand of prepaid cards using services from existing wholesale providers
    • Start a new prepaid card provider company
    • Create new software package for prepaid card services
    • Create a Free Phone Booth
    • etc.