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Configuring
Mediatrix 4108, 4116, 4124, 1102, 1104 FXS SIP Gateway with the Asterisk
IP PBX System
Introduction
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 5.0.18.113
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:
[101]
type=friend
host=dynamic
nat=yes
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context=sip
username=101
secret=num101
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Parameter |
Description
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[101] |
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.
|
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type |
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.
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host |
How to
find the client - IP # or host name. Using the keyword
dynamic
indicates
that the phone
registers
itself.
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nat |
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.
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qualify |
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
setting.
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canreinvite |
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.
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|
dtmfmode |
You have
the choice between rfc2833, info, and inband.
You will have to enter similar
information in the Mediatrix 41xx.
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|
context |
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).
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secret |
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.
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2.
In the
/etc/asterisk/extensions.conf file, add the following:
[sip]
exten=>101,1,Dial(SIP/101)
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
Asterisk.
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:
https://support.mediatrix.com/DownloadPlus/Download.asp
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
192.168.0.1). The unit’s IP address can be found by dialing
*#*0
from a phone
connected to port 2 or
above.
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
details
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.
Done!
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:
https://support.mediatrix.com/DownloadPlus/Download.asp
The MIB parameter
is:
mediatrix.mediatrixExperimental.sipInteropMIB.sipInteropReplacesConfig.
This MIB has to be set to
useReplacesNoRequire.
Default value is “useReplacesWithRequire”:
References
For more
information on the Asterisk configuration parameters and the Mediatrix
products, visit these links:
http://www.voip-info.org/
https://support.mediatrix.com/DownloadPlus/Download.asp
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