Introduction to voip by nikeborome

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									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-1.2.9.1
   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-2.6.2.2.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:160@192.168.0.160
  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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