Grandstream GXW410x and Elastix Server

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Dario Zarate
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1 Grandstream GXW410x and Elastix Server Setup Guide

2 1. Setup Diagram Figure 1-1 is a setup diagram for a single gateway Grandstream GXW410x configuration. The gateway is setup as a SIP telephony device. Figure 2-1. Setup Diagram 2. Host PC Environment Table 2-1. Host Server Environment Details Description Hardware Type Elastix Appliance ELX-Series Hardware Version ELX-3000 Software Type Elastix Software Version Test Setup Equipment Table 3.1. Test Setup Equipment Equipment Model Version Video Call IP (SIP) Phone N/A N/A Grandstream Gateway GXW410x Switch N/A N/A 2

Host Server Environment Details Description Hardware Type Elastix Appliance ELX-Series Hardware Version ELX-3000 Software Type

3 4. Setup Procedure To set up the Elastix Server for the Grandstream GXW410x 1. Go to the web address of the Elastix Server Login page. The web address is determined by the customer, for this guide we have used the IP address On the Login page, type the username and password for an administrative user into the Username and Password fields, see Figure 4-1. The username is admin and the password is determined by the customer. Figure 4-1. Login 3. Press Enter or click on the Submit button to go to Elastix s Dashboard 4. Once inside, click on the PBX tab on the menu at the top of the screen. Figure 4-2. Dashboard 3

On the Login page, type the username and password for an administrative user into the Username and Password fields, see Figure 4-1.

4 5. Click on the Submit button to add an extension so we can test the gateway using it when we finish the entire configuration, see Figure 4-3. This will take you to the Add SIP Extension page. Figure 4-3. Add an Extension 6. On the Add SIP Extension page (Figure 4-4), fill in the following information: User Extension (302 in this example) Display Name ( IPPhone in this example) secret ( h7dka3rf9si0t in this example) Figure 4-4. Add SIP Extension 7. Click on the Submit button at the end of the page and you will see a similar page on Figure 4-5 displaying the Apply Configuration Changes Here pink ribbon on top of the screen. Click on there to apply changes. 4

On the Add SIP Extension page (Figure 4-4), fill in the following information: User Extension (302 in this example) Display Name ( IPPhone in this example) secret (

5 Figure 4-5. Apply Configuration 8. Configure an IP Phone with the same settings to register it with Elastix Server. 9. Now we will configure an Outbound Route for outgoing calls depending on a prefix. For this we have to configure a SIP Trunk first. Go to PBX => PBX Configuration => Trunks. Click on Add SIP Trunk, then Submit (Figure 4-6). Set the following: General Settings Trunk Name: (trunk801 in this example) Outgoing Settings Trunk Name: (801 in this example) Peer Details: host=dynamic username=( 801 in this example) secret=( qbp567bbd4gt in this example) type=peer insecure=very qualify=yes Figure 4-6. Add SIP Trunk 5

Go to PBX => PBX Configuration => Trunks. Click on Add SIP Trunk, then Submit (Figure 4-6).

6 10.Click on the Submit button at the end of the page and the SIP Trunk will be created. 11.Now, go to PBX => PBX Configuration => Outbound Routes to configure the outbound route using this trunk. Fill in the following information: (Figure 4-7) Route Settings Route Name: ( 6_Line1_gateway in this example) Dial patterns Prefix: ( 6 in this example) Match pattern: (. in this example) Trunk Sequence for Matched Routes 0: ( trunk801 in this example) Figure 4-7. Add Route 12.Click on the Submit button at the end of the page and Apply changes. With this configuration when you want to dial and external number, just dial the number with the prefix 6 and the call will go out through the port 1 of the gateway. 13.Now, we ll create an incoming route for the calls from PSTN that pass through the gateway. We re going to use an IVR for incoming calls. Go to PBX => PBX Configuration => IVR. Click on the link Add IVR (Figure 4.8). Set the following: Name: Name of IVR (WelcomeIVR in this example) Announcement: Record which will be played for incoming calls. Options: o * - Phone book. o Extension o t - Repeat the options of IVR (Add this option by modifying the IVR after creation) 6

in this example) Trunk Sequence for Matched Routes 0: ( trunk801 in this example) Figure 4-7. Add Route 12.Click on the Submit button at the end of the page and Apply changes.

7 Figure 4-8. IVR 14.Click on Save and Apply changes by clicking on the pink ribbon that appears at the top of the page. Now go to PBX => PBX Configuration => Inbound Routes. Click on Add Incoming Route. (Figure 4.9). Set the following: Description: Name of inbound route ( Incoming_Calls in this example) DID Number: Set destination: Where the call will be routed. ( WelcomeIVR IVR in this example) Figure 4-9. Incoming Route 15.Click on Submit button and apply changes. Now when we receive calls from PSTN to the number , the IVR will be played to the caller giving him choices to interact with our Elastix Server. 16.To configure the GXW410x you will need to enter the information from the sip trunk created on the Elastix Server into the gateway. 7

Set the following: Description: Name of inbound route ( Incoming_Calls in this example) DID Number: 999999 Set destination: Where the call will be routed.

8 For the initial configuration, refer to the Grandstream GXW410x User Manual found at: Table 4.1. Factory default settings IP Addressing Web Access Password DHCP admin 17.Log in to the Grandstream GXW410x WebUI by pointing your browser to the IP address of the gateway. 18.When prompted, enter the Web Access Password to access to the WebUI (Figure 4-10). Figure Grandstream WebUI 19.Use the information from the Add SIP Extension page (Figure 4-4) to enter the following necessary information on Channels section of the Grandstream Gateway WebUI (Figure 4-11): Phone Number Settings Channels: 1 (The number of FXO port where the PSTN line is connected) SIP User ID: 801 in this example Authenticate ID: 801 in this example Authen Password: qbp567bbd4gt in this example Profile ID: Profile 1 (We are going to set this profile later) Channel Specific Setting DTMF Methods (1-7): ch1-4:2; 8

9 Figure Channels 20.Once you have entered the required information, click on Update button located at the end of the page. Now go Profile 1 to edit some parameters (Figure 4-12). Activate Profile: Yes SIP Server: Elastix s IP Address ( ) User ID is phone number: No SIP Registration: Yes Figure Profile 1 21.Click on Update button at the end of the page. 22.Go to FXO Lines and set the Channel Dialing to PSTN section See Figure Channel Dialing to PSTN Stage Method (1/2): ch1-4:1; (This option is for dialing a number directly to the PSTN, the option is for obtaining a dial tone to dial the number). Channel Dialing to VoIP User ID: ch1-4:999999; (This number has to match the DID number in the Incoming Route in Elastix Server, See Figure 4.9) PSTN to VoIP Caller ID Setting 9

Go to FXO Lines and set the Channel Dialing to PSTN section See Figure 4-13.

10 Number of Rings Before Pickup: ch1-4:4; Caller ID Scheme: ch1-4:1; (Choose the option that better fits to your PSTN line) Caller ID Transport Type: ch1-4:4; Figure FXO Lines 23.Click on the Update button and then POWER OFF the gateway and POWER ON again. When it is on, go to Status section and check out the channel 1 is registered and the PSTN line is connected and idle (the PSTN line must be connected in the first port). See Figure

Click on the Update button and then POWER OFF the gateway and POWER ON again.

11 Figure Status 24.If the phone number is not registered please check out all the confirmation. 25.To test the outbound calls, pick up the previously configured IP (SIP) Phone and call to an external number using the prefix 6 (i.e ). The call will use the registered port of the gateway to connect to the call and you should be able to hear the ring tone. To test the inbound calls, from the PSTN call to your public number which should be connected to a registered port in GXW410x gateway, and after some rings you should be able to reach the IVR configured in Elastix. This step completes the procedure of configuration. Make sure you have configured all the parameters correctly if you are facing problems. For advance configuration please refer to the gateway manual cited in previous pages. 11

The call will use the registered port of the gateway to connect to the call and you should be able to hear the ring tone.

12 IN CASE THE CALLER ID OF INCOMING CALLS IS NOT RECEIVED CORRECTLY 1. Log in to the console in Elastix Server. You can access remotely or directly from the server. 2. Go to /etc/asterisk/extensions_additional.conf and search the context of the inbound route configured in Elastix. As a reference try to find the DID number we set. In this case the name of the context is [ext-did-0002]. See Figure Figure Copy the context and paste it at the end of the file /etc/asterisk/extensions_custom.conf and change the context s name to [route-gselx]. In addition to this, add the following lines to the context starting from the third line: exten => ,n,NoOp(Obtaining CallerID...) exten => ,n,Set(foo=${SIP_HEADER(P-asserted-identity)}) exten => ,n,Set(cadnum=${CUT(foo,:,2)}) exten => ,n,Set(CALLERID(number)=${CUT(cadnum,@,1)}) exten => ,n,Set(CALLERID(name)=${CUT(foo,\",2)}) The context should end as shown in Figure Figure Add the line context=route-gs-elx in the Peer Settings of the SIP Trunk configuration in Elastix Server. Refer to Figure 4.6 on step 9. At the end the SIP trunk configuration must be as shown in Figure

In this case the name of the context is [ext-did-0002]. See Figure 4-15. Figure 4-15 3. Copy the context and paste it at the end of the file /etc/asterisk/extensions_custom.

13 Figure

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