Connecting with Medicatrix 2102The Mediatrix 2102 is a high-quality and cost efficient VoIP gateway connecting SOHOs to an IP network, while preserving investment in analog telephones and faxes. It connects up to two analog phones and/or faxes, as well as a PC or a home router to an IP network over a single broadband connection. www.mediatrix.com
Configure Mediatrix 2102Once the IP address is configured you can access Mediatrix 2102 FXS Gateway webpage by http://192.168.10.1/ (example of selected IP address).Please follow the following steps once the webpage opened.
- Enter User name: admin (default)
- Enter Password: 1234 (default)
- Click on [OK] button
SIP Setting and RestartThe FXS Gateway will need to register to the SIP Server, so it needs to know the SIP Server's IP address. Please follow the following steps to complete the configuration.
1. SIP Configuration Setting
- Open the Unit Manager Network software which you can obtain from Mediatrix web site and discover the device.
- Enter the “SIP Registrar” :192.168.0.167 (Ex)
- Enter the “SIP Proxy”:192.168.0.167(ex)
- Enter the “User Name”/Phone number: 222/333 (Ex)
- Click on [OK] button
2. Networking Setting
2.1 System > WAN
- Enter the “WAN IP Address”: 192.168.0.99(Ex)
- Enter the “WAN Network Mask”:255.255.255.0 (Ex)
- Enter the “Default Gateway”:192.168.0.2 (Ex)
- Enter the “Primary DNS”:XXX.XXX.XXX.XXX
- Enter the “Secondary DNS”:XXX.XXX.XXX.XXX
- Click on [Submit] button
Brekeke SIP Server’s Registration PageClick the Registered tab of Brekeke SIP Server admintool.
Mediatrix 2102 is registered with Brekeke SIP Server.
Additional Telephony Setting
1. Call Pickup: Telephony > Digit MapsCall Pickup is a function of Brekeke PBX that allows users to dial “*” to answer incoming calls from any extension in the same Call Pickup group. By default, * is not allowed to dial. Please complete the following setting to allow *.
- Entry the “Digit Map Allowed” in the first row: *.T
- Digit Map Characters:
T X . * The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished dialing and the SIP Server can make the call Matches any digit, excluding “#” and “*” Indicates a choice of matching expressions (OR). use “ *” to answer incoming calls from any extension in the same Call Pickup group
2. DTMF Setting: Telephony > CODECSet DTMF Transport Using SIP INFO, if RTP relay = off in Brekeke PBX. The following explains how to set Mediatrix to use SIP INFO.
To enable DTMF Transport Using SIP INFO by Unit Manager Network
- In the voiceIfMIB, set the DTMF transport type in the voiceIfDtmfTransport variable (voiceIfDtmfTransportTable group): outOfBandUsingSignalingProtocol
- In the sipInteropMIB> sipInteropDtmfTransportBySipProtocol, set the DTMF transport type in the sipInteropDtmfTransportMethod variable : infoDtmfRelay and in the sipInteropDtmfTransportDuration variable:160
- Set the DTMF duration sent in the INFO message when using the infoDtmfRelay method to transmit DTMFs in the sipInteropDtmfTransportDuration variable. This value is expressed in milliseconds (ms). The default value is 100 ms
- In the analogScnGwMIB, set the DTMF duration when using the infoDtmfRelay method to receive. DTMFs in the analogScnGwDtmfDuration variable. This is the duration, in milliseconds (ms), a DTMF is played when dialing the destination phone number.
- Set an inter-digit dial delay in the analogScnGwInterDigitDial Delay variable. This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone number. This is useful when the Mediatrix 2102 receives DTMFs out-of-band faster than it can signal them.
- Restart the Mediatrix 2102 so that the changes may take effect.