This document describes the configuration to be applied to AudioCodes' MP-11X ATA box range to interconnect it with the Rainbow solution in SIP TLS over RTP.
Equipment version
| AudioCodes MP-112 FXS | 6.60A.371.002 |
Access to the equipment web interface
Access the AudioCodes ATA box web interface by entering its IP address in your web browser:
Enter the default ATA box user name and password:
Username: Admin
Password: Admin
Warning: Failure to edit the default password exposes the device to unauthorized access, data breaches and fraud, which can compromise the security of the entire network and system. This also increases the risk of service interruption and makes it difficult to monitor and prevent security breaches.
Network configuration
Network card configuration
To configure the network card, go to the VoIP > Network > IP Interfaces Table menu.
From this menu, edit the network card settings with the following parameters:
IP Address: IP address of the ATA box
Gateway: ATA box default gateway
Primary DNS Server IP Address: DNS server in use
NTP configuration
NTP must be configured before certificates can be imported into the device. To do this, go to the System > Application Settings menu and edit the following values:
NTP Server Address (IP or FQDN): IP address of NTP server to be used
Certificate configuration
First, you need to download the certificate chain from the Rainbow platform. To do this, follow this link
Once you've retrieved the certificate chain, you need to import it onto the device by going to the System > Certificates menu, then to the Upload certifical files from your computer > Send Trusted Root Certificate Store file from your computer to the device submenu .
Click on Browse, select the Alcatel certificate chain and click on Send File.
SIP configuration
User configuration
You need to enter the authentication information for the Rainbow user. To do this, go to the GW and IP to IP > Hunt Group > Endpoint Phone Number menu, then enter the information and profile to be used.
User authentication
To log-in users, go to menu Analog Gateway > Authentication
Fill in the following fields:
User Name: Rainbow platform user name
Password: Rainbow platform user password
SIP profile configuration
Proxy configuration
To configure the proxy, go to menu Control Network > Proxy Sets Table
In the Proxy Address tab, enter the IP address of the Rainbow platform, and in the Transport Type tab, enter TLS.
Destination port configuration
In the SIP Definitons > General parameters menu, set the"SIP Destination Port" value to 5061.
Proxy usage configuration
In the SIP Definitons > Proxy & Registration menu, set Use default Proxy to Yes.
Registration server configuration
Enable registration: Yes
Registrar name: "Rainbow platform IP address".
Registrar IP Address: "Rainbow platform IP address".
Registrar Transport Type: TLS
Call routing configuration
Configuration of outgoing calls
In the Routing > Tel to IP Routing menu, enter the following information:
Dest. IP Address: Outbound proxy domain of the Rainbow Platform
Configuring incoming calls
In the Routing > IP to Hunt Group Routing menu, enter the following information:
Media configuration
Activate SRTP
To activate SRTP, go to the Media > Media security menu :
Then set media security to enable and media security behavior to preferable .
In the SRTP Offered Suites submenu, check the first two choices as shown below:
Codec configuration
ATA box codec configuration
In the Coders and Profiles > Coders menu, fill in the codecs in the following order:
Group codec configuration
In the Coders and Profiles > Coders Group Settings menu, fill in the codecs in the following order:
Configuration of additional services
Setting up calls to voicemail
To enable the analog phone to call voicemail, you need to configure a prefix and transform the user part of the SIP To header into a "voicebox". To do this, go to the GW and IP to IP > Manipulations > Dest Number Tel -> IP menu, then edit the rule as described below:
When the user dials 99, he'll be able to consult his voicemail.
Flash-hook code configuration
To enable the ATA box to interpret the flash-hook, you need to set a sequence of digits that will be sent in DTMF to simulate a flash-hook (initially not compatible):
This flash-hook can be used to put calls on hold.
To do this, go to the DTMF and Supplementary > Supplementary Services menu and edit the hook-flash Code value to 100.
Key configuration for call forwarding
In the Analog gateway > Keypad Features menu, enter a prefix to forward calls and cancel call forwarding.