Login       Monday, January 18, 2021      Search  


Configuring Mediatrix 4108, 4116, 4124, 1102, 1104 FXS SIP Gateway with the Asterisk IP PBX System





This document outlines the configuration steps to set up Mediatrix 4108, 4116, 4124, 1102, 1104 series FXS gateways with the Asterisk open-source

telephone system. It assumes that you have an Asterisk server properly installed with the necessary modules. If you need

technical assistance to configure your Asterisk server, the Mediatrix technical team can provide appropriate support to help

you realizing your VoIP projects.


About Mediatrix 4100 Series FXS Gateways


The Mediatrix® 4100 Series products are high quality

and cost-efficient VoIP gateways

connecting larger branch offices and multitenant

buildings to an IP network, while

preserving investment in analog telephones

and faxes.

The Mediatrix® 41xx access devices allow

Service Providers to deploy rapidly and

economically their solutions in medium-size

premises and they are the ideal solution for

branch office connectivity to larger private

networks. The following Mediatrix 41xx

models are available:

_ 4108 – Enterprise 8-port FXS gateway

_ 4116 – Enterprise 16-port FXS gateway

_ 4124 – Enterprise 24-port FXS gateway

Benefits of using the Mediatrix 41xx over other Asterisk FXS solutions:

_ Superior Voice quality with dedicated DSP hardware

_ Adaptive Jitter Buffer, G.168 Echo Cancellation

_ T.38 Fax Relay support

_ Can be used to extend the Asterisk system in remote locations

_ Highly flexible, easily adaptable to VoIP systems other than Asterisk

_ Simple configuration to work with Asterisk



Application Scenario


This is a typical deployment scenario of Mediatrix 41xx units in a SIP-based MTU/MDU environment. The units provide

connectivity to analog phones and fax machines and analog trunking to legacy PBX/KSU (for small business) in each

building floor, whereas the Asterisk IP PBX provides call control and telephony services for all gateways.

FXS Gateway (41xx) – connectivity to analog phones and fax machines, analog trunking to legacy PBX/KSU.

The Asterisk IP PBX – provides:

_ Call routing, Dial Plan

o Including routes to local PSTN gateways

_ Telephony services (voicemail, call forwarding etc)

_ SIP Endpoints management

_ Auto-Attendant


Features Supported

The following features are supported by both Asterisk and Mediatrix 41xx:

_ RFC 2833, SIP INFO, and inband DTMF transports

_ SIP Authentication

_ Blind and Supervised Call Transfer

_ Call Forward On Busy/On No Answer/Unconditional

_ Call Waiting (via Asterisk only)

_ Voice Mail

_ Conference Call

_ Music on Hold

_ Caller ID

_ G.711 (recommended), G.723.1 and G.726 codecs

_ Fax transmission

Versions Supported

This configuration note was written and validated using the following platforms and versions.

_ Asterisk: 1.2.4 running on RedHat Linux 9

_ Mediatrix 41xx: SIP

The following configuration notes are not a substitute for the Mediatrix Administration Documentation for Mediatrix 41xx

products. Please have the following manuals available for reference:

_ Mediatrix 4108, 4116, and 4124 SIP Reference Manual

_ Mediatrix 4108, 4116, and 4124 SIP Quick Start

_ Mediatrix Unit Manager Network Administration Manual

_ All relevant Asterisk Manuals


Asterisk Configuration

The following sections describe special configuration you must perform in Asterisk in order to properly work with the

Mediatrix 41xx. The configuration parameters are located in various configuration files.

Creating an Extension

In Asterisk, an extension is the equivalent of the SIP user in the Mediatrix 41xx.

1. In the /etc/asterisk/sip.conf file, scroll to the bottom of the file and add the following:
















This is the extension number. Use a unique number. The Mediatrix 41xx will use this number to

authenticate to the system and users will dial it to ring the Mediatrix 41xx.



Attribute of the SIP object. There are three values available:

_ peer: A SIP entity to which Asterisk sends calls (a SIP provider for example). If

you want a user (extension) to have multiple phones, define an extension that

calls two SIP peers. The peer authenticates at registration.

_ user: A SIP entity which places calls through Asterisk.

_ friend: An entity which is both a user and a peer. This make sense for most desk

handsets and other devices.

If a peer is defined with host=dynamic it is allowed to register with Asterisk to tell Asterisk

where it can be found (IP address/host name) and that it is reachable from now on.




How to find the client - IP # or host name. Using the keyword dynamic indicates that the phone

registers itself.



When entering yes, Asterisk will change the behaviour, addressing, etc. of communication with

the client (SIP UA) that is behind a NAT device, to make communication possible.



Entering yes indicates that Asterisk keeps a UDP session open to a device that is located

behind a network address translator (NAT). This can be used in conjunction with the nat=yes




Entering no indicates that Asterisk does not issue a reinvite to the client and acts as the RTP

portal. The Mediatrix 41xx supports both yes and no options.



You have the choice between rfc2833, info, and inband. You will have to enter similar

information in the Mediatrix 41xx.



Page 6 of 16

Refers to the context to which the extension belongs as defined in the extensions.conf file (see

Step 2 for details).



If you want to use authentication for this extension, enter the proper information in the secret

field. You will have to enter similar information in the Mediatrix 41xx. See SIP

AuthenticationError! Reference source not found. on page 10 for more details.



2. In the /etc/asterisk/extensions.conf file, add the following:



3. Execute the following command on the server to reload the configuration:

asterisk -rx "reload"


Mediatrix 41xx Configuration

The following sections describe special configuration you must perform in the Mediatrix 41xx in order to properly work with


All models in the Mediatrix 41xx series feature an embedded Web server. Most of the commonly used parameters are

accessible from the web interface. The Mediatrix Unit Manager Network (UMN) software is needed if access to full unit

configuration is required. The UMN can be downloaded from the Mediatrix Portal:


It has a default 3-units limit upon installation. This will suffice for most configurations. Additional unit license can be

purchased. Please contact your Mediatrix reseller for more details.

Using the Web interface

The web interface may be used to:

_ View the status of the Mediatrix 41xx.

_ Set numerous parameters of the Mediatrix 41xx.

Mediatrix recommends that you use the latest version of the Microsoft® Internet Explorer web browser to properly access

the web interface. To use the web interface configuration:

1. In the web browser’s address field, type the IP address of the Mediatrix 41xx (if you have performed a recovery

mode, this is The unit’s IP address can be found by dialing *#*0 from a phone connected to port 2 or


2. Enter the default login name admin and password 1234.

 3. The system information screen then appears, giving information about the firmware version, hardware revision,

system up-time and MAC address.



Network Parameters Configuration

Network parameters are located in the Management. / Network Settings page.

1. Click Management and Network Settings to go the Network Settings web page.

 2. In the IP Address Source option, select DHCP or Static.

DHCP is the default selection.

3. If you are using a Static IP address, select Static and enter the IP address, subnet mask, default router and DNS

server IP in the proper fields.

4. If SNTP is required, set the SNTP Enable option to Enable and enter the appropriate SNTP server IP address in the

SNTP Host field.

5. Click Submit to apply the changes.





Page 9 of 16

SIP Configuration

In Asterisk, an extension is the equivalent of the SIP user in the Mediatrix 41xx. You must match the extension you have

created in Asterisk in section Creating an Extension on page 5.

1. Click SIP and Configuration to go the SIP Configuration web page.

 2. Choose Static as the SIP Server Source.

3. In the Registrar Host and Proxy Host fields, enter the address of the PC that hosts Asterisk.

4. In the User Name (equivalent to the phone number) column, enter a user name as defined in Asterisk.

5. You can also enter a Friendly Name for each port.

6. Click Submit to apply the changes.




SIP Authentication

The next step is to enter SIP authentication information for each port.

1. Click SIP and Authentication to go to the SIP Authentication web page.

You can enter up to 5 credentials for each port, but only one is needed in most cases.

2. Type asterisk in the Realm field.

3. Enter the Username 101 and Password num101 for port 1.

4. Click Submit to save the changes.


Codec and DTMF Configuration

1. Click Telephony and Codec to go to the Codec page.

 Codec and DTMF settings are configured on a port-by-port basis. 

2. Set the DTMF Transport drop-down menu according to the DTMF transport mode you have defined in Asterisk.

_ If you have used rfc2833: set the parameter to outOfBandUsingRtp. ( Default)_

_ If you have used info, set the parameter to outOfBandUsingSignalingProtocol.

_ If you have used inband, set the parameter to inBand.

In this case, OutOfBandUsingRTP is used

 3. Set the Payload Type field to 101.

 4. G.711 PCMU is the default codec.

 5. Disable G.711 VAD (aka Silence Suppression):

You must turn off the Silence Suppression feature in the Mediatrix 41xx according to this Asterisk website:

http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf ,

Asterisk uses the incoming RTP Stream as a timing source for sending its outgoing Stream. If the

incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So

in conclusion, you cannot use silence suppression. Make sure ALL SIP phones have disabled

silence suppression. There is a solution for the silence suppression problem, see bug 5374 for



6. Disable G711 Comfort Noise.


7. Enable (default) / Disable T.38 Fax if you are not using it in your VoIP setup. 

8. Click Submit to apply the changes.


Reboot the gateway.

1. Click Advanced and Reboot to go to the Reboot page.

2. Click the Reboot button on the page that displays.

The changes you just made will become effective after the unit reboots.

After the unit comes back, observe that the Ready LED should now light up (or blink if not all the ports are configured and

registered with the SIP server). You can then hook up a telephone and make some test calls with the Mediatrix 41xx.

Good Luck!




Advanced Settings

* Star Code and # Key Dial Map

By default, the Mediatrix 41xx dial map does not allow * and # keys. To do that, you must add the following dial map:

_ x.# - for speed dialling, e.g. 267#, call will be dialled right away once the # key is hit

_ *xx – for * code, e.g. *69

_ #xx – for # code, e.g. #21

Here the first dial map is (*xx|#xx). The second dial map is x.# with # removed before the number is sent to Asterisk.


Call Transfer SIP Interop Setting for Asterisk 1.4

The SIP behaviour of Asterisk in the Call transfer scenario in version 1.4 is different over previous Asterisk versions. The

following MIB on the Mediatrix 41xx must be changed. To access this MIB parameter, Mediatrix Unit Manager Network

(UMN) is required. UMN can be downloaded from the Mediatrix Download Portal:


The MIB parameter is: mediatrix.mediatrixExperimental.sipInteropMIB.sipInteropReplacesConfig. This MIB has to be set to

useReplacesNoRequire. Default value is “useReplacesWithRequire”:


For more information on the Asterisk configuration parameters and the Mediatrix products, visit these links:






Call Today

29771 Stagecoach Blvd
Evergreen CO 80439




Meditrix Asterisk configuration notes 66 Block 25 Pair Wiring Color Codes