This document outlines the configuration steps required to connect a Mediatrix unit to an Asterisk open-source telephone system.
In this scenario, the Mediatrix unit is used to:
The Asterisk IP-PBX provides:
Before starting to use these configuration notes, complete the following table to make sure you have the required information to complete the different steps.
If you are not familiar with the meaning of the fields and buttons, click Show Help, located at the upper right corner of the Web page. When activated, the fields and buttons that offer online help will change to green and if you hover over them, the description will be displayed.
The Mediatrix unit must be reinitialized to its factory default settings to make sure the configuration can be successfully executed.
|Information||Value||Used in Step|
|IP address of unit||Configuring the Asterisk - PBX Trunk and Configuring the Asterisk - PSTN Lines|
|Listening port of unit||Configuring the Asterisk - PBX Trunk and Configuring the Asterisk - PSTN Lines|
|The SIP username used for calls coming from the PSTN||Configuring the Asterisk - PSTN Lines|
|The SIP username used for calls coming from the PBX||Configuring the Asterisk - PSTN Lines|
|SIP Username and Password to authenticate the gateway||Authenticating the SIP Default Gateway|
|Asterisk server IP address|
|Asterisk server SIP listening port|
|DTMF transport method||Configuring the Asterisk - PBX Trunk and Configuring the Asterisk - PSTN Lines and Configuring DTMF Transport|
|[PBXTRUNK]||SIP username used for calls coming from the PBX.|
|[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=ip address of Mediatrix unit||This means the extension will not register to the Asterisk server. This is the IP address of the Mediatrix unit.|
|port=xxx||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=xxxx||This is the context where the call from this extension is sent. It is the same context for the IP phone and the Mediatrix Unit.|
|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 and the Mediatrix unit .|
Setting the address to 0.0.0.0:0 or leaving the field empty disables the outbound proxy host.
The default value is 0 which stands for 5060.
Mediatrix units are supplied with an exhaustive set of documentation.
Mediatrix user documentation is available on the Documentation Portal .
Several types of documents were created to clearly present the information you are looking for. Our documentation includes:
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