vim /etc/apache2/mods-available/mpm_prefork.conf
<IfModule mpm_prefork_module>
StartServers 50
MinSpareServers 25
MaxSpareServers 100
MaxRequestWorkers 500
MaxConnectionsPerChild 0
ServerLimit 500
</IfModule>
service apache2 restart
Updated: March-25-2019
Note: Below steps only cover standalone server installation on Ubuntu 14.04.
I am using Digitalocean VPC. Installation should be similar in AWS EC2 instances.
Make sure to open 8088,8089,80,443 TCP and 10000 -20000 UDP ports Open in your firewall..
Standard Packages:
sudo apt-get install build-essential
apt-get install linux-headers-`uname -r`
apt-get install subversion libjansson-dev sqlite autoconf automake libxml2-dev libncurses5-dev -y
sudo add-apt-repository ppa:ondrej/php
#Press ENTER
sudo apt-key adv --keyserver keyserver.ubuntu.com --recv-keys E5267A6C
#Press ENTER
sudo apt update
sudo apt-get install apache2 apache2-bin apache2-data libapache2-mod-php5.6 php5.6 php5.6-cli php5.6-common php5.6-json php5.6-mysql php5.6-readline sox lame screen libnet-telnet-perl php5.6-mysqli -y
apt-get install libasterisk-agi-perl
apt-get install mysql-server
apt-get install libelf-dev
apt-get install autogen libtool shtool
Special package for ASTblind and ASTloop(ip_relay need this package)
apt-get install libc6-i386
#This Setup is painful, but you need to do it.(will try to add the packages)
[email protected]:~# cpan Loading internal logger. Log::Log4perl recommended for better logging There seems to be running another CPAN process (pid 9731). Contacting... Other job is running. Shall I try to run in downgraded mode? (Y/n) [y] Running in downgraded mode (experimental). Please report if something unexpected happens cpan shell -- CPAN exploration and modules installation (v2.26) nolock_cpan> #install below modules install CPAN::Meta::Requirements install CPAN reload cpan install YAML install MD5 install Digest::MD5 install Digest::SHA1 install readline install Bundle::CPAN reload cpan install DBI force install DBD::mysql install Net::Telnet install Time::HiRes install Net::Server install Switch install Mail::Sendmail install Unicode::Map install Jcode install Spreadsheet::WriteExcel install OLE::Storage_Lite install Proc::ProcessTable install IO::Scalar install Spreadsheet::ParseExcel install Curses install Getopt::Long install Net::Domain install Term::ReadKey install Term::ANSIColor install Spreadsheet::XLSX install Spreadsheet::Read install LWP::UserAgent install HTML::Entities install HTML::Strip install HTML::FormatText install HTML::TreeBuilder install Time::Local install MIME::Decoder install Mail::POP3Client install Mail::IMAPClient install Mail::Message install IO::Socket::SSL install MIME::Base64 install MIME::QuotedPrint install Crypt::Eksblowfish::Bcrypt
Install Asterisk and Dhadi:
#Install dahdi wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.11.1-rc1+2.11.1-rc1.tar.gz tar -xvf dahdi-linux-complete-2.11.1-rc1+2.11.1-rc1.tar.gz cd dahdi-linux-complete-2.11.1-rc1+2.11.1-rc1/ mnake all && make install && make config cp /etc/dahdi/system.conf.sample /etc/dahdi/system.conf #reboot system reboot #test dahdi modprobe dahdi [email protected]:~# /usr/sbin/dahdi_cfg -vvvvvvvvvvvvv DAHDI Tools Version - 2.11.1-rc1 DAHDI Version: 2.11.1-rc1 Echo Canceller(s): Configuration ====================== Channel map: 0 channels to configure. #Install Asterisk wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz tar -xvf asterisk-13-current.tar.gz cd asterisk-13.25.0/ ./contrib/scripts/install_prereq install && ./bootstrap.sh #This could take time.. # if you need WebRtc Support ./configure --libdir=/usr/lib --with-gsm=internal --enable-opus --enable-srtp --with-ogg=/usr/lib64/ --with-ssl --enable-asteriskssl --with-pjproject-bundled make && make menuconfig #select app_meetme in Applications(Vicidial Need this for conference) #select res_http_websocket,res_srtp for WebRTC support. # If you want Regualr setup with Softphone ./configure && make && make menuconfig #select app_meetme in Applications(Vicidial Need this for conference) make install && make samples asterisk asterisk -r vici*CLI> dahdi show channels Chan Extension Context Language MOH Interpret Blocked In Service Description pseudo default default Yes #WebRTC Extra steps(IF YOU DONT NEED WEBRTC SKIP BELOW STEPS) Note: Since WebRTC need SSL i am using my Public Domain to Generate SSL from Let's Encrypt. How to Generate SSL: Link here sample http.conf cat /etc/asterisk/http.conf [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 enablestatic=yes tlsenable=yes tlsbindaddr=YOURPUBLICIP:8089 #(replace j.thojutam.com with your domain) tlscertfile=/etc/letsencrypt/live/j.thoutam.com/cert.pem tlsprivatekey=/etc/letsencrypt/live/j.thoutam.com/privkey.pem sample sip.conf cat /etc/asterisk/sip.conf [general] transport=udp,ws,wss avpf=yes srvlookup=yes udpbindaddr=0.0.0.0:5060 context=trunkinbound allowguest=no allowoverlap=no realm=j.thoutam.com ; REPLACE YOUR DOMAIN HERE bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw mohinterpret=default mohsuggest=default language=en #check http show status
cd /usr/src mkdir astguiclient/ cd astguiclient/ svn checkout svn://svn.eflo.net/agc_2-X/trunk cd trunk/
#start MySQL service mysql start mysql -uroot -pPASSWORD # Create Dialer DB create database DialerDB; #Grant all privileges to DialerDB cron which used in astguiclient.conf file GRANT ALL PRIVILEGES on DialerDB.* to 'cron'@'localhost' IDENTIFIED by '1234'; #Grant all privileges to custom this needed for custom tables GRANT ALL PRIVILEGES on DialerDB.* to 'custom'@'localhost' IDENTIFIED by 'custom1234'; exit; # Load the initial database cd /usr/src/astguiclient/trunk/extras mysql -uroot -p DialerDB < MySQL_AST_CREATE_tables.sql #Load First instalation setup database mysql -uroot -p DialerDB < first_server_install.sql #Load sample sip and iax phones(this contain default user pass, carefull when you installing in vpc) mysql -uroot -p DialerDB < sip-iax_phones.sql #Do back to root Directory of vicidial cd .. perl install.pl #Fallow the setup with appropriate value.#Configiguration example
#Populate ISO country codes cd /usr/src/astguiclient/trunk/bin perl ADMIN_area_code_populate.pl #update the Server IP with latest IP address.(VICIDIAL DEFAULT IP IS 10.10.10.15) perl /usr/src/astguiclient/trunk/bin/ADMIN_update_server_ip.pl --old-server_ip=10.10.10.15 #Say 'Yes' to all
#Add crontab entries(Conf generation, Autodialing, Audio mix and other keepalive processes run from here.
### 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_mix.pl --MIX 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 --GSM #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 --GSM ### keepalive script for astguiclient processes * * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl --cu3way ### 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 and the full day run at night 33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl 50 0 * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --last-24hours ## uncomment below if using QueueMetrics #*/5 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --only-qm-live-call-check ## uncomment below if using Vtiger #1 1 * * * /usr/share/astguiclient/Vtiger_optimize_all_tables.pl --quiet ### 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 ### 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/sbin/ntpdate -u pool.ntp.org 2>/dev/null 1>&2 ### VICIDIAL agent time log weekly and daily summary report generation 2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl 22 0 * * * /usr/share/astguiclient/AST_agent_day.pl ### VICIDIAL campaign export scripts (OPTIONAL) #32 0 * * * /usr/share/astguiclient/AST_VDsales_export.pl #42 0 * * * /usr/share/astguiclient/AST_sourceID_summary_export.pl ### remove old recordings more than 7 days old #24 0 * * * /usr/bin/find /var/spool/asterisk/monitorDONE -maxdepth 2 -type f -mtime +7 -print | xargs rm -f ### roll logs monthly on high-volume dialing systems #30 1 1 * * /usr/share/astguiclient/ADMIN_archive_log_tables.pl ### 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 30 0 * * * /usr/bin/find / -maxdepth 1 -name "screenlog.0*" -mtime +4 -print | xargs rm -f ### cleanup of the scheduled callback records 25 0 * * * /usr/share/astguiclient/AST_DB_dead_cb_purge.pl --purge-non-cb -q ### GMT adjust script - uncomment to enable #45 0 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --list-settings ### Dialer Inventory Report 1 7 * * * /usr/share/astguiclient/AST_dialer_inventory_snapshot.pl -q --override-24hours ### inbound email parser * * * * * /usr/share/astguiclient/AST_inbound_email_parser.pl
Note: Lookout for the recording removal cron. It might fill your server.
#Add entries to rc.local file to start vicidial and related services
# OPTIONAL enable ip_relay(for same-machine trunking and blind monitoring) /usr/share/astguiclient/ip_relay/relay_control start 2>/dev/null 1>&2 ### roll the Asterisk logs upon reboot /usr/share/astguiclient/ADMIN_restart_roll_logs.pl ### clear the server-related records from the database /usr/share/astguiclient/AST_reset_mysql_vars.pl ### load dahdi drivers modprobe dahdi /usr/sbin/dahdi_cfg -vvvvvvvvvvvvv ### sleep for 20 seconds before launching Asterisk sleep 20 ### start up asterisk /usr/share/astguiclient/start_asterisk_boot.pl
[email protected]:~# screen -ls There are screens on: 2240.ASTVDremote (03/21/2019 02:16:03 AM) (Detached) 2237.ASTVDauto (03/21/2019 02:16:03 AM) (Detached) 2234.ASTlisten (03/21/2019 02:16:02 AM) (Detached) 2231.ASTsend (03/21/2019 02:16:02 AM) (Detached) 2228.ASTupdate (03/21/2019 02:16:02 AM) (Detached) 2025.ASTconf3way (03/21/2019 02:15:02 AM) (Detached) 2019.ASTVDadapt (03/21/2019 02:15:02 AM) (Detached) 1826.asterisk (03/21/2019 02:14:51 AM) (Detached) 1819.astshell20190321021448 (03/21/2019 02:14:49 AM) (Detached) 9 Sockets in /var/run/screen/S-root.
cd /var/www/html/agc/
git clone https://github.com/chornyitaras/PBXWebPhone.git
# Add Web Socket URL to Server in Server settings from 6666(Admin login).
Replace j.thoutam.com with your domain.
# Goto system settings and Add WebPhone URL path
#Goto Templates --> SIP_generic and add below content to template.
type=friend
context=default
host=dynamic
trustrpid=yes
sendrpid=no
qualify=yes
qualifyfreq=600
transport=ws,wss,udp
encryption=yes
avpf=yes
icesupport=yes
rtcp_mux=yes
directmedia=no
disallow=all
allow=ulaw,opus,vp8,h264
nat=yes
directmedia=no
dtlsenable=yes
dtlsverify=no
;Replace j.thoutam.com with your domain.
dtlscertfile=/etc/letsencrypt/live/j.thoutam.com/cert.pem
dtlsprivatekey=/etc/letsencrypt/live/j.thoutam.com/privkey.pem
dtlssetup=actpass
#Goto to Phones --> Select cc100 --> Set As a WebPhone --> Y
And Set Template ID: SIP_Generic
Wait for few minutes to Generate config..
m=audio 12548 RTP/AVP 0 8 101
It’s a field from SDP protocol, describing parameters of media (“m” is for “media”). Next, the type of media is “audio”, not video, for example. (m=audio). 12548 is a port address for streaming media. “RTP/AVP” means “RTP Audio/Video
Profile” and representing one of RTP profiles, which are coded by 0, 8 and 101. 0 is PCMU 8000 Hz, 8 is PCMA 8000 Hz, and 101 is payload type for DTMF digits sending.
OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others…
Platfrom : Ubuntu 14.04 +
Let’s Begin…
apt-get install build-essential openssl bison flex apt-get install perl libdbi-perl libdbd-mysql-perl libdbd-pg-perl libfrontier-rpc-perl libterm-readline-gnu-perl libberkeleydb-perl ncurses-dev apt-get install mysql-server libmysqlclient-dev wget http://opensips.org/pub/opensips/latest/opensips-2.3.0.tar.gz tar -xvf opensips-2.3.0.tar.gz cd opensips-2.3.0/ make all make install mkdir /var/run/opensips cd packaging/debian/ cp opensips.default /etc/default/opensips cp opensips.init /etc/init.d/opensips chmod +x /etc/init.d/opensips useradd opensips update-rc.d opensips defaults 99
vim /etc/default/opensips Here you need to replace the 'RUN_OpenSIPS' to 'Yes'. You can also change the user and group and name that you wish to use for the Opensips services and also change the shared memory to minimum 128 which is recommended for the OpenSIPS server.
we also need to update the daemon on OpenSIP and change it location in its startup script and update its state from ‘off’ to ‘on’ and then close the file after making changes.