Practical Guide. How to setup VoIP Infrastructure using AsteriskNOW

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1 Practical Guide How to setup VoIP Infrastructure using AsteriskNOW

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3 Table of Contents 1. Background The VoIP scenarios Before getting started Training Kits Software requirements Conventions Known issues Virtualization versus dedicated hardware Installing AsteriskNOW Installation Screenshots discussed Configuring AsteriskNOW for Scenario Configuration though the Asterisk GUI Setup Wizard Step 1: Hardware detection Step 2: Local extensions settings Step 3: Configuring service providers Step 4: Outbound calling rules Step 5: Voic settings Step 6: User extensions Step 7: Incoming calls rules Advanced options: Asterisk GUI Configuration of ATAs Quick Installation Guide Scenario Scenario Scenario Verify your results Scenario Scenario Scenario Background The first edition of the VoIP 4D Primer, Building voice infrastructure in developing regions released in December 2006 covered the basic aspects of IP Telephony and provided configuration guidelines for the Asterisk PBX for three basic scenarios. This document aims to make the installation of such scenarios even easier. While in the first version of the Guide we configured Asterisk by editing the configuration files, in this guide we are going to use a graphical user interface (GUI). We have reviewed several initiatives that provide a graphical interface to Asterisk and decided to prepare this practical tutorial based on the AsteriskGUI available in Asterisk 1.4.x series. A new distribution known as AsteriskNOW, includes a straightforward installer and all the software packages for Asterisk production and development. Although the distribution is still in beta stage (beta5 in November 2007), it has been designed with a very clean interface and a very intuitive wizard. 1

4 An analysis of other initiatives is available in the document: Making IP telephony knowledge accessible (prestudy1). 2. The VoIP scenarios The three scenarios described in this document are similar to the ones described in detailed in the VoIP 4D Primer. Scenario 1 Creating a local private telephony network in a rural community This scenario consists of a single PBX with a set of clients. Clients can be either softphones, VoIP phones or ATAs. Scenario 2 Interconnecting communities In this scenario we interconnect two PBXs. Local extensions of one PBX are made available to the other and vice versa. Scenario 3 Connecting communities to the PSTN In this final scenario, we have interconnected two PBXs and allow the possibility of reaching the PSTN from any of them. 1 Can be downloaded from under Documentation 2

5 3. Before getting started If you have not read the VoIP 4D Primer ( have a look to the first chapter as it will provide you with the necessary background to understand the basic concepts of IP Telephony. The very minimum hardware requirements for Scenario 1 and 2 is a single PC running Windows and hosting two virtualized installations of Asterisk. You can test the calls using a softphone and the voic service. For Scenario 3, you will need two computers, one of them with a dedicated communication card TDM400p. Alternatively, you can use two Asterisk appliances such as the IP04s Training Kits If you want to run a VoIP training session based on this material, consider at least having one training kit per group as follows: 2 PCs with network cards 2 4 ATAs or (2 4 VoIP Phones) 2 4 analogue phones (if using ATAs) 1 TDM400p card with 1 FXS and 1 FXO ports 1 4 port switch (better a hub, if you can find one!) Access to a PSTN line Alternatively you can use the following training kit 2 IP04 (3 FXS, 1 FX0) 2 4 analogue phones 1 4 port switch Access to a PSTN line 3.2 Software requirements ISO Image of AsteriskNOW VMware Image of Asterisk NOW VMware Player Softphones ; X Lite, Kiax, etc info.org/wiki Asterisk+IAX+clients Wireshark (for debugging, advance users) DHCP Server 2 3

6 Latest Firefox version >= Documentation AsteriskNOW quickstart guide VoIP 4D Primer This document ( Setting up VoIP Infrastructure using AsteriskNOW ) 3.3 Conventions This documentation has used the IP address for the majority of the screenshots. Be aware that you need to use your own IP addresses for your setup. 1. We will create four local extensions in each PBX, with the names 1000, 2000, 3000, We will use the same number as username, callerid and password, i.e. username = callerid = password = 1000 (or 2000, 3000, 4000) ). 3. The voic extension is Scenario 2 and 3 include two different PBXs that should have different IP addresses. 5. Each of the PBXs sees the other PBX as a VoIP Service Provider. 6. The account username: 4646 password: 4646 is created in each of the PBXs for the purpose of routing calls between them 3.4 Known issues These are some of the issues found during the preparation of this tutorial: 1. If you have problems during authentication, consider using the latest Firefox version and/or removing the cache and the cookies of your browser. 2. VMplayer can not boot your image if you have a Windows machine with FAT16 filesystem with a size bigger than 2 GB. Include the line disklib.sparsemaxfilesizecheck= "false" at the end of the VMX file to overcome the problem. 3. AsteriskNOW is still in beta stage. In some cases it is not possible to edit entries after running the wizard. Consider deleting and recreating the entry instead of editing it. 4

7 4. Virtualization versus dedicated hardware There are several ways to install AsteriskNOW on a computer. The method to use depends on your answers to the following two questions: 1. Does you VoIP setup need to be connected to the PSTN (TDM support)? 2. Do you have a dedicated machine for the VoIP setup? If you answer is Yes to the first question, you need to install AsteriskNOW in a dedicated machine. If you do not need to be connected to the PSTN, you have two options depending on if you have a machine available for the implementation (Dedicated machine). If you do not have a dedicated machine you need to install VMware player in your machine and the boot the VMware AsteriskNOW ISO. Thereafter you can install AsteriskNOW virtually, using your VMware installation. If you have a dedicated machine, boot from a CD that contains the AsteriskNOW ISO. 5. Installing AsteriskNOW Install AsteriskNOW The distribution is available in three main flavours: 1. A version that runs on the x86, 32 bit/64 bit processors such as Intel P4 and AMD Athlon XP. 2. A version that runs on the Xen virtual machine. 3. A version that runs on the VMware Player. If you do not have a dedicated machine available or you want to test the software distribution, you should consider using the VMware ISO image. Please note that using the VMware image will not allow you to use any specialized hardware as the PCI TDM400p card. Although, it is not mandatory, consider having a DHCP server available on the network. Log into the web interface Open a browser and go to: 5

8 The web interface provides you access to three main configuration areas: 1. Asterisk GUI configuration wizard A step by step configuration wizard that guide you through 7 steps to get your PBX up and running 2. The AsteriskGUI Once you have run the wizard for the first time you can edit and modify the parameters using the URL The Appliance Platform Configuration Wizard This wizard allows to configure parameters that are not Asterisk specific, for example the root password of the system, the IP address, backup schedule, etc. :8003/rAA/ Important notice! This tutorial covers only how to use the Asterisk GUI configuration wizard to set up the scenarios presented. For a complete description of all options available in the other graphical interfaces, consult the Asterisk Quickstart Guide

9 5.1 Installation Screenshots discussed Areas with a grey background are advanced tips. If you are not familiar with Asterisk, simply ignore them. GETTING STARTED Installing a VMware image If you install AsteriskNOW using a ISO image: Install VMware player and make sure that your Ethernet is in bridge mode. By putting the interface in Bridge Mode, your AsteriskNOW will fetch and IP address via DHCP after booting. Important: You need to have a DHCP server available in your network. asterisk x86.vmx CONFIGURATION FILES If during the process of booting the Configuration file starts by VMware player complains about the #!/usr/bin/vmplayer size of your FAT filesystem (> 2 GB): Append this line: Locate the configuration file of the disklib.sparsemaxfilesizecheck= "false" guest application, a file that finishes with VMX. Use a text editor like Wordpad, and append an extra line. 7

10 GETTING STARTED Make sure AsteriskNOW has an IP address If your DHCP is working, your AsteriskNOW will inform you of the IP address that your box has obtained. The AsteriskNOW Console allows you to update, restart and shutdown the system. Using the Console you can also operate a command line interface (the asterisk CLI>). Accessing the box via SSH CONFIGURATION FILES You can access the AsteriskNOW If you log into the box using SSH, please have a look box at any time via SSH. at the /etc/password and /etc/sudores files. You can see that the user admin can get administrative The account is admin with the privileges. Consider changing the default password of default password password. the user admin. You can get admin privileges using It is important to notice that there are 3 different sudo. admin users in each installation: (1) The admin user that let you log into the box via SSH. (2) The admin user that have access to the AsteriskGUI via HTTPS and (3) The admin user that can configure the appliance settings (rpath). Yes, three different passwords! 8

11 GETTING STARTED Log into AsteriskNOW web interface Open a browser and go to the IP address that AsteriskNOW has obtained. In our example Log into the interface using the user admin and the default password password. /etc/asterisk/manager.conf CONFIGURATION FILES [general] AsteriskGUI uses Asterisk manager displaysystemname = yes commands (Asterisk Manager API) enabled = yes webenabled = yes to communicate with Asterisk. The user and password of the AsteriskGUI is available in the manager.conf configuration file port = 5038 bindaddr = [admin] secret = password read = system,call,log,verbose, command,agent,user,config write = system,call,log,verbose,command, agent,user,config 9

12 6. Configuring AsteriskNOW for Scenario In a nutshell, the process of configuring each of the PBXs can be summarized in 7 basic steps: (Step 1) Verify if any zaptel card has been detected. Any zaptel compatible card should be detected. Cards supported by zaptel include: Digium, Sangoma, Xorcom Astribank (in beta 6.5), Rhino and Openvox cards. This step will report no hardware detected if you are configuring a PBX without zapata compatible hardware or running the VMware version of AsteriskNOW. Only Scenario 3 will use and detect a zaptel card. (Step 2) Indicate the first extension number and the length of the local extensions. Here you indicate the number of digits that your local extensions have and what the first extension number is. In our scenario we will use 4 digits and extension 1000 as the first one. (Step 3) Create Service Providers In this step we specify who the service providers of outgoing calls are. Scenario 1: the PBX is standalone and has no external service providers. Scenario 2: each PBX sees the other PBX as VoIP service provider. Scenario 3: the PBX with a TDM card needs to be configured with two different service providers. The first provider is the other PBX (VoIP) and the second provider is the Analogue Port of the TDM Card. (Step 4) Configure (Outbound) Calling Rules In this step we specify what the calling rules are to reach the different service providers. Scenario 1: does not need any outbound calling rules. Scenario 2: need to indicate that to reach the other PBX's local extensions we need to dial 9 + <local extension #>. Scenario 3: Same calling rule as in Scenario 2 to reach the other PBX. Also, we need to add an outgoing calling rule that indicates how to reach the PSTN. To reach the PSTN, we need to append a 0 to the PSTN number ( 0 + <PSTN #>). (Step 5) Voic extension In this step we will configure the extension number used to reach the voic . The default number for all three scenarios is (Step 6) Users extensions Here we configure all local extensions associated to each of the PBXs. We need to create four local extensions in all three scenarios (1000, 2000, 3000 and 4000). The local extensions can be either VoIP clients running IAX or SIP, or analogue ports if available. For scenario 2 and 3, we will add the special extension 4646 that is used to route calls between the PBXs. 10

13 (Step 7) Configure (Inbound) Calling Rules In Scenario 1 and 2, we do not need to create any special rules for incoming calls as all calls will be generated locally. In Scenario 3, we will need to indicate which local extension will ring when there is an incoming call from the PSTN. 6.1 Configuration though the Asterisk GUI Setup Wizard This section guides you though the graphical configuration setup wizard provided by AsteriskNOW. If it is the first time that you log into the AsteriskGUI, you will be redirected straight to the setup wizard. The setup wizard will guide you through seven (7) steps to configure your VoIP setup. This guide includes both basic and advanced configuration tips. Areas with grey background are advanced tips. If you are not familiar with Asterisk, please ignore them Step 1: Hardware detection STEP 1 HARDWARE DETECTION (Scenario 1 and 2) This screenshoot shows Step 1 of the wizard for Scenario 1 and 2, where our PBX does not include any PCI expansion cards. CONFIGURATION FILES It is possible to run the wizard again by accessing the following All the static web pages of the wizard are available in URL: the following path: setup/install.html /var/lib/asterisk/static http 11

14 STEP 1 HARDWARE DETECTION (2/2) (Scenario 3) If you have a PCI card as the TDM400, the wizard will detect the modules automatically. In the example, we have 1 FXO and 1 FXS port with the following functionality: FXO port: we can attach an external PSTN line. FXS port: we can attach a phone. /sbin/zapscan CONFIGURATION FILES The zapscan utility detects the ports #grep v "#" /etc/zaptel.conf and generates the /etc/zaptel.conf loadzone = us configuration file. defaultzone=us The configuration files shows the type of signalling needed for each of the ports. fxsks=1 fxoks=2 fxsks=1 means that port #1 is a FXO that needs FXS Kewlstart signalling. 12

15 6.1.2 Step 2: Local extensions settings STEP 2 LOCAL EXTENSIONS SETTINGS (Scenario 1, 2 and 3) In the second step of the configuration we indicate the length of the local extensions. In our setup we are going to use four digits and the extension number 1000 as the first extension. This configuration is common to all three Scenarios. /etc/asterisk/users.conf CONFIGURATION FILES This parameter that we set up in the wizard can be found in the users.conf with the name userbase inside of the section [general] [general] userbase = 1000 localextenlength = 4 13

16 6.1.3 Step 3: Configuring service providers STEP 3 CONFIGURING SERVICE PROVIDERS (1/3) (Scenario 2 and 3) In Scenario 2 and 3, we want our PBX to be able to communicate with another PBX and vice versa. To do that, we need to create a new Service Provider. In this example, we add a new service provider that is reachable at the IP address We indicate that we want to communicate using the protocol IAX using an account with username 4646 and password This account will be used between the PBXs for authentication and routing calls. /etc/asterisk/users.conf /etc/asterisk/extensions.conf The creation of new service provider involves: (1) A new section in the users.conf file and (2)A new entry point in the extensions.conf (dialplan) In our example we are creating a service provider [trunk_1] reachable at We are using the account user: 4646 password: Incoming calls from this provider fall in the section [DID_trunk_1] of the dialplan CONFIGURATION FILES /etc/asterisk/users.conf [trunk_1] disallow = allow = all callerid = contact = context = DID_trunk_1 dialformat = ${EXTEN:1} fromdomain = fromuser = group = hasexten = no hasiax = yes hassip = no host = insecure = port = 4569 provider = registeriax = yes registersip = no secret = 4646 trunkname = Custom InterIAX Calls trunkstyle = customvoip username = 4646 /etc/asterisk/extensions.conf [DID_trunk_1] include = default 14

17 STEP 3 CONFIGURING SERVICE PROVIDERS (2/3) (Scenario 3) If your PBX contains a TDM card with a FXO port (Scenario 3) we can communicate with the telephone network (PSTN). In the third scenario, you need to create a new Service Provider associated to the Analog Port. In our example, the TDM card contains a FXO port in slot #1. /etc/asterisk/users.conf /etc/asterisk/extensions.conf The creation of new service provider via the PSTN also modifies two files: (1) a new section in the users.conf file and (2) a new entry point in the dialplan. In our example the AsteriskGUI creates a new service provider with the name [trunk_2] reachable via the analog port #1 Incoming calls from this provider fall in the section [DID_trunk_2] of the dialplan CONFIGURATION FILES /etc/asterisk/users.conf [trunk_2] disallow = allow = callerid = asreceived contact = context = DID_trunk_2 dialformat = fromdomain = fromuser = group = 1 hasexten = no hasiax = no hassip = no host = dynamic insecure = port = provider = registeriax = registersip = secret = trunkname = Port 1 trunkstyle = analog username = zapchan = 1 /etc/asterisk/extensions.conf [DID_trunk_2] include = default 15

18 STEP 3 CONFIGURING SERVICE PROVIDERS (3/3) (Scenario 3) You can create as many service providers as you wish. One of the PBX of the Scenario 3, has two different Service Providers. One provider is the other PBX that can be reached via a VoIP connection (Custom VoIP) and the second provider is reachable via the Analog (TDM400) expansion card. /etc/asterisk/users.conf CONFIGURATION FILES The configuration file users.conf The users.conf contains three types of sections was introduced in the Asterisk 1.4 [general] series. This section includes default values. In the 1.2.x series, each user or peer was defined in sip.conf or iax.conf. The entity was classified depending on the protocol. The users.conf merges iax.conf, sip.conf and some of the options of zapata.conf into one single file. [trunk_#] These sections include the configuration of the different service providers. [XXXX] These sections include the configuration of the local extensions (1000, 2000, 3000, 4000). They can be analog ports or IAX or SIP users. 16

19 6.1.4 Step 4: Outbound calling rules STEP 4 OUTBOUND CALLING RULES (1/2) (Scenario 2 and 3) Once we have configured the service providers we can configure our dialplan. In Scenario 2 and 3, we need to create a rule to be able route calls between the PBXs. In the example, we create a calling rule with the name InterIAX Calls. In this menu, we describe the dialing rules that need to be applied when we want to reach the extensions of the VoIP provider (the other PBX) that we peer with. We indicate that to reach the other PBX, we need to dial 9 before the extension number. To reach the extension 1000 in the other PBX, we need to dial /etc/asterisk/extensions.conf CONFIGURATION FILES AsteriskNOW allows you to create different dialplans. The default DialPlan associated to the context of local extensions is numberplan custom 1. /etc/asterisk/extensions.conf [numberplan custom 1] plancomment = DialPlan1 include = default exten=_9xxxx.,1,macro(trunkdial,${trunk_1}/ ${EXTEN:1}) comment = _9XXXX.,1,InterIAX Calls,standard Outgoing calls between the PBX are routed using the trundial Macro, that places a call using: Dial(IAX2/4646:4646@ /${EXTEN:1}) and uses the account 4646 data for authentication. 17

20 STEP 4 OUTBOUND CALLING RULES (2/2) (Scenario 3) In Scenario 3, we need to create another calling rule that indicates that any local extensions can reach the PSTN by Port #1 (the analog service provider). Select Define a custom pattern and fill in your outbound calling rule according to the screenshot to the right. In the example, a call is placed by appending a 0 to a valid PSTN number, which is defined to be 6 or more digits. /etc/asterisk/extensions.conf CONFIGURATION FILES In this example we have two service providers. The first service provider is a VoIP provider (another PBX) and the second provider is the analog PSTN line. [numberplan custom 1] plancomment = DialPlan1 include = default ;Calls between PBXs. 9 + <local extension #> exten = _9XXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1}) comment = _9XXXX!,1,InterIAX Calls,standard To reach the VoIP provider: 9 + extension # To reach the PSTN via analogue port: 0 + PSTN # ;Calls to the PSTN. 0 + <PSTN #> exten = _0XXXXXX.,1,Macro(trunkdial,${trunk_2}/${EXTEN:1}) comment = _0XXXXXX.,1,outgoing PSTN,standard 18

21 6.1.5 Step 5: Voic settings STEP 5 VOIC SETTINGS (Scenario 1, 2 and 3) The default extension for voic is The default password for voic is the password of the extension it is associated with. When the configuration wizard is completed, you can change the password of your voic to any sequence of digits. You will find the option VW password under Users in the main menu. /etc/asterisk/voic .conf /etc/asterisk/users.conf When voic is activated in a local extension the setting hasvoic is set to yes. By settings the voic we also modified the way that extensions are called. Instead of a normal Dial(), Asterisk 1.4.x will call macro the [macro stdexten]. If not other value is specified the default the Voic password is the same that your account secret. The vmsecret option allows you to set a different password for your voic . In the example extension 3000 uses the secret 3000 for authentication of calls and the password 1234 to reach the mailbox. CONFIGURATION FILES [3000] callwaiting = yes cid_number = 3000 context = numberplan custom 1 = fullname = 3000 group = hasagent = yes hasdirectory = no hasiax = yes hasmanager = no hassip = yes hasvoic = yes host = dynamic mailbox = 3000 secret = 3000 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = yes registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 disallow = allow = 19

22 6.1.6 Step 6: User extensions STEP 6 USER EXTENSIONS (1/3) (Scenario 1, 2 and 3) In this step we will create the four local user extensions. The extensions can be associated to an IAX or SIP device such as an ATA or VoIP Phone, or associated to a analogue port available in the PBX. /etc/asterisk/users.conf CONFIGURATION FILES Each of the new extensions will have entry of the type [1000], [2000], [3000], etc. If the local extension is a SIP or IAX device it will be indicated with the values: hassip = yes hasiax = yes 20

23 STEP 6 USER EXTENSIONS (2/3) (Scenario 2 and 3) In Scenario 2 and 3 we need to create extension 4646, which needs to be available in both PBXs. This extension is created to accept incoming calls from the other PBX. /etc/asterisk/users.conf CONFIGURATION FILES This extension is not visible to the users and it is used for the purpose of routing and authenticating calls between the PBXs. [4646] callwaiting = yes cid_number = 4646 context = numberplan custom 1 = fullname = 4646 group = hasagent = yes hasdirectory = no hasiax = yes hasmanager = no hassip = yes hasvoic = yes host = dynamic mailbox = 4646 secret = 4646 threewaycalling = yes vmsecret = zapchan = registeriax = yes registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 disallow = allow = In Scenario 3 we are using IAX as the protocol for interconnecting the PBXs. IAX is more NAT friendly and efficient in terms of bandwidth. 21

24 STEP 6 USER EXTENSIONS (3/3) (Scenario 3) In Scenario 3, we need to define which local extension that should be associated with the Analog Port #2 (the phone). In this example, we have chosen Extension 1000 for that task. We define the association by editing the existing User Extension for Extension 1000 and select Analog Port #2 as Analog Phone. /etc/asterisk/users.conf CONFIGURATION FILES Although it might look surprising, it is possible to have an extension associated to more than one communication technology. [1000] callwaiting = yes cid_number = 1000 context = numberplan custom 1 = fullname = 1000 group = hasagent = no hasdirectory = no hasiax = yes hasmanager = no hassip = yes hasvoic = yes host = dynamic mailbox = 1000 secret = 1000 threewaycalling = yes zapchan = 2 registeriax = yes registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 In the example, extension 1000 is reachable in the Analogue Port #2 and via SIP and IAX. hasiax = yes hassip = yes zapchan = 2 This allows us to have as many as three devices associated to the same extension number. The three devices will ring simultaneously. 22

25 6.1.7 Step 7: Incoming calls rules STEP 7 INCOMING CALLS RULES (Scenario 3) In Scenario 3, we need to decide what to do with the calls originated in the PSTN. In the example, we indicate that all calls from the PSTN should be forward to the local extension /etc/asterisk/users.conf /etc/asterisk/extensions.conf Port #1 is a FXO port connected to the PSTN (zapchan = 1). Incoming calls fall in the context DID_trunk_2. In the dialplan, under the context [DID_trunk_2] we see that by default all calls (_X.,s) are forward to extension 1000 Goto(default ) CONFIGURATION FILES [trunk_2] disallow = allow = callerid = asreceived contact = context = DID_trunk_2 dialformat = fromdomain = fromuser = group = 1 hasexten = no hasiax = no hassip = no host = dynamic insecure = port = provider = registeriax = registersip = secret = trunkname = Port 1 trunkstyle = analog username = zapchan = 1 [DID_trunk_2] include = default exten = _X.,1,Goto(default ) exten = s,1,goto(default ) 23

26 6.1.8 Advanced options: Asterisk GUI ADVANCED OPTIONS ASTERISK GUI After completing the seven steps you can have a look to the AsteriskGUI interface. This interface allows to modify your entries via the wizard and create more advance services. When you have made changes to the configuration, do not forget to press the button Active Changes, in order apply the changes. 7. Configuration of ATAs No matter which ATA or IP Phone you need to configure, you will find that they can be configured in a similar manner. This example shows the configuration process of a Linksys PAP2 Internet Phone Adapter. The configuration is the same for a Sipura (SPA 3000). IP settings The ATA needs to have an IP address in order to be able to communicate with other devices on the LAN or the Internet. The IP address can be static or dynamic. In this example, we have chosen to obtain an IP address through DHCP. All IP settings of the ATA are configured using the handset Attach an analog phone to the ATA Connect the ATA to the LAN where the DHCP is running Enter the configuration menu of the ATA by pressing **** on the phone. Enable DHCP by pressing 101# followed by 1. Make sure that the ATA has obtained an IP address by pressing 110#. Extension number The extension number of the ATA is configured through its web interface. Direct your browser to address of your ATA>/admin/advanced Go to the tab Line 1, and fill in the following fields: 24

27 Proxy and Registration Proxy: (the IP address of the PBX you want to register) Register: Yes Subscriber Information Display Name: 1000 User ID: 1000 Password: 1000 Use Auth ID:yes Auth ID: 1000 You can verify from the web based Asterisk Configuration Panel that the ATA is registered in the PBX. 1. Go to Asterisk CLI in the left menu 2. On the bottom of the page (in the pink text field), write sip show peers 3. All registered phones and ATAs will be listed with IP address and extension number. 25

28 8. Quick Installation Guide This section includes a 7 step quick installation guide for Scenario 1, 2 and 3. Please note that the red crosses in the table indicate steps in the configuration procedure that are not needed for that specific scenario. 8.1 Scenario 1 26

29 8.2 Scenario 2 27

30 8.3 Scenario 3 28

31 9. Verify your results This section includes a set of Checkpoints for each Scenario that you should be able to do with your current VoIP setup. If you successfully manage all checkpoints listed for your Scenario, your Asterisk based VoIP setup has been configured correctly. 9.1 Scenario 1 Checkpoint 1: Place local phone calls from one extension to another within the same PBX. Checkpoint 2: Call a local extension and leave a voice messages (don't pick up the phone!). Use the voic to fetch the voice message you just left. 9.2 Scenario 2 Checkpoint 1: Place local phone calls from one extension to another within the same PBX. Checkpoint 2: Place phone calls between the two PBX's by using the prefix 9 before the extension number. Checkpoint 3: Call a local extension and leave a voice messages (don't pick up the phone!). Use the voic to fetch the voice message you just left. 9.3 Scenario 3 Checkpoint 1: Place local phone calls from one extension to another within the same PBX. Checkpoint 2: Place phone calls between the two PBX's by using the prefix 9 before the extension number. Checkpoint 3: Call a local extension and leave a voice messages (don't pick up the phone!). Use the voic to fetch the voice message you just left. Checkpoint 4: Call to the PSTN from any of the PBXs (try both). Checkpoint 5: Call in to the PBX from the PSTN. 29