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Contents

 
  1. Introduction
  2. Features Supported
  3. Asterisk Configuration
  4. Mediatrix Units Configuration
  5. References

Introduction

This document outlines the configuration steps to set up Mediatrix analog units with the Asterisk open-source telephone system. It assumes that you have an Asterisk server properly installed with the necessary modules.

Features Supported

The following features are supported by both Asterisk and Mediatrix analog units:

  • 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
  • Music on Hold
  • Caller ID
  • PSTN Trunk Calls (via Mediatrix 1204)
  • G.711 (recommended), G.723.1 and G.726 codecs
  • Fax transmission
  • IVR

Asterisk Configuration

The following sections describe special configuration you must perform in Asterisk in order to properly work with the Mediatrix analog units. 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 units.
1. In the /etc/asterisk/sip.conf file, scroll to the bottom of the file and add the following:

[101]
type=friend
host=dynamic
mailbox=101@VoiceServer
nat=yes
qualify=yes
canreinvite=no
dtmfmode=info
context=sip
secret=num101



CN279 P3 1.png

CN279 P4 1.png

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

[sip]
exten=>101,1,Dial(SIP/101,20,RtT)
exten=>101,2,Voicemail,u101
exten=>101,102,Voicemail,b101
exten=>981,1,VoicemailMain


Notes:

 The T and t characters are used to allow blind and supervised call transfer. When the T or t option is in the Dial command,
Asterisk will ignore the canreinvite=yes command. It will thus be used as a RTP portal. See Blind Transfer and Supervised
(Attended) Transfer for more details.
You can also replace the last 101 extension of the expression with the ${EXTEN} variable (for instance, exten=>101,1,Dial(SIP/${EXTEN},20,RtT).

3. In the /etc/asterisk/voicemail.conf file, add the following:

[VoiceServer]
101=>1234,John Smith,jsmith@test.com.example

4. Execute the following command on the server to reload the configuration:
asterisk -rx "reload"



Authentication via Password

The following are the tasks you must perform in Asterisk if you want to have authentication via password:
1. In the [sip] section of the /etc/asterisk/extensions.conf file, modify the section as follows (in red bold):

[sip]
exten=>101,1,Authenticate(1234)
exten=>101,2,Dial)SIP/101,20,RtT)
exten=>101,3,Voicemail,u101

exten=>101,102,Voicemail,b101

exten=>981,1,VoicemailMain


Notes:
1234 is the password to enter.
Be sure to put the exten=>101,102,Voicemail,b101 line in comment by adding a “;” at the beginning of the line.
2. Execute the following command on the server to reload the configuration:
asterisk -rx "reload"

 Note: You can also perform authentication via voice mail. See the Asterisk documentation for more details.

Blind Transfer

Asterisk has a built-in blind transfer feature. To activate this feature, do the following:
1. In the [featuremap] section of the /etc/asterisk/features.conf file, remove the “,” at the beginning of the following line:
blindxfer=>#1
Note that #1 corresponds to the default digits a user must dial to transfer a call. You can put anything in place of “#1”.
During a call, the calling user simply has to dial #1 followed by the proper telephone number to perform a blind call transfer.
2. Execute the following command on the server to reload the configuration:
asterisk -rx "reload"

Supervised (Attended) Transfer
Asterisk has a built-in supervised transfer feature. The following are the tasks you must perform in Asterisk:
1. In the [featuremap] section of the /etc/asterisk/features.conf file, remove the “,” at the beginning of the following line:
atxfer => *2
Note that *2 corresponds to the default digit a user must dial to signal a call transfer. For instance, with an active call, the user must dial *2 followed by the proper telephone number to perform a supervised call transfer. You can put anything in place of “*2”.
2. Execute the following command on the server to reload the configuration:
asterisk -rx "reload"

Conference

If you want to add a password to the conference, proceed as follows:
1. In the [rooms] section of the /etc/asterisk/meetme.conf file, add the following line:
conf => 8101,1234
where 1234 is the password you want to use for the conference on extension 101. The 101 extension must exist.
2. In the /etc/asterisk/extensions.conf file, add the following section:
[conference]
exten=>8101,1,Meetme(8101)
Notes:
The first 8101 corresponds to the number to dial.
The second 8101 corresponds to the room number.
3. In the [sip] section of the /etc/asterisk/extensions.conf file, add the following line:
include => conference
4. Execute the following command on the server to reload the configuration:
asterisk -rx "reload"

 Note: If the conference number is invalid, make sure that the ztdummy module is properly installed and configured.

Outgoing Calls via a Mediatrix 1204

You must create a new SIP trunk:
1. In the /etc/asterisk/sip.conf file, scroll to the bottom of the file and add the following:
[1204Gateway]
type=peer
host=192.168.0.10
mask=255.255.255.255
canreinvite=yes
dtmfmode=info
context=sip
CN279 P6 1.png

CN279 P7 1.png

2. In the [sip] section of the /etc/asterisk/extensions.conf file, add the following calling route:
exten=>_1418XXXXXXX,1,Dial(SIP/${exten:4}@1204Gateway)
Notes:
1 represents the priority of the route. With several routes, define the priority carefully. For routes that match long distance numbers, for example, you'd want to pick the cheapest routes for long distance (ie, VoIP trunks first) followed by more expensive routes (POTS lines).
exten:4 indicates that when dialing, Asterisk will strip the first four digits of the number.
Add any other calling rule as required by your system.
3. Execute the following command on the server to reload the configuration:
asterisk -rx "reload"

Mediatrix Units Configuration

The following sections describe special configuration you must perform in the Mediatrix units in order to properly work with Asterisk.
Running the Unit Manager Network Software
The Unit Manager Network (UMN) software is a configuration and management tool for Mediatrix units. The UMN is provided on the CD included with the unit. It has a default 3-units limit upon installation. This will suffice for most configurations.
It is recommended that the administrator uses the UMN to configure the Mediatrix units. Please refer to the UMN Quick Start guide for the installation of the software. Once the UMN software has been installed on your PC, proceed with the following steps.
1. Start the UMN by selecting Start Menu > Programs > Unit Manager Network 3.2 > Unit Manager Network.
2. In the Administrator login window (Connect to Unit Manager), a User Name and Password are not required. Click OK to proceed.
3. On the left pane, right-click the Unit Manager level, then select Autodetect.
CN279 P8 1.png

4. Set the IP Address Range to minimize the time it takes to auto-detect the unit. Click Start to begin Mediatrix unit detection. When the unit is detected, the Result section lists the unit.
5. Select the unit and click OK. If no DHCP server is used in your subnet, you must connect one unit at a time since they will start by using the default IP address 192.168.0.1 after a recovery reset. You will have to set a different static IP address for every unit. Please see the 1100 series and 1204 SIP Quick Start Guide for more details on initial setup.
CN279 P9 1.png

SIP Configuration
In Asterisk, an extension is the equivalent of the SIP user in the Mediatrix units. You must match the extension you have created in Asterisk in section Creating an Extension on page 3.
1. Once the Mediatrix unit has been auto-detected in the UMN, it should appear under the Unit Manager level. Right-click the Sip level of the Mediatrix unit and select the Edit menu.
2. In the SIP Registrar and SIP proxy fields, enter the address of the PC that hosts Asterisk.
3. In the User Name column, enter a user name as defined in Asterisk.
4. If you have entered a secret in the Asterisk authentication information, click the Authentication button.
5. Enter asterisk as the Realm.
6. In the Password column, enter the same secret as set in Asterisk.
7. Click OK.
8. Click OK once more.
CN279 P10 1.png

9. Right-click the unit and select Edit SNMP.
CN279 P10 2.png

10. On the top menu bar, check the option Automatic GET.
11. Go to iso>org>dod>internet>private>enterprises>mediatrix>mediatrixConfig>voiceIfMIB>voiceIfMIBObjects> voiceIfDtmfTransportTable>voiceIfDtmfTransport and set this parameter according to the DTMF transport mode you have defined in the AMP.
CN279 P10 3.png

 Note : Mediatrix recommends to use the outOfBandUsingSignalingProtocol method.

12. If you have used the info DTMF transport, go to iso>org>dod>internet>private>enterprises>mediatrix> mediatrixExperimental>sipExperimentalMIB>sipExperimentalMIBObjects>sipInteropDtmfTransportBySipProtocol> sipInteropDtmfTransportmethod and set this parameter to infoDtmfRelay.
CN279 P11 1.png

Silence Suppression
You must turn off the Silence Suppression feature in the Mediatrix units.
1. In the UMN, right-click the Port level of the Mediatrix unit and select the Edit menu.
2. In the G.711 Silence Suppression column, change all settings to OFF.
3. Click OK.
CN279 P11 2.png

Conference

You must set the ptime to 20 ms.
1. In the UMN, right-click the unit and select Edit SNMP.
CN279 P12 1.png

2. Go to iso>org>dod>internet>private>enterprises>mediatrix>mediatrixConfig>voiceIfMIB>voiceIfMIBObjects> voiceIfCodectTable>voiceIfCodecPcmuMinPTime and set this parameter to 20.
3. Set the voiceIfCodecPcmuMaxPTime parameter to 20.
CN279 P12 2.png