Lab02 Asterisk – SIP Server Installation and Setup I. IntroductionThe purpose of lab02 is to install and configure a SIP proxy server and to make phone calls through the proxy server. To...

Please see the attached file!


Lab02 Asterisk – SIP Server Installation and Setup I. Introduction The purpose of lab02 is to install and configure a SIP proxy server and to make phone calls through the proxy server. To demonstrate the success of the lab work, students need to monitor and capture the call flows in tcpdump and present the screenshots in the lab report. The installation is on the virtual machine environment, and the platform is Oracle VirtualBox which is available for both Windows and Mac. The software distribution for the VM is SNG7-PBX-64bit-1805-1 which is based on the following components: · FreePBX 14 · CentOS 7.5 (Linux) · Asterisk 13 (SIP Proxy) II. Task 1: Oracle VirtualBox Step 1: From the Oracle VirtualBox download site, download the latest version of VirtualBox for your OS. http://www.virtualbox.org/wiki/Downloads Step 2: Install VirtualBox · Install · Choose the default setting. · Choose “not to start” after installation Confirmation: After the installation is complete, you will see the Oracel VM Virtual Box on your desktop. III. Task 2: FreePBX/Asterisk Installation Step 1: familiar with the Asterisk on-line guide Introduction: https://wiki.asterisk.org/wiki/display/AST/Beginning+Asterisk Step 2: Download Asterisk Server (aka FreePBX): https://www.freepbx.org/downloads/. Note that I tried and tested a newer version but failed to install it. You may try the newer version and check if it works on your environment. After download, you should get the following disk image file. SNG7-FPBX-64bit-1904-2 Option 1: keep two copies of the FreePBX file. One copy is for the installation, and it must be removed after the installation. Option 2 (recommended): save the file on a USB drive, and do the installation from the drive. After the installation, remove the drive. Step 3: Open VirtualBox and create a new VM: Linux (64-bit) Allocate RAM and disk space for the new VM. a. RAM: 2G (preferred) or 1G (minimum) b. Disk space (VDI or VMSK): 20G (fixed or dynamic) You will have a new VM under VirtualBox. Note that Asterisk is not installed yet. Step 4: Start the new VM and install Asterisk from the downloaded disk image file (1904-2). Click the folder to select the disk image file Reference the Asterisk guide for the details of the installation https://wiki.asterisk.org/wiki/display/AST/Installing+AsteriskNOW Step 5: After clicking the button, the installation will start. Choose the default (Asterisk 13) for your installation. ►The installation will ask to set the root password, and save the password in a secured place. The installation process takes a long time (> 15 min) so be patient. Step 6: After the installation, do not reboot. Do not reboot. Close the VM Step 7: Close and power off the VM. Step 8: Remove the file SNG7-FPBX-64bit-1904-2. This step is very important; otherwise, the installation process will start all over again. Step 9: Change the network setting to “Bridged Adapter” Then restart the VM of FreePBX(Asterisk). Ignore the warning message. Step 8: Use the root account to access the VM. This is the console session of FreePBX. Step 9: Check the service of asterisk and its version. Must show the date. Screenshot-1 Screenshot-2 Option: instead of using the console for administration, the recommendation is to use a terminal session (e.g., putty) to login to FreePBX. You can create multiple terminal sessions for various tasks. You can also change the size of the terminal session to get better-look screenshots. IV. Task 3: Asterisk Configuration Step 1: Lab02 does not use /etc/asterisk/pjsip.conf, and move it out. cd /etc/asterisk mv pjsip.conf pjsip.conf.bak Step 2: Use an editor (nano or vi) to edit two Asterisk files in /etc/asterisk: extensions.conf, sip.conf, You should remove all existing content (except comments) in these two files and add the content as specified below. Note: the secret is the password of the SIP account (6001). It will be used in the X-Lite configuration. After editing the configuration files, restart asterisk. After that, check the asterisk service is running. Note that whenever you modify the configuration files, you need to restart asterisk. service asterisk { stop | start | restart | status } V. Asterisk Testing Step 1: Setup a simple testing environment with a single computer. The FreePBX is on the virtual machine, and X-lite is on the physical machine. Figure- 1 Step 2: Back to your host computer (Windows). Start X-lite and setup the user account. Asterisk IP address (VM IP address) Step 2: Save the setting and X-lite should show the registered ID. Close X-lite. Step 3: On the FreePBX server, start tcpdump to monitor the traffic. Note: it is recommended that you use a terminal session to login to the FreePBX server. You need to get the IP address of FreePBX and use putty (or any terminal software) to login (the root account) to FreePBX. You can easily change the size and font of your terminal session and it is more convenient than the console. It is also better to take the screenshots from the terminal session than from the console. Step 4: start X-lite. Step 5: You should capture the SIP REGISTER message (in tcpdump) Screenshot-3 Note: you may ignore the SUBSCRIBE messages in your screenshot. Step 6: You may also try to call yourself and will get a recording that the phone is not available. Step 7: From Xlite, dial 100 and you should hear the call back message: Hello World. Do it again (dial 100) and use tcpdump to capture the traffic (as Step-5). Screenshot-4 VI. Asterisk - Proxy Calls Step 1: Setup you network with two SIP clients. Figure 2 Step 2: Follow the example of the 6001 SIP client, edit sip.conf and extensions.conf to add two new extensions that are assigned to you in lab01. (2016, 2017) Call each other. Capture the SIP messages on the caller. Step 3: On FreePBX, start tcpdump. Step 4: Make a phone call from Client-X to Client-Y (or from Client-Y to Client-X). If you hear ringing, answer the phone. Wait for a few seconds, hang-up (on-hook) from the callee side. Note: it is important to terminate the call from the callee side so that you can capture the SIP address of caller. Step 5: Capture the SIP traffic on the Server. Note the SIP messages and compare them with the call flow discussed in the class. Screenshot-5 · This conclude the successful installation of Asterisk as a SIP server. ================= CUT HERE ======= Lab02 Report (Asterisk) Name: ______Nebal__________ Assigned Phone Numbers: ___2016_____ ____2017_______ 1. Screenshot-1: Time stamp of Asterisk service 2. Screenshot-2: Asterisk Version 3. Network Diagram (Figure 1): Single Computer Testing (must show the IP addresses) 4. Screenshot-3 (SIP Registration). The IP addresses must be consistent with Figure 1. 5. Screenshot-4 (SIP Call, playback) 6. Network Diagram (Figure 2): two-computer testing (must show the IP addresses) 7. Screenshot-5 (SIP calls) – the call flow must show a. Both of your assigned phone numbers (as assigned in lab01) b. IP addresses must be consistent with Figure 2. c. must show INVITE, 180 Ringing, 200OK, and BYE (BYE is from callee). 8. Conclusion a. discuss if the learning goal is met. b. problems of doing the lab and how you resolve the problems c. interesting findings d. miscellaneous learning experience 10/11/2022NET464-lab02 (Extra Credit)9 | Page