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Mediatrix FXO Unit with Asterisk
Mediatrix 3000 series, Mediatrix C7 series, and Mediatrix Sentinel
Revision 296
2015-06-29
Table of Contents
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Table of Contents
Mediatrix FXO unit with Asterisk 4
Requirements 5
Configuration of the PBX Trunk 6
Configuring the Asterisk - PBX Trunk 6Adding the PBX Gateway 7Restarting Services 8Authenticating the PBX Gateway 8Creating a Route From E1T1 to PBX Gateway 9Creating a Route From PBX Gateway to E1T1 10
Configuration of the PSTN Lines 12
Configuring the Asterisk - PSTN Lines 12Adding the PSTN Gateway 13Restarting Services 14Authenticating the PSTN Gateway 15Creating a Route from FXO Ports to PSTN Gateway 16Creating a Hunt Group - PSTN Gateway 17Creating a Route from PSTN Gateway to a Hunt Group 18Enabling Automatic Call Activation - Asterisk 19
General Configuration of the Mediatrix Unit with Asterisk 20
Configuring DTMF Transport - Out of Band Using Signalling Protocol 20
Table of Contents
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Disabling Voice Activity Detection 21
Available Documentation 22
Copyright Notice 23
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Mediatrix FXO unit with AsteriskThis document outlines the configuration steps required to connect a Mediatrix FXO unit with a6FXO/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.
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Requirements
• Asterisk server properly installed
• Mediatrix 4102S and the IP phone are correctly configured with the IP-PBX and are able tomake calls.
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Configuration of the PBX Trunk
Configuring the Asterisk - PBX Trunk
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) In the sip.config configuration file, create a new extension by adding the following:
Extension Description
[PBXTRUNK] The SIP username used for calls comingfrom the PBX.
type=peer This is part of the method used by theAsterisk server to match incoming INVITES tothis user.
host=192.168.1.249 This means the extension will not register tothe Asterisk server. This is the IP address ofthe Mediatrix 3732.
port=5061 Port used for requests to and from theMediatrix 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 INFOmessages.
context=FromPBX This is the context where the call from thisextension is sent. It is the same context forthe 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 for the IP phone andthe Mediatrix 4102S.
2) In the extensions.conf, add a context for calls coming from the PBX:
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Extension Description
[FromPBX]exten =>_1XX,1,Dial(SIP{EXTEN},40,)
This allows the PBX users to reachextensions 100 to 199. The Asterisk serverwill make the extension ring for 40 seconds.
3) Still in extensions.conf, modify the context of the extensions to allow them to send callsto 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 Mediatrix4102S to reach the PBX extensions 200 to299.
4) Reload the Asterisk settings by connecting to the Asterisk CLI (asterisk -r) and typing thereload command.
5) If you want to use the T.38 protocol to transfer faxes, you can follow the procedure of theConfiguration Notes 259 - Mediatrix 41xx With Asterisk web
Next StepAdding the PBX Gateway (p. 7)
Adding the PBX Gateway
Before you startMake sure your PRI interface has both Physical and Signalling up before starting.
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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 5061.5) Click Apply.
Result
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Next StepRestarting Services (p. 8)
Restarting Services
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) Go to System > Services.2) In the Restart required services table, click Restart required services.
Next StepRegistering FXS Port - Fax
Authenticating the PBX Gateway (p. 8)
Authenticating the PSTN Gateway (p. 15)
Authenticating the PBX Gateway
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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 PBX-Gateway.
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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 and Refresh Registration.
Result
Next StepCreating a Route From E1T1 to PBX Gateway (p. 9)
Creating a Route From E1T1 to PBX Gateway
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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.
ResultThe route will be added to the Route table.
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Next StepCreating a Route From PBX Gateway to E1T1 (p. 10)
Creating a Route From PBX Gateway to E1T1
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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.
ResultThe route will be added to the Route table.
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Next StepConfiguring DTMF Transport - Out of Band Using Signalling Protocol (p. 20)
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Configuration of the PSTN Lines
Configuring the Asterisk - PSTN Lines
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) In the sip.conf configuration file, create a new extension by adding the following:
Extension Description
[PSTNTrunk] SIP username used for calls coming from thePSTN.
type= peer This is part of the method used by theAsterisk server to match incoming INVITES tothis user.
host=192.168.1.249 This means the extension will not register tothe Asterisk server. This is the IP address ofthe Mediatrix 3732.
port=5062 Port used for requests to and from theMediatrix 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 INFOmessages.
context= FromPSTN Context where the call from this extension issent. It is the same context for the IP phoneand 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 for the IP phone andMediatrix 4102S.
2) In the extensions.conf, add a context for calls coming from the PSTN:
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Extension Description
[FromPSTN]exten=> 900,1,Answer()
exten=> 900,2,Background(Company-Prompt)
This plays a prompt (IVR) to the calling partyand 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 Mediatrix4102S to reach the PBX extensions 200 to299.
3) In the extensions.conf, modify the context of the PBX to allow the PBX users to send callsto 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 thePSTN and removes the 9 before dialling thenumber.
4) In the extensions.conf, modify the context of the extensions to allow them to send calls tothe 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 Mediatrix4102S to dial on the PSTN and removes the9 before dialling the number.
5) You can reload the Asterisk settings by connecting to the Asterisk CLI (asterisk –r) and typingthe reload command.
Next StepAdding the PSTN Gateway (p. 13)
Adding the PSTN Gateway
Steps1) Go to SIP > Gateways.
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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.
ResultThe PSTN-Gateway gateway will be available under the SIP > Servers page.
Next StepRestarting Services (p. 8)
Restarting Services
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) Go to System > Services.2) In the Restart required services table, click Restart required services.
Next StepRegistering FXS Port - Fax
Authenticating the PBX Gateway (p. 8)
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Authenticating the PSTN Gateway (p. 15)
Authenticating the PSTN Gateway
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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 and Refresh Registration.
Result
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Next StepCreating a Route from FXO Ports to PSTN Gateway (p. 16)
Creating a Route from FXO Ports to PSTN Gateway
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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
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Next StepCreating a Hunt Group - PSTN Gateway (p. 17)
Creating a Hunt Group - PSTN Gateway
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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
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Next StepCreating a Route from PSTN Gateway to a Hunt Group (p. 18)
Creating a Route from PSTN Gateway to a Hunt Group
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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
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Next StepEnabling Automatic Call Activation - Asterisk (p. 19)
Enabling Automatic Call Activation - Asterisk
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) 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 Enablec) 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
Next StepConfiguring DTMF Transport - Out of Band Using Signalling Protocol (p. 20)
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General Configuration of the Mediatrix Unit withAsterisk
Configuring DTMF Transport - Out of Band Using SignallingProtocol
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) Go to Media > Misc.2) In the DTMF Transport table, set the Transport Method to Out-of-Band using SIP.3) Click Apply.
Result
Next StepDisabling Voice Activity Detection (p. 21)
Disabling Voice Activity Detection (p. 21)
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Disabling Voice Activity Detection (p. 21)
Disabling Voice Activity Detection
InformationIf you are not familiar with the meaning of the fields, click Show Help, located at the upper rightcorner of the Web page, to display field description when hovering over the field name.
Steps1) Go to Media > Codecs.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
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Available Documentation
For more details, refer to the Mediatrix Documentation.
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Copyright NoticeCopyright © 2016 Media5 Corporation.
This document contains information that is proprietary to Media5 Corporation.
Media5 Corporation reserves all rights to this document as well as to the Intellectual Property ofthe document and the technology and know-how that it includes and represents.
This publication cannot be reproduced, neither in whole nor in part, in any form whatsoever, withoutwritten prior approval by Media5 Corporation.
Media5 Corporation reserves the right to revise this publication and make changes at any time andwithout the obligation to notify any person and/or entity of such revisions and/or changes.