Knowledge Updates

Voice Broadcasting Solution: Benefits which will help in growing business

Voice broadcasting solution makes a great impact when it comes to the VoIP services. A lot of businesses need the best voice broadcasting service. The voice broadcasting means the voice which should be sent to a mass people. The people receive the messages or call for the promotion, surveys, notification, brand awareness and what not in the voice broadcasting solution.

The voice broadcasting communication goes to the massive scale. Commercially and as a community establishment both use the voice broadcasting solution as it is cost effective and as well as easy. The broadcast also makes sure to send the messages to the customers at once.

Benefits of Voice broadcasting solution

There are more ways in which the Voice Broadcasting Solutions can be used. Some of them are:

Non-intrusive and personal

Voice broadcasting is not like the direct marketing, it is less intrusive. The personal touch can be added to the communication. The personal voice creates the aura where it shows the personal side of the brand and it let establishing the trust in the brand for the connection.

Large-scale communication

What makes the voice broadcasting solution best? The answer is the power of sending the messages to a hundred and thousands of peoples at once. Hosted voice broadcasting solution helps customer utilize this service as a potential communication. The business automation is at its best in the voice broadcasting service.

Reach the non-tech savvy

Everyone in the world is already using the mobile phones. And the voice broadcasting solutions makes it easy when it comes to the mobile phones. As if the people are receiving the messages or calls then actually they are listening and it will also make an effect. Voice broadcasting is actually the oldest business technology. Voice broadcasting, in this case, means that if there is a non-tech savvy person then they are also the ones who can easily get what the brand is saying.

Brand-building

Send the messages directly to the customers telling about your brand. It is very easy to tell them what you want to portray. Once you have messaged or call them then the name of the brand will be in their mind if they’re going to purchase.

Expand Your Reach and Increase Profits with Voice broadcasting solution

There are already the businesses which are opting out of the voice broadcasting solution to increase the profits and also increase the engagement. There are also the other ways in which the company can take the help of this best voice broadcasting service. Let us have a look at that:

Surveys

For a small business, it is quite difficult to know what the customer wants, as it is needed for the growth. Now here the voice broadcasting comes in handy. A company can conduct the survey by implementing the IVR into the voice broadcasting solution. In the voice broadcasting solution, the question can be asked about the products and get the feedback. It also makes the product good in many scenarios as the customer feedback is very important. In this case, the brand will be always be connected to the customer and will also fulfill their needs and solve their problems.

Webinar invites

Invite the peoples to the webinars. Webinars are run to generate the leads or sales. Use voice broadcasting to increase the participation. If half of the messages are delivered for the webinars and attendees will become the leads and then the sale can be grown. And if the leads and sale are worth, it will generate a lot of additional revenues.

Notifications

Voice broadcasting helps in engaging for the promotions, sales, and product launches. With the help of voice broadcasting solution let the audience know and get them excited. Any time if there is a launch or promotion, send a message or call to the customer to alert them. Generate excitement and build momentum with the help of voice broadcasting for the product launch.

Speed up the marketing processes, connect with your potential customer and build the brand. Voice broadcasting solution is the small and viable solution for the small business. Engage the audience in the business and grow it with the help of voice broadcasting solution.

About Gventure,

Interested in voice broadcasting solution? Talk about your project with Gventure’s experts. Asterisk is required to build custom communications solutions and to build innovative communications products such as voice broadcasting and IVR. Whether you are using VoIP software, Gventure can help you with various VoIP software solutions which you’ll agree on the next steps of your project.


It’s All about Voice Broadcasting & White Label Hosted PBX

Voice is our fundamental correspondence asset yet particularly is the tool more essential for chipping away at radio. The voice on radio transmits the clear content, as well as all the importance and the full of feeling extent. Subsequently, the crowd’s thoughtfulness regarding the radio message and in addition understanding and absorbing of it will rely upon a right and expressive utilization of voice. Various affordable voice broadcasting software’s are there in the market which can communicate the Voice. Asterisk Hosted PBX Solution has its own particular benefits with regards to Voice broadcasting programming. A considerable amount of the organizations provide white label and hosted PBX.

Voice broadcasting is a mass communication technique which broadcasts telephone messages to millions of call recipients at once. Both commercial and community applications are made from this technology as everywhere there is a need for Voice broadcasting. Targets such as subscribers, members of an association, employees, group of local residents can be contacted by the Voice broadcasting solution. When used by government authorities, it may be known as an emergency notification system as these notifications are intended only for use in emergencies mainly in a disaster management.

The Broadcasting software is developed in the Asterisk and with any of the programming language as a User interface. These Affordable Voice Broadcasting Solutions are always customizable for every client and easily integrate with ERP, CRM, billing systems. Voice broadcasting solutions are usually used by banks and credit companies so that they remember their clients about payment day as a lot of customers need to remind about the payment.

Affordable Voice Broadcasting Software: Types of campaign

Affordable Voice Broadcasting Solution is completely adjustable to every customer’s specific needs and takes into consideration exceptionally configurable call-flow according to your particular business objectives. Automated voice calling takes care of media transmission issues like bill updates, notices, alarms feedbacks and business promotions and so forth. Proactive dialing increment telemarketing approach, voice communication enhances communication with customers. There are numerous manners by which Voice Broadcasting solution can be utilized. Some of them are:

Business to Business(B2B): Campaign where one business is centered around calling different organizations. Voicemail campaign message battles after business hours are regularly viable for calling organizations.

Business to Consumer(B2C): Campaigns where one business is centered around calling residences and the people or consumers. The broadcasting is incredible for thank you or follow up calls.

Live message and Voice mail campaign: Similar to a live message the only campaign this alternative will play two unique messages. One for live answers and one for voicemails live messages will Offer the alternative to press a key to be exchanged a live representative and a choice to be removed from the call list. The voice message messages essentially leave a number to get back to.

Surveys: Ask questions and permit responses by pressing keypad choices. The voice broadcasting system will record reactions for reporting purposes.

Voice message only campaigns: Primarily utilized for Business to Business calls. This is a decent way to produce callbacks. A large portion of the supplier’s software will leave voice message messages with a number to get back to. That gets back to will be directed to operators.

Live message delivery only campaigns: Play your message just if the autodialer recognizes that the telephone has been replied by a live human voice Live messages can offer the alternative to press a key to be exchanged to a live representative or and a choice to be removed the call list.

Message Campaign: A message will be picked to communicate to the recipients from telephone list. There are two sorts of messages, one being a message with an exchange/Voicemail option and another with none.

Asterisk hosted PBX solution: Why in Asterisk?

Asterisk is a software implementation of a telephone private branch trade (PBX). It was made in 1999 by Mark Spencer of Digium. It’s an open source language which means the developers can develop affordable voice broadcasting solutions. Like any PBX, it enables appended phones to make calls to each other, and to interface with other telephone utilities including people public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. It is mainly used to develop the White Label Hosted PBX Solution.

Custom voice and text message broadcasting software solution accompany advanced features and particulars that assist and help the remote audiences. The solution is most appropriate to every single measured business. It is a perfect and Affordable Voice Broadcasting Solution for running event promotions, campaigns, making polls and surveys, sending updates, warnings, and reports. Sending voice and SMS broadcasting arrangement can kill the necessity of voice agents. Along these lines, organizations can spare 30% to 70% cost. Clients can transfer their own voice recording or utilize its “text-to-speech” feature. Additionally, IVR scripts can likewise be dealt with utilizing this voice broadcasting solution. This arrangement is anything but not difficult to set up, easy to use and maintain.

White Label and Hosted PBX: Why every business should have PBX?

Putting resources into a product based IP PBX makes both budgetary and great business sense, for new organizations obtaining another business telephone system, and furthermore for organizations who are hoping to replace a current PBX. IP communication is the method for the future and an IP-PBX will convey to your business cost reserve funds in administration, upkeep, and telephone call costs, in addition to extra money-saving advantages from included efficiencies. Any of the business can go for either White Label or Hosted PBX. Some of the reason why business should go for IP PBX is:

Twice the phone system features for half the price Better customer service & productivity Eliminate phone wiring Easier to manage because of web/GUI based configuration interface Much easier to install & configure than a proprietary phone system

About GVenture Technology,

Interested in White Label and Hosted PBX or asterisk hosted PBX solution? Talk about the project with the experts of the top notch development company. Gventure uses Asterisk to build custom communications solutions and to build innovative communications products such as Voice broadcasting software or the hosted PBX which consists of IVR and many more. GVenture can help with various VoIP software solutions which you’ll agree on the next steps of the project.


Double Your Profit With Gventure's free hosted PBX solution

When it involves selecting the appropriate PBX system to use for your business, there are varieties of things concerned that may bend your conclusion toward employing a specific system. A free hosted PBX solution is the need of the time. The most efficient things to get a handle of those factors are to look at the advantages and disadvantages of the various sorts of PBX systems that are offered.

PBX (Private Branch Exchange) is a multiline telephone system, for the most part, used within businesses, which allows users to communicate internally as well as making calls outside of the business, in simplest terms. In the first place, a hosted PBX usually comes with options that permit unlimited free calls to be placed to assure locations around the globe. Also, not like associate degree on-the-spot IP PBX, there are not any initial investment prices for equipment and software solution. Well, looking at the benefits of a free hosted PBX solution for your business some of them are listed below.

Costs: With a hosted PBX, cut the value of line rental, on-the-spot maintenance, etc. As everything is managed over Broadband, there’s no line rental at all! Moreover, as this, IP telecom train you in our online web-portal, thus you’ll be ready to manage your system moreover as implement any changes to call flows or ring teams as required. It’s additionally cheaper to form international calls through VoIP.

Flexibility & Scalability: Distinctive proprietary phone systems are all too easy to outgrow as they only provide a certain number of physical lines or ports. Upgrading can be time-consuming and expensive. It is a lot easier to grow your business with a Hosted PBX, and IP Telecom will help you every step of the way to ensure any growth is seamless and easily implemented.

Integration: As your voice traffic is all done through your Broadband connection, any device with a data connection will be used anyplace within the world to create and receive calls. This suggests that you just will install soft-phone applications on your mobile or a desktop.

More Control: An IP PBX system is managed from one central web-based configuration interface to create maintenance and upgrades straightforward. Compare that to ancient phone systems that use advanced interfaces that have got to be accessed and managed by proprietary phone technicians solely.

No disruption in business setup: An IP PBX phone system runs on a computer without the requirement for complex telecommunication equipment or added copper wiring to be run into your office. Instead, the phone lines are routed over the internet. The IP PBX system switches local calls to over the information network within your business and permits all users to share identical external phone lines. You even get to stay your regular phone numbers.

Boost to Productivity: IP PBX phone systems generally cost less than traditional systems. However, they additionally deliver the options to extend workplace productivity. This concept extends to remote operating too.

Loads of New Features: A Hosted PBX comes with a variety of options that may build running your business a lot easier such as Calling Features, Call Management & Reporting, Call Centre Features, etc.

If you own a business or enterprise, give us a call. We will do an assessment of your needs and make a proposal that fits your requirements and budget. GVenture’s Free GPBX- a free hosted PBX solution is a best in class PBX system and is worth considering.

Gventure’s free GPBX Solution provides latest features and functionality with affordable price. Features we tend to give to our clients are:

3rd Party CRM Integration: End-to-end customer engagement with all the sales, marketing, and support apps your customer-facing teams need to engage with customers.

Hosted IVR: To answer the callers and assist those with the correct solution or call transfer, install machine-based IVR.

ACD PBX: Answer the call automatically to greet the customers and offer a customizable pre-recorded menu while your attendants are busy on other calls by a digital receptionist.

Inbuilt Billing: Bill anything and everything with billing software offer real-time rating & mediations, accurate taxation, prepaid & post-paid telecom billing, multi-location billing, residential & business telecom billing, and more comprehensive & scalable telecom billing platform.

Graphical & Statistical Report: The Whole host of graphical representations as well as statistics available plus, assess your current team resources utilisation at a glance.

DID Management: A number of extensions can be customized, manage and maintain on the same phone line. Music on Hold Configuration: Playing music, current offers, etc. to keep your call entertained even on hold make more business.

API Integration: API integration allows to easily generating automated workflows to help you sell, deliver, manage, and invoice more effectively.

Gventure provides a free hosted PBX Solution which is known as Free GPBX. A cloud-hosted Business Phone System that carries all the potential features of traditional VoIP Phone along with the extraordinary advantages of the cloud. Get this in action and see the business connecting to the globe, like never before.


Query related to Kamailio Installation? Here is the one you were looking for

The Kamailio SIP server is the main Open Source software program for building SIP services like a SIP proxy, SIP Presence Server, SIP location server and much more. Kamailio used to handle thousands of call setups per second. Kamailio is the choice for building enterprise as well as carrier solutions with a rich configuration language, popularity and continues development. Kamailio runs on UNIX and Linux systems, ranging from embedded systems to huge scale multi-core servers.

Pre-requites for Kamailio installation:

The following packages are required before proceeding to the next steps.

  • git client: apt-get install git-core – it is recommended to have a recent version if your Linux distro has an old version.
  • gcc compiler: apt-get install gcc
  • flex – apt-get install flex
  • bison – apt-get install bison
  • libmysqlclient-dev – apt-get install libmysqlclient-dev
  • make – apt-get install make
  • if you want to enable more modules, some of them require extra libraries:
  • 1. libssl – apt-get install libssl-dev
  • 2. libcurl – apt-get install libcurl4-openssl- dev
  • 3. libxml2 – apt-get install libxml2-dev
  • 4. libpcre3 – apt-get install libpcre3-dev

Note: Kamailio (version 4.3.0) uses the directory /var/run/kamailio/ for creating FIFO and UnixSocket control files. You have to complete the section related to the installation of init.d script for creating /var/run/kamailio even if you plan to start Kamailio manually from command line. The alternative is to set different paths via parameters of mi_fifo and ctl modules.

Step1: First, Getting sources from GIT:

First of all, you have to create a directory on the file system where the sources will be stored.

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  mkdir -p /usr/local/src/kamailio-devel
  cd /usr/local/src/kamailio-devel

you can Download the sources from GIT using the following commands.

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  git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio kamailio
  cd kamailio
  git checkout -b 4.4 origin/4.4

Step2. Tuning Make files

The first step is to generate build config files.

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  make cfg

Next step is to enable the MySQL module. Edit modules.lst file:

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  nano -w modules.lst

Add db_mysql to the variable include_modules.

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include_modules= db_mysql

Save the modules.lst and exit.

NOTE: this is one mechanism to enable modules which are not compiled by default, such as lcr, dialplan, presence – add the modules to include_modules variable inside the modules.lst file.

Alternative is to set ‘include_modules’ variable to specify what extra modules to be included for compilation when building Makefile cfg:

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make include_modules="db_mysql dialplan" cfg

NOTE: If you want to install everything in one directory (so you can delete all installed files at once), say /usr/local/kamailio-devel, then set PREFIX variable to the install path in ‘make cfg …’ command:

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make PREFIX="/usr/local/kamailio-devel" include_modules="db_mysql dialplan" cfg

NOTE: Previous release series 3.x.x used FLAVOUR mechanism to set the name of the application to kamailio. Starting with 4.0.0, the default is kamailio. From that version, if you want to build ser flavour, use FLAVOUR=ser in the make command to generate the build configs.

Step3. Compile Kamailio

Once you added the mysql module to the list of enabled modules, you can compile Kamailio:

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  make all

You can get full compile flags output using:

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  make Q=0 all

Step4. Install Kamailio

When the compilation is ready, install Kamailio with the following command:

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  make install

Step5. Kamailio installation Path

The binaries and executable scripts were installed in:

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  /usr/local/sbin

These are kamailio, kamdbctl, kamctl, kamcmd commond to manage SIP Server.

To be able to use the binaries from command line, make sure that ‘/usr/local/sbin’ is set in PATH environment variable.

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  PATH=$PATH:/usr/local/sbin
  export PATH

Kamailio (OpenSER) modules are installed in:

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  /usr/local/lib/kamailio/modules/

Note: On 64 bit systems, /usr/local/lib64 may be used.

The documentation and readme files are installed in:

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  /usr/local/share/doc/kamailio/

The man pages are installed in:

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  /usr/local/share/man/man5/
  /usr/local/share/man/man8/

The configuration file was installed in:

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  /usr/local/etc/kamailio/kamailio.cfg

Step6. Create MySQL database

To create the MySQL database, first edit kamctlrc file to set the database server type:

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  nano -w /usr/local/etc/kamailio/kamctlrc

Locate DBENGINE variable and set it to MYSQL:

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DBENGINE=MYSQL

You can change other values in kamctlrc file, at least it is recommended to change the default passwords for the users to be created to connect to database. Once you are done updating kamctlrc file, run the script to create the database used by Kamailio:

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  /usr/local/sbin/kamdbctl create

Step7. Edit configuration file

To fit your requirements for the VoIP platform, you have to edit the configuration file.

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  /usr/local/etc/kamailio/kamailio.cfg

Follow the instruction in the comments to enable usage of MySQL. Basically, you have to add several lines at the top of config file, like:

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#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB

Step8. The init.d script

The init.d script can be used to start/stop the Kamailio server in a nicer way. A sample of init.d script for Kamailio is provided at:

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/usr/local/src/kamailio-devel/kamailio/pkg/kamailio/deb/debian/kamailio.init

Just copy the init file into the /etc/init.d/kamailio. Then change the permisions:

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  chmod 755 /etc/init.d/kamailio

then edit the file updating the $DAEMON and $CFGFILE values:

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  DAEMON=/usr/local/sbin/kamailio
  CFGFILE=/usr/local/etc/kamailio/kamailio.cfg

You need also setup a configuration file in the /etc/default/ directory. This file can be found at:

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  /usr/local/src/kamailio-devel/pkg/kamailio/debian/kamailio.default

You need to rename the file to ‘kamailio’ after you’ve copied it. Then edit this file and set RUN_KAMAILIO=yes. Edit the other options at your convenience. Create the directory for pid file:

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mkdir -p /var/run/kamailio

Default setting is to run Kamailio as user “kamailio” and group “kamailio”. For that you need to create the user:

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adduser --quiet --system --group --disabled-password \
        --shell /bin/false --gecos "Kamailio" \
        --home /var/run/kamailio kamailio
 
# set ownership to /var/run/kamailio
chown kamailio:kamailio /var/run/kamailio

Then you can start/stop Kamailio using the following commands:

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  /etc/init.d/kamailio start
  /etc/init.d/kamailio stop

Step9. Ready to use

Now everything is in place. You can start the VoIP service, creating new accounts and setting the phones. A new account can be added using ‘kamctl’ tool via ‘kamctl add ‘. (or try without the email)

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kamctl add test testpasswd test@mysipserver.com
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kamctl add test testpasswd test@mysipserver.com

If you are asked for SIP_DOMAIN environment variable do one of the following option.

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  export SIP_DOMAIN=mysipserver.com
  edit '/root/.kamctlrc' and add:
  SIP_DOMAIN=mysipserver.com

Step10. Maintenance

The maintenance process is very simple right now. You have to be user ‘root’ and execute following commands:

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  cd /usr/local/src/kamailio-devel/kamailio
  git pull origin
  make all
  make install
  /etc/init.d/kamailio restart

Now you have the latest Kamailio devel running on your system.

Kamailio is the choice for building enterprise as well as carrier solutions with a rich configuration language, popularity and continues development. We at GVenture Technology will fulfil all your necessary requirements.

Tagged:  

OpenSIPS Development: An important resource for joined communication

GVenture Technology takes on a pleasant ride to develop OpenSIPS applications. OpenSIPS (Open SIP Server) is a developed Open Source implementation of a SIP server that runs on Linux platforms and play in the infrastructure of an Internet Telephony Service Provider and it includes application-level functionality.

Why use OpenSIPS?

  • No vendor trap
  • Faster development cycle
  • Split work between parties
  • Easy synchronization with the mainstream via contributions ⇒ Unified effort for development
  • Performance and flexibility

Prerequisites:

  • Since OpenSIPS is known to work on most nix based systems, installing and running OpenSIPS requires some very basic Linux knowledge.
  • Also, since OpenSIPS is a SIP proxy, you will most likely need some basic SIP knowledge in order to deploy/debug OpenSIPS in some more advanced scenarios.
  • Last but not least, you will need some basic programming logic knowledge – The OpenSIPS configuration file is text-based, written in an OpenSIPS custom language, very similar to the C language.

Requirements:

gcc / suncc / icc : gcc >= 2.9x; 4.[012] recommended (it will work with older version but it might require some options tweaking for best performance)

  • bison or yacc (Berkley yacc)
  • flex
  • GNU make (on Linux this is the standard “make”, on FreeBSD and Solaris is called “gmake”) version >= 3.79.
  • sed and tr (used in the makefiles)
  • GNU tar (“gtar” on Solaris) and gzip if you want “make tar” to work
  • GNU install or BSD install (on Solaris “ginstall”) if you want “make install”, “make bin”, “make sunpkg” to work
  • openssl if you want to compile the TLS support
  • libsctp if you want to compile the SCTP support
  • libmysqlclient & libz (zlib) -libs and devel headers- if you want mysql DB support (the db_mysql module)
  • libpq / postgresql -libs and devel headers- if you want postgres DB support (the db_postgres module)
  • unixodbc -libs and devel headers- if you want unixodbc DB support (the db_unixodbc module)
  • libexpat if you want the jabber gateway support (the jabber module) or the XMPP gateway support
  • libxml2 if you want to use the cpl-c (Call Processing Language) or the presence modules (presence and pua*)
  • libradius-ng -libs and devel headers- if you want to use functionalities with radius support – authentication, accounting, group support, etc
  • unixodbc – libs and devel headers – if you want UNIXODBC support as DB underlayer
  • libxmlrpc-c3 – libs and devel headers – if you want to have XML-RPC support for the Management interface (MI)
  • libperl – libs and devel headers – if you want PERL connector to support perl scripting from you config file (perl module)
  • libsnmp9 – libs and devel headers – if you want SNMP client functionality (SNMP AgentX subagent) for opensips
  • libldap libs and devel headers v2.1 or greater – if you want LDAP support
  • libconfuse and devel headers – if you want to compile the carrier route module

Installation:

By following above 2 processes we have installed the OpenSIPS server on our system now we have to configure the server corresponding to our requirement.

OpenSIPS Installation:

1. System Update:  Update your system with latest updates and security patches using the following command.

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# apt-get update

2. Installing Dependencies: After system update you need to install the missing packages using the ‘apt-get’ command if these are not already installed.

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# apt-get install build-essential openssl bison flex

3. Installing MySQL Server: Let’s run the following commands to install MySQL server and it development libraries.

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# apt-get install mysql-server libmysqlclient-dev

4. Download and extract OpenSIPS Package: We are going to download the package into the following directory using ‘wget’ command and the extract within th esame directory.

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# cd /usr/src
# wget http://opensips.org/pub/opensips/latest/opensips-2.1.2.tar.gz
# tar -zxvf opensips-2.1.2.tar.gz

5. Compiling OpenSIPS Source: Moving step ahead let’s move into the opensips directory and run the command below to start the compilation process.

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# cd opensips-2.1.2/
# make all

6. Installing OpenSIPS Source: Once the compilation process completes, you can start making its installation as shown below.

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# make install

7. OpenSIPS Configuration: The OpenSIPS has been installed, now we are going to configure some of its basic parameters and startup script. Let’s first create a new directory for the OpenSIPS run files.

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# mkdir /var/run/opensips

Now move to the following ‘debians’ directory and list the files in it.

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# cd packaging/debian/

8. Starting OpenSIPS Service: All necessary configurations has been setup to proceed on starting OpenSIPS service. Simply run the following command to start OpenSIPS and the check its status, that should be active and running as shown in the image.

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# /etc/init.d/opensips start
# systemctl status opensips

You can also check the state of OpenSIPS services and the port on which its running that is ‘5060’ by using the below commands.

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# ps -ef | grep opensips
# netstat -alnp | grep opensips

CONFIGURATION

We have following directory where all files has be installed:

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/usr/local/lib/opensips/ directory where all modules are installed
/usr/local/etc/opensips/ directory where all configuration you have make

FOR LOAD

Under Load-balancing format The below setting will create alternate call from one server and other alternately it will create same number of calls from each gateway.

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log_stderror=yes
log_facility=LOG_LOCAL0
fork=yes
children=4
debug=2
listen=5060

MODULE PATH

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mpath="/usr/local/lib/opensips/modules/"

Modules Section

DB_MYSQL

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loadmodule "db_mysql.so"
modparam("db_mysql", "ping_interval", 300)
modparam("db_mysql", "timeout_interval", 10)
modparam("db_mysql", "auto_reconnect", 1)

SL

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loadmodule "sl.so"
modparam("sl", "enable_stats", 1)

TM

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loadmodule "tm.so"
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
modparam("tm", "wt_timer", 5)
modparam("tm", "delete_timer", 2)
#modparam("tm", "noisy_ctimer", 0)
modparam("tm", "ruri_matching", 1)
modparam("tm", "via1_matching", 1)
modparam("tm", "unix_tx_timeout", 2)
modparam("tm", "restart_fr_on_each_reply", 1)
modparam("tm", "pass_provisional_replies", 0)
modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)")

RR

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loadmodule "rr.so"
modparam("rr", "enable_full_lr", 0)
modparam("rr", "append_fromtag", 1)
modparam("rr", "enable_double_rr", 0)
modparam("rr", "add_username", 0)

MAXFWD

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loadmodule "maxfwd.so"

USRLOC

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loadmodule "usrloc.so"
modparam("usrloc", "user_column", "username")
modparam("usrloc", "domain_column", "domain")
modparam("usrloc", "contact_column", "contact")
modparam("usrloc", "expires_column", "expires")
modparam("usrloc", "q_column", "q")
modparam("usrloc", "callid_column", "callid")
modparam("usrloc", "cseq_column", "cseq")
modparam("usrloc", "methods_column", "methods")
modparam("usrloc", "flags_column", "flags")
modparam("usrloc", "user_agent_column", "user_agent")
modparam("usrloc", "received_column", "received")
modparam("usrloc", "socket_column", "socket")
modparam("usrloc", "use_domain", 0)
modparam("usrloc", "desc_time_order", 0)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url" , "mysql://opensips:opensips@database/opensips")
modparam("usrloc", "matching_mode", 0)
modparam("usrloc", "cseq_delay", 20)
modparam("usrloc", "nat_bflag", 6)

REGISTRAR

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loadmodule "registrar.so"
modparam("registrar", "method_filtering", 1)
modparam("registrar", "default_expires", 3600)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 0)
modparam("registrar", "default_q", 0)
modparam("registrar", "append_branches", 1)
modparam("registrar", "case_sensitive", 0)
modparam("registrar", "received_param", "received")
modparam("registrar", "max_contacts", 10)
modparam("registrar", "retry_after", 0)
modparam("registrar", "path_mode", 2)
modparam("registrar", "path_use_received", 0)
#modparam("registrar", "received_avp", "$avp(i:801)")
modparam("registrar", "aor_avp", "$avp(i:3223)")
modparam("registrar", "received_avp", "$avp(s:rcv)")

TEXTOPS

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loadmodule "textops.so"

MI_FIFO

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loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
#modparam("mi_fifo", "fifo_mode", 0660)
#modparam("mi_fifo", "fifo_group", "opensips")
#modparam("mi_fifo", "fifo_user", "opensips")
#modparam("mi_fifo", "reply_dir", "/tmp/")
#modparam("mi_fifo", "reply_indent", "\t")

URI

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loadmodule "uri.so"

URI_DB

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loadmodule "uri_db.so"
modparam("uri_db", "db_url", "mysql://opensips:opensips@database/opensips")
modparam("uri_db", "db_table", "uri")
modparam("uri_db", "use_uri_table", 1)
modparam("uri_db", "use_domain", 1)
modparam("uri_db", "user_column", "username")
modparam("uri_db", "domain_column", "domain")
modparam("uri_db", "uriuser_column", "uri_user")

xlog

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loadmodule "xlog.so"
modparam("xlog", "force_color", 1)
modparam("xlog", "buf_size", 4096)

acc

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loadmodule "acc.so"
modparam("acc", "db_url", "mysql://opensips:opensips@database/opensips")
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 1)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "db_extra" , "ua=$hdr(User-Agent); src_user=$avp(i:100);
src_domain=$fd; src_port=$sp; dst_user=$rU; dst_domain=$rd; dst_port=$rp;
message=$mf; unixtime=$Ts; avp_var=$avp(s:can_uri); callstate=$avp(s:callstate);
usercode=$avp(i:1000); recieved=$avp(s:rcv); aor_avp=$avp(i:3223)")

auth

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loadmodule "auth.so"
modparam("auth", "nonce_expire", 300)
modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes")
modparam("auth", "rpid_avp", "$avp(s:rpid)")

AVPOPS

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loadmodule "avpops.so"
modparam("avpops", "db_url", "mysql://opensips:opensips@database/opensips")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops", "use_domain", 0)
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
#modparam("avpops", "avp_aliases", "c_uri" )

DIALOG

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loadmodule "dialog.so"
modparam("dialog", "enable_stats", 1)
modparam("dialog", "default_timeout", 21600)
modparam("dialog", "db_url", "mysql://opensips:opensips@database/opensips")
modparam("dialog", "db_mode", 1)
modparam("dialog", "dlg_flag", 4)

SIPTRACE

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loadmodule "siptrace.so"
modparam("siptrace","db_url","mysql://opensips:opensips@database/opensips")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_flag", 22)

MI_DATAGRAM

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loadmodule "mi_datagram.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_datagram", "socket_name", "udp:10.10.2.144:8080")

Routing Logic

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 main request routing logic
 
route{
        sip_trace(); # Recording all sip signal for each calls connect through this server
        xlog("Recive call variable : $avp(s:rcv) $avp(i:3223)\n"); # prints the cli log
 
        if($fU==NULL){
                $avp(i:100)="0";
        }
        else
                $avp(i:100)=$fU;
 
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }
 
        if (has_totag()) {
                if (loose_route()) {
                        if(is_method("BYE")){
		        #setting for call accounting
                                setflag(1);
                                setflag(3);
                        }
                        route(1);
                } else {
                        if ( is_method("ACK") ) {
                                if ( t_check_trans() ) {
                                        t_relay();
                                        exit;
                                } else {
                                        exit;
                                }
                        }
                        sl_send_reply("404","Not here");
                }
                exit;
        }
 
        if (is_method("CANCEL"))
        {
                if (t_check_trans())
                        t_relay();
                exit;
        }
 
       t_check_trans();
 
        # record routing
        if (!is_method("REGISTER|MESSAGE")){
                record_route();
        }
 
        # account only INVITEs
        if (is_method("INVITE")) {
	   if(from_uri=~".*@10.10.2.136" || from_uri=~".*@10.10.2.134"){
 
                }
                else{
                        avp_db_query("select id_skillgroup from subscriber u where username='$rU'", "$avp(i:2001)");
                        avp_db_query("select id from ast_skillgroup where id='$avp(i:2001)'", "$avp(i:2002)");
                        avp_db_query("select id, serverip, callerid from ast_servers where id_skillgroup='$avp(i:2002)'
                                      order by calls asc limit 1", "$avp(i:2003);$avp(i:2004);$avp(i:2005)");
                        avp_db_query("Update ast_servers SET calls=calls+1 WHERE id=$avp(i:2003)");
 
                        avp_pushto("$ru/domain","$avp(i:2004)");
 
                        if (!t_relay()) {
                                sl_reply_error();
                        };
                        exit;
                }
        }
 
         if (!uri==myself)
        {
                append_hf("P-hint: outbound\r\n");
                route(1);
        }
 
        if (is_method("PUBLISH"))
        {
                sl_send_reply("503", "Service Unavailable");
                exit;
        }
 
        if (is_method("REGISTER"))
        {
                # authenticate the REGISTER requests (uncomment to enable auth)
                xlog("User credentals : $avp(i:1000)\n");
                if (www_authorize("", "subscriber"))
                {
                        xlog("User credentals : $avp(i:1000)\n");
                        if (!save("location"))
                                sl_reply_error();
 
                        exit;
                }
                else{
                        www_challenge("", "0");
                        exit;
                }
 
                if (!check_to())
                {
                        sl_send_reply("403","Forbidden auth ID");
                        exit;
                }
        }
 
        if ($rU==NULL) {
                sl_send_reply("484","Address Incomplete");
                exit;
        }
 
        if (!lookup("location")) {
                switch ($retcode) {
                        case -1:
                        case -3:
                                t_newtran();
                                t_reply("404", "Not Found");
                                exit;
                        case -2:
                                sl_send_reply("405", "Method Not Allowed");
                                exit;
                }
        }
 
        setflag(2);
        xlog("Script is at the end\n");
        route(1);
   }
 
   route[1] {
        # for INVITEs enable some additional helper routes
        if (is_method("INVITE")) {
                avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)");
                $avp(s:can_uri) = $ru;
                xlog("User credentals : $avp(s:callee_uuid)\n");
                t_on_branch("2");
                t_on_reply("2");
                t_on_failure("1");
        }
 
        if (!t_relay()) {
                sl_reply_error();
        };
        exit;
   }
 
    branch_route[2] {
        xlog("new branch at $ru\n");
        $avp(s:can_uri) = $ru;
        $avp(s:callstate)="CALL REROUTED";
        add_rr_param(";foo=true;user=vivek");
        xlog("Call is forwared to asterisk\n");
   }
 
    onreply_route[2] {
        xlog("incoming reply\n");
   }
 
    failure_route[1] {
        if (t_was_cancelled()) {
                exit;
        }
   }

Above configuration is sampled corresponding to load balanced format where calls are forwarded to different gateway corresponding to our requirements and availability of gateway.

GVenture Technology will provide with vicinity to make yourself convenient and to have carefree talk to tell us about your project and then we can offer you all the bits and pieces essential for it.


Got Stuck in FreeSwitch Installation? Let us help you in the crisis.

FreeSWITCH whereas is an extensible open source cross-platform telephony platform designed to route and interconnects prominent communication protocols using audio, video, text or any other form of media. It can be used as a soft-client, carrier-class SoftSwitch or even as PBX.

FreeSwitch Installation:

Following are the needed steps to get it up and running.

Installing FreeSwitch Dependencies:

  • 1. Pre-requisites: We require using git to drag FreeSwitch from the GIT repo. To install GIT we require the EPEL repository installed:
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[root@voip ~]# rpm -Uvh http://download.fedoraproject.org/pub/epel/6/i386/epel-release-6-5.noarch.rpm
[root@voip ~]# yum install git
  • 2. Install the required packages:
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[root@voip src]# yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make ncurses-devel unixODBC-devel openssl-devel gnutls-devel libogg-devel libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel expat-devel zlib zlib-devel bzip2 which
  • 3. Change directory and drag the source from the git repo :
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[root@voip src]# cd /usr/local/src
[root@voip src]# git clone git://git.freeswitch.org/freeswitch.git
  • 4. Change into the recently downloaded source code directory:
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[root@voip src]# cd freeswitch/
  • 5. Run the bootstrap and configure scripts to create the required modules.conf and get the code ready for compiling:
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[root@voip freeswitch]# ./bootstrap.sh
[root@voip freeswitch]# ./configure --without-libcurl --without-pgsql
  • 6. Edit the modules.conf file. Remove the # in front of any modules you want to compile:
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[root@voip freeswitch]# vim modules.conf
  • 7. Make and Install Freeswitch:
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[root@voip freeswitch]# make && make install
  • 8. Install the default sounds:
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[root@voip freeswitch]# make uhd-sounds-install && make uhd-moh-install
  • 9. Copy the init script and set the correct permissions:
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[root@voip freeswitch]# cp build/freeswitch.init.redhat /etc/init.d/freeswitch
[root@voip init.d]# chmod 755 /etc/init.d/freeswitch
  • 10. Configure Freeswitch to start on boot:
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[root@voip freeswitch]# chkconfig --add freeswitch
[root@voip freeswitch]# chkconfig freeswitch on
  • 11. You require modifying the default init script. It points to Freeswitch being installed in /opt/freeswitch/ however we installed it in /usr/local/freeswitch/ You require changing the lines:
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FS_FILE=${FS_FILE-/opt/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/opt/freeswitch}
<strong>To</strong>
FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/usr/local/freeswitch}
  • 12. Next, we require changing the FreeSWITCH directory permissions as we will be running it under the FreeSWITCH account:
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[root@voip local]# cd /usr/local
[root@voip local]# chown -R freeswitch freeswitch
[root@voip local]# chgrp -R freeswitch freeswitch
  • 13. For performance reasons, it is recommended to change the kernel boot parameters to disable the tick-less feature. To do this you require adding nohz=off to the end of your boot parameters in your grub.conf.
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[root@voip freeswitch]# vim /boot/grub/grub.conf
  • Add to the end of the line starting with kernel to add nohz=off at the end. Mine becomes
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kernel /vmlinuz-2.6.32-71.el6.x86_64 ro root=/dev/mapper/vg_voip-lv_root rd_LVM_LV=vg_voip/lv_root rd_LVM_LV=vg_voip/lv_swap rd_NO_LUKS rd_NO_MD rd_NO_DM LANG=en_US.UTF-8 SYSFONT=latarcyrheb-sun16 KEYBOARDTYPE=pc KEYTABLE=us crashkernel=auto rhgb quiet nohz=off
  • 14. Restart for the kernel parameters to take effect.

At Gventure Technology, Our FreeSWITCH Engineers will be providing an environment where your issues can be resolved ASAP.

Tagged:  

Having a problem with the installation of the Asterisk server? Read us for a simple procedure.

Asterisk is an open source framework build for communication applications. A communication server made of a normal computer by using Asterisk. IP PBX systems, VoIP gateways, conference server etc. are powered by Asterisk. It is used by small businesses, large businesses, call centres, carriers and government agencies, worldwide. Asterisk is the free and open source. Asterisk Development is one of the more user-friendly communications frameworks that are equipped with up to date networks and features enhanced media capabilities.

Asterisk Server Installation:

Pre-requisites for Asterisk installation:

  • System running with kernel-1.2.6 and their header files must be present in our system directory to compile asterisk on our system.
  • It should be written in c we require gcc with their supporting lib such as termcap, and open SSL.
  • Asterisk add-on requires MySQL header file to install the MySQL library, MySQL client and header to compile asterisk add-ons.
yum install bison bison-devel ncurses ncurses-devel zlib zlib-devel OpenSSL OpenSSL-devel gnutls-devel gcc gcc-c++ MySQL-server MySQL-client which flex make.

Download all files: Download below mentioned files as pre-requisites steps for the asterisk installation.

  1. Zaptel (http://downloads.asterisk.org/pub/telephony/zaptel/zaptel-1.4.12.1.tar.gz)
  2. libpri (http://downloads.digium.com/pub/libpri/libpri-1.4-current.tar.gz)
  3. Asterisk (http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-1.8.30.0.tar.gz)
  4. Asterisk-sounds (http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-sounds-1.2.1.tar.gz)
  5. Asterisk add-ons (http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addons-1.4.7.tar.gz)

Step for Installation of an asterisk:

Copy all the files into your server (assuming you have copied all files into /us/src/) 30 Steps for installing asterisk on the system:
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tar -xzf zaptel-1.4.12.1.tar.gz
tar -xzf libpri-1.4.9.tar.gz
tar -xzf asterisk-1.4.20.tar.gz
tar -xzf asterisk-sounds-1.2.1.tar.gz
tar -xzf asterisk-addons-1.4.7.tar.gz
cd zaptel-1.4.12.1
./configure
make
make install
make config
service zaptel start
cd ..
cd libpri-1.4.9
make
make install
cd ..
cd asterisk-1.4.20
./configure
make
make install
make samples
make config
cd ..
cd asterisk-sounds
make install
cd ..
cd asterisk-addons-1.4.7
./configure
make
make install
service asterisk start

Step for Installation of an asterisk:

Copy all the files into your server (assuming you have copied all files into /us/src/) 30 Steps for installing asterisk on the system:
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tar -xzf zaptel-1.4.12.1.tar.gz
tar -xzf libpri-1.4.9.tar.gz
tar -xzf asterisk-1.4.20.tar.gz
tar -xzf asterisk-sounds-1.2.1.tar.gz
tar -xzf asterisk-addons-1.4.7.tar.gz
cd zaptel-1.4.12.1
./configure
make
make install
make config
service zaptel start
cd ..
cd libpri-1.4.9
make
make install
cd ..
cd asterisk-1.4.20
./configure
make
make install
make samples
make config
cd ..
cd asterisk-sounds
make install
cd ..
cd asterisk-addons-1.4.7
./configure
make
make install
service asterisk start
If all above commands run well then we have been installed new asterisk server on the current system.

Creating first sip extension:

Please add following line into sip.conf (/etc/asterisk/sip.conf)
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[common](!) ; this is template.
type=friend
context=internal
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723
allow=g729
dtmfmode=rfc2833
 
[1000](common)
username=1000
secret=1000
 
[1001](common)
username=1001
secret=1001
 
[1002](common)
username=1002
secret=1002
 
[1003](common)
username=1003
secret=1003
 
[1004](common)
username=1004
secret=1004
Above we have created 5 extensions that can be used any sip client (xlite,cisco sip phone, ATA). All users will get registered. If it does not work then check out the firewall settings. Please disable those settings until setup is completed.

Creating First Dialplan:

As all extension cannot talk each other unless we did not configure its dial plan. We have to open extension.conf (/etc/asterisk/extension.conf). Add following lines [internal] exten=> _XXXX,1,Dial(SIP/${EXTEN}) Now the entire configured phone can talk. This makes asterisk simple platform in PBX, not many skills require to develop office PBX.

Creating First SIP Trunk:

Asterisk can make outbound and inbound calls, for outbound we require providing to terminate our calls, and did to get call routed to our system so for that we require having public IP.

Add following code to sip.conf:

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[trunk]
type=friend
context=internal
host=
disallow=all
allow=ulaw
allow=alaw
allow=g723
allow=g729
dtmfmode=rfc2833
 
After the update our sip.conf looks as follows:
 
[common](!) ; this is template.
type=friend
context=internal
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723
allow=g729
dtmfmode=rfc2833
 
[1000](common)
username=1000
secret=1000
 
[1001](common)
username=1001
secret=1001
 
[1002](common)
username=1002
secret=1002
 
[1003](common)
username=1003
secret=1003
 
[1004](common)
username=1004
secret=1004
 
[trunk]
type=friend
context=internal
host=
disallow=all
allow=ulaw
allow=alaw
allow=g723
allow=g729
dtmfmode=rfc2833
 
Now you have to add one line to extension.conf:
 
exten =&gt; _XXXXXXX.,1,Dial(SIP/trunk1/${EXTEN})
 
So our extension.conf looks like:
 
[internal]
exten=&gt; _XXXX,1,Dial(SIP/${EXTEN})
exten =&gt; _XXXXXXX.,1,Dial(SIP/trunk1/${EXTEN})
As the above setting, it is simple to create an IP-PBX with the outbound trunk. Gventure will provide you one of the best services and will fulfil all of your demands in the Asterisk Development.

Tagged:  

Cyclic Replication Between MySQL 5.1 DB

Currently we are doing replication between two servers i.e. VPN3 and VPN4. Our project needed persistent backup of the business critical database on another server as we don't want to loose any data while server goes down. So we have implemented below mentioned steps. We are using MySQL 5.1 version of mysql. Any higher version is also suitable, as performance of replication has been increased in MySQL 5.6 and above. So below mentioned steps are tested successfully on MySQL5.1, however the same steps might work for other version, that you need to sort out.

  • Step 01: Add my.cnf files as mentioned in the /etc/my.cnf (this holds several mysql environment variables and values for smooth performance of the mysql).Run below queries on both the server to create a user which will handle replications between these two servers. mysql > create user 'replicator' identified by 'replicator';
  • Step 02: Once added these files according to server replication architecture you are following.Reboot the services of mysqld to take effect of changes. mysql> Service mysqld restart on all server you want to do replication settings.(Currently we are doing between two servers, but cyclic replication can be done between multiple servers as per project need).
  • Step 03: Create GRANT access for replication user on each server i.e. VPN3 and VPN4 On VPN3 run following command on mysql CLI MYSQL>GRANT REPLICATION SLAVE ON *.* TO 'replicator'@'10.3.140.70' IDENTIFIED BY 'replicator'; MYSQL> FLUSH PRIVILEGES; On VPN4 run following command on mysql CLI MYSQL> GRANT REPLICATION SLAVE ON *.* TO 'replicator'@'10.34.188.148' IDENTIFIED BY 'replicator'; MYSQL> FLUSH PRIVILEGES; You can verify the replication user added on mysql by running following commands. mysql> select user,host from mysql.user; Also verify the permissions status by running below command mysql> select * from mysql.user where user='replicator';
  • Step 04: Run mysql query on VPN4 server MYSQL> SHOW MASTER STATUS; +------------------+----------+--------------+------------------+ | File             | Position | Binlog_Do_DB | Binlog_Ignore_DB | +------------------+----------+--------------+------------------+ | mysql-bin.000002 | 35271224 |              |                  | +------------------+----------+--------------+------------------+ 1 row in set (0.00 sec) and then run below mentioned mysql command on slave i.e. on VPN3 server with above mentioned details of File and Position in it. MYSQL> CHANGE MASTER TO MASTER_HOST='192.168.1.21', MASTER_PORT=3306, MASTER_USER='replicator', MASTER_PASSWORD='replicator', MASTER_LOG_FILE='mysql-bin.000002', MASTER_LOG_POS=35271224;
  • Step 05: Run mysql query on VPN3 server MYSQL> SHOW MASTER STATUS; +------------------+----------+--------------+------------------+ | File             | Position | Binlog_Do_DB | Binlog_Ignore_DB | +------------------+----------+--------------+------------------+ | mysql-bin.000003 | 37416567 |              |                  | +------------------+----------+--------------+------------------+ 1 row in set (0.00 sec) and run this mysql command on slave i.e. on VPN4 server with above mentioned details of File and Position in it. MYSQL> CHANGE MASTER TO MASTER_HOST='192.168.1.21', MASTER_PORT=3306, MASTER_USER='replicator', MASTER_PASSWORD='replicator', MASTER_LOG_FILE='mysql-bin.000003', MASTER_LOG_POS=37416567;
  • Step 06: Run mysql query on VPN4 server MYSQL> START SLAVE; Wait 5 seconds , then on Server VPN4 MYSQL> SHOW SLAVE STATUS\G If Slave_IO_Running=Yes and Slave_SQL_Running=Yes, replication is working on VPN4
  • Step 07: Run mysql query on VPN3 server MYSQL> START SLAVE; Wait 5 seconds,then  on Server VPN3 MYSQL> SHOW SLAVE STATUS\G If Slave_IO_Running=Yes and Slave_SQL_Running=Yes, replication is working.

Now replication is successfully set on both the servers. Changes from one server will automatically copied to another mysql server. Changes always flow from Master Server to Slave Server. As this is asynchronous process so there is slight delay in the copy of data from one server to another server. Generally this process is used for persistent backup of the DB to another server. Since we have created both servers as Master and Slave as well. So any changes on server VPN4 will be reflected back to VPN3 and vice versa.

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