Goautodial Install from Scratch CentOS 7.

 

Install Goautodial From Scratch (using CentOS 7)

 Installing the GOautodial app (v4) on a CentOS 7.0 based server.

Update system

yum update -y
yum install -y nano wget
yum groupinstall -y "Development Tools" 

Install the Goautodial yum repository

cd /etc/yum.repos.d/
wget http://downloads2.goautodial.org/centos/7/goautodial.repo

Install MariaDB 10 PHP 7, Asterisk 13, Kamailio 5, RTPengine and other dependencies

yum install MariaDB-server MariaDB-devel php70w-mysql php70w-mcrypt php70w-devel 
php70w-mbstring php70w-common php70w-xml php70w-pear php70w-cli php70w-imap php70w-fpm 
php70w-gd php70w-opcache php70w-pdo php70w-process php70w php70w-intl php70w-pear.noarch 
php70w-xmlrpc asterisk-mysql-13.17.2-vici.el7.centos.x86_64 asterisk-perl-0.08-2.go.x86_64 
asterisk-voicemail-plain-13.17.2-vici.el7.centos.x86_64 asterisk-devel-13.17.2-vici.el7.centos.x86_64 
asterisk-voicemail-13.17.2-vici.el7.centos.x86_64 asterisk-alsa-13.17.2-vici.el7.centos.x86_64 
asterisk-sip-13.17.2-vici.el7.centos.x86_64 asterisk-13.17.2-vici.el7.centos.x86_64 
asterisk-dahdi-13.17.2-vici.el7.centos.x86_64 asterisk-iax2-13.17.2-vici.el7.centos.x86_64 
asterisk-mp3-13.17.2-vici.el7.centos.x86_64 kamailio-tls kamailio kamailio-mysql kamailio-ims kamailio-utils 
kamailio-websocket  kamailio-json perl-Math-Round perl-Net-Server perl-File-Touch perl-Sys-RunAlone 
perl-Switch perl-Time-Local ngcp-rtpengine ngcp-rtpengine-kernel ngcp-rtpengine-dkms dkms 
dahdi-linux dahdi-linux-devel kernel-devel perl-Crypt-Eksblowfish perl-DBI perl-DBD-mysql perl-Net-Telnet lame 
httpd mod_ssl screen crontabs mailx net-tools glibc.i686

Enable EPEL repository

yum install -y epel-release
yum install perl-Crypt-Eksblowfish perl-Sys-RunAlone vim

Edit /etc/yum.conf:

vim /etc/yum.conf

Append to the bottom:

exclude=dahdi-tools*

Note: there's an asterisk (*) at the end of the line above. It's not a typo error.

This will avoid dahdi-tools package conflict with the EPEL repo.

yum update

Enable at system startup and start the service

systemctl enable php-fpm 
systemctl enable httpd
systemctl enable mariadb
systemctl enable kamailio
systemctl enable ngcp-rtpengine

Disable firewalld (IMPORTANT!)

systemctl stop firewalld
systemctl disable firewalld

We're using Iptables (customize /etc/sysconfig/iptables if needed)

Create missing Kamailio runtime directory

mkdir /var/run/kamailio
chown kamailio /var/run/kamailio

Install GOautodial

yum -y install goautodial-ce
cd /usr/src/goautodial
./install.sh

Install CPAN

yum install -y cpan
cpan install Net::Server
cpan install Asterisk::AGI

Configure RTPengine

vim /etc/rtpengine/rtpengine.conf

Change 123.234.345.456 to your public IP address

### a single interface:
interface = 123.234.345.456

Configure Kamailio

vim /etc/kamailio/kamailio.cfg

Change 10.10.100.19 to your public IP address

/* add local domain aliases */
alias="10.10.100.19" 

#!substdef "!MY_IP_ADDR!10.10.100.19!g" 

listen=udp:10.10.100.19:5060

Update GOautodial web application

cd /var/www/html
git config --global user.email "root@localhost" 
git stash
git pull

cd /var/www/html/goAPIv2
git stash
git pull

Reboot your server (very important!)

reboot

Edit /var/www/html/php/Config.php

<?php
// database configuration
define('DB_USERNAME', 'goautodialu');
define('DB_PASSWORD', 'goautodialu1234');
define('DB_HOST', 'localhost');
define('DB_NAME', 'goautodial');
define('DB_PORT', '3306');
define('DB_NAME_ASTERISK', 'asterisk');
define('DB_USERNAME_KAMAILIO', 'kamailiou');
define('DB_PASSWORD_KAMAILIO', 'kamailiou1234');
define('DB_HOST_KAMAILIO', 'localhost');
define('DB_NAME_KAMAILIO', 'kamailio');
define('DB_PORT_KAMAILIO', '3306');

// other configuration parameters
define('CRM_ADMIN_EMAIL', 'admin@localhost.com');
?>

SSL Configuration

yum install -y certbot
certbot certonly -d pbx.pstncall.com
cd /etc/letsencrypt/live/pbx.pstncall.com/
cp privkey.pem /etc/pki/tls/private/localhost.key
cp fullchain.pem /etc/pki/tls/certs/localhost.crt

service httpd restart

Access your GOautodial application (Google Chrome is recommended)

https://pbx.pstncall.com

User: goadmin
Pass: G0autodial2018

Notes

Kamailio default port 5060/UDP

Asterisk default port 5070/UDP

Please use the community forum boards (https://goautodial.org/projects/goautodialce/boards) for questions and issues.

Vicidial installation step by step

Vicidial Installation Guide

Below is an easy step for Vicidial installation. PSTN CALL. If you face any issue contact us at support@pstncall.com

Open a terminal on the system

$ sudo su

$ passwd (set the root user's password)

“apt-get install apache2 build-essential iftop lame libncurses5-dev libploticus0-dev libsox-fmt-all linux-source mpg123 mytop  ntp openssh-server php php-cli php-dev php-mysql phpmyadmin ploticus screen sipsak sox  subversion subversion-tools unzip

removed linux-headers mysql-client-5.0 mysql-doc-5.0  mysql-server-5.0 mtop libmysqlclient15-dev apache2-mpm-prefork

password given test123

apt install mysql-server

Go to terminal:

$ cd /usr/src

$ tar -xjf linux-source-*.tar.bz2 (where * is the kernel version)

$ cpan

(press enter to go through the prompts. If you have a multi cored system you should enter the -j option when specified with n+1 as the value, where n is the number of CPUs you have in your system. Also enter UNINST=1 when asked. until you get to the mirror selection portion)

(select 3 mirror sites in your area)

> install MD5

> install Digest::SHA1

> install readline

> install Bundle::CPAN   (do not change settings)

> quit

$ cpan   (enter through questions until you get to the cpan prompt)

> o conf commit (saves the config changes)

> force install Scalar::Util

> install DBI

> force install DBD::mysql

> install Net::Server

> install Time::HiRes

> install Net::Telnet

> install Unicode::Map

> install Jcode

> install OLE::Storage_Lite

> install Spreadsheet::WriteExcel

> install Proc::ProcessTable

> install Spreadsheet::ParseExcel

> install Mail::Sendmail

> quit

cd /usr/src

wget http://download.vicidial.com/packages/asterisk-perl-0.08.tar.gz

tar xzf asterisk-perl-0.08.tar.gz

cd asterisk-perl-0.08

perl Makefile.PL

make all

make install

cd /usr/src

wget http://www.daveltd.com/src/util/ttyload/ttyload-0.5.tar.gz

tar xzf ttyload-0.5.tar.gz

cd ttyload-0.5

make

make install

vim /etc/php/7.2/apache2/php.ini and

;opcache.enable=0 to opcache.enable=0

mkdir /usr/src/asterisk

cd /usr/src/asterisk

****FOR 1.2 asterisk run the following

wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.30.2.tar.gz

wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.27.tar.gz 

wget http://ftp.digium.com/pub/libpri/releases/libpri-1.2.8.tar.gz

tar xzf asterisk-1.2.30.2.tar.gz

tar xzf zaptel-1.2.27.tar.gz 

tar xzf libpri-1.2.8.tar.gz

cd libpri-1.2.8

make clean; make; make install

cd ../zaptel-1.2.27

make clean; make; make install

cd ../asterisk-1.2.30.2

cd apps

wget http://www.eflo.net/files/app_amd2.c

mv app_amd2.c app_amd.c

$ vi Makefile

 replace this line(line 32):

 app_mixmonitor.so app_stack.so

 with this line:

 app_mixmonitor.so app_stack.so app_amd.so

wget http://www.eflo.net/files/amd2.conf

mkdir /etc/asterisk

mv amd2.conf /etc/asterisk/amd.conf

wget http://www.eflo.net/files/meetme_DTMF_passthru-1.2.23.patch

patch -p1 < ./meetme_DTMF_passthru-1.2.23.patch

 File to patch: app_meetme.c

wget http://www.eflo.net/files/meetme_volume_control_1.2.16.patch

patch -p1 < ./meetme_volume_control_1.2.16.patch

 File to patch: app_meetme.c

cd ../

wget http://www.eflo.net/files/cli_chan_concise_delimiter.patch

patch -p1 < ./cli_chan_concise_delimiter.patch

   File to patch: cli.c

$ wget http://www.eflo.net/files/app_waitforsilence.c

$ mv app_waitforsilence.c apps/app_waitforsilence.c

$ wget http://www.eflo.net/files/enter.h

$ wget http://www.eflo.net/files/leave.h

$ mv -f enter.h apps/enter.h

$ mv -f leave.h apps/leave.h

$ vi codecs/gsm/Makefile

   add “OPTIMIZE=-O2” to the file  before the ifneq section, to fix GSM audio problems

$ make clean; make; make installation

$ make samples

$ modprobe zaptel

$ modprobe ztdummy

****FOR 1.4 asterisk do the following:

wget http://downloads.digium.com/pub/asterisk/old-releases/asterisk-1.4.21.2.tar.gz

wget http://downloads.digium.com/pub/zaptel/zaptel-1.4.12.1.tar.gz

wget http://downloads.digium.com/pub/libpri/libpri-1.4.9.tar.gz

tar xzf asterisk-1.4.21.2.tar.gz

tar xzf zaptel-1.4.12.1.tar.gz 

tar xzf libpri-1.4.9.tar.gz

cd libpri-1.4.9

make clean; make; make install

cd ../zaptel-1.4.12.1

./configure; make clean; make; make install

cd ../asterisk-1.4.21.2

$ wget http://www.eflo.net/files/enter.h

$ wget http://www.eflo.net/files/leave.h

$ mv -f enter.h apps/enter.h

$ mv -f leave.h apps/leave.h

$ vi codecs/gsm/Makefile

   add “OPTIMIZE=-O2” to the file  before the ifneq section, to fix GSM audio problems

./configure; make clean; make; make install

make samples

modprobe zaptel

modprobe ztdummy

$ asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvgc      (to see if Asterisk runs)

> show version

> zap show status

> show application meetme

> stop now

*** for asterisk 1.2

cd /var/lib/asterisk/mohmp3/

mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-sunshine.mp3 > /var/lib/asterisk/mohmp3/fpm-sunshine.raw

sox -r 44100 -w -s -c 1 fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav

sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm

sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm

mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-calm-river.mp3 > /var/lib/asterisk/mohmp3/fpm-calm-river.raw

sox -r 44100 -w -s -c 1 fpm-calm-river.raw -r 8000 -c 1 fpm-calm-river.wav

sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm

sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm

mpg123 -s --rate 44100 --mono /var/lib/asterisk/mohmp3/fpm-world-mix.mp3 > /var/lib/asterisk/mohmp3/fpm-world-mix.raw

sox -r 44100 -w -s -c 1 fpm-world-mix.raw -r 8000 -c 1 fpm-world-mix.wav

sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm

sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm

mkdir ../orig-mp3

mv -f *.mp3 ../orig-mp3/

mkdir ../quiet-mp3

cd ../quiet-mp3

sox -r 44100 -w -s -c 1 ../mohmp3/fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav vol 0.25

sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm

sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm

sox -r 44100 -w -s -c 1 ../mohmp3/fpm-calm-river.raw -r 8000 -c 1 fpm-calm-river.wav vol 0.25

sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm

sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm

sox -r 44100 -w -s -c 1 ../mohmp3/fpm-world-mix.raw -r 8000 -c 1 fpm-world-mix.wav vol 0.25

sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm

sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm

rm -f ../mohmp3/*.raw

**** for asterisk 1.4

cp -a /var/lib/asterisk/moh /var/lib/asterisk/mohmp3/

cd /var/lib/asterisk/mohmp3/

sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm

sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm

sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm

sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm

sox -r 44100 -w -s -c 1 fpm-world-mix.raw -r 8000 -c 1 fpm-world-mix.wav

sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm

sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm

mkdir ../quiet-mp3

cd ../quiet-mp3

sox ../mohmp3/fpm-sunshine.wav -r 8000 -c 1 fpm-sunshine.wav vol 0.25

sox fpm-sunshine.wav -t gsm -r 8000 -b -c 1 fpm-sunshine.gsm

sox fpm-sunshine.wav -t ul -r 8000 -b -c 1 fpm-sunshine.pcm

sox ../mohmp3/fpm-calm-river.wav -r 8000 -c 1 fpm-calm-river.wav vol 0.25

sox fpm-calm-river.wav -t gsm -r 8000 -b -c 1 fpm-calm-river.gsm

sox fpm-calm-river.wav -t ul -r 8000 -b -c 1 fpm-calm-river.pcm

sox ../mohmp3/fpm-world-mix.wav -r 8000 -c 1 fpm-world-mix.wav vol 0.25

sox fpm-world-mix.wav -t gsm -r 8000 -b -c 1 fpm-world-mix.gsm

sox fpm-world-mix.wav -t ul -r 8000 -b -c 1 fpm-world-mix.pcm

**** for SVN 2.0.4 branch:

$ mkdir /usr/src/astguiclient

$ cd /usr/src/astguiclient

$ svn checkout svn://svn.eflo.net:43690/agc_2-X/branches/agc_2.0.4

$ cd agc_2.0.4

$ perl install.pl

**** for SVN 2.0 trunk:

$ mkdir /usr/src/astguiclient

$ cd /usr/src/astguiclient

$ svn checkout svn://svn.eflo.net:43690/agc_2-X/trunk

$ cd trunk

$ perl install.pl

 : manual configuration [y]

 : press enter until you get to webroot and set that to the following: /var/www

 : press enter through to the “Sample configuration files” and set that to 'y'

 : press enter through to the end of the script

 : add the highlighted lines to the top of your [default] context:

$ vi /etc/asterisk/extensions.conf

$ cd /var/lib/asterisk/sounds

$ wget http://downloads.vicidial.com/sounds/conf.gsm

$ cp conf.gsm park.gsm

$ mysql

> CREATE DATABASE `asterisk` DEFAULT CHARACTER SET utf8 COLLATE utf8_unicode_ci;

> GRANT SELECT,INSERT,UPDATE,DELETE,LOCK TABLES on asterisk.* TO cron@'%' IDENTIFIED BY '1234';

> GRANT SELECT,INSERT,UPDATE,DELETE,LOCK TABLES on asterisk.* TO cron@localhost IDENTIFIED BY '1234';

> use asterisk;

> \. /usr/src/astguiclient/trunk/extras/MySQL_AST_CREATE_tables.sql

> \. /usr/src/astguiclient/trunk/extras/first_server_install.sql

> \. /usr/src/extras/sip-iax_phones.sql

> quit

$ /usr/share/astguiclient/ADMIN_update_server_ip.pl –old-server_ip=10.10.10.15

$ /usr/share/astguiclient/ADMIN_area_code_populate.pl

$ cp /usr/src/astguiclient/trunk/extras/performance_test_leads.txt /usr/share/astguiclient/LEADS_IN/

$ /usr/share/astguiclient/VICIDIAL_IN_new_leads_file.pl --forcelistid=107 –forcephonecode=1

$ crontab -e

  Add the following lines:

### recording mixing/compressing/ftping scripts

0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl

#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl

1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl –MP3

#2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --MP3

### keepalive script for astguiclient processes

* * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl

### kill Hangup script for Asterisk updaters

* * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl

### updater for voicemail

* * * * * /usr/share/astguiclient/AST_vm_update.pl

### updater for conference validator

* * * * * /usr/share/astguiclient/AST_conf_update.pl

### flush queue DB table every hour for entries older than 1 hour

11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q

### fix the vicidial_agent_log once every hour

33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl

### updater for VICIDIAL hopper

* * * * * /usr/share/astguiclient/AST_VDhopper.pl -q

### adjust the GMT offset for the leads in the vicidial_list table

1 1,7 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug --postal-code-gmt

### reset several temporary-info tables in the database

2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl

### optimize the database tables within the asterisk database

3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl

## adjust time on the server with ntp

30 * * * * /usr/local/bin/ntpdate -u pool.ntp.org 2>/dev/null 1>&2

### VICIDIAL agent time log weekly summary report generation

2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl

### remove old recordings more than 7 days old

# 24 0 * * * /usr/bin/find /var/spool/asterisk/monitor -maxdepth 2 -type f -mtime +7 -print | xargs rm -f

### remove old vicidial logs and asterisk logs more than 2 days old

28 0 * * * /usr/bin/find /var/log/astguiclient -maxdepth 1 -type f -mtime +2 -print | xargs rm -f

29 0 * * * /usr/bin/find /var/log/asterisk -maxdepth 3 -type f -mtime +2 -print | xargs rm -f

cd /etc/init.d/

cp /usr/src/astguiclient/trunk/extras/init.d/vicidial ./

chmod 0755 vicidial

cd /usr/bin

ln -s /bin/grep grep

cd ../rc2.d

ln -s ../init.d/vicidial S35vicidial

$ vim /etc/apache2/sites-available/default                  (add the following lines)

 Alias /RECORDINGS/ "/var/spool/asterisk/monitorDONE/"

 <Directory "/var/spool/asterisk/monitorDONE">

 Options Indexes MultiViews

 AllowOverride None

 Order allow,deny

 Allow from all

 <files *.mp3>

 Forcetype application/forcedownload

 </files>

 </Directory>

$ chmod 0777 /var/spool/asterisk/monitorDONE/

$ /etc/init.d/apache2 restart

$ vi /etc/fstab             (add the following line to the end of the file)

tmpfs /var/spool/asterisk/monitor tmpfs rw  0 0

$ shutdown -r 0

$ screen -ls          (should show at least 6 screens, one of which should be asterisk)

In a web browser, go to (http://YOUR_SERVER_IP_ADDRESS/vicidial/admin.php) to see if everything is working. You should also reboot at this point to make sure everything will start back up properly.

From here on you should follow the tutorials in the VICIDIAL Manager Manual(available at eflo.net)

/var/www/html/

/usr/src/astguiclient/trunk

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Think of a Public Switched Telephone Network (PSTN) as a combination of telephone networks used worldwide, including telephone lines, fiber optic cables, switching centers, cellular networks, as well as satellites and cable systems. These help telephones communicate with each other.

Put simply, when you dial a phone number your call moves through the network to reach its destination – and two phones get connected. To fully understand how it actually works, consider what happens when you dial a number from your own phone.

Step #1 – Your telephone set converts sound waves into electrical signals. These signals are then transmitted to a terminal via a cable.

Step #2 – The terminal collects the electrical signals and transmits these to the central office (CO).

Step #3 – The central office routes the calls in the form of electrical signals through fiber optic cable. The fiber optic conduit then carries these signals in the form of light pulses to their final destination.

Step #4 – Your call is routed to a tandem office (a regional hub responsible for transmitting calls to distant central offices) or a central office (for local calls).

Step #5 – When your call reaches the right office, the signal is converted back to an electrical signal and is then routed to a terminal.

Step #6 – The terminal routes the call to the appropriate telephone number. Upon receiving the call, the telephone set converts the electrical signals back to sound waves.

This may sound complicated, but the thing to remember is that it takes a few seconds for your call to reach its destination. This process is facilitated by using fiber optic cables and a global network of switching centers.

Now, let’s have a look at each of the four types of switching which take place at different levels.

1. The Local Exchange

A local exchange – which may consist of one or more exchanges – hooks up subscribers to a PSTN line. Also known as a central office or a switching exchange, a telephone exchange may have as many as 10,000 lines.

All telephones are connected to the local exchange in a specific area. Interestingly, if you were to dial the number of your supplier located in the building next to yours, the call won’t leave your local exchange and will be routed to the supplier as soon as it reaches the exchange.

The exchange then identifies the number dialed so it can route the call towards the correct end destination. This process works as follows:

The first three digits of a phone number represent the exchange (the local switch), while the last four digits identify the individual subscriber within that exchange.

This means that when you dial a number and it reaches your local exchange, your call is immediately linked to the subscriber without the need for any further routing.

2. The Tandem Office

Also known as a junction network, a tandem office serves a large geographical area comprising several local exchanges while managing switches between local exchanges.

Let’s say you dialed the number of a client who lives in the same city but in another suburb. In this case your call will be routed to a tandem office from your local exchange, and the tandem office will route the signal on to the local exchange near your client’s location.

3. The Toll Office

This is where any national long-distance switching takes place.

A toll office is connected to all the tandem offices. For instance, if you have an office in another city you’ll find that, whenever you dial that branch’s number, your call will be switched through a toll office.

4. The International Gateway

International gateways manage international call switching, routing domestic calls to the appropriate countries.

PSTN – How Much Does It Cost?

Exactly how much would it cost you if you were to install and use a PSTN phone system in your home or office?

Cost for Consumers

If you require a PSTN phone connection in your home, you’ll need to have a telephone set and a PSTN provider.

Now, a decent phone set can be purchased for less than $60 from Walmart or Amazon, while a basic phone service with unlimited local calls will cost up to $30 per month.

Alternatives to PSTN?

The Plain Old Telephone Service has many robust features, but when it comes to businesses, POTS tends not to be a good fit because choosing this option costs a lot in the long-run (and let’s not forget that these services run on an old technology).

After all, the switching technology itself hasn’t changed much since the last century. This, is a potential drawback of PSTN phone networks as they don’t allow you to transmit other data types.

What’s more, this downside has led to a new and modern telephone service known as VoIP, which is proving to be nothing less than a game-changer in the telephone industry.

The Voice Over Internet Protocol (VoIP) is considered to be the best-known alternative to the PSTN system as it isn’t just cost-effective but also has several other benefits that businesses (and consumers) love.

What is VoIP?

Voice over IP (VoIP) is also known as IP telephony, broadband telephony, or internet telephony—but it means the same thing: your voice transmitted through the internet. The voice signal is converted into a digital signal and it then travels over the internet and reaches the destination.

Bria Configuration

This guide will walk you through configuring the X-Lite softphone to register directly to your PSTN SIPTRUNK! and make a call.

1-X-Lite Setting

•  Download X-Lite from https://www.counterpath.com/x-lite/
•  Once you have X-Lite downloaded and open, navigate to Preferences and then to the Accounts tab.
•  Create a new SIP account and fill in the following fields with the appropriate information: User ID = Your Caller ID , Domain = sbc.pstncall.com and Save it.

2-Whitelist your IP via PSTN portal

•  Login to portal and click on https://portal.pstncall.com/user/user_ipmap/
•  Add your IP details with some name. You can get the computer IP by seaching "what is my ip" on google search.

•  Click on "Map IP" to save the configuration

3-Make Call using X-Lite

•  Dial 91XXXXXXXXXX
•  If you hear conference IVR then Your X-Lite is now ready to make call.

•  If your call does not work then please send mail to suuport@pstncall.com.

Hosted PBX

What is Hosted PBX?

Unlike a traditional PBX, or Private Branch Exchange, which requires a large investment and ongoing maintenance and training, a hosted PBX is a cloud-based PBX system accessible via an IP network. Rather than being responsible for hardware, software, training, maintenance, and more, a hosted PBX provider takes care of it all. In addition to being completely managed off-site, resulting in no IT or installation costs, a hosted PBX system also provides businesses with the ability to manage their phone systems via a user-friendly control panel. For these reasons and more, hosted PBX systems are becoming increasingly popular solutions for today’s growing small to medium-sized business owners.

Features of Hosted PBX

If you own a business with fewer than 300 lines, hosted PBX can provide you with a multitude of features and benefits. For starters, a hosted PBX system is much less expensive than a traditional on-site system. With hosted PBX, there is no need to buy expensive hardware and software, pay for installation, and manage the system. You simply pay a monthly fee and enjoy all of the benefits of the service.

While less costly than standard on-site PBX systems, hosted PBX solutions are teeming with valuable features, such as on-hold music, call waiting, call routing, transfers, and more. Moreover, as the popularity of hosted PBX continues to grow, additional features like auto attendants, extension dialing, and ACD queues are being introduced.

A hosted PBX can also be deployed immediately. In fact, most business owners are able to have their hosted PBX setup and running in under a day. Best of all, newer and more flexible hosted PBX applications are being deployed as well. Adding these applications and features can typically be done with a simple download or click of the mouse.

The ability to add new features as they roll out is just one example of the scalability of hosted PBX. Additional lines, phones, and even an entire new department can effortlessly be added, which would be much more complex with a traditional on-site PBX.

When you opt for a hosted PBX and outsource all the techie stuff, you will find your stress levels lowering as well. With a hosted PBX, a mountain of responsibility will be taken off your shoulders, allowing you to focus your attention on more important matters, like improving your bottom line.

Hosted PBX Tips and Considerations

Hosted PBX systems are incredibly easy to setup and utilize, but there are a few things you need to do when opting for one of these modern business phone solutions, such as:

Make Phone Number Arrangements – A temporary phone number is necessary for the porting process, but if you already have an existing number to port over, you must remain with your current provider until the number is ported to your new PBX provider. If you are getting a new number, it’s important to update others and let them know of the change through an email blast or similar means. While brief, all calls can be conveniently routed to cell phones during the transition period. Just remember, if you cancel your service with your previous carrier prior to porting your number, the number will no longer be yours to keep.

Arrange Your Dial Plan – When switching over to a hosted PBX, organizing the routing of your calls is one of the first things that must be done. In order to do so, you’ll need to define various rules for your calls, including:

• The buttons used to activate voicemail and other features
• The hours of operation for your desk phones
• Setting up your directory
• The handling of faxes
• Sequential and simultaneous ringing
• How off-hour calls are handled
• Configure your emergency 911 settings

Review Your Bill

As you can see, setting up a hosted PBX is remarkably easy. However, after doing so, you should review the bills for the first and second month to ensure you’re receiving and utilizing all of the services you’re paying for. Depending on which company you choose as your hosted PBX provider, the first month’s bill may or may not reflect activation fees, setup fees, and number porting fees. The second month’s bill should be a typical bill without added charges for the setup process. By this point, you’ll be well on your way to enjoying one of the most streamlined and hassle-free business phone systems available for small and medium-sized business owners.

Hosted PBX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.

“Hosted” means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP provider’s data center, thus the term: hosted PBX. Hosted PBX is also sometimes referred to as Hosted VoIP.

Benefits of Hosted PBX

There are many benefits to using Hosted PBX rather than a traditional phone system, or an on-premise PBX. The main benefit is cost- a Hosted PBX system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted PBX system. Purchasing and setting up an on-premise PBX can cost tens of thousands of dollars. Hosted PBX phone systems fall under operational expenditure rather than capital expenditure, which also makes hosted PBX service attractive to businesses. With hosted PBX service, you pay a monthly fee, and the hosted PBX service provider takes care of the rest.

Another benefit to a hosted PBX system over an on-premise PBX is that hosted PBX service providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get a hosted PBX system. A downside to a hosted PBX is you may have a little less of an ability to customize your solution to your business, but many hosted PBX service providers can achieve a deep level of customization.

Hosted PBX Providers

PSTN CALL provides Hosted VoIP solution for all type for usage. We Install all kinds of VoIP PBX at very less cost at any location. Please visit www.pstncall.com for more details

VoIP Termination

What is VoIP Termination?

VoIP call termination is used to refer to the procedures that are used for routing telephone calls from one provider to the next provider until the call has been routed to the last telephone company and has been received by the recipient. Voice termination is another term that is used for call termination. The telephone companies are also referred to as providers or carriers.

Called Party

The called party is the person who has received the telephone call. The endpoint of the route may be on the Internet or may be at a point that was reached by routing the call through the public switched telephone network. The procedures for routing the call stop when the call has been received by the recipient. The process may seem simple to individuals who do not experience problems with making telephone calls but is not so simple to individuals who make calls that will not connect such as calls to a different country.

Calling Party

The calling party is the person who has initiated the call and who wants to be connected to the called party. There may be problems with the telephone calls that begin on the Internet and end at a cellular phone. The sound of the voices may seem to be amplified with echoes and have a scratchy quality. The calls that are initiated with Google Voice may never connect or could be delayed.

VoIP

Voice over Internet Protocol (VoIP) is a term that is used to describe a call that was initiated on the Internet such as Skype calls or calls made with Google Voice. The calls that were initiated on the Internet usually end at a point that is not on the Internet. Most of the recipients of telephone calls receive the calls with a landline phone or with a cellular phone. The route may begin on the Internet but will end at a point that was reached by routing the call from the public switched telephone network (PSTN), which is the common description for call termination.

Internet Networks

A tier-one operator is licensed and registered to operate an Internet protocol (IP) network for Internet telephony services. Tier-one operators can handle call origination and call termination. A tier-two operator can lease services from a tier-one operator. The tier-three operators can lease services from either a tier-one or two-two operator. There are also resellers of VoIP services and wholesalers in the market. The quality of the services is not very high because of the inconsistencies in the market such as fluctuations with demand, fraud and problems with doing business on an international level.

Call Origination

Call origination is used to refer to telephone calls that originate from the public switched telephone network and end the route on the Internet. Call termination is considered to be the opposite of call origination because the direction of the paths is reversed. The terms are associated with the starting point of the calls, path of the route of the calls and termination point of the calls. The operators of IP telephony services can handle calls that originate or terminate on the Internet.

Fees

The fees for services are subject to the regulations of several countries because the routes of the calls will cross over more than one country. Those countries can use legislation to control the fees for the services. Termination rates are usually very high for the countries in the Middle East and Asia. The rates are intentionally high because there are more incoming calls than outgoing calls from the countries, which is caused by the diaspora effect of a migrating population.

VoIP Termination Providers

PSTN Communication offers SIP termination that is reliable, redundant and built for call quality. If you need to urgently get out 1 million calls in an hour, we give you access to 45,000+ channels at your fingertips. We have helped SMBs, Large Call Centers, VoIP carriers, CLECs and we look forward to assisting you! Visit our site below for more information and pricing.

• Flat Rate Termination
• Domestic Conversational Termination
• A-Z Conversational Termination
• Toll-Free Termination
• Retail Origination (DID)
• Toll-Free Origination (18YY)
• Hosted VoIP Billing Solution

For more detail contact at support@pstncall.com. 

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