Mediatrix FXO Unit with Asterisk

Mediatrix 3000 series, Mediatrix C7 series, and Mediatrix Sentinel Revision 296 2015-06-29

Table of Contents Table of Contents Mediatrix FXO unit with Asterisk 4 Requirements 5 Configuration of the PBX Trunk 6 Configuring the Asterisk - PBX Trunk 6 Adding the PBX Gateway 7 Restarting Services 8 Authenticating the PBX Gateway 8 Creating a Route From E1T1 to PBX Gateway 9 Creating a Route From PBX Gateway to E1T1 10 Configuration of the PSTN Lines 12 Configuring the Asterisk - PSTN Lines 12 Adding the PSTN Gateway 14 Restarting Services 14 Authenticating the PSTN Gateway 15 Creating a Route from FXO Ports to PSTN Gateway 16 Creating a Hunt Group - PSTN Gateway 17 Creating a Route from PSTN Gateway to a Hunt Group 18 Enabling Automatic Call Activation - Asterisk 19 General Configuration of the Mediatrix Unit with Asterisk 20 Configuring DTMF Transport - Out of Band Using Signalling Protocol 20 Disabling Voice Activity Detection 20

Table of Contents Available Documentation 22 Copyright Notice 23

4 Mediatrix FXO unit with Asterisk This document outlines the configuration steps required to connect a Mediatrix FXO unit with a 6FXO/2FXS card to an Asterisk open-source telephone system. In the scenario used throughout this document, the Mediatrix 3732 (1 PRI, 6 FXO/2FXS) is used to: Interface a PBX with an IP-PBX; Provide PSTN access via analog lines; Provide IP connectivity to analog phones and fax machines. The Asterisk IP-PBX provides: Call routing, Dial Plan # Including routes to local PSTN gateways Telephony services (voicemail, call forwarding, etc,) to IP users; SIP Endpoints management; Auto-Attendant.

Requirements Asterisk server properly installed Mediatrix 4102S and the IP phone are correctly configured with the IP-PBX and are able to make calls. 5

6 Configuration of the PBX Trunk Configuring the Asterisk - PBX Trunk 1) In the sip.config configuration file, create a new extension by adding the following: Extension Description [PBXTRUNK] The SIP username used for calls coming from the PBX. type=peer This is part of the method used by the Asterisk server to match incoming INVITES to this user. host=192.168.1.249 This means the extension will not register to the Asterisk server. This is the IP address of the Mediatrix 3732. port=5061 Port used for requests to and from the Mediatrix unit. nat=no The Mediatrix unit is not behind a NAT. qualify=no No keep alive is used. canreinvite=no No Re-Invite is sent to this extension. dtmfmode=info The DTMF is sent/received in SIP INFO messages. context=frompbx This is the context where the call from this extension is sent. It is the same context for the IP phone and the Mediatrix 4102S. secret=trunkpassword The SIP authentication password t38pt_udptl=yes This allows T.38 fax to be sent by this trunk. This setting is set to no

Extension 7 Description for the IP phone and the Mediatrix 4102S. 2) In the extensions.conf, add a context for calls coming from the PBX: Extension Description [FromPBX] exten =>_1XX,1,Dial(SIP{EXTEN},40,) This allows the PBX users to reach extensions 100 to 199. The Asterisk server will make the extension ring for 40 seconds. 3) Still in extensions.conf, modify the context of the extensions to allow them to send calls to the PBX: Extension Description [InternalExtensions] exten =>_1XX,1,Dial(SIP/ ${EXTEN},40,) exten =>_2XX,1,Dial(SIP/ ${EXTEN}@PBXTrunk,40,) This allows the IP phone and Mediatrix 4102S to reach the PBX extensions 200 to 299. 4) Reload the Asterisk settings by connecting to the Asterisk CLI (asterisk -r) and typing the reload command. 5) If you want to use the T.38 protocol to transfer faxes, you can follow the procedure of the Configuration Notes 259 - Mediatrix 41xx With Asterisk web Adding the PBX Gateway (p. 7) Adding the PBX Gateway Before you start Make sure your PRI interface has both Physical and Signalling up before starting. 1) Go to SIP > Gateways. 2) In the General Configuration table, type PBX-Gateway in the blank field under the Name column. 3) Click on. 4) In the Port field, type 5061.

8 5) Click Apply. Result Restarting Services (p. 8) Restarting Services 1) Go to System > Services. 2) In the Restart required services table, click Restart required services. Registering FXS Port - Fax Authenticating the PBX Gateway (p. 8) Authenticating the PSTN Gateway (p. 15) Authenticating the PBX Gateway 1) Go to SIP > Authentication. 2) In the Authentication table, click. 3) On the row that just appeared, click on.

9 4) Set the following parameters: a) Set Criteria to Gateway. b) Set Gatewayto PBX-Gateway. c) Set Validate Realm to Disable. d) Set the User Name to reflect your configuration. e) Set the Password to reflect your configuration. 5) Click on Apply & Refresh Registration. Result Creating a Route From E1T1 to PBX Gateway (p. 9) Creating a Route From E1T1 to PBX Gateway 1) Go to Call Router > Route Config. 2) In the Route table click located on the same row as an existing route to add a route above or, click located at the bottom of the table to add a route at the end of the table. 3) In the Configure Route End table, set the following parameters: a) Set Sources to isdn-slot1/e1t1. b) Set Properties Criteria to None. c) Set Destination to sip-pbx-gateway. 4) Click Save. Result The route will be added to the Route table.

10 Creating a Route From PBX Gateway to E1T1 (p. 10) Creating a Route From PBX Gateway to E1T1 1) Go to Call Router > Route Config. 2) In the Route table click located on the same row as an existing route to add a route above or, click located at the bottom of the table to add a route at the end of the table. 3) In the Configure Route End table, set the following parameters: a) Set Sources to sip-pbx-gateway. b) Set Properties Criteria to None. c) Set Destination to isdn-slot1/e1t1. 4) Click Save. Result The route will be added to the Route table.

Configuring DTMF Transport - Out of Band Using Signalling Protocol (p. 20) 11

12 Configuration of the PSTN Lines Configuring the Asterisk - PSTN Lines 1) In the sip.conf configuration file, create a new extension by adding the following: Extension Description [PSTNTrunk] SIP username used for calls coming from the PSTN. type= peer This is part of the method used by the Asterisk server to match incoming INVITES to this user. host=192.168.1.249 This means the extension will not register to the Asterisk server. This is the IP address of the Mediatrix 3732. port=5062 Port used for requests to and from the Mediatrix unit. nat=no The Mediatrix unit is not behind a NAT. qualify=no No keep alive is used. canreinvite=no No Re-Invite is sent to this extension. dtmfmode=info The DTMF is sent/receive in SIP INFO messages. context= FromPSTN Context where the call from this extension is sent. It is the same context for the IP phone and the Mediatrix 4102S. secret= TrunkPassword SIP authentication password. t38pt_udptl=yes This allows T.38 fax to be sent by this trunk. This setting is set to no

Extension 13 Description for the IP phone and Mediatrix 4102S. 2) In the extensions.conf, add a context for calls coming from the PSTN: Extension Description [FromPSTN] exten=> 900,1,Answer() exten=> 900,2,Background(CompanyPrompt) This plays a prompt (IVR) to the calling party and allows it to dial extensions. [FromPSTN] exten=> 900,3,Hangup() exten=> _1XX,1,Dial(SIP/ ${EXTEN},40,) exten=> _2XX,1,Dial(SIP/ ${EXTEN}@PBXTrunk,40,) This allows the IP phone and Mediatrix 4102S to reach the PBX extensions 200 to 299. 3) In the extensions.conf, modify the context of the PBX to allow the PBX users to send calls to the PSTN: Extension Description [FromPBX] exten=> _1XX,1,Dial(SIP/ ${EXTEN},40,) exten=> _9X.,1,Dial(SIP/ ${EXTEN:1}@PSTNTrunk,40,) This allows the PBX users to dial on the PSTN and removes the 9 before dialling the number. 4) In the extensions.conf, modify the context of the extensions to allow them to send calls to the PBX. Extension Description [InternalExtensions] exten=> _1XX,1,Dial(SIP/ ${EXTEN},40,) exten=> _2XX,1,Dial(SIP/ ${EXTEN}@PBXTrunk,40,) exten=> _9X.,1,Dial(SIP/ ${EXTEN:1}@PSTNTrunk,40,) This allows the IP phone and Mediatrix 4102S to dial on the PSTN and removes the 9 before dialling the number. 5) You can reload the Asterisk settings by connecting to the Asterisk CLI (asterisk r) and typing the reload command.

14 Adding the PSTN Gateway (p. 14) Adding the PSTN Gateway 1) Go to SIP > Gateways. 2) In the Gateway Configuration table, in the Name field, enter PSTN-Gateway. 3) Click. 4) Complete the fields as follows: From the Type selection list, select Trunk. From the Signaling Network selection list, select Uplink. In the Port field, enter 5062. 5) Click Apply. Result The PSTN-Gateway gateway will be available under the SIP > Servers page. Restarting Services (p. 8) Restarting Services 1) Go to System > Services. 2) In the Restart required services table, click Restart required services.

15 Registering FXS Port - Fax Authenticating the PBX Gateway (p. 8) Authenticating the PSTN Gateway (p. 15) Authenticating the PSTN Gateway 1) Go to SIP > Authentication. 2) In the Authentication table, click. 3) On the row that just appeared, click on. 4) Set the following parameters: a) Set Criteria to Gateway. b) Set Gatewayto PSTN-Gateway. c) Set Validate Realm to Disable. d) Set the User Name to reflect your configuration. e) Set the Password to reflect your configuration. 5) Click on Apply & Refresh Registration. Result

16 Creating a Route from FXO Ports to PSTN Gateway (p. 16) Creating a Route from FXO Ports to PSTN Gateway 1) Go to Call Router > Route Config. 2) In the Route table click located on the same row as an existing route to add a route above or, click located at the bottom of the table to add a route at the end of the table. 3) In the Configure Route End table, set the following parameters: a) Add all the FXO ports in the Sources field. b) Set Properties Criteria to None. c) Set Destination to sip-pstn-gateway. 4) Click Save. Result

17 Creating a Hunt Group - PSTN Gateway (p. 17) Creating a Hunt Group - PSTN Gateway 1) Go to Call Router > Route Config. 2) In the Hunt table, click 3) In the Configure Hunt End table, set the following: a) Set the Name field to PSTN-Gateway. b) Using the Destinations' dropbox, add all the FXO ports used in your configuration. c) Leave the other fields with their default value. 4) Click Save. Result

18 Creating a Route from PSTN Gateway to a Hunt Group (p. 18) Creating a Route from PSTN Gateway to a Hunt Group 1) Go to Call Router > Route Config. 2) In the Route table click located on the same row as an existing route to add a route above or, click located at the bottom of the table to add a route at the end of the table. 3) In the Configure Route End table, set the following parameters: a) Set Sources to sip-pstn-gateway. b) Set Properties Criteria to None. c) Set Destination to hunt-pstn-gateway. 4) Click Save. Result

19 Enabling Automatic Call Activation - Asterisk (p. 19) Enabling Automatic Call Activation - Asterisk 1) Go to Telephony > Services. 2) In the Select Endpoint dropdown menu, select a FXO port you want to configure. 3) In the Automatic Call section, set the following parameters: a) Set Endpoint Specific to Yes. b) Set Automatic Call Activation to Enable c) Set the Automatic Call Target to the number of the IVR configured in the Asterisk server. 4) Repeat the steps 2 and 3 for the other FXO ports you need to configure. Result Configuring DTMF Transport - Out of Band Using Signalling Protocol (p. 20)

20 General Configuration of the Mediatrix Unit with Asterisk Configuring DTMF Transport - Out of Band Using Signalling Protocol 1) Go to Media > Misc. 2) In the DTMF Transport table, set the Transport Method to Out-of-Band using SIP. 3) Click Apply. Result Disabling Voice Activity Detection (p. 20) Disabling Voice Activity Detection (p. 20) Disabling Voice Activity Detection (p. 20) Disabling Voice Activity Detection 1) Go to Media > Codecs.

21 2) In the Select Endpoint dropdown menu, select the endpoint you want to configure. 3) In the Generic Voice Activity Detection (VAD) table, set Enable (G.711 and G.726) to Disable. Result

Available Documentation For more details, refer to the Mediatrix Documentation. 22

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