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Design and Implementation of IP-PBX Server Using Asterisk 1 2 HARSHADA JAGTAP , PROF

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Design and Implementation of IP-PBX Server Using Asterisk 1 2 HARSHADA JAGTAP , PROF ISSN 2319-8885 Vol.04,Issue.17, June-2015, Pages:3235-3238 www.ijsetr.com Design and Implementation of IP-PBX Server using Asterisk 1 2 HARSHADA JAGTAP , PROF. D.G.GAHAN 1PG Scholar, Dept of E & T, Priyadarshini College of Engineering, Nagpur, India. 2Professor, Dept of E & T, Priyadarshini College of Engineering, Nagpur, India. Abstract: An Internet Protocol Private Branch Exchange (IP-PBX) is a complete telephony system. It resides on your network using existing LAN. IP-PBX provide free of cost, without SIM card calling. This system uses only one computer system as a server with Linux based operating system called “CentOS”. The telephony communication is based on Session Initiation Protocol (SIP) which enables the communication between user softphones managed by Asterisk one of the first open source PBX software package that runs on a wide variety of operating system. By implementing this server we can make communication by using laptop, computer having software called softphone and also using mobile phone having Wi-Fi facility. The traditional EPABX telephony system can be replaced by this IP-PBX system which is advance technology of communication. Keywords: Internet Protocol Private Branch Exchange (IP-PBX), Session Initiation Protocol (SIP), CentOS, Asterisk. I. INTRODUCTION delete user extension and is flexible to move the user Many organizations use Electronic Private Automatic extension without disturbing the existing network. This Branch Exchange (EPABX) for telephony communication operating system consists of telephony package called with internal employees and with the outside word. The user “Asterisk”. Asterisk is one of the first open sources PBX extension is connected to central electronic system with long software package that runs on a wide variety of Linux based copper cables. This system works like a mini telephony operating system. This package consists of several features exchange. If we have two telephone lines it can be used by such as Interactive Voice Response (IVR) , Phonebook, Ring eight lines and calling can be done from these eight numbers group, Voice mail, Call waiting, Conference calling, Call at the same time by having only two telephone line or hold, Call transfer etc.[3][2]. Asterisk supports audio number. But EPABX system uses lot of maintenance work as protocols such as SIP which is Session Initiation Protocol well as requires extra hardware and wiring for the new user used for audio communication [4]. extension. Also EPABX is less secured and is less flexible. To overcome, traditional EPABX is replaced by Internet II. NETWORK ARCHITECTURE Protocol Private Branch Exchange (IP-PBX). Voice over Internet Protocol (VoIP) has become a very popular technology during last decade. VoIP originated in mid-90, when hobbyists began to realize the potential of sending voice data packet over internet. The great development of VoIP protocols and codec has enabled a remarkable change in the transmission of vocal communications, evolving from classical circuit switching to a more efficient packet switching model.[5][6] The VoIP PBX system for organization use the backbone of LAN on which the extensions where configured using computer system [1]. There is a central server called IPPBX server and the users get connected through their computers and/or laptops via software called softphone. The softphone can act as a user extension number which could also run on mobile phones who has Wi-Fi facility. The project aim is to Fig1. Network Architecture. design IP PBX server, for this a Linux based operating system called “CentOS” is used. The main feature of the Fig1 shows network architecture of IP-PBX server. operating system is sophisticated installation and Architecture consists of IP-PBX server, Wi-Fi router and SIP configuration of the system. Moreover it is easy to add and phones. SIP phones can be hard phones which work like a Copyright @ 2015 IJSETR. All rights reserved. HARSHADA JAGTAP, PROF. D.G.GAHAN normal telephone and soft phone which are our laptops or o # make install computers. Xlite software is used for soft phone calling. Compile dahdi- Mobile phone having Wi-Fi facility is also used for calling. o #Cd /usr/src/dahdi_version For Smartphone CSIPSIMPLE application is required. o # make clean A Linux based operating system called “CentOS” is used. o # make This operating system consists of telephony package called o # make install “Asterisk”. Asterisk is one of the first open sources PBX software package that runs on a wide variety of Linux based B. SIP Framework operating system. Session Initiation Protocol (SIP) commonly used in VoIP phones takes care of setup and teardown of calls. Figure 3 shows SIP framework for IP-PX. The transaction starts with user1 who sends the invite request to user2 via asterisk server. The sip phone on receiving the request starts ringing and the feedback message is sent to user1. Once user 2 receives the call „ok‟ response is sent to user1 and the connection is setup between user1 and user2. Fig2. X-Lite Softphone. SIP phone configuration is done on windows, Linux or Mac system with Xlite software. Configuration is done by entering SIP proxy setting. Figure 2 shows Xlite softphone with SIP account. After entering the details of the extension i.e. user ID, password, domain etc. extension is created. A. Asterisks Implementation During the installation of Asterisk some packages are Fig3. SIP framework for IP-PBX. required such as Zapata Telephony drivers (zaptel), PRI libraries (libpri), Asterisk sound package. Once the III. SMARTPHONE CONFIGURATION installation is completed extensions can be created. Mobile phone having Wi-Fi facility is also used for calling. Following are the commands used for installation of Asterisk For Smartphone CSIPSIMPLE application is required. package-configure is used in order to determine what CSIPSIMPLE is Google Android VoIP application using applications and libraries are installed on system. Make clean SIP. It is a open source software. User extension is created by is not always necessary but it is good to run before entering the details of the user. CSIPSIMPLE configuration recompiling any of the modules. Make sample command is done as follows- install the default configuration files. Account name: (name of user) Compile Zaptel- Caller ID: (Extension number) o #Cd /usr/src/zaptel_version Server: (IP Address of server) o # make clean Username: (Extension number) o ./configure Password: (Created in server) o #make menuselect o # make o # make install Compile asterisk- o #Cd /usr/src/asterisk_version o # make clean o ./configure o #make menuselect o # make install o Make samples Compile libpri- o #Cd /usr/src/libpri_version o # make clean o # make Fig4. Smartphone Configuration. International Journal of Scientific Engineering and Technology Research Volume.04, IssueNo.17, June-2015, Pages: 3235-3238 Design and Implementation of IP-PBX Server using Asterisk IV. IVR AND RING GROUP Interactive voice response is a technology that allows us to interact with computers through the use of voice and DTMA tones input via keypad. Customer calls on IVR number, call is answered by an automatic attendant. On answering the customer gets a recorded voice and on dialing the required extension number the customer can reach the required person. To dial multiple extensions at the same time ring group module is used. All the multiple extensions will ring at the same time. For example, we can create a ring group of required members and assign that group a single contact number. On dialing that contact number all the members of the group will receive call. Ring group and IVR can be used together. Ring group can be set as destination in IVR, so that when a user presses a number, the number assigned to that particular group will ring. Fig5 shows Fig6. X-lite softphone showing incoming call from user id working of Interactive Voice Response and Ring Group. 5000. When the caller calls on IVR number, interactive voice Fig 6 shows successful calling from user id 5000 to user response is activated and the caller gets the recorded voice. 5003 using X-lite softphone. Once all the required extensions This is followed by menu1 where the recording is playing are created calling can be done between the users. like press 1 for Bachelor of Engineering or press 2 for Master of Technology. As the caller press the number call is diverted to the respective menu option. Then in menu 2 there are three choices- press 1 for project, press 2 for training, press 3 for others. According to the choice of caller the call is diverted. For each project, training and others group of members is created and single number is assigned respectively. On dialing that number all the members of the ring group will ring at the same time. Fig7. X-lite calling user id 5003 and asterisk console showing all the activities of users. Fig7 shows X-lite calling user id 5003 and asterisk console showing all the activities of users. Every single detail of the user is shown on asterisk console. The transaction starts with user 5000 making an invite request for user 5003. The sip phone on the receiving the invite request, starts ringing. Once Fig5. Working of IVR and Ring Group. the call is accepted asterisk console shows 5003 answered 5000. In this way all the activities of users are recorded on V. RESULT AND DISCUSSION Asterisk console. The IP-PBX system resides on our network using our VI. CONCLUSION existing LAN. The server is only a short distance away, so This paper describes design and implementation of IP- signaling distance and time (latency) is very short. A PBX server using Asterisk, which is an open source PBX business owned IP-PBX will usually result in lower averaged software package. By implementing this server we can monthly cost especially for system with higher no.
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