Mediatrix 4108, 4116, 4124, 1102, 1104 FXS SIP Gateway with the Asterisk
IP PBX System
outlines the configuration steps to set up Mediatrix 4108, 4116, 4124,
1102, 1104 series FXS gateways with the Asterisk open-source
It assumes that you have an Asterisk server properly installed with the
necessary modules. If you need
assistance to configure your Asterisk server, the Mediatrix technical
team can provide appropriate support to help
you realizing your
Mediatrix 4100 Series FXS Gateways
products are high quality
branch offices and multitenant
buildings to an IP
investment in analog telephones
to deploy rapidly and
solutions in medium-size
premises and they
are the ideal solution for
connectivity to larger private
following Mediatrix 41xx
Enterprise 8-port FXS gateway
Enterprise 16-port FXS gateway
Enterprise 24-port FXS gateway
Benefits of using
the Mediatrix 41xx over other Asterisk FXS solutions:
Voice quality with dedicated DSP hardware
Jitter Buffer, G.168 Echo Cancellation
_ T.38 Fax
_ Can be
used to extend the Asterisk system in remote locations
flexible, easily adaptable to VoIP systems other than Asterisk
configuration to work with Asterisk
This is a typical
deployment scenario of Mediatrix 41xx units in a SIP-based MTU/MDU
environment. The units provide
analog phones and fax machines and analog trunking to legacy PBX/KSU
(for small business) in each
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
The Asterisk IP
PBX – provides:
routing, Dial Plan
routes to local PSTN gateways
services (voicemail, call forwarding etc)
features are supported by both Asterisk and Mediatrix 41xx:
_ RFC 2833,
SIP INFO, and inband DTMF transports
_ Blind and
Supervised Call Transfer
Forward On Busy/On No Answer/Unconditional
Waiting (via Asterisk only)
_ Music on
_ Caller ID
(recommended), G.723.1 and G.726 codecs
note was written and validated using the following platforms and
1.2.4 running on RedHat Linux 9
41xx: SIP 188.8.131.52
configuration notes are not a substitute for the Mediatrix
Administration Documentation for Mediatrix 41xx
have the following manuals available for reference:
4108, 4116, and 4124 SIP Reference Manual
4108, 4116, and 4124 SIP Quick Start
Unit Manager Network Administration Manual
relevant Asterisk Manuals
sections describe special configuration you must perform in Asterisk in
order to properly work with the
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.
/etc/asterisk/sip.conf file, scroll to the bottom of the file and
add the following:
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
of the SIP object. There are three values available:
A SIP entity to which Asterisk sends calls (a SIP provider for
you want a
user (extension) to have multiple phones, define an extension
SIP peers. The peer authenticates at registration.
A SIP entity which places calls through Asterisk.
An entity which is both a user and a peer. This make sense for
and other devices.
If a peer
is defined with
allowed to register with Asterisk to tell Asterisk
can be found (IP address/host name) and that it is reachable
from now on.
find the client - IP # or host name. Using the keyword
that the phone
Asterisk will change the behaviour, addressing, etc. of
(SIP UA) that is behind a NAT device, to make communication
that Asterisk keeps a UDP session open to a device that is
network address translator (NAT). This can be used in
conjunction with the
that Asterisk does not issue a reinvite to the client and acts
as the RTP
The Mediatrix 41xx supports both yes and no
the choice between rfc2833, info, and inband.
You will have to enter similar
information in the Mediatrix 41xx.
the context to which the extension belongs as defined in the
extensions.conf file (see
Step 2 for
want to use authentication for this extension, enter the proper
information in the secret
will have to enter similar information in the Mediatrix 41xx.
Reference source not found.
on page 10
for more details.
/etc/asterisk/extensions.conf file, add the following:
following command on the server to reload the configuration:
asterisk -rx "reload"
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
the web interface. The Mediatrix Unit Manager Network (UMN) software is
needed if access to full unit
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
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.
numerous parameters of the Mediatrix 41xx.
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:
In the web
browser’s address field, type the IP address of the Mediatrix 41xx (if
you have performed a recovery
mode, this is
192.168.0.1). The unit’s IP address can be found by dialing
from a phone
connected to port 2 or
Enter the default
system information screen then appears, giving information about the
firmware version, hardware revision,
system up-time and
Network Parameters Configuration
are located in the Management. / Network Settings page.
the Network Settings web page.
In the IP
Address Source option, select
DHCP is the
If you are using a
Static IP address, select
and enter the IP
address, subnet mask, default router and DNS
server IP in the
If SNTP is
required, set the SNTP Enable option to
and enter the
appropriate SNTP server IP address in the
to apply the
Page 9 of 16
In Asterisk, an
extension is the equivalent of the SIP user in the Mediatrix 41xx. You
must match the extension you have
Asterisk in section Creating an Extension on page 5.
the SIP Configuration web page.
SIP Server Source.
Registrar Host and Proxy Host fields, enter the address of
the PC that hosts Asterisk.
In the User
Name (equivalent to the phone number) column, enter a user name as
defined in Asterisk.
You can also enter
a Friendly Name for each port.
to apply the
The next step is
to enter SIP authentication information for each port.
to the SIP Authentication web page.
You can enter up
to 5 credentials for each port, but only one is needed in most cases.
for port 1.
to save the
Codec and DTMF Configuration
to go to the
Codec and DTMF
settings are configured on a port-by-port basis.
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
_ If you
have used info, set the parameter to
_ If you
have used inband, set the parameter to
In this case,
Payload Type field to
G.711 PCMU is the
Disable G.711 VAD
(aka Silence Suppression):
You must turn off
the Silence Suppression feature in the Mediatrix 41xx according to this
the incoming RTP Stream as a timing source for sending its outgoing
Stream. If the
is interrupted due to silence suppression then musiconhold will be
you cannot use silence suppression. Make sure ALL SIP phones have
suppression. There is a solution for the silence suppression problem,
see bug 5374 for
Enable (default) /
Disable T.38 Fax if you are not using it in your VoIP setup.
to apply the
to go to the
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
the SIP server). You can then hook up a telephone and make some test
calls with the Mediatrix 41xx.
* 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:
speed dialling, e.g. 267#, call will be dialled right away once the #
key is hit
– for *
code, e.g. *69
– 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
This MIB has to be set to
Default value is “useReplacesWithRequire”:
information on the Asterisk configuration parameters and the Mediatrix
products, visit these links: