Dr. Adeel Akram Telecommunication Engineering Department UET Taxila Lecture Overview Introduction 12 Steps to VoIP Case Studies Asterisk Trixbox Brekeke Free PBX miniSipServer Introduction In 2004, with voice-over-IP capturing over 35% of all new enterprise voice shipments Since then, most enterprises have begun their road to implement VoIP, ranging from new all-IP PBXs to service provider hosted solutions (IP Centrex). With the problems encountered by early adopters largely resolved and mainstream organizations moving from “If VoIP” to “When VoIP” decisions Lets begin to layout a roadmap for successful VoIP implementation. VoIP RoadMap The lessons learnt from deployment of large data networks can help organizations in developing a roadmap to help guide enterprises to move to VoIP successfully. It is not a simple recipe to follow, but it outlines a thoughtful approach containing an admixture of the ingredients that make a successful plan. The skills and determination of the team leader and the team members (including vendors and service providers, at the right times), will determine the success of the overall project. 12 Steps to VoIP The key success factor is building a plan to guide the implementation from initial brainstorming about what your enterprise can accomplish with VoIP through solution selection and implementation, closing with a measurement program to evaluate the benefits-to determine how successful your plan attained its objectives. And how to evolve the initial implementation to further success over its lifetime. Successful users do not select a box, but have implemented a business solution that will provide increasing benefits over time to the enterprise. Step 1 Create and educate a cross-organization project team telecom, datacom, financial, planning, business, marketing, sales, customer support, maybe even customers, business partners and suppliers, etc. Step 1 (Contd.) A committed interdisciplinary team is the key to project success. The goals of this team are to determine what to do, how to do it and to build performance benchmarks to evaluate progress and measure the value received from what will be a substantial investment over many years. The need for an interdisciplinary, cross-functional project team is critical. It must not only understand the technologies involved in convergence but, more importantly, ensure that the project will support and enable the business goals of your enterprise. Step 2 Survey capabilities and applications Step 2 (Contd.) Key areas of investigation will include: networking capabilities, system and device features and functions, open interfaces to business applications, multimedia messaging, web-based applications, mobility capabilities and, where appropriate, contact centers. Of emerging importance is employee presence (enabled by instant messaging) and its opportunities for improved collaborative work. Step 2 (Contd.) Conduct a survey of the breadth and depth of the capabilities being offered and planned by the various vendors. Not just today’s availability, but a thorough look at multi-year solution roadmaps should be considered. Vendors who support both embedded systems evolution as well as new pure IP-systems should be considered. Step 3 Determine how to apply VoIP within your enterprise Step 3 (Contd.) Understanding the business plan and future directions of your organization is an early gating step. As Benjamin Franklin (a very early VoIP planner) is credited with saying, “If you don’t know where you’re going, any road will get you there!” Consider how the offerings of the various suppliers might be applied in your business today and tomorrow and where there are existing or potential opportunities to achieve business benefits. Step 3 (Contd.) Benefits include, for example: Effectively manage geographical dispersion Extend, coordinate or disperse contact centers Support mobile/remote workers, road warriors and tele- workers Manage acquisitions of new locations Improve customer service Reduce real estate costs Step 4 Audit data network (LAN and WAN) Step 4 (Contd.) Voice places special performance requirements on your underlying network infrastructure. Be certain that your infrastructure can support the real-time, quality, class-of-service and reliability needs of business voice communications. Fortunately, this is a service that most vendors and providers now offer and audits are conducted routinely and draw upon the many experiences learned during VoIP’s gestation period. Step 5 Find the business hook(s)! Step 5 (Contd.) Now that you know what VoIP offers and what are the likely additional capabilities coming down the road, your team needs to apply this knowledge to identify how your enterprise can benefit from a VoIP implementation. The questions about how and where to use VoIP solutions and applications need to be answered for your enterprise. And this view needs to be broad and multi-year. The implementation will not be static and, in many cases, will extend over several years. Step 6 Develop the business case(s) for your enterprise Step 6 (Contd.) To develop an analysis based upon a quantification of the benefits that your organization can achieve by implementing VoIP and each of the VoIP-enabled applications your team identified as having business value. This evaluation of business value begins to point up the true value of having an interdisciplinary team charged with the overall VoIP analysis. All of the key organizations are represented, and they can apply their organizational knowledge towards making the analyses “real” and implementable. Step 7 Develop a detailed functional and implementation plan Step 7 (Contd.) While the business case(s) are being developed, a plan to implement the solution(s) needs to be developed. A good idea might be to develop the implementation plans in parallel with the business case(s). These detailed implementation plans will identify the key dependencies and major work programs that will be needed for your organization to obtain the benefits proposed by the applications posed in the business cases. They are two sides of the same sheet of paper: What can be achieved, and The steps to get there. Step 7 (Contd.) The implementation plan serves two additional purposes: It develops a prioritization of what can be accomplished over what time frames. It develops a view of the resources required to implement the solutions. Both of these plan elements are critical to the success of obtaining executive buy-in to the plan and identifying and receiving the required budgetary commitments. Step 8 Obtain internal commitments and budget Step 8 (Contd.) Because we have already evaluated the overall impacts on the organization--the costs, benefits and financial implications (investment dollars over time and the returns on those investments) Budgetary approvals, while certainly never easy for a major technology investment, are certainly more likely to be obtained if the team making the recommendations represents all of the involved and affected organizations, have identified both the benefits and costs and proposes a timeframe that the decision makers are likely to view as both reasonable and prudent. Step 8 (Contd.) It is likely that only the first few elements of the overall project may be approved in the initial request, and that the full plan may require several budget cycles to be fully approved. Because of the focus on infrastructure upgrades, it is also likely that the early approval elements may not have the best return on investment. This also shows the value of having undertaken a full, multi-year plan. Your organization will be making an investment with returns that become realized only after several phases are implemented. Step 9 Implement your plan, and be prepared to adjust Step 9 (Contd.) Having obtained the approvals to proceed, implement the plan developed earlier. This should include making the technology choices and vendor acquisition decisions, business process and systems re-design, as required. Since it may have been some time since the earlier network assessment (Step 4) was conducted and changes may have been made to applications or volumes, etc., an update assessment should be included as an early part of the implementation. Step 10 Make feedback loops built into your roadmap, and adjust appropriately Step 10 (Contd.) Frequently overlooked is the ability to monitor progress and adjust the roadmap as things progress We all know from experience that major projects generally take more time than estimated, cost more than budgeted and often return less than expected. Having appropriate feedback embedded into your process makes the required adjustments less dramatic and more manageable than when not anticipated and folded back into the implementation program. Such additional forced communication across the organization can serve to improve the quality of results. Step 11 Determine how well the benefits track the expectations Step 11 (Contd.) Taking the time to do this evaluation is an important step that should not be overlooked. This is especially true if the project was broken into several phases for implementation. The next-phase results need to be based upon a good understanding of the results of the prior implantation phases. Step 12 Celebrate your success Step 12 (Contd.) Having successfully brought VoIP into your enterprise, take a little time to celebrate your successes. But be careful to not celebrate too much...or you may need to enroll in another kind of 12-step program Step by Step Practical Implementations of VoIP Solutions Asterisk The Linux based Open Source Software Asterisk has become the de facto standard in modern VoIP PBX systems. Because of its powerful and flexible structure Asterisk is also being used as the VoIP engine in commercial PBX products, partly because some PBX manufacturers have realized that it would not make much sense to compete against the development momentum of this open source project and end up having an expensive look-alike that no one wants to write interface software for. Asterisk The flexibility of Asterisk comes with a price, though. There is no user friendly interface included and the command language and syntax have a very steep learning curve. Even though some VoIP enthusiasts are configuring their Asterisk PBX box from the command line interface, this is not practical for a commercial product. Managing a PBX system this way would be just as absurd as trying to sell a fax machine that needs a computer science diploma to operate. Asterisk O.K, this fax machine comparison is not quite fair because PBX systems in general need much more complex configuration, but this is why the Asterisk PBX Manager Web GUI was developed. It allows configuring and operating an Asterisk based VoIP system as conveniently as with conventional PBX boxes but leaving the door open for much more sophisticated telephony and interface functions. Asterisk Asterisk is basically a a complete PBX engine in software. It runs on Linux, BSD and Mac OSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Even though it behaves as a classical voice exchange there is a major difference in dataflow when using the popular SIP protocol for connecting telephones: Asterisk In principle there is no difference of a telephone being external or internal. This means that a person using a mobile IP telephone, for example a SIP softphone installed on a notebook or pda, this telephone will ring no matter where it is globally located. When using the popular SIP protocol it is even possible to have all voice or video data flowing directly between caller and callee if both are located somewhere external to the Asterisk PBX. Installing Suse Linux 10 and Asterisk Here we show you how to install a complete VoIP Server, including Linux, Asterisk and PBX Manager - Step by Step. Suse 10 Linux Installation Preparing Linux for Asterisk Install MP3 Support Asterisk Installation Procedure PBX Manager Installation Suse Linux 10 Installation Download your all five Suse 10 Open Source CD images and burn CDs from these ISO files. Alternatively you may buy a single DVD from Novell. Insert CD #1 (or the Novell DVD) and install Linux to your local hard disk. For realtime performance reasons be sure to install only the minimal text based Linux operating system and no graphicai user interface such as GNOME or KDE. Other than that, the basic installation steps are quite straigthforward and all proposed default settings can be accepted. Preparing Linux for Asterisk After booting from hard disk log on as user root at the command line. Now we have to use the installation tool yast to install a C Compiler and some additional modules. You may do this interactively by calling yast without any parameters or by entering the following commands: yast -i ncurses-devel yast -i zlib-devel yast -i glibc-devel yast -i gcc yast -i kernel-source yast -i openssl-devel yast -i samba Preparing Linux for Asterisk The Samba Server comes in handy to access the Linux file system directly from any Windows PC over the network. In order to get Samba working you have to configure /etc/smb.conf, which can be done with the vi editor: cd /etc/samba vi smb.conf Look for the line workgroup = xxx and replace it with the name of your Windows Workgroup or Microsoft Domain, for example: workgroup = uettaxila Preparing Linux for Asterisk At the end of this file add the following lines: [root] path = / valid users = root public = no writable = yes directory mask = 777 Save the edited file and leave vi by pressing the following keys exactly as noted: (Esc):wq(Enter). To leave vi without saving any changes enter: (Esc):qa!(Enter) Preparing Linux for Asterisk Now we need to add root as a valid Samba user, give him a password and ensure that Samba runs automatically at system start: smbpasswd -a root (you will be prompted for a password) chkconfig smb on chkconfig nmb on In order to access the Linux partition over the network you have to modify or shut off the Suse Linux firewall. Use yast, go to Security.. / Firewall, stop the firewall and set its startup mode to manual. Preparing Linux for Asterisk After your Linux Server is restarted by entering reboot or shutdown –r now you should be able to see the root share in the Network Neighborhood of any Windows PC on the local network. If not, try to map a network drive similar to: \\192.168.1.100\root, replacing 192.168.1.100 with the IP Address of your Linux server. Use this drive to copy the downloaded Webmin installation file to /usr/src and install it according to the following example (adjust the version numbers accordingly): cd /usr/src/webmin-1.260-1 rpm -i webmin-1.260-1.noarch.rpm Preparing Linux for Asterisk Now you are ready to administer your Linux server from any Web Browser on the LAN. By entering the following URL: http://192.168.1.100:10000 Again, replace 192.168.1.100 with the actual IP Address of your server. Install MP3 Support In order to play Music-On-Hold (MOH) Linux needs to be able to decode and handle the MP3 file format. This can be done in two different ways: by installation of the classic mpg123 module or by using the new format_mp3 module contained in Asterisk Addons. Be sure to install only one of these solutions and not both! Follow the procedure in next slide to install either of the above modules: MP3 playback with mpg123 This is a proven and reliable solution. Without applying the patch below there is a known security issue, but the most serious drawback is the fact that mpg123 is not being actively maintained any more. Download both the latest mpg123 installation source and security patch, copy both to /usr/src und install them as follows: cd /usr/src rpm -i mpg123-0.59s-513.i586.rpm rpm -i mpg123-0.59s-513.i586.patch.rpm MP3 playback with format_mp3 This is a new light-weight mp3 module found in Asterisk Addons V.1.2.3. This is a solution with a more promising future, even though it is not quite ready for production yet. The author states that is has only been tested with Solaris 2.6 and there are known issues with MP3 files encoded with LAME that may lead to Linux crashes. You should also make sure to convert your mp3 files to 8 kHz Mono format in order to avoid high CPU load. For latest installation instructions see the README inside the Asterisk Addons package. Asterisk Installation Procedure Download the latest Asrerisk installation sources (zaptel, libpri, asterisk, addons and sounds) and unpack them into subdirectories. Go to the Linux command line interface and install them in the following order: cd /usr/src/asterisk-zaptel-1.2.6 make Linux26 make config make install cd /usr/src/asterisk-libpri-1.2.3 make make install Asterisk Installation Procedure cd /usr/src/asterisk-126.96.36.199 make make install make samples cd /usr/src/asterisk-addons-1.2.3 make make install cd /usr/src/asterisk-sounds-1.2.1 make install Asterisk Installation Procedure If you have managed to get all these Asterisk modules compiled, congratulations! You are now ready to start Asterisk from the command line simply by entering asterisk. In order to interact with Asterisk from the Asterisk Command Line Interface (CLI) enter: asterisk -r Asterisk Installation Procedure You will see the Asterisk welcome message and this is where you can do all Asterisk debugging and realtime modifications. Try help to see a list of the supported CLI commands. Use the restart now command to restart Asterisk after a major configuration change or (Strg)-z to go back to the Linux command level leaving Asterisk running in the background. The installation scripts should have configured Linux so that Asterisk will be automatically started after system boot. OS Specific instructions Asterisk installation on Fedora Core 2 final Asterisk installation for CentOS 4.x CentOS 5 and Asterisk 1.4.x installation CentOS 5.2 and Asterisk 1.6.x installation Asterisk installation tips How to install on all kinds of operating systems and distributions PBX Manager Installation Point your web browser to Webmin (http://<Linux server>:10000), go to Webmin Configuration / Webmin Modules and enter the PBX Manager installation file that you have downloaded from here. After a few seconds you will have a new Asterisk entry in the Webmin Menu Server page that may also be accessed directly via: http://<Linux server>:10000/asterisk If this is the first time you started PBX Manager you should go to the Files Menu and deploy the standard configuration shipped with PBX Manager in order to have a basic, working default configuration to start out with. PBX Manager Installation Point your web browser to Webmin (http://<Linux server>:10000), go to Webmin Configuration / Webmin Modules and enter the PBX Manager installation file that you have downloaded from here. After a few seconds you will have a new Asterisk entry in the Webmin Menu Server page that may also be accessed directly via: http://<Linux server>:10000/asterisk If this is the first time you started PBX Manager you should go to the Files Menu and deploy the standard configuration shipped with PBX Manager in order to have a basic, working default configuration to start out with. Online Demo User: demo, PW: insecure Trixbox PBX Installation Trixbox Installation Requirements Dedicated Machine or Virtual machine with the following specifications: 700 MHz processor 10 GB Hard disk At least256 mb of Ram (512 Recommended) Trixbox (IP Phone Server Software) Any Softphone or Hardphone Trixbox Installation Procedure 1) Download trixbox CE 2.6.2 (Stable) from the following link: http://master.dl.sourceforge.net/sourceforge/asteriskatho me/trixbox-188.8.131.52.iso After downloading if you are gonna use it on dedicated machine burn the image into CD otherwise you can use ISO with Vmware or any other virtualization software. Trixbox Installation Procedure 2) Here we assume that we are using virtual machine, although the process is same for dedicated machine. 3) Use .iso with Vmware or any other virtualization software and start the virtual machine. You will be greeted with a cool green screen of trixbox installation. Now press enter to install trixbox. 4) It will prompt you to select language so select the language you wanna use and press OK. 5) It will now ask for your timezone. This is an important step so please select the correct timezone. Trixbox Installation Procedure 6) Now you will need to select the password for root user select the password and rewrite it to confirm and press OK. 7) Installation will be started in 1 minute and it will reformat your harddisk and install trixbox. 8) When installation is done your will be restarted and you will have the following screen very soon: Trixbox Installation Procedure Trixbox Installation Procedure 9) There are 2 username and password you will need to use to login to trixbox. Username and password you will need to use for logging in to console. Username : root Password is the password you supplied during installation. Username and password you will need to use for logging in to Gui of trixbox. Username : maint Password : password Trixbox Installation Procedure 10) We need to enter an IP for IP Phone server here. Enter the following command: system-config-network and enter an static IP which you want to use. It is always better to use LAN IP at this point. 11) After assigning an IP you can login to GUI. Open up IE or any other browser and enter the ip you selected in step 10. 12) After logging in to GUI You will receive the Trixbox interface as shown in the following figure: Trixbox Installation Procedure Trixbox Configuration 13) Now it is the time to add first extension. Click on PBX > PBX Settings > Extension Select Generic SIP Device and click submit. You will need to enter the following details to add an extension. User extension (It can be anyone like 202,302,402 and so on. Display Name (Enter any name you want) Secret (Write any word you want to be used as a secret for this extension). Trixbox Configuration 14) You can see all the extensions on right side. Trixbox Configuration 15) Now its the time to configure softphone. Lets suppose you have downloaded and installed X-Lite. When you will run X-Lite for first time it will ask for you SIP account. You will have the following dialog box. Trixbox Configuration Trixbox Configuration In the first box of display name enter the display name you entered while adding an extension on trixbox. Second box is for username.It is the extension number you entered while adding an extension on trixbox. Third box is for password enter the secret you entered while adding an extension on trixbox. Fourth is for authorization username it is also as extension number you entered while adding an extension on trixbox. In the fifth box of domain enter the IP address of your trixbox server. Check register with domain and receive incoming calls. select domain and click apply then OK Trixbox Configuration You are now done and you will have a final screen looks like the following: Trixbox Configuration Now let me send a test call to my number and i will have the following: Brekeke SIP Server Requirements A simple/inexpensive LAN switch (or hub) for setting up a small LAN environment A Windows PC (i.e., Windows XP, Windows Vista) BudgeTone-100 from Grandstream X-Lite Softphone*. If you do not have X-Lite Softphone on your PC, you can download it from CounterPath’s webpage. (Download X-Lite Softphone) *When using X-Lite Softphone or any other SIP softphones, you need a microphone and speaker for voice communication. Brekeke SIP Server Installation Setting up Windows PC with a static IP address: Configure your PC with a static IP address: 192.168.0.102 and subnet mask of 255.255.255.0: Click [Start]->[Control Panel], click the [Network Connection] icon. Double click [Local Area Connection], click the [Properties] button. In the "Local Area Connection Properties" dialog box, double- click the [Internet Protocol (TCP/IP)] item. In the "Internet Protocol (TCP/IP) Properties" dialog box, select [Use the following IP address]. Set: [IP address]: 192.168.0.102; [Subnet mask]: 255.255.255.0 Click [Ok] to save the settings Make sure your PC is connected to the LAN switch/Hub with an Ethernet cable. Brekeke SIP Server Installation Download and Install Brekeke SIP Server: Download the Brekeke SIP Server installer from its website download page. Free trial is available. After successfully installing the product, a Brekeke SIP Server icon will appear on your PC's desktop. Activate Brekeke SIP Server Start the Brekeke SIP Server by double-clicking the [Brekeke SIP Server] icon from your PC desktop. Activate Brekeke SIP Server with product ID. Need a Product ID? Login Enter "sa" in both the [User ID] and [Password] fields from the Brekeke SIP Server Admintool screen. Start Brekeke SIP Server. The Status field should indicate that the Brekeke SIP Server is Active. Brekeke SIP Server Installation Setting Up Brekeke SIP Server for User Authentication: The authentication for REGISTER and INVITE is enabled by default on Brekeke SIP Server, which is set at Brekeke SIP Server > [Configuration] > [SIP] >[Authentication] To register SIP UA with Brekeke SIP Server authentication on, create [User Authentication] account for each SIP UA is needed. Do the following to set up authentication accounts at Brekeke SIP Server: Brekeke SIP Server Installation Choose Brekeke SIP Server >[User Authentication] >[New User], and set. [User]: UA UserName. [Password]: SomePassword. [(Confirm)]: SamePassword. [Name]: Description to remind whose auth account it is. Click [Add]. To make sure the user authentication information is added for this user, check from [View Users]. Brekeke SIP Server Installation In this example, we will create Authentication accounts from SIP UA Grandstream and xLite. In the following setup, the authentication user IDs will be the same as the phone numbers (SIP user ID). It is requied by default settings of Brekeke SIP Server authentication. Grandstream BudgeTone-100 User Authentication account at Brekeke SIP Server: xLite User Authentication account at Brekeke SIP Server: [User]: 160 [Password]: 1234 [User]: 555 [(Confirm)]: 1234 [Password]: 5678 [(Confirm)]: 5678 [Name]: Grandstream [Name]: xLite Setting up Grandstream BudgeTone-100 Set up Grandstream BudgeTone-100 with a fixed IP address (Ex: 192.168.0.160) and subnet mask of 255.255.255.0: Set SIP registrar IP, user/phone number and authentication information from [Advanced Settings] tab on the phone SIP Server: 192.168.0.102 (Brekeke SIP Server's IP address set in step 1- 2) Outbound Proxy: 192.168.0.102 (Brekeke SIP Server's IP address set in step 1-2) SIP User ID: 160 (the same as the authentication user set in step 3-2a) Authenticate ID: 160 (authentication user set in step 3-2a) Authenticate Password: 1234 (password set in step 3-2a) Name: any Update the settings and reboot the phone Setting up Grandstream BudgeTone-100 Check Grandstream manual for more information on setting up your BudgeTone-100 phone. Most of the default settings in the Grandstream BudgeTone-100 should work. Now make sure the phone is connected to the LAN switch/Hub with an Ethernet cable From Brekeke SIP Server Admintool -> [Registered Clients], this phone is registered as [User:] 160; [Contact URL:] sip:email@example.com Setting up X-Lite as a SIP UA Set up X-Lite as a SIP UA on your PC: Start X-Lite from your PC by choosing [Start]->[All Programs]- >[X-Lite]. From the X-Lite dialog box, go to [▼]->[SIP Account Settings]- >[Properties], enter: Display Name: any User name: 555 (the same as the authentication user set in step 3-2b) Password: 5678 (password set in step 3-2b) Authorization user name: 555 (the authentication user set in step 3- 2b) Domain: 192.168.0.102 (Brekeke SIP Server's IP address set in step 1-2) Check [Register with domain and receive incoming calls] and select [domain] under [Send outbound via:] In order for X-Lite to work as a voice communication device, Brekeke SIP Server Installation Making VoIP Calls To make a call from the Grandstream BudgeTone-100 to X-Lite, follow these steps: Press [Speakerphone] button. Wait for a steady dial tone. Dial 555, and press [Send] button. To make a call from X-Lite to the Grandstream BudgeTone-100, follow these steps: Enter 160 and click on [Dial] button To hang up, click on [hang up] button Free PBX Installation FreePBX Install Guide (CentOS v5.x, Asterisk v1.6.x, FreePBX v2.6.x) Includes every detail in the form of step by step instructions from bare metal to a running VoIP PBX in about 2 hours. miniSipServer IP PBX In normal, we need to setup three main components: IP-PBX, phones (or soft-phones) and VoP carriers' service that let you call other peoples on the PSTN (Public Switched Telephone Network, it is our traditional telephone network). miniSipServer IP PBX IP PBX There are lots of IP-PBX. Some of them are hardware based devices, some of them are software based server. miniSipServer is a professional SIP PBX for Windows system. It has all features we need. Most important, it is so easy that we can setup and run it in 10 minutes! miniSipServer IP PBX SIP Phones We can buy SIP phones from Grandstream, Cisco, Linksys, etc. If we still want to use our traditional analog telephone, we can buy a SIP adapter. To get started, we use a softphone and run it in the same computer or another computer. We will try Xlite softphone. miniSipServer IP PBX VoIP Carriers Lots of VoIP carriers can provide SIP services. The following VoIP providers are suggested: Sipgate. VoiceTrading. BroadVoice. Once register to these VOIP providers, they will give you a SIP account information, such as SIP server address, account name and password, etc. These information will be used when we configure our IP-PBX. miniSipServer IP PBX Scenario Following figure describes a simple environment for small business or home based business miniSipServer IP PBX In our demo scenario, the small company only has two member, Holly and G.T. Holly's extension number is 100, and G.T's extension phone number is 101. Both of them have PCs based on windows system. The IP address of Holly's PC is 192.168.1.100. The IP address of G.T's PC is 192.168.1.101. miniSipServer will be installed on another PC whose IP address is 192.168.1.110. The company establish connection with PSTN through a VoIP carrier's network. miniSipServer IP PBX We will follow subsequent steps to establish our VoIP network: Step 1: Setup miniSipServer. Following figure describe it. miniSipServer IP PBX Step 2: Connect local users to miniSipServer. miniSipServer IP PBX Step 3: Connect miniSipServer to VoIP providers' network. miniSipServer Configuration Step 4: Configure some wonderful advance services for our small company, such as auto-attendant, ring group and pick- up. miniSipServer Installation Step1: setup miniSipServer This step could be the easiest one in our configuration. We can download the miniSipServer SETUP file from http://www.myvoipapp.com/download. The latest version is 2.5.4 now. There are several kinds of miniSipServer, such as 20 client, 50 clients etc. For example, "100 clients" version can support 100 extensions. Since there are only two members in our scenario, "20 clients" version is enough. miniSipServer Installation Before we install miniSipServer, please make sure that the PC based on Windows and network are work well. After that, click the .EXE file for miniSipServer setup and run it! There is no configuration required during setup process. If everything goes well, the miniSipServer should run as show in the following figure. miniSipServer Installation miniSipServer Installation Step2: connect local users to miniSipServer When miniSipServer is installed, it will create three default extensions automatically. Please click button "local users" in main window to check it. The default password for these extensions are also 100, 101 and 102. The users information is displayed in the following figure: miniSipServer Installation miniSipServer Installation Configuring Softphones We begin to configure Holly's softphone to connect miniSipServer. Holly use Xlite as her softphone. The Xlite is a wonderful SIP client and can be download from http://www.counterpath.net/x-lite.html. After install Xlite and begin to run it, it will prompt to configure a SIP account. The following figure shows the SIP account setup window of Xlite: miniSipServer Installation miniSipServer Installation Click "Add..." button to add a SIP account as shown in figure: miniSipServer Installation The key configurations are described below: User name 100 Password 100 Authorization user name 100 Domain 192.168.1.110 Register with domain and receive incoming calls Yes Send outbound via domain miniSipServer Installation Click 'OK' button to complete the Xlite configuration. Xlite will try to register to miniSipServer. If it successes, it should display 'Ready' information. miniSipServer Installation Now, Holly's extension has connected to miniSipServer . We can follow the same step to configure G.T' extension. If you have other kinds of SIP clients/phones, they should also be configured with same information. Both Holly and G.T‘s extensions have been connected to miniSipServer. We can check miniSipServer‘s local user information to check their status. Their icons should be blue indicating their online status. miniSipServer Installation After we finish this step, the basic VoIP network is established. Holly and G.T can call each other. Holly can dial '101' to call G.T, and G.T can also dial '100' to call Holly. miniSipServer Configuration Add a new extension in miniSipServer In above configuration, we use the default extensions '100' and '101'. In future, with the growth of company, more and more people will join with us, we need add more extensions to support them. So we can do as following: In the 'local users information' window, please click 'Add' button to add a new extension. miniSipServer Configuration Step 3: connect miniSipServer to VOIP provider We have establish internal VoIP network of Holly and G.T. It is time to establish connection with customers now. Usually, if we want to make call to outside or receive a call from outside, we need a VoIP gateway connect our miniSipServer and traditional telephone or we need VoIP provider to do it for us. We decide to connect our miniSipServer to VoIP providers' network and we select BroadVoice as our VoIP provider. After we request a SIP account from BroadVoice, for example, the account is '723123456', we will use use this account to configure miniSipServer to connect BroadVoice. miniSipServer Configuration In the miniSipServer main window, please click button 'External lines' to add an external line information. miniSipServer Configuration In the pop up window, please click button 'Add' to add an external line with BroadVoice account information. miniSipServer Configuration The key informations is described in example below: External line 7321234567 Password 1234 Peer server address sip.broadvoice.com Peer server port 5060 Share outgoing call with other local users Yes Auto attendant Yes miniSipServer Configuration Since we assume that both Holly and G.T can make outgoing calls, we select 'share outgoing call with other local users'. Here we configure 'auto attendant' to support receiving incoming call from outside. If the external line success to connect to peer server ( VoIP provider's network or VOIP gateway), the icon of the external line should be gray and without cross flag. Now, we discuss some details about making outgoing call and receiving incoming call. miniSipServer Configuration Make outgoing call As we have confirmed in previous sections, Holly and G.T can call each other by dialing their extensions number directly. If we want to make outgoing call to our customers, we must be connected to our VoIP provider. Since the external line is connected to VoIP provider's network ( or VoIP gateway), we can call outside customers, by adding prefix '9' before we dial our customers numbers. In miniSipServer, prefix '9' is the default outgoing prefix which is used to distinguish call type. For example, if the customer's number is '7321234568', we dial '97321234568'. miniSipServer Configuration Receive incoming call When we configured external line before, we has indicated 'auto attendant' at the same time. So when the customer calls in, miniSipServer will prompt him/her to enter extension number. For example, once the customer calls "7321234567" (the external line number provided by VoIP provider and configured in miniSipServer), he/she will hear "Welcome, please enter extension number" and the customer can enter '100' to call Holly or enter '101' to call G.T. The benefit of using "auto attendant" to receive incoming call is that the company can provide only one public telephone number to customers and assign different extensions to each employee. Questions ????
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