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Getting started
Danger, Warning, Caution, and Note Definitions
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Logon
Logging on to the Mediatrix Unit Web Interface
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Mediatrix Unit Reset
Basic Reset Concepts
Partial Reset
The partial reset provides a way to contact the Mediatrix unit in a known and static state while keeping most of the configuration unchanged.
A partial reset can be performed at the initial start-up of the Mediatrix unit or on a unit already in use where the configuration was modified in such a way that the user can no longer access the system by the Web page or otherwise. In both cases, the user can manage the Mediatrix unit through its Rescue interface, which is bound to the unit's WAN port (wan for the Mediatrix 4102S, and ETH1 for all other Mediatrix units). The IP address of the Rescue interface is 192.168.0.1 (IPv4) or an IPv6 Link Local address.. These connections give access to the Rescue Management Interface where the configuration of a new unit can be completed and where an existing configuration can be modified.
By default the Rescue Network Interface is disabled. When a partial reset is performed, the Rescue network Interface becomes enabled and the "Power" and "Ready" LEDs are blinking at 1Hz with 75% duty and all other LEDs are off. Once the configuration has been modified to solve the problem that required the partial reset, it is important to disable the Rescue Network Interface to make sure that you are no longer working in the Rescue Network Interface.
- Rollback Local Firewall settings that are not yet applied.
- Add a Local Firewall rule to allow complete access to the Rescue interface.
- Rollback NAT settings that are not yet applied.
- Add NAT rule to allow complete access to the Rescue interface.
- Cancel the changes that were being modified but not yet applied to the configuration.
- Disable any Network Interface in conflict with the Network Rescue Interface.
- Configure and enable the Rescue Network Interface to:
- use the link as the default value used by the Uplink Network Interface
- set the IP address to 192.168.0.1 and the Network Mask to 255.255.255.0.
- set the IPv6 link-local address on all network links. The IPv6 link-local address can be found underneath the unit.
Service | Parameter | Default Value |
---|---|---|
AAA | Users.Password | User(s) from profile are restored with their factory password. All other usernames keep their password. |
Users.AccessRights | User(s) from profile are restored with their factory rights. | |
ServicesAaaType (table) | Each service will be configured to use Local authentication and no accounting mechanism. | |
CLI | EnableTelnet | Disable |
TelnetPort | 23 | |
EnableSsh | Enable | |
SshPort | 22 | |
InactivityTimeOut | 15 | |
Ha | servicesInfoExecState (Scm) | Stopped |
servicesInfoStartupType (Scm) | Manual | |
HOC | ManagementInterface | Rescue |
SNMP | Port | 161 |
EnableSnmpV1 | Enable | |
EnableSnmpV2 | Enable | |
EnableSnmpV3 | Enable | |
Web | ServerPort | 80 |
SecureServerPort | 443 |
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Factory Reset
The Factory reset reverts the Mediatrix unit back to its default factory settings.
- User files stored in the File service
- Certificates, except for factory installed ones
- Log files of the File service
- Directly on the unit. Refer to Performing a Factory Reset.
- Via the web interface of the Mediatrix unit (Management/Firmware Upgrade).
- Via the Command Line Interface of the Mediatrix unit by using the fpu.defaultsetting parameter.
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RESET/DEFAULT Button
The Reset/Default button is a switch that can be used to perform a partial or factory reset while the unit is running.
- Cancel an action that was started.
- Revert to known factory settings if the Mediatrix unit refuses to work properly for any reason or the connection to the network is lost.
- Reconfigure a unit.
LED Pattern | Action | Comment |
---|---|---|
Power1 blinking, all other LEDs OFF | Restarts the Mediatrix unit. | No changes are made to the Mediatrix unit settings. |
All LEDs blinking, 1cycle per second, 50% duty | Initiates a Partial Reset of the Mediatrix unit. Note: The partial reset is
optional as it can be disabled with the CLI
Hardware.ResetButtonManagement parameter. For more details,
refer to the DGW Configuration Guide - Reference Guide
published on the Media5 Documentation Portal. |
Restarts the unit in a known and static state while keeping most of the configuration unchanged. |
All LEDs steady ON | Initiates a Factory Reset of the Mediatrix unit. | Reverts the unit back to its default factory settings. |
All LEDs will become OFF after blinking and being steady on. | No action is taken. This is useful if you accidentally pushed the button and do not need and action to be applied. | The action is ignored. |
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Basic Reset Tasks
Performing a Partial Reset
The Rescue Network Interface is displayed when accessing the Management Interface. Several parameters and services are modified, refer to Partial Reset. Do not forget to perform the Disabling the Rescue Interface step.
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Disabling Partial Reset - ResetButtonManagement
- Open CLI (Command Line Interface).
- Set ResetButtonManagement to DisablePartialReset.
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Performing a Factory Reset
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Personnal Data Usage and Protection
Personal Data Exposure
Personal Data Collection
Mediatrix products collect the basic personal data required for the proper delivery of the telecommunication service. The actual collected data depends on the type of users and how the Mediatrix products are administrated.
Type of users | Collected Personal Information | Collected Activity Information |
---|---|---|
End-Users | Name and phone number used to register to the telecommunication provider service. | Calls history for billing purposes and call details and recordings for
troubleshooting purposes. For example:
|
System Administrators and Technical Support | Account name and password used to access the product for administrative and troubleshooting purposes. |
|
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Personal Data Processing
Personal data is processed in Mediatrix products through the following activities:
- Configuring and storing end-user data
- Recording voice and fax calls
- Logging call history (CDR)
- Logging administration audit trails
- Access of the personal data by an authorised system administrator
- Provisioning data
- Maintenance, administration and technical support records
- Audit trails
- End-user activity records
- End-User personal content
- Recording voice and fax calls for troubleshooting
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Personal Data Transfers
The following collected personal data may be transferred to other systems, depending on how the device administrators configure the Mediatrix products.
- Call Details Records (CDR) may be sent to an external call accounting system.
- Logs may be sent over an external monitoring system for live troubleshooting.
- Administration activity logs may be sent over an external monitoring system for auditing.
- Backups of the Mediatrix products, containing collected personal data, may be retrieved by an authorised system administrator.
- Network captures from the Mediatrix products, containing collected personal data, may be retrieved by an authorised system administrator for troubleshooting purposes.
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Personal Data Protection
System and Data Protection
To protect the end-user personal data stored inside the Mediatrix devices, the device administrator should control and restrict access to the management interfaces by:
- Forcing the use of a strong authentication password
- Authorising LAN access only
- Using the device firewall service to limit the remote access to the device to only authorized peers and authorised services
- Using an external firewall
- Enabling IEEE 802.1x authentication of Ethernet link
The device administrator may also enforce the use of encryption and authentication for a secure administration of the Mediatrix devices:
- Authenticated Management Interfaces:
- Web Interface: HTTPS with trusted certificates
- CWMP: HTTPS with trusted certificates
- CLI: SSH
- Secure Management Operations:
- Consult or retrieve the stored personal data: HTTPS with trusted certificates
- Provisioning: HTTPS with trusted certificates
- Firmware upgrades: HTTPS with trusted certificates
- Backup/restore: HTTPS with trusted certificates
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Communications Protection (VoIP Calls)
The device administrator may configure the encryption of the data that transits through Mediatrix products:
- Call signalling: SIP over TLS with trusted certificates
- Media packets: SRTP
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Access and Communications
- Administrator
- User (no password access)
- Observer (read-only)
All the management interfaces are restricted to authorised accounts only, verified by username and password. Refer to the System and Data Protection section for the list of management interfaces and how to protect them.
The account credentials may be stored locally in the Mediatrix devices or in an external RADIUS authentication server.
In all cases, the device administrator should restrict the physical access to the Mediatrix products.
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Data Deletion
The Mediatrix products allow an authorised system administrator to delete end-user registration information (name and number).
- call history
- call recordings
- network captures
A factory reset can be performed by a system administrator to revert a Mediatrix device back to its default factory state through a factory reset, thus erasing all the collected data and configuration.
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Audit
Audit trail logs of the system administrator activities may be enabled by the device administrator. These audit logs may be temporarily stored locally or sent through syslog to an external monitoring system.
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Management Interfaces

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IPv6
Basic IPv6 Concepts
IPv6
IPv6 (Internet Protocol version 6) is the successor to the most common Internet Protocol today (IPv4).
This is largely driven by the fact that IPv4 32-bit addresses are quickly being consumed by the ever-expanding sites and products on the Internet. IPv6 128-bit address space should not have this problem for the foreseeable future.
IPv6 addresses, in addition to being longer, are distinguished from IPv4 addresses by the use of colons ":", e.g. 2001:470:8929:4000:201:80ff:fe3c:642f. An IPv4 address is noted by 4 sets of decimal numbers separated by periods ".", e.g. 192.168.10.1.
Please note that IPv6 addresses should be written between [ ] to allow port numbers to be set.
For instance, [fd0f:8b72:5::1]:5060.
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IPv6 Availability
- CPE WAN Management Protocol (CWMP)/TR-069
- DHCP embedded sever
- IP Routing
- Local Firewall (LFW)
- Network Firewall (NFW)
- Network Address Translation (NAT)
- Online Certificate Status Protocol (OSCP)
- Remote Authentication Dial In User Services (RADIUS)
- Session Border Controller (SBC)
- Simple Network Management Protocol (SNMP)
- PPPoE
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IPv6 link-local Addresses
IPv6 link-local addresses start with fe80 and must include the scope identifier
Therefore, the format of a link-local address is: [IPv6 link-local%ScopeIdentifier].
- On Windows: the network link used to contact the IPv6 link-local address.
- On Linux: the link name or the interface number.
For example, if the unit must contact a server at the IPv6 link-local fe80::201:80ff:fe3c:642f address, you must check on which network link the server is available. Some units have WAN or LAN. If it is on the WAN link, the IP address would then be "[fe80::201:80ff:fe3c:642f%wan]".
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IPv6 Basic Tasks
Locating the Scope Identifier of fe80 IPv6 Addresses on Windows

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Locating the Scope Identifier of fe80 IPv6 Addresses on Linux

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Naming Conventions
When defining a name for a parameter, only ascii characters are authorised.
This is valid when defining a name for a parameter in a Web Page of the Management interface, but also for parameters accessed via the CLI, the MIB, or a script.
For example, to be valid, the Service Name defined during PPPoE configuration must only contain ascii characters. Special characters such as " " (space), """ (double quote), "“" (left double quote), "‘" (left single quote), "#", "£", "¢", "¿", "¡", "«", "»" will cause the system to display a syntax error message.
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ASCII Special Characters
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Unit Macros
Macro | Description |
---|---|
%mac% | the MAC address of the unit |
%version% | the MFP version of the unit (firmware version) |
%product% | the Product name of the unit |
%productseries% | the Product series name of the unit. |
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Command Line Interface (CLI)
CLI Basic Concept
Command Line Interface (CLI)
The Command Line Interface (CLI) provides an access to interactively configure all the Mediatrix unit parameters.
The command interpreter interface of the CLI allows the user to browse the unit parameters, write the command lines, and display the system's notification log.
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CLI Basic Tasks
Creating the Telnet Session Activation Configuration Script

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Accessing the CLI via a Telnet Session
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Accessing the CLI via an SSH Session
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File Servers
Configuring the FTP Server
Perform this procedure if you plan to use the FTP transport protocol.
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Configuring the HTTP Server
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Configuring the HTTPS Server
Make sure the unit is set to the proper date (refer to Configuring the Mediatrix Unit to Use an SNTP Server.
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Configuring the TFTP Server
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System
Information
Activating a Licence Using the Web Interface

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Services
Basic Service Concepts
Services
The Mediatrix unit uses many services to carry out tasks and support features.
- system services : You cannot perform service commands on system services. The service is
restarted by using the Reboot
button located under the Reboot
tab of the DGW Web interface.Note: If the unit is in use when clicking Reboot, all calls are terminated.
- user services: You can perform service commands on user services. The service is restarted by using the start

When a service needs to be restarted, the restart required services is systematically displayed. If you are not able to restart a service because it is a system service, click the Reboot link in the top menu. The Reboot page then opens. You must click Reboot. This restarts the Mediatrix unit. If the unit is in use when you click Reboot, all calls are terminated
Services can also be restarted via the CLI using the Scm.ServiceCommandsRestart parameter.
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System Services vs User Services
The Mediatrix unit uses many services to carry out tasks and support features.
- Start
- Stop
- Restart
- system services: the user cannot perform service commands. The service can only be restarted by rebooting the unit. Refer to Restarting a System Service.
- user services: the user can perform service commands. The service can be restarted by the user. Refer to Setting the Service Start-up Type
When a system or a user service needs to be restarted, the Some changes require to restart a service to apply new configuration message is systematically displayed on the DGW Web pages. Also, every service has the NeedRestartInfo CLI/MIB parameter to indicate if the service needs to be restarted for the configuration to fully take effect.
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DGW User Services
User Service | Description |
---|---|
Basic Network Interface (Bni) | Manages the layer 3 network interfaces. |
Call Detail Record (Cdr) | Allows the administrator to generate custom call notifications with information such as endpoints, point of origin, duration, etc. |
Command Line Interface (Cli) | Allows the administrator to manage the unit using the SSH or TELNET protocols. |
Call Routing (Crout) | Transforms properties and routes calls between telephony interfaces and SIP endpoints. |
CPE WAN Management Protocol (Cwmp) | Allows the administrator to manage the unit using the TR-069 protocol. |
DHCP Server (Dhcp) | Manages a DHCP server on each network interface. |
E&M Channel Associated Signaling (Eam) | Manages the E&M CAS telephony interfaces. |
Element Management System for Virtuo (EMS) | Makes the unit compatible with the Virtuo EMS infrastructure. |
Endpoint Administration (EpAdm) | Allows for high-level management of telephony endpoints. |
Endpoint Services (EpServ) | Manages the telephony services of each endpoint. |
IP routing (IpRouting) | Manages the unit's IP routing table. |
Integrated Services Digital Network (Isdn) | Manages the ISDN parameters for BRI and PRI telephony interfaces. |
Local Firewall (Lfw) | Allows the administrator to filter the network with the unit as final destination. |
Link Layer Discovery Protocol (Lldp) | Manages the IEEE 802.1ab protocol used for advertising the unit's capabilities on the network. |
Media IP Transport (Mipt) | Manages the voice and data encodings over the IP network. |
Music on Hold (Moh) | Manages the option to play an audio file when a telephony endpoint is on hold. |
Network Address Translation (Nat) | Allows the administrator to change the source or destination IP address of a packet. |
Network Firewall (Nfw) | Allows the administrator to filter traffic that is routed between networks. |
Network Traffic Control (Ntc) | Allows the administrator to perform traffic shaping on the network interfaces. |
Plain Old Telephony System Line (Pots) | Manages the FXS and FXO analog telephony interfaces. |
R2 Channel Associated Signaling (R2) | Manages the E1 CAS telephony interfaces |
Session Border Controller (Sbc) | Allows the administrator to perform SIP to SIP normalization, call routing, NAT traversal, and survivability. |
SIP Endpoint (SipEp) | Allows the administrator to associate telephony endpoints with SIP user agents. |
SIP Proxy (SipProxy) | The SIP Proxy (SipProxy) service is used to add local survivability for local endpoints and SIP phones. |
Simple Network Management Protocol (Snmp) | Allows the administrator to manage the unit using the SNMP protocol. |
Telephony Interface (TelIf) | Manages tone generation and detection on the telephony interfaces. |
Virtual Machine (Vm) | Allows the administrator to manage virtual machines. |
Web Service (Web) | Manage the unit using HTTP(S) web pages. |
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DGW System Services
System Service | Description |
---|---|
Authentication, Authorization and Accounting (Aaa) | Manages the administrator accounts and grants or denies access to various parameters. |
Certificate Manager (Cert) | Manages the security certificates used for the authentication of the unit and its peers before establishing a secure connection. |
Configuration Manager (Conf) | Allows executing configuration scripts as well as performing backup/restore of the unit configuration. |
Device Control Manager (Dcm) | Manages the hardware properties as well as the licence activation keys. |
Ethernet Manager (Eth) | Manages the unit Ethernet link interfaces. |
File Manager (File) | Allows the administrator to manage the files stored on the unit. |
Firmware Pack Updater (Fpu) | Manages firmware upgrade, downgrade and rollback operations. |
Host Configuration (Hoc) | Manages the IP host parameters and other system settings. |
Local Quality Of Service (LQos) | Manages the QoS parameters applicable to the unit. |
Notifications and Logging Manager (Nlm) | Manages the routing and filtering of the unit's event notification messages. |
Process Control Manager (Pcm) | Manages the startup and shutdown sequence of the system. |
Service Controller Manager (Scm) | Allows the administrator to enable or disable services. |
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Basic Service Tasks
Setting the Service Start-up Type
- Go to System/Services.
-
In the User Service
table, from the Startup Type selection list
located next to the service you wish to set.
- choose Auto if you wish the service to start automatically when the system starts, or
- choose Manual to start to the service manually when needed.
- Click Apply.
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Starting/Stopping/Restarting a User Service Using the DGW Web Page
- If you clicked
, the tab from which you can access the service from the Web pages are greyed out
- If you clicked
, the tab from which you can access the service from the Web pages are no longer greyed out.
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Disabling a Service
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Restarting a Service with a Grace Delay
- Go to System/Services.
- In the Restart Required Services table, set the Graceful Delay (min) field.
- Click Restart Required Services.

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Restarting a System Service
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Advanced Service Tasks
Starting/Stopping/Restarting a User Service Using a MIB Browser
- Open a MIB Browser
- Navigate to the Service that needs to be restarted.
- Locate the needRestartInfo parameter to determine if the service needs to be restarted.
- In the scmMIB, locate the serviceCommandsTable .
-
In the serviceCommandsName column, locate the service to
restart.
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Starting/Stopping/Restarting a Service Using the CLI
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Hardware
Hardware Basic Tasks
Selecting the Source of the Clock Reference
- Go to System/Hardware.
- From the Clock Reference Configuration table, select from the Suggestion list, several clock reference sources.
- Click Apply.

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Selecting the Port Used for Synchronisation
- Access the DGW Web interface of your unit.
- Go to System/Hardware.
- In the Clock Reference Configuration table, from the Suggestion selection list, choose SYNCIN.

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Associating a PRI Port to a Line Type and Protocol
- Go to System/Hardware.
- In the PRI Ports Configuration table, from the Line Type selection list, select either E1 or T1.
- From the Signaling selection list, associate a type of signaling to the PRI port.
- Click Apply.
- Restart the unit.
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Setting the Mediatrix Unit to Use the R2 Signaling Protocol

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Setting the Mediatrix Unit to Use the E&M Signaling Protocol

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Cabling Several Units for TDM Synchronisation
- Connect a standard Ethernet cable to the SYNC OUT port of the first device.
- Connect the other end of the Ethernet cable to the SYNC IN port of another device.
- Connect all your units in the same fashion.

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Hardware Advanced Parameters
Some aspects of the Hardware configuration can only be completed with the MIB parameters.
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration variables
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Endpoints State Configuration
Basic Endpoints State Concepts
Administrative State of Unit
The administrator can disable the use of specific endpoints or all endpoints of a unit through the administrative state configuration. It is also possible to select the initial endpoint state to be applied on unit start-up.
- in a multi-tenant environment, if a tenant stops its service subscription, the administrator can lock the unit so the FXS port can give either a fast busy tone or no tone to signal the line is decommissioned. Then later the administrator can send a technician on site to re-wire and make the port available to another tenant.
- if a user does not pay his service, the administrator can simply lock the endpoint.
- to prevent calls during unit maintenance
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Basic Endpoint State Tasks
Locking All Unit Endpoints - Gracefully
- Go to System/Endpoints.
- In the Unit States table, from the Action selection list, choose Lock.
- If the state of the unit is Idle or Idle Unusable, the unit is locked right away.
- If the unit is either Busy or Active, the unit will be
locked only when it will become Idle.
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Locking All Unit Endpoints - Immediately
- Go to System/Endpoints.
- In the Unit States table, from the Action selection list, choose Force Lock.

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Unlocking All Unit Endpoints
- Go to System/Endpoints.
- In the Unit States table, from the Action selection list, choose Unlock.

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Locking an Endpoint - Gracefully
- Go to System/Endpoints.
- In the Endpoint States table, from the Action selection list of an endpoint, choose Lock.
- If the state of the endpoint is Idle or Idle Unusable, the endpoint is locked right away.
- If the endpoint is either Busy or Active, the endpoint will be
locked only when the unit will become in Idle.
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Locking an Endpoint- Immediately
- Go to System/Endpoints.
- In the Endpoint States table, from the Action selection list of an endpoint, choose Force Lock.
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Unlocking an Endpoint
- Go to System/Endpoints.
- In the Endpoint States table, from the Action selection list of an endpoint, choose Unlock.
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Setting the Endpoint Behavior after a Unit Restart
- Go to System/Endpoints.
- In the Endpoint States table, from the Initial Administrative selection list of an endpoint, choose Lock or Unlock.
- If the Initial Administrative selection list is set to Lock, when the unit restarts, the endpoint will remain locked, therefore unusable.
- If the Initial Administrative selection list is set to Unlocked, when the unit restarts, the endpoint will become usable.
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Disabling the Unit Endpoints when No Gateways are Ready
- Go to System/Endpoints.
- In the Administration table, select Enable next to Disable Unit (All Endpoints) when No Gateways Are In State Ready .

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Shutting Down Endpoint if in Idle-Unusable State

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Disabling All Gateways when Trunk Lines are Down

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Advanced System/Endpoints Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
For more details on the following advanced parameters, refer to the DGW Configuration Guide - Reference Guide published on the Media5 Documentation Portal.
Setting the toggle delay to disable the SIP gateways when trunk lines are down
EpAdm.DisableSipGatewaysWhenTrunkLinesDownToggleDelaySetting the Behavior of the unit While in Shutting Down State
EpAdm.BehaviorWhileInUnitShuttingDownStateTop
Event Log
Basic Event Log Concepts
Event Notifications
An event is something that happens in the system and that needs to be reported. Event notifications are formatted text messages issued by the DGW software to signal something of interest to the unit administrator.
- for which service event notifications will be reported;
- which event notifications will be reported based on the severity level of the event;
- if a specific event notification should be reported or not;
- where the event notifications will be logged.
- Internal destination
- Log to File to a file in the DGW File management system (not available on the 4102S and the Mediatrix C7 series).
- Log Locally to the Local Logs page of the DGW Web interface.
- External destination
- to a SIP server via NOTIFY.
- to a Syslog server via syslog transport.
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Send via Syslog
Event notifications can be logged by a Syslog server.
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Send via SIP
It is possible to send the event notifications to a SIP server.
The event notifications sent to a SIP server are included in the SIP Notify messages. When choosing to send event notifications to a SIP server, it is not allowed to send all notifications, i.e. select the All criteria when configuring event notifications. This has been implemented to avoid sending high volumes of traffic and risking the overload of the SIP server.
The destination can be configured with the sipEp.sipNotificationsGateway parameter and the sipEp.maxNotificationsPerNotify parameter is used to define how many messages can be sent in a single "SIP NOTIFY" request the
For more details, refer to DGW Configuration Guide - Reference Guide published on the Media5 documentation portal at https://documentation.media5corp.com
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Log Locally
The DGW Local Log displays the Event Notifications related to the events occurring on the Mediatrix unit that are generated by the Mediatrix Notifications and Logging Manager (Nlm) service.
Notifications will be displayed in the Local Log Entries table of the DGW Web interface provided the issuing of events for a service is enabled, and if the event meets the selected severity level chosen to be reported.
- Number of displayed event notifications is limited
- Newer notifications replace older ones.
- The routing criteria should be designed to avoid overloading the log.
- The local log has no persistence. Its content is erased when the unit or the Nlm service is restarted.
- The local log does not accept messages from the SNMP service with a ‘Debug’ severity level. This is to prevent an issue when reading the local log with SNMP. The SNMP walk through the table would not catch up with the increasing index because of the DEBUG events generated by the SNMP service itself.
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Log to File
The event notifications can be logged in a file saved in the DGW File Management system.
Event notifications logged to a file will be available in the DGW Web interface, under Management/File tabs. The log to file feature is not available on the Mediatrix 4102S and C7 Series as they do not have more than 1MB or more user storage. The number of files stored in the unit is limited. When the maximal number is reached, the oldest stored file is deleted. The limit is configured in the LogFileMaxNb Mib parameter.
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Send via SNMP
The event notification can be sent via the SNMP transport.
- The traps will be sent according to the Trap Destination(s) and SNMP Protocols configured in the Management/SNMP tab of the unit.
- By default, Notifications are only sent at severity level Warning or higher. This means you will only get traps in case of an error. To also get the recovery events for a particular service, set its severity to « Info » in System/Event Log/Service Notification Configuration.
- For SNMP traps, the notification queue is limited to 100 notifications per second for bandwidth limitation purposes.
- Newer traps replace older ones.
- The routing criteria should be designed to avoid overloading the queue
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Basic Event Notification Tasks
Logging Event Notifications

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Logging Specific Event Notifications

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Disabling Event Notification Reporting for a Service
- Go to System/Event Log.
- In the Service Notification Configuration table, from the drop-down list located next to a service name, select Disable.
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Modifying the Severity Level Triggering the Reporting of a Notification
- Go to System/Event Log.
- In the Service Notification Configuration table, from the drop-down list located next to a service name, select the severity level an event should have to issue a notification.
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Local Log
Basic Local Log Tasks
Clearing the Local Logs
- Go to System/Local Log.
- Click Clear Local Log.
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Updating the Local Logs
- Go to System/Local Log.
- Click Refresh Local Log.
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Packet Capture
Basic Packet Capture Concepts
Packet Captures
Packet captures are data packets intercepted when passing through a specific computer network.
Captured packets can be sent to a specific location where they can be analysed. The content of the capture can therefore be used to diagnose or troubleshoot network problems and determine if network security policies are being followed.
- With the pcapture CLI command available only via the CLI. This method displays the captured packet directly in the CLI or allows streaming the captured packet to a SSH tunnel to a remote Wireshark client.
- With the Nlm.PCaptureStart command. This is a muse command, it can be executed via SNMP, a script, and the CLI. This is the same command used when performing packet captures via the DGW Web page. This method sends the captured file to a file or to a HTTP server via a standard HTTP upload.
- With the DGW Web Interface, under System/Packet Capture.
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Basic Packet Capture Tasks
Starting a Network Capture
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Starting a Network Capture on a Specific VLAN
This method is performed with the PCaptureStart command of the Nml service.
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Starting a Network Capture Remotely On Windows
- You must know the IP address of the unit running the DGW software.
- The Mediatrix unit must be running a DGW v2.0.39.689 firmware or higher.
- You must have a PC running Wireshark.
- The first time the unit is connected via plink/wireshark, do not forget to answer y to the Store key in cache? (y/n) question displayed in the CMD window.
- Make sure there are no other plink sessions already running.
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Starting a Network Capture Remotely On MacOS or Linux
- The Mediatrix unit must be running a DGW v2.0.17.285 firmware or higher.
- You must know the IP address of the unit running the DGW software.
- You must have a PC running Wireshark.
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Examples
Filter Examples
- Filter: port 5060
- Captures all traffic on (either source or destination) port 5060 (SIP)
- Filter: port 5060 and host 192.168.0.99
- Captures all traffic on port 5060 and source or destination IP 192.168.0.99
- Filter: port 5060 and dst host 192.168.0.99
- We can enter “dst” or “src” before “host” (or “port”) to specify the destination or source host (or port
- Filter: not broadcast and not multicast
- Filter out the broadcast and multicast traffic
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Examples of pcapture Commands for Windows
plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -raw -i eth1 not broadcast and not multicast" | wireshark -k -i -
Capture
from the uplink interface of the Mediatrix unit, the packets of the VLan for which the VlanId
is 100
only.plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -raw -i eth1.100" | wireshark -k -i -
Capture
from the uplink interface of the Mediatrix unit, the packets going through the Ethernet port
eth1, but using RTP
only.plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -raw -i eth1 -t rtp " | wireshark -k -i -
Capture
from the uplink interface of the Mediatrix unit, the packets going through the Ethernet port
eth1, but using port 5060 only (either source or
destination).plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -raw -i eth1 port 5060 " | wireshark -k -i -
Capture
from the uplink interface of the Mediatrix unit, the packets going through the Ethernet port
eth1, but using port 5060 as the source
only.plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -raw -i eth1 src port 5060 " | wireshark -k -i -
Capture
from the uplink interface of the Mediatrix unit, the packets going through the Ethernet port
eth1, but using port 5060 as the destination
only.plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -raw -i eth1 dst port 5060 " | wireshark -k -i -
Capture
the packets going through the Ethernet port eth1, for traffic for which the source or the
destination is the unit with the 00:90:F8:07:5A:6D MAC
address.plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -i eth1 ether host 00:90:F8:07:5A:6D " | wireshark -k -i -
Capture
the packets going through the Ethernet port eth1, for traffic for which the source or the
destination is the units whit the 10.5.128.11 or host 10.5.128.4 IP
addresses.plink.exe -ssh -no-antispoof -pw "administrator" admin@192.168.0.10 "pcapture -i eth1 host 10.5.128.11 or host 10.5.128.4 " | wireshark -k -i -
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Examples of pcapture Commands on MacOs and Linux
ssh admin@192.168.0.10 "pcapture -raw -i eth1 not broadcast and not multicast" | wireshark -k -i -
ssh admin@192.168.0.10 "pcapture -raw -i eth1.100" | wireshark -k -i -
Forces capture to interpret all packets as rtp packeta. Typically, this is used with a filter that only keeps rtp packets.
ssh admin@192.168.0.10 "pcapture -raw -i eth1 -T rtp " | wireshark -k -i -
Capture only rtp packets, going through the Ethernet port eth1, but using port 5006 only (either source or destination)
ssh admin@192.168.0.10 "pcapture -raw -i -T rtp eth1 port 5006 " |wireshark -k -i -
ssh admin@192.168.0.10 "pcapture -raw -i eth1 port 5060 " | wireshark -k -i -
ssh admin@192.168.0.10 "pcapture -raw -i eth1 src port 5060 " | wireshark -k -i -
ssh admin@192.168.0.10 "pcapture -raw -i eth1 dst port 5060 " | wireshark -k -i -
ssh admin@192.168.0.10 "pcapture -raw -i eth1 ether host 00:90:F8:07:5A:6D " | wireshark -k -i -
ssh admin@192.168.0.10 "pcapture -raw -i eth1 host 10.5.128.11 or host 10.5.128.4 " | wireshark -k -i -
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Diagnostic
Basic Diagnostic Concept
Diagnostic Traces
Diagnostic Traces are specifically used for troubleshooting purposes. As for Event Notifications, they report errors, warnings, or system information.
Diagnostic Traces do not need to be activated, except at the specific demand of Media5 Technical Assistance Center.
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Basic Diagnostic Trace Tasks
Enabling the Automatic Diagnostic Log Dump
- Go to System/Diagnostic.
- In the Diagnostic Log Configuration table, select Enable.
- Click Apply.
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Manually Starting a Diagnostic Log Dump
- Go to System/Diagnostic.
- In the Diagnostic Log Configuration table, select Dump Now.
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PCM Traces
Basic PCM Traces Concepts
PCM Traces
The PCM traces are two different RTP streams made specifically to record all analog or digital signals that are either sent or received by the telephony ports of the Mediatrix unit.
These RTP streams are sent to a configurable IP address, normally an IP address on your network where it can be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP streams of the VoIP call. On the analog devices, the streams are sent instantly at device start-up, with an average ptime of 5 ms. The resulting streams, depending on the model, are around 15 kB/s.
Only the configured port, port #1 and/or #2 send the PCM traces for a maximum of four simultaneous RTP streams.
All streams are sent instantly at start-up with an average ptime of 15 ms. This means that until the PCM traces are disabled, even an idle unit will continuously send up to 66.6 packets/s X 4 streams = 267 packets/s using approximately 174 bytes each, for a total of 46 Kbytes of upstream bandwidth.
On digital devices, the streams will be sent once a call is in process of being established (ISDN SETUP, SIP INVITE). This means no data will be sent if the gateway is idle with no calls in progress.
- Echo in the network
- DTMF signals
- Caller ID signals
- Fax signals (or false Fax detection)
- Message Waiting Indicator signals
- Any other analog signal
- Any voice quality issue
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Basic PCM Tasks
Enabling the PCM Traces - SIP 5.0


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Enabling the PCM Traces of a Port Using UMN

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Endpoint Examples
Endpoint Name | Description |
---|---|
Bri1-2 | BRI port 1, channel 2 |
Slot2/E1T1-3 | Channel 3 of the E1 port located in slot 2 |
Port09 | Port 09 of a Mediatrix 4108-16-24 unit |
Phone-Fax1 | Port 1 of a Mediatrix 4102S unit |
FXS1 | Port 1 of the FXS card of a Mediatrix C7 unit |
FXO1 | Port 1 of the FXO card of a Mediatrix C7 unit |
All possible endpoint names are listed in the Endpoint table displayed in the DGW Web interface (System/Endpoints). You may also access this table via the CLI by using the EpAdm.Endpoint command or directly via UMN.
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Enabling PCM Traces of a Port Using the Configuration Script

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VM
Virtual Machine Basic Concepts
Important Information on Virtual Machines
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RAM and SSD Sizes
Description | Possible Values |
---|---|
RAM size 1 |
|
SSD size 2 |
|
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RAM Allocation to Virtual Machines
To reduce the wear-and-tear of the Solid State Drive, make sure to allocate the maximum amount of RAM possible to the virtual machine.
Installed RAM on Mediatrix Units | Available RAM for Virtual Machine |
---|---|
2 Gb | 1.5 Gb |
4 Gb | 3.5 Gb |
8 Gb | 7 Gb ( 87.5% of available RAM) |
16 Gb | 10 Gb |
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VM name
The name is set when adding a new virtual machine with the CreateVm command. The user cannot modify the name after.
When the CreateVm command is called without a name, the index is used to generate a unique name such as VM_Index.
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VM Memory
When adding a VM with the CreateVm command, the amount of allocated memory is set; this amount cannot be modified after adding the VM.
When the CreateVm command is called without an amount of allocated memory, a minimal value of 128 MB is set.
- a minimum of 512 MB, or
- 12.5% of the total volatile memory capacity if more than 512 MB is available
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USB Usage
The VM config allows the user to associate none or all USB Ports to a virtual machine.
A USB port can be associated with one VM. The first VM that is configured with USB can use all available USB ports.
If an another VM tries to use a USB port already in use, the Vm service ignores this config and starts the VM as if it was configured with NONE, and sets its configuration status (ConfigStatus) to USBNotAvailable.
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Virtual Switch
Enabling the Virtual Switch with the Eth.Links.VirtualSwitch parameter grants network access to the VM. Once enabled, the virtual switch creates a bridge between the VM and the associated Ethernet link.
When the Virtual Switch is enabled, the Vm.Vm.NetworkAdapter parameter configures its virtualised network adapter.
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Behaviour on Factory Reset
The unit can be preinstalled with a factory-installed VM stored in the vm/images/factory folder. This folder can only be created in factory and must have the factory-installed VM files.
- When one or more factory-installed VM is present, VM images and configurations are returned to their original factory state.
- When no factory-installed VM is present, the VM images and configurations stay unchanged, i.e. the files present in vm/images/ are not erased.
- The files in the vm/images/ folder are erased, which removes the VM snapshots and all VMs created, modified, or installed by users.
- Note: this is done even if the vm/images/factory folder exists and is empty.
- For each factory-installed VM (visible in the vm/images/factory folder), the configuration file (.cfg) is copied in the vm/images folder and a snapshot of the VM image is also created in the vm/images folder. The snapshot file is given the .snapshot extension and is always in a QCOW2 format. When this VM is used, the snapshot file changes over time but the base image (located in vm/images/factory) is never modified, allowing the next factory reset to restore the factory VMs to their original state again.
- The admin can use, configure, convert, and delete a VM with a snapshot image like any other VM, but after a factory reset, the snapshot image is deleted and a new one is created.
- When a snapshot file is converted into a VM image file, the resulting file is a new image file combining the base VM plus the history contained in the snapshot file.
- Users cannot add or delete files on the vm/images/factory folder. This can only be done in factory.
- The VM images under vm/images/factory can have either the RAW or QCOW2 format.
- The RestoreAfterFactory parameter in the VM configuration file is ignored. The factory-installed VMs are always restored.
In all cases, the content of /vm/drives is erased.
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What are the Meltdown and Spectre Security Vulnerabilities
These vulnerabilities allow a non-privileged process to read sensitive data in memory, thus accessing privileged information from the kernel or other processes.
A Virtual Machine (VM) running inside the Sentinel 400 may be vulnerable to Meltdown (CVE-2017-5754) and to the two variants of Spectre (CVE-2017-5753 and CVE-2017-5715).
For more information on these vulnerabilities:
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How to Protect my VM against Spectre
There are different mitigation techniques against Spectre:
- Mitigation #1: A microcode update from the CPU vendor for better control over the branch speculation. Also need an updated kernel to enable these new features (IBRS and IBPB).
- Mitigation #2: Different techniques (like "retpoline" and "LFENCE") that require recompiling the kernel, packages and applications.
As the time this document was written, Mitigation #1 could not be applied, as Intel had not yet released a microcode update for the CPU of the Sentinel 400.
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How to Protect my VM against Meltdown
Linux kernels have a new feature called KPTI (previously known as KAISER) that protects against Meltdown.
If your Virtual Machine is vulnerable, Media5 recommends that you upgrade your kernel to a version that supports KPTI, and enable it.
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SSD Lifespan
Mediatrix units running a Virtual Machine are equipped with various sizes of Solid State Drive (SSD) for storage.
A Solid State Drive (SSD) is a form of flash-based storage. Media5 uses high quality, enterprise grade SSDs in its products. However due to its technical nature, flash memory can handle so many read/write cycles. Beyond that, the performance may degrade or in extreme cases, the drive may fail. For customers using a Mediatrix unit running a Virtual Machine, special attention should be paid to the number of writes caused by the Operating System running in the virtual machine, as well as by the application. If the Virtual Machine is not optimised, it will lead to excessive read/write access to the SSD, and hence significantly reduce the lifespan of the SSD. For more details, please refer to Technical Bulletins - Virtual machine installation published on the Media5 Documentation Portal.
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SSD Lifespan Extrapolation
SSD lifespan can be extrapolated and Virtual Machine optimisation can be validated.
For example, let the virtual machine run for a month (or a week), read and compare the WearPercentage parameter at different times during this interval. Use the collected information to extrapolate the lifespan.
For example: the Wear Percentage increased from 19% to 20% in 2 months. Take the remaining 80% divided by a monthly increase of 0.5% gives 160 months left. Therefore at least 13 years left at the current increase rate.
- disable some of the logging
- log to an external device
- assign more RAM to the Virtual Machine
- consider ordering a bigger SSD for future installations
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Is all the System Vulnerable?
The DGW firmware in the Mediatrix system, by itself, is not vulnerable since it does not allow running rogue code:
But it is theoretically possible, for a Virtual Machine compromised by the Spectre vulnerability, to read memory outside the Virtual Machine and access sensitive data of the Mediatrix system. The best protection against this is to secure your VM, to make sure there is no known means an attacker can use to break into your VM.
Media5 also recommends to always keep your Sentinel 400 up-to-date with to the latest DGW firmware version.
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Basic Virtual Machine Actions
Stopping the Virtual Machine
- Go to System/VM.
-
In the Virtual Machine Configuration
table, click
located on the same row as the VM you wish to stop.
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Stopping the Virtual Machine - Graceful Stop
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Rebooting a VM
- Go to System/VM.
-
Click
.
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Rebooting a VM - Graceful Reboot
Although the virtual machine can be rebooted via the Web page, rebooting the virtual machine using a VNC Client is the preferred way to reboot the virtual machine.
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Setting the Virtual Machine to Automatic Start
If the virtual machine you wish to start requires resources equivalent to the available resources on the unit, then it will not be possible to start another virtual machine. It is only possible to start a virtual if there are enough resources on the unit.
- Go to System/VM.
- In the Virtual Machine Configuration table, from the Startup dropdown list, select Auto.
- Click Apply.

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Setting the Virtual Machine to Manual Start
- Go to System/VM.
- In the Virtual Machine Configuration table, from the Startup dropdown list, select Manual.
- Click Apply.

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Deleting a VM
- Go to System/VM.
-
In the Virtual Machine Configuration
table, click
located on the same row as the virtual machine you wish to delete.
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Monitoring the SSD Wear Percentage Using the CLI
- Open the Command Line Interface ( CLI).
- At the prompt, enter the following command: Dcm.PersistentWearPercentage
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Monitoring the SSD Wear Percentage Using a MIB Browser
The Mediatrix UMN Mib browser can be used for this procedure.
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Virtual Machine Installation
Adding a Virtual Machine
You must have a virtual machine licence and the VM service must be started.

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Configuring the VM Network Adapter (VirtIO)
The virtual machine Network Adapter will be set to VirtIO.

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Installing the OS on the Virtual Machine Using an ISO image

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Installing the Virtual Machine OS using a USB External Device
The virtual machine will be started only if it is started manually

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Disabling Swap on Linux
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Virtual Machine Modification
Modifying the Virtual Machine Configuration
- Go to System/VM.
- In the Virtual Machine Configuration table, modify the fields as required.
- Click Apply.
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Network
Host
Basic Network Host Concepts
DNS Servers
The DNS server list is the ordered list of DNS servers that the device uses to resolve network names.
Up to four servers can be used. The DNS servers can be specified statically or obtained automatically (for example through DHCPor PPP). DNS query results are cached on the system to optimise name resolution time. For more details, refer to DGW Configuration Guide - DNS Behavior with Mediatrix Gateways published on the Media5 Documentation Portal.
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Simple Network Time Protocol (SNTP)
The Simple Network Time Protocol (SNTP) is used to update and synchronise the clock of the Mediatrix unit (day, month, time) when it is restarted.
Mediatrix units do not all include a real time clock allowing them to maintain accurate time when they are shutdown. Your system needs to have access to accurate time, for example if you are using HTTPS or for the caller ID feature. The Mediatrix unit implements a SNTP client, which can synchronise the local clock with remote NTP/SNTP servers. The configuration can be automatic (through DHCP for example), with fallback, or static, with up to four servers.
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Time Format
The time format (also known as 'TZ' format) is based on the format described by the IEEE 1003.1 standard (i.e. POSIX specification).
- The first part, mandatory, is the system timezone expressed in the IEEE 1003.1 POSIX format (also known as 'TZ' format).
- The second part, available since 46.0 is optional. It is the timezone used for the time displayed in some of the SBC Web pages (Live Calls, Events, and Registration). This part is only useful for units with the Sbc service. This string must be expressed in the IANA format. If this part is not present, the UTC time zone is used on the SBC Web pages.
STDOFFSET[DST[OFFSET],[START[/TIME],END[/TIME]]]
where: STD / DST
: Three or more characters for the standard (STD) or alternative daylight saving time (DST) time zone. Only STD is mandatory. If DST is not supplied, the daylight saving time does not apply. Lower and upper case letters are allowed. All characters are allowed except:- digits
- leading colon (:)
- comma (,)
- minus (-)
- plus (+), and
- ASCII NUL.
OFFSET
: Difference between the GMT time and the local time. The offset has the format h[h][:m[m][:s[s]]]. If no offset is supplied for DST, the alternative time is assumed to be one hour ahead of standard time. One or more digits can be used; the value is always interpreted as a decimal number.- The hour value must be between 0 and 24. IMPORTANT: If preceded by a minus sign (-), the time zone is east of the prime meridian, otherwise it is west, which can be indicated by the preceding plus sign (+). For example, New York time is GMT 5.
- The minute and second values, if present, must be between 0 and 59.
- The hour value must be between 0 and 24.
START / END
Indicates when to change to and return from the daylight saving time. The START argument is the date when the change from the standard to the daylight save time occurs; END is the date for changing back. If START and END are not specified, the default is the US Daylight saving time start and end dates. The format for start and end must be one of the following:- n where n is the number of days since the start of the year from 0 to 365. It must contain the leap year day if the current year is a leap year. With this format, you are responsible to determine all the leap year details.
- Jn where n is the Julian day number of the year from 1 to 365. Leap days are not counted. That is, in all years – including leap years – February 28 is day 59 and March 1 is day 60. It is impossible to refer to the occasional February 29 explicitly. The TIME parameter has the same format as OFFSET but there can be no leading minus (-) or plus (+) sign. If TIME is not specified, the default is 02:00:00.
- Mx[x].y.z where x is the month, y is a week count (in which the z day exists) and z is the day of the week starting at 0 (Sunday). For instance: M10.4.0 is the fourth Sunday of October. It does not matter if the Sunday is in the 4th or 5th week. M10.5.0 is the last Sunday of October (5 indicates the last z day). It does not matter if the Sunday is in the 4th or 5th week. M10.1.6 is the first week with a Saturday (thus the first Saturday). It does not matter if the Saturday is in the first or second week. The TIME parameter has the same format as OFFSET but there can be no leading minus (-) or plus (+) sign. If TIME is not specified, the default is 02:00:00.
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Time Format Examples

Time Zone | String | IANA format (Optional, 46.0+) |
---|---|---|
Atlantic Time (Canada) | AST4ADT,M3.2.0,M11.1.0 | America/Halifax |
Australia Eastern Standard Time | AEST-10AEDT,M10.1.0,M4.1.0/3 | Australia/Sydney |
Central European Time | CET-1CEST,M3.5.0,M10.5.0/3 | Europe/Brussels |
Central Time (Canada & US) | CST6CDT,M3.2.0,M11.1.0 | America/Chicago |
China Standard Time | CST-8 | Asia/Shanghai |
Eastern Time Canada & US) | EST5EDT,M3.2.0,M11.1.0 | America/Toronto |
Greenwich Mean Time | GMT0BST,M3.5.0/1,M10.5.0 | Europe/London |
Mountain Time (Canada & US) | MST7MDT,M3.2.0,M11.1.0 | America/Denver |
Pacific Time (Canada & US) | PST8PDT,M3.2.0,M11.1.0 | America/Los_Angeles |
Japan Standard Time | JST-9 | Asia/Tokyo |
UTC (Coordinated Universal Time) | UTC0 | Etc/UTC |
Western Europe Time | WET0WEST,M3.5.0/1,M10.5.0 | Europe/Lisbon |
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Basic Network Host Tasks
Choosing the Network Providing the IPv4 Automatic configuration
- Go to Network/Host.
- In the Automatic Configuration Interface table, from the Automatic IPv4 config source network selection list, choose a network.
- Click Apply.
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Choosing the Network Providing the IPv6 Automatic configuration
- Go to Network/Host.
- In the Automatic Configuration Interface table, from the Automatic IPv6 config source network selection list, choose a network.
- Click Apply.
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Configuring the Host Name and Domain Name of the Mediatrix Unit
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Configuring the Default Network Gateway to a Static IP Address
- Go to Network/Host.
- In the Default Gateway Configuration table, from the IPv4/Configuration Source selection list, select Static.
- In the IPv4/Default Gateway field, enter the IP address used as the Static Default Router for the Uplink Network Interface.
- In the Default Gateway Configuration table, from the IPv6/Configuration Source selection list, select Static.
- In the IPv6/Default Gateway field, enter the IP address used as the Static Default Router for the Uplink Network Interface.
- Click Apply.

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Configuring the Default Network Gateway to an Automatic IP Address

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Configuring DNS Servers - Automatically
- Go to Network/Host.
- In the DNS Configuration table, from the Configuration Source selection list, choose Automatic IPv4 or Automatic IPv6 .
- Click Apply.

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Configuring DNS Servers - Manually
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Configuring the SNTP Server to a Static IP Address

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Configuring the SNTP Server to an Automatic IP Address

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Configuring the SNTP Server to an Automatic IP Address with Fallback

.
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Configuring the Mediatrix Unit to Use an SNTP Server

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Selecting the Unit's Time Zone
Any DGW parameter referring to a time value will use the local time described by this time zone reference. The Hoc.SystemTime will return the unit local time in accordance with the configured time zone.
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Advanced Network Host Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Configuring Dns Cache Randomisation:
- Hoc.DnsCacheRandomization
Configuring Pre-resolved Static FQDNs
Up to 10 pre-resolved FQDNs can be configured. The StaticHosts table allows configuring FQDNs with static IP addresses. When a device attempts to reach a FQDN configured in this table, the static IP addresses will be used instead of resolving the FQDN.- Hoc.InsertStaticHost: To insert a new static host
- Hoc.StaticHosts.Delete: To delete a static host:
Updating the system name or system location
The name and location of the Mediatrix unit can be specified. This information is for display purposes only and does not affect the behavior of the unit.- Hoc.SystemName: To set the system name
- Hoc.SystemLocation: Set the system location
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Interfaces
Basic Interface Concepts
IP Address Reservation
Before connecting the Mediatrix unit to the network, Media5 strongly recommends to reserve an IP address in your network server – if using one – for the unit you are about to connect.
This way, the IP address associated with a particular unit will be known. Network servers generally allocate a range of IP addresses for use on a network and reserve IP addresses for specific devices using a unique identifier for each device. The Mediatrix unit unique identifier is the media access control (MAC) address. Refer to Locating the MAC Address of Your Mediatrix Unit.
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Locating the MAC Address of Your Mediatrix Unit
- printed on a label located under the Mediatrix unit
- displayed in the Current Status table of the Web Interface (System/Information)
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Important Information About Network Interfaces
Naming
- The name of the network interface is case sensitive.
- Using the special values All, Loop, LoopV6, and Rescue are not allowed to name a network interface
- A valid network interface name:
- must start with a letter
- cannot contain characters other than letters, numbers, and underscores
Configuration
- It is not possible to have different IP addresses from the same subnet on one interface.
- It is possible to create up to 48 network interfaces.
- LLDP cannot be activated on multiple network interfaces simultaneously.
- If no network is configured in IPv6, the unit does not have any IPv6 address, not even the Link-Local address. When a network is configured in IPv6, the Link-Local (FE80 ::...) address is automatically created and displayed in the Network Status information.
- In case of address conflicts between two or more network interfaces, the network interface with the highest priority will remain enabled and the other interfaces will be disabled. If the priority is the same, only the first enabled network interface will be able to use the IP address. When a conflict ends, all network interfaces concerned automatically return to an operational state.
- Media5 recommends to reserve an IP address with an infinite lease for each Mediatrix unit on the network.
- The Rescue Network Interface cannot be deleted.
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Default Network Interfaces
There are four Network Interfaces created by default on the Mediatrix unit: Uplink, Lan 1, UplinkV6, and Rescue.
- The Uplink network interface defines the uplink information required by the Mediatrix unit to properly connect to the WAN. (By default eth1 for all platforms, except for the 4102S which is WAN . By default, this interface uses the IpDhcp (IPv4 DHCP) connection type. If you are using only one Network Interface, you must use Uplink.
- The Lan1 network interface defines the information required by the Mediatrix unit to properly connect to the LAN.(By default eth2-5 for all platforms, except for the 4102 which is LAN) By default, the Lan1 Network Interface uses the IpStatic (IPv4 Static) connection type. The Lan1 network interface can only be added on units with 2 network ports.
- The Rescue network interface,
is used to display the Rescue
Management Interface when a partial reset of the unit is performed. By default, the
Rescue network interface
- is disabled and automatically enabled when a partial reset is performed.
- uses the IpStatic (IPv4 Static) or the Ip6Static (IPv6 Static) addresses.
- The UplinkV6 network interface defines the IPv6 uplink information required by the Mediatrix unit to properly connect to the WAN. By default, this interface uses the IP6autoConf (IPv6 Auto-Conf) configuration mode.
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Link Default Values for the Uplink Network Interface
Unit Type | Link Default Value |
---|---|
Sentinel 400 | eth1 |
Sentinel 100 | eth1 |
Mediatrix G7 | eth1 |
Mediatrix S7 | eth1 |
Mediatrix C7 series | eth1 |
Mediatrix 4102S | Wan |
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Link Default Values for the Lan1 Network Interface
Unit Type | Link Default Value |
---|---|
Sentinel 400 | eth2-5 |
Sentinel 100 | eth2-5 |
Mediatrix G7 series | eth2-5 |
Mediatrix S7 series | eth2-5 |
Mediatrix C7 series | eth2 |
Mediatrix 4102 | lan |
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Link Layer Discovery Protocol (LLDP)
The Link Layer Discovery Protocol (LLDP) service is used by network devices for advertising their identity, capabilities, and neighbors on a IEEE 802 local area network, usually wired Ethernet.
LLDP cannot be activated on more than one network interface at a time.
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Link Connectivity Detection
Each Ethernet port of the Mediatrix unit is associated with an Ethernet link.
An Ethernet link has connectivity if at least one of its port status is not disconnected. The link connectivity is periodically polled (every 500 milliseconds). It takes two consecutive detections of the same link state before reporting a link connectivity transition. This avoids reporting many link connectivity transitions if the Ethernet cable is plugged and unplugged quickly.
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PPP Negotiation
When the Mediatrix unit restarts, it establishes the connection to the access concentrator in conformance with RFC 2516 section 5.1.
- Discovery phase: The Mediatrix unit broadcasts the value of the Service
Name field. The access concentrator with a matching service name answers the Mediatrix
unit.
- If no access concentrator answers, this creates a PPPoE failure error.
- If more than one access concentrators respond to the discovery, the Mediatrix unit tries to establish the PPP connection with the first one that supports the requested service name.
- Authentication phase: If the access concentrator requests authentication, the Mediatrix unit sends the ID/secret pair configured in the User Name and Password fields. If the access concentrator rejects the authentication, this creates an “authentication failure” error.
- Network-layer protocol phase: The Mediatrix unit negotiates an IP address. The requested IP address is the one from the last successful PPPoE connection. If the Mediatrix unit never connected by using PPPoE (or after a factory reset), it does not request any specific IP address.
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IPv6 Autoconfiguration Interfaces
When the Type drop-down menu is set to IPv6 Auto-Conf, the network interface is an IPv6 over Ethernet connection with IP parameters obtained by stateless auto-configuration or stateful (DHCPv6) configuration.
- The router explicitly required stateful autoconfiguration by setting the “managed” or “other” flag of the router advertisement.
- No router advertisement was received after 3 router solicitations. RFC 4861 defines the number of router solicitations to send and the 4 seconds interval between the sent router solicitations.
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Stateless Autoconfiguration
All IPv6 addresses present in the router advertisements are applied to the network interface
- GU (Global Unique)
- UL (Unique Local)
- LL (Link-Local)
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Spanning Tree Protocol vs Stateless Autoconfiguration
Many network switches use the Spanning Tree Protocol (STP) to manage Ethernet ports activity.
STP uses a detection timeout before a router advertisement is sent to the Mediatrix unit. The default value for this timeout is usually 30 seconds. However, when the unit wants to get an IPv6 address in Stateless autoconfiguration, this timeout is too long and the unit falls into Stateful Autoconfiguration mode before it receives the router advertisement. This results in the unit receiving a DHCPv6 address. To solve the issue, check if the default STP detection timeout value in your router can be modified. If so, set it to a value of 8 s or less. If you cannot modify the timeout value, Media5 recommends to disable the Spanning Tree Protocol on the network to which the unit is connected.
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Statefull Autoconfiguration
Stateful autoconfiguration is managed by DHCPv6. The DHCPv6 lease is negotiated according to the limitations listed in section 1.5 of RFC 3315.
- IPv6 addresses (when the router advertisement “managed” flag is set)
- Other configuration (when the router advertisement “other” flag is set)
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Speed and Duplex Detection Issues
There are two protocols for detecting the Ethernet link speed: parallel detection and auto-negotiation (IEEE 802.3u).
The auto-negotiation protocol allows the detection of the connection speed and duplex mode. It exchanges capabilities and establishes the most efficient connection. When both endpoints support the auto-negotiation, there are no problems. However, when only one endpoint supports auto-negotiation, the parallel detection protocol is used. This protocol can only detect the connection speed; the duplex mode cannot be detected. In this case, the connection may not be established. The Mediatrix unit has the possibility to force the desired Ethernet link speed and duplex mode by disabling the auto-negotiation and selecting the proper setting. When forcing a link speed at one end, be sure that the other end (a hub, switch, etc.) has the same configuration. To avoid any problem, the link speed and duplex mode of the other endpoint must be exactly the same.
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Basic Interface Tasks
Creating a Network Interface
- Media Interface Configuration table under the SBC/ Configuration tabs (provided you have the Sbc service)
- Signaling Interface Configuration table under SBC/ Configuration tabs (provided you have the Sbc service)
- DHCP Server Configuration table under the Network/ DHCP Server tabs
- Signaling Network table under the SIP/Gateways tabs
- Network Interface table under the SIP Proxy/Configuration tabs.
- Network Interface table under the Management/Misc tabs.
- Forward To Network table under the Network/IP Routing tabs.
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Configuring a Network Interface
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Associating an Ethernet Link to a Network Interface

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Configuring the PPPoE Connection Type
- Go to Network/Interfaces.
- In the PPPoE Configuration table, complete the fields as required.
- Click Apply.
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Configuring the Link Layer Discovery Protocol (LLDP)
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Configuring a Link as a Virtual Switch
- Go to Network/Interfaces.
- In the Ethernet Link Configuration table, from the Virtual Switch selection list, select Enable located on the same row as the link you wish to enable for the virtual switch.
- Click Apply.

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Configuring the Ethernet Link linked to a Network Interface
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Selecting the IEEE 802.1x Version
- Go to Network/Interfaces.
- In the EAP 802.1x Configuration table, select the IEEE 802.1x version.
- Click Apply if you do not need to set other parameters.

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Advanced Network Interface Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Network Interfaces Priority
Refer to eth.networkInterfacesPriorityDHCP Client Identifier Presentation
Refer to bni.dhcpClientIdentifierPresentationEthernet Connection Speed
Refer to eth.portsSpeedTop
VLAN
Basic VLAN Tasks
Creating a VLAN
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Associating a VLAN to an Ethernet Link
- Go to Network/VLAN.
- From the Link selection list, select the Ethernet link the VLAN interface will use.
- In the VlanId field, set the VLAN ID used by the VLAN interface.
- In the VLAN configuration table, click +.
- Set the default user priority value.
- To create another VLAN, repeat steps 1 to 5.
- Click Apply.

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Configuring the Default User Priority on an Existing VLAN
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Configuring the Default User Priority on a New VLAN
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QoS
Basic QoS Concepts
Quality of Service (QoS)
QoS (Quality of Service) features enable network managers to decide on packet priority queuing.
- Differentiated Services (DS) Field (for IPv4)
- Traffic Class Field (for IPv6)
- 802.1Q taggings
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Differentiated Services (DS) Field (for IPv4 only)
Differentiated Services (DiffServ, or DS) is a protocol for specifying and controlling network traffic by class so that certain types of traffic (for example voice traffic which requires a relatively uninterrupted flow of data) might get precedence over other kinds of traffic.
DiffServ replaces the first bits in the ToS byte with a differentiated services code point (DSCP). It uses the existing IPv4 Type of Service byte. In DGW the entire ToS byte is currently configurable, thus the ToS decimal value is used. Please refer to:
- https://tools.ietf.org/html/rfc2474 for the definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers.
- https://en.wikipedia.org/wiki/Differentiated_services for the differentiated services.
TOS (Dec) | TOS (Hex) | TOS (Bin) | TOS Precedence (Bin) | TOS Precedence (Dec) | TOS Precedence Name | TOS Delay flag | TOS Throughput flag | TOS Reliability flag | DSCP (Bin) | DSCP (Hex) | DSCP (Dec) | DSCP/PHB Class |
---|---|---|---|---|---|---|---|---|---|---|---|---|
0 | 0x00 | 00000000 | 000 | 0 | Routine | 0 | 0 | 0 | 000000 | 0x00 | 0 | none |
4 | 0x04 | 00000100 | 000 | 0 | Routine | 0 | 0 | 1 | 000001 | 0x01 | 1 | none |
8 | 0x08 | 00001000 | 000 | 0 | Routine | 0 | 1 | 0 | 000010 | 0x02 | 2 | none |
12 | 0x0C | 00001100 | 000 | 0 | Routine | 0 | 1 | 1 | 000011 | 0x03 | 3 | none |
16 | 0x10 | 00010000 | 000 | 0 | Routine | 1 | 0 | 0 | 000100 | 0x04 | 4 | none |
32 | 0x20 | 00100000 | 001 | 1 | Priority | 0 | 0 | 0 | 001000 | 0x08 | 8 | cs1 |
40 | 0x28 | 00101000 | 001 | 1 | Priority | 0 | 1 | 0 | 001010 | 0x0A | 10 | af11 |
48 | 0x30 | 00110000 | 001 | 1 | Priority | 1 | 0 | 0 | 001100 | 0x0C | 12 | af12 |
56 | 0x38 | 00111000 | 001 | 1 | Priority | 1 | 1 | 0 | 001110 | 0x0E | 14 | af13 |
64 | 0x40 | 01000000 | 010 | 2 | Immediate | 0 | 0 | 0 | 010000 | 0x10 | 16 | cs2 |
72 | 0x48 | 01001000 | 010 | 2 | Immediate | 0 | 1 | 0 | 010010 | 0x12 | 18 | af21 |
80 | 0x50 | 01010000 | 010 | 2 | Immediate | 1 | 0 | 0 | 010100 | 0x14 | 20 | af22 |
88 | 0x58 | 01011000 | 010 | 2 | Immediate | 1 | 1 | 0 | 010110 | 0x16 | 22 | af23 |
96 | 0x60 | 01100000 | 011 | 3 | Flash | 0 | 0 | 0 | 011000 | 0x18 | 24 | cs3 |
104 | 0x68 | 01101000 | 011 | 3 | Flash | 0 | 1 | 0 | 011010 | 0x1A | 26 | af31 |
112 | 0x70 | 01110000 | 011 | 3 | Flash | 1 | 0 | 0 | 011100 | 0x1C | 28 | af32 |
120 | 0x78 | 01111000 | 011 | 3 | Flash | 1 | 1 | 0 | 011110 | 0x1E | 30 | af33 |
128 | 0x80 | 10000000 | 100 | 4 | FlashOverride | 0 | 0 | 0 | 100000 | 0x20 | 32 | cs4 |
136 | 0x88 | 10001000 | 100 | 4 | FlashOverride | 0 | 1 | 0 | 100010 | 0x22 | 34 | af41 |
144 | 0x90 | 10010000 | 100 | 4 | FlashOverride | 1 | 0 | 0 | 100100 | 0x24 | 36 | af42 |
152 | 0x98 | 10011000 | 100 | 4 | FlashOverride | 1 | 1 | 0 | 100110 | 0x26 | 38 | af43 |
160 | 0xA0 | 10100000 | 101 | 5 | Critical | 0 | 0 | 0 | 101000 | 0x28 | 40 | cs5 |
176 | 0xB0 | 10110000 | 101 | 5 | Critical | 1 | 0 | 0 | 101100 | 0x2C | 44 | voice-admit |
184 | 0xB8 | 10111000 | 101 | 5 | Critical | 1 | 1 | 0 | 101110 | 0x2E | 46 | ef |
192 | 0xC0 | 11000000 | 110 | 6 | InterNetwork Control | 0 | 0 | 0 | 110000 | 0x30 | 48 | cs6 |
224 | 0xE0 | 11100000 | 111 | 7 | Network Control | 0 | 0 | 0 | 111000 | 0x38 | 56 | cs7 |
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Network Traffic Control
It is possible to apply bandwidth limitations to the network interfaces.
The limitations are applied on the raw data of the physical link and not only on the payload of the packets. All headers, checksums and control bits (TCP, IP, CRC, etc.) are considered in the actual bandwidth. A bandwidth limitation is applied on a physical link and not on a virtual network interface. All high-level network interfaces (including VLANs) using the same physical link are affected by a configured limitation. This limitation is applied to outgoing traffic only (egress). Bandwidth limitation is an average of the amount of data sent per second. Thus, it is normal that the unit sends a small burst of data after a period of silence.
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Basic QoS Tasks
Creating the Default Unit QoS
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Configuring the Default User Priority on Physical Links (802.1Q Tagging)
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Overriding the DiffServ and QoS Service Class Default Values
- Go to Network/QoS.
- In the Service Class Configuration table, for each service class, set for IPv4 packets the DiffServ value or the Traffic Class value for IPv6 packets, .
- Set a specific User Priority for each class under the User Priority column.
- Click Apply.
- Click restart required services, located at the top of the page.
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Configuring Network Traffic Control
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Example
Default Unit QoS and Service Class Configuration for IPv4
Any IPv4 packet sent from the unit has the value applied in the Default DiffServ (IPv4) field of the Differentiated Services Field Configuration table under Network/QoS tab. This default value is overridden on what concerns the specific service classes defined under the same area and in the Service Class Configuration table.



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Local Firewall
Basic Local Firewall Concepts
Local Firewall
The local firewall allows you to create and configure rules to filter incoming packets that have the Mediatrix unit as destination.
The Local Firewall is therefore a security feature that allows you to protect your Mediatrix unit from receiving packets from unwanted or unauthorized peers. As a best practice, the way the Local Firewall should work is to, by default, drop all incoming packets (i.e. not forward the packet to its destination) and let incoming packets go through only if they match a rule requirements. However, incoming packets for an IP communication established by the Mediatrix unit are always accepted (Example : If the Mediatrix unit sends a DNS request, the answer will be accepted).
When configuring the Local Firewall, enabling the default policy to drop all incoming packets should be the last step you perform otherwise, you may lose contact with the Mediatrix unit, even if you are performing the initial configuration of your system. Therefore, start by creating the rules that allow the Mediatrix unit to accept some packets. This way communication will not be lost and you will not need to perform a partial or factory reset to reconnect with the Mediatrix unit.
You can use a maximum of 20 rules, but the more rules are enabled, the more overall performance is affected.
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Firewall Rule Order - Important
The order in which the incoming packets are tested against the rules is important if you want to make sure that they actually have a filtering effect on incoming packets.
Rules can be configured to accept or to decline packets. But, always put the most restrictive rules first as they are executed sequentially starting with the first one listed at the top of the table i.e. make sure that the order in which the rules are executed does not cause a rule to be systematically excluded.
- If the first rule excludes all packets coming from a specific net mask, providing a second rule for an IP address with that same net mask will have no effect.
- If the first rule indicates actions to be taken for a specific IP address with a given net mask, and the second rule indicates to exclude all IP addresses with that net mask, both rules will be applied and have a result on the incoming packets.
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Basic Local Firewall Tasks
Configuring the Local Firewall
If a rule with the Black listing enable box checked matches a packet and no Rate Limit Value was set, then the source address of the packet will be black listed and all packets coming from this address will be blocked for the duration of the Blacklist Timeout.
If a rule with the Black listing enable box checked matches a packet and the Rate Limit Value has been reached, then the source address of the packet will be black listed and all packets coming from this address will be blocked for the duration set for the Blacklist Rate Limit Timeout.
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Disabling the Local Firewall
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Setting the No Match Local Firewall Default Policy
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Configuring Black Listing Duration
If a rule with the Black listing enable box checked matches a packet and no Rate Limit Value was set, then the source address of the packet will be black listed and all packets coming from this address will be blocked for the duration of the Blacklist Timeout.
If a rule with the Black listing enable box checked matches a packet and the Rate Limit Value has been reached, then the source address of the packet will be black listed and all packets coming from this address will be blocked for the duration set for the Blacklist Rate Limit Timeout.
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Local Firewall Examples
Generic Whitelist
All incoming calls are dropped unless they match one of the firewall rules which are acting on the incoming packets going towards the Mediatrix gateway.

Result:
Rule # | |
---|---|
1 | Any incoming packet from the LAN subnet having the unit's LAN host IP address as a destination is allowed. |
2 | Any incoming packet from the Uplink subnet is allowed (assuming this is a private network). |
3 | Any HTTP incoming packet from the selected IP address having the unit's Uplink IP address as a destination through TCP port 80 is allowed. |
4 | Any HTTPS incoming packets from the selected IP address having the unit's Uplink IP address as a destination through TCP port 443 is allowed, but rate limited to 10 new connection attempts per 60 sec. |
5 | Any SSH incoming packets from the selected subnet having the unit's Uplink IP address as a destination through TCP port 22 is allowed. |
6 | Any SIP incoming packets from the selected subnet having the unit's Uplink IP address as a destination through UDP port 5060 is allowed. |
7 | Any RTP and T.38 incoming packet from the selected subnet having the unit's Uplink IP address as a destination through UDP port range 5004-6020 is allowed. |
Default | All other incoming packets are rejected. |
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Whitelist for Internet Hacker Protection
Simple Local Firewall rules to protect the unit from Internet hackers. All incoming calls are dropped unless they match one of the local firewall rules which are acting on the incoming traffic towards the Mediatrix gateway.

Result:
Rule # | Description |
---|---|
1 | Any incoming packet from the LAN subnet is allowed. |
2 | Any incoming packet from the Uplink subnet is allowed (assuming this is a private network). |
3 | Any incoming packet from selected IP address is allowed (e.g. this is the management server). |
4 | Any incoming packet from the selected subnet is allowed (e.g. this is the Core SIP server, SBC and its media gateways). |
Default | Any incoming packet not meeting the criteria of these rules is dropped. |
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Generic Blacklist
The default policy is set to "Accept" but the firewall rules are Blacklists acting on incoming traffic towards the Mediatrix gateway:

Result:
Rule # | Description |
---|---|
1 | Any incoming packet going to the Uplink interface through TCP port 22 (SSH) is dropped. |
2 | Any incoming packet coming from the specified subnet is dropped. |
3 | Any HTTP incoming packet coming from the specified IP address to the Uplink interface through TCP port 80 is dropped. |
4 | Any incoming packet from the specified subnet to the Lan1 interface is rejected, and an ICMP message is returned. |
5 | Any SIP incoming packets from the specified IP address to the Lan1 interface through UDP port 5060 is rate limited to 10 new connection attempts per 60 sec. |
Default | All other incoming packets are accepted. |
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IP Routing
Basic IP Routing Concepts
Important Information About IP Routing
- The Mediatrix unit’s IP Routing settings do not support IPv6.
- A packet matching a route uses the custom routing table first and then the main routing table if no route in the custom routing table was able to send the packet to the desired destination IP address.
- When creating an advanced IP routing Rule, leaving the Source Address or/and Source Link fields empty, indicate that any source of address or/and link will match the rule
- IP Routing works together with the following services:
- Network Firewall
- NAT
- DHCP server
- Network Traffic Control
- When the IP Routing service is started and the IPv4 Forwarding is enabled, IP routing is activated even if there is no configured rule (the Mediatrix unit will forward received packets). If the IP Routing service is stopped, IP forwarding is disabled, this tab is greyed out and the parameters are not displayed.
- Enabling the IP routing service and adding rules has an impact on the Mediatrix unit’s
overall performance as IP routing requires additional processing. The more rules are
enabled, the more overall performance is affected. Note: Media5 recommends to use a 30 ms packetization time when IP routing is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels available on the unit.
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IP Routing Rule Order - Important
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Static IPv4 vs Advanced IP Routing
Network IP routing defines the routes for outgoing network traffic, where each route is associated with a network interface. The selection of which route a network packet should follow is generally based on the destination IP address criteria.
The Static IPv4 Routes are used to specify additional routes from the default ones automatically created by the configuration of the various network interfaces (see the BNI service).
The Advanced IP Routes are used only when IPv4 Forwarding is enabled, and allow to select a route, not just from the destination IP address, but also from the source IP address and interface criteria.
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Basic IP Routing Tasks
Enabling IPv4 Forwarding
- Go to Network/IP Routing.
- In the IP Routing configuration table, select Enable.
- Click Save.

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Creating an IP Routing Rule

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Creating a Static IPv4 Route

The current routes available are displayed in the Network/Status under the IP4 Routes IPv4 Routes table.

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Advanced IP Routing Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
- To define whether or not the Classless Static Route Option is enabled: Bni.DhcpClientClasslessStaticRouteOption
- To define a list of user classes: Bni.DhcpClientUserClass
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IP Routing Configuration Examples
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Network Firewall
Basic Network Firewall Concepts
Network Firewall
The Network firewall allows you to dynamically create and configure rules to filter incoming packets forwarded by the Mediatrix unit among its network interfaces, when the unit is used as a router. Its main functionality is to secure the traffic routed from and to the devices inside the local network.
Since this is a network firewall, rules only apply to incoming packets forwarded by the unit. The traffic is analyzed and filtered by all the rules configured. If no rule matches the incoming packet, the default policy is applied. A rule's priority is determined by its index in the table. Rules using Network Names are automatically updated as the associated IP addresses and network mask are modified. If the Network Firewall service is stopped, all forwarded traffic is accepted, this tab is greyed out and the parameters are not displayed.
Of course, the more rules are enabled, the more overall performance is affected. You can use a maximum of 20 rules.
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Firewall Rule Order - Important
The order in which the incoming packets are tested against the rules is important if you want to make sure that they actually have a filtering effect on incoming packets.
Rules can be configured to accept or to decline packets. But, always put the most restrictive rules first as they are executed sequentially starting with the first one listed at the top of the table i.e. make sure that the order in which the rules are executed does not cause a rule to be systematically excluded.
- If the first rule excludes all packets coming from a specific net mask, providing a second rule for an IP address with that same net mask will have no effect.
- If the first rule indicates actions to be taken for a specific IP address with a given net mask, and the second rule indicates to exclude all IP addresses with that net mask, both rules will be applied and have a result on the incoming packets.
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Basic Network Firewall Tasks
Configuring the Network Firewall

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Disabling the Network Firewall
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Configuring Black Listing Duration
If a rule with the Black listing enable box checked matches a packet and no Rate Limit Value was set, then the source address of the packet will be black listed and all packets coming from this address will be blocked for the duration of the Blacklist Timeout.
If a rule with the Black listing enable box checked matches a packet and the Rate Limit Value has been reached, then the source address of the packet will be black listed and all packets coming from this address will be blocked for the duration set for the Blacklist Rate Limit Timeout.
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Firewall Port Opening Example
This generic example shows how to allow remote clients to communicate with the IP office located at the LAN side of the Mediatrix unit.
- The default policy is Drop, meaning that any packet that does not match the network firewall rules configured in the Network Firewall Configuration table will be dropped.
- To use the network firewall, IPv4 Forwarding (under IP Routing" tab), must be enabled. Without the forwarding, the network firewall is irrelevant because no packet will get passed from Uplink to LAN.

Rule | |
---|---|
1 | All packets matching an existing connexion are accepted. |
2 | All packets coming through UDP are accepted. |
3 | New packets coming from the IP address 1 and port 1 with a destination to IP address 9 and port 7 through TCP, will be allowed. |
4 | New packets coming from Subnet 2 and port 2 with a destination to Subnet 10 and port 8 through TCP, will be allowed. |
5 | New packets coming from the IP address 3 and port 3 with a destination to IP address 11 with any port through TCP, will be allowed. |
6 | New packets coming from the Subnet 4 and port 4 with a destination to Subnet 12 with any port through TCP, will be allowed. |
7 | Any packet coming from IP address 5 and port 5 to any destination and port through TCP, will be allowed. |
8 | Any packet coming from Subnet 6 and port 6 to any destination and port through TCP, will be allowed. |
9 | This rule will not be applied as it is disabled. |
10 | Any packet coming from Subnet 8 and any port to any destination and port through TCP, will be allowed. |
Default | Packets are dropped. |
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NAT
Basic Concepts
Network Address Translation (NAT)
Network Address Translation (NAT, also known as network masquerading or IP masquerading) rewrites the source and/or destination addresses/ports of IP packets as they pass through a router or firewall. It is most commonly used to connect multiple computers to the Internet (or any other IP network) by using one IP address. This allows home users and small businesses to cheaply and efficiently connect their network to the Internet.
The basic purpose of NAT is to multiplex traffic from the internal network and present it to the Internet as if it was coming from a single computer having only one IP address. The Mediatrix unit’s NAT service allows the dynamic creation and configuration of network address translation rules. Depending on some criteria, the packet matching the rule may see its source or destination address modified.
- Source rules: They are applied on the source address of outgoing packets.
- Destination rules: They are applied on the destination address of incoming packets.
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Understanding Network Address Translation Rules
A NAT rule specifies a set of matching conditions for network packets.
- Source Address
- Destination Address
- Protocol (All, TCP, UDP or ICMP).
- Source Port
- Destination Port
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NAT Rule Order - Important
The NAT rules are applied on a first match basis, in the order they appear in the configuration.
Because the first match is applied, you must ensure that specific rules come before more general rules, or the specific rules might not be applied as desired.
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Destination or Source IP Address Format
- An empty string, meaning that the rule will match any IP address
- An IP address, for example 192.168.0.11
- A network address, for example 192.168.1.0/24, which corresponds to all IP addresses in the range 192.168.1.0 to 192.168.1.255
- Specifying the interface name without a trailing slash represents the IP address
- The same name with the trailing slash represents the network.Note: This is case sensitive (the first letter must be uppercase)
- Lan1 will be replaced by the current IP address of the lan interface, 192.168.0.10
- Lan1/ will be replaced with the network of the lan interface, /24, meaning the range from 192.168.0.0 to 192.168.0.255.
If the specified interface is disabled or removed, the rule is automatically disabled thus removed from the NAT. When the network interface is enabled or added back, the rule is automatically enabled and applied in the NAT.
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Source or Destination Port Format
- An empty string, meaning that the rule will match any port
- Single port, for example for a web server: 80
- A range of ports, for example to forward RTP: 5004-5099
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Interaction of NAT rules with the Firewall Service
When using the Network Firewall service, it is important to configure it with respect to the Destination NAT rules because:
- Source NAT (SNAT) rules are executed after the routing decision, before the packet leaves the unit.
- Destination NAT (DNAT) rules are executed before the routing decision, as the packet enters the unit.
An example of this would be port forwarding where the DNAT changes the routed address of a packet to a new IP address/port. The Network Firewall must also accept connection to this IP/port in order for the port forwarding to work.
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Basic Tasks
Starting/Stopping/Restarting the NAT Service Using the DGW Web Page
- If you clicked
, the tab from which you can access the service from the Web pages are greyed out
- If you clicked
, the tab from which you can access the service from the Web pages are no longer greyed out.
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Creating a Source NAT Rule
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Creating a Destination NAT Rule
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NAT Rule Examples
Creating a Source NAT Rule to Forward the Lan to the Uplink Network Interface
- Go to Network/NAT.
-
In the Source Network Address Translation Rules, click
.
- From the Activation selection list, click Enable.
- In the Source Address field, enter Lan1/.
- From the Protocol selection list, choose All.
- In the New Address field, enter Uplink.
- Click Save & Apply.
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DHCP Server
Basic Concepts
DHCP Service
The DHCP service allows the Mediatrix unit to act as a DHCP server. The Mediatrix will be able to allocate a range of IP addresses to use on a network, reserve, and distribute the IP addresses and network configuration parameters for specific devices using the MAC address as unique identifier for each device.
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DHCP Server
The Mediatrix unit contains an embedded DHCP server that allocates IP addresses and provides leases to the various subnets that are configured
These subnets could have PCs or other IP devices connected to the unit’s LAN Ethernet connectors. These devices could be any combination of switches, PCs, IP phones, etc. If the DHCP service is stopped (which is the default configuration), the DHCP Server tab is greyed out and the parameters are not displayed.
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Default vs Specific DHCP Server Configurations
- Default configurations that apply to each subnet of all network interfaces of the Mediatrix unit
- Specific configurations that override the default configurations. You can define specific configurations for each subnet in your Mediatrix unit. For instance, you could define a lease time for all the subnets of the Mediatrix unit and use the specific configuration parameters to set a different value for one specific subnet.
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Available Configuration Sources
A parameter’s configuration source can be toggled. Here are the possible configuration sources:
Static | Parameter is defined as a static parameter locally |
Automatic | Parameter is obtained from the Uplink network via DHCP or IPCP (PPPoE) |
HostConfiguration | Parameter is obtained from the HOC Service |
HostInterface | Parameter is obtained from network interface matching the subnet |
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Parameters Configuration Sources
The following table lists the configuration parameters and their available configuration sources:
Parameter name | Configuration sources | |||
---|---|---|---|---|
Static | Automatic | HostConfiguration | HostInterface | |
Domain name | X | X | ||
Lease time | X | |||
Default router | X | X | ||
List of DNS servers | X | X | X | |
List of NTP servers | X | X | X | |
List of NBNS servers | X |
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Subnet Server
The DHCP server manages hosts’ network configuration on a given subnet. Each subnet can be seen as having a distinct DHCP server managing it, which is called a subnet server.
To activate a subnet server for a given network interface, the name of that network interface and the name of the subnet must match, the subnet enable option must be enabled and the configuration of the subnet must be valid. Only one subnet can be defined per network interface.
The network interface can be a physical interface or a logical interface (ex: sub-interface using VLAN). The subnet server status is updated dynamically according to many parameters.
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Sending Configuration Parameters to a Client
When an address is leased to a host, several network configuration parameters are sent to that host at the same time according to the options found in the DHCP request. Parameters are set to default at subnet creation time. A parameter can be defined with a subnet specific configuration.
The subnet server will not send a parameter with an empty value. This means that if a client requests a domain name and the subnet server domain name parameter contains an empty field, the subnet server will not add the domain name option in the DHCP response.
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Lease Assignment
In order to assign leases, the subnet server draws from an IP address pool (or subnet scope) defined by a start address and an end address. The subnet mask assigned to hosts is taken directly from the network interface. All hosts on the same subnet share the same configuration. The maximum number of supported hosts on a subnet is 254.
Specific IP addresses (static leases), designated by their MAC address, can be defined as reserved for specific hosts.
The subnet server will always assign leases within the IP address pool with an exception for static leases. When a static lease is defined for the host requesting an address, this lease will be assigned to the host if the IP address is within the subnet range even if the address is not within the IP address pool.
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DHCP Server Configuration Options
- Lease Time (Option 51): The Mediatrix unit DHCP server offers a lease time to its subnets. You can use a default lease time for all subnets or define a lease time specific to one or more subnets.
- Domain Name (Option 15): The Mediatrix unit DHCP server offers a domain name to its subnets. You can use a default domain name for all subnets or define a domain name specific to one or more subnets.
- Default Gateway (Option
3): The
Mediatrix unit DHCP server offers a default gateway (also called default router) to its
subnets.Note: The default gateway parameters are not available in the Default interface. You must access the specific subnets configuration to set its parameters.
- DNS (Option 6): The Mediatrix unit DHCP server offers up to four DNS addresses to its subnets. You can use the default DNS addresses for all subnets or define static DNS addresses specific to one or more subnets.
- NTP (Option 42): The Mediatrix unit DHCP server offers the addresses of up to four NTP (Network Time Protocol) servers to its subnets. You can use the default NTP addresses for all subnets or define static NTP IP addresses specific to one or more subnets.
- NBNS (Option 44):The NetBIOS Name Server (NBNS) protocol, also called Windows Internet Name Service (WINS) can be configured through this option. This is only needed if you need file sharing on old Windows 95/98/Me/NT PCs. The Mediatrix unit DHCP server offers up to four NBNS addresses to its subnets. You can use the default NBNS addresses for all subnets or define static NBNS addresses specific to one or more subnets.
- Server Name (Option 66): The Mediatrix unit DHCP server offers a server name to its subnets. You can use a default server name for all subnets or define a server name specific to one or more
- Bootfile Name (Option 67): The Mediatrix unit DHCP server offers a Bootfile name to its subnets. You can use a default Bootfile name for all subnets or define a Bootfile name specific to one or more.
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Basic DHCP Tasks
Configuring the LAN1 DHCP Server Subnet
- Go to Network/DHCP Server.
- In the Select Subnet drop down menu located at the top of the page, select Lan1.
- In the DHCP Server Configuration table, set the DHCP Server Enable to Enable.
- In the Start IP Address and End IP Address fields, indicate the IP range to use
- Set Automatic Configuration Interface to Uplink.
-
Complete the fields of the Lease Time (Option 51) section.
Make sure to set the
- Subnet Specific selection list to Yes
-
Complete the fields of the Domain Name (Option 15) section.
Make sure to set the
- Enable Option selection list to Enable
- Subnet Specific selection list to Yes
- Configuration Source selection list to Static
-
Complete the fields of the Default Gateway (Option
3)
section. Make sure to set the
- Enable Option selection list to Enable
- Configuration Source selection list to Host Interface.
-
Complete the fields of the DNS (Option 6) section. Make
sure to set the
- Enable Option selection list to Enable
- Subnet Specific selection list to Yes
- Configuration Source selection list to Static
-
Complete the fields of the NTP (Option 42) section. Make
sure to set the
- Enable Option selection list to Enable
- Subnet Specific selection list to Yes
- Configuration Source selection list to Static
-
Complete the fields of the NBNS (Option 44) section. Make
sure to set the
- Enable Option selection list to Enable
- Subnet Specific selection list to Yes
- Do not enable Option 66 or 67.
- Click Apply.
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Advanced DHCP Tasks
Creating A VLAN DHCP Server Subnet

The Vlan10 subnet will be available for configuration from the Select Subnet field under Network/DHCP Server.
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Adding a Static Lease Using the DGW Web Interface
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Configuring the DHCP Server Subnet Lease Time (Option 51)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 51 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
-
Under the Lease Time (Option 51) section,
from the Subnet Specific selection list, choose
- No to use the default lease time
- Yes to override the default lease time.
- If you chose Yes, complete the Lease Time field.
-
- Click Apply.
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Configuring the DHCP Server Domain Name Parameters (Option 15)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 15 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
-
Under the Domain Name (Option 15) section,
from the Enable Option
selection list, choose
- Disable to use the default domain name
- Enable to override the default domain name.
- If you chose Enable, complete the fields as required.
- Click Apply.
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Configuring the DHCP Server Default Gateway (Option 3)
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Configuring the DHCP Server DNS Parameters (Option 6)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 6 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
-
Under the DNS (Option 6)
section, from the Enable Option selection list,
choose
- Disable to use the default DNS address
- Enable to override the default router.
- If you chose Enable, complete the fields as required.
- Click Apply.
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Configuring the DHCP Server NTP Parameters (Option 42)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 42 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
-
Under the NTP (Option 42)
section, from the Enable Option selection list,
choose
- Disable to use the default NTP servers
- Enable to override the NTP servers.
- If you chose Enable, complete the fields as required.
- Click Apply.
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Configuring the DHCP Server NBNS (Option 44)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 51 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
- Under the NBNS (Option 44) section, make sure the Enable Option is set to Enable
-
from the Subnet Specific selection list, choose
- No to use the default configuration,
- Yes to override the specific configuration as defined in the following parameters: SpecificNbnsServers.StaticNbns1 , SpecificNbnsServers.StaticNbns2 , SpecificNbnsServers.StaticNbns3 , and SpecificNbnsServers.StaticNbns4.
- If you chose Yes, complete the fields as required.
- Click Apply.
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Configuring the DHCP Server Name Parameters (Option 66)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 66 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
- Under the Server Name (Option 66) make sure the Enable Option is set to Enable,
-
From the Subnet Specific selection list, choose
- No to use the default server name
- Yes to override the default server name.
- If you chose Yes, complete the fields as required.
- Click Apply.
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Configuring the DHCP Server Bootfile Parameters (Option 67)
- Go to Network/DHCP Server.
- From the Select Subnet selection list, choose the subnet requiring Option 67 configuration.
- In the DHCP Server Configuration table, make sure the DHCP Server Enable selection list is set to Enable.
- Under the Bootfile Name (Option 67) section, from the Enable Option selection list, choose Enable to override the default DHCP Server Bootfile Name.
-
From the Subnet Specific selection list, choose
- No to use the default configuration.
- Yes to override the default configuration.
- If you chose Yes, complete the Bootfile Name field.
- Click Apply.
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SIP Proxy
Configuring the SipProxy Service

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SBC
Sentinel on the LAN
The Sentinel 100 or 400 is designed to fit different network roles and topologies. It can be deployed inside a LAN behind a NAT firewall.

In this scenario, the Sentinel is usually configured as follows:
- Private (local) IP assigned to LAN port, Internal SIP clients (e.g. IP phones and IP PBX) also on the same LAN network.
- The Uplink Network interface is associated with the Wan/Eth1 physical link.
- The Lan1 Network interface is associated with the LAN/Eth2-5 physical link.
- The LAN signaling and media interfaces are not used.
- A signaling and media interface (pbx_s and pbx_m) wil be created to avoid port conflicts when configuring the Call agents. They will be assigned associated to the LAN/Eth2-5 network interface
- Local firewall rules created to protect the SBC from outside attack (to complement the Edge NAT firewall router, optional). For more details refer to the Configuring Local Firewalls Configuration guide published on the Media5 documentation portal at https://documentation.media5corp.com
- Port forwarding for SIP and RTP ports set up on the edge NAT firewall router
- SBC SIP near end NAT traversal is configured
- SBC rules to process VoIP calls
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Sentinel on the Edge
The Sentinel 100 or 400 is designed to fit different network roles and topologies. It can be deployed on the network Edge, with a public IP address and firewall enabled.

The Sentinel located on the Edge is usually configured as follows:
- Public IP assigned to WAN/Eth1 port
- Uplink Network interface is associated with the WAN/Eth1
- Lan1 network interface is associated with the LAN/eth2-5
- Private (local) IP assigned to LAN port
- Local firewall rules created to protect the SBC from outside attack, for reference: https://documentation.media5corp.com/display/DGWLATEST/Configuring+Local+Firewalls
- NAT, IP forwarding and Network Firewall are enabled if Sentinel is used as a router (this is optional, the Sentinel by default does not forward any IP packets between the WAN and LAN) for local IP clients, for reference: https://documentation.media5corp.com/display/DGWLATEST/Configuring+a+Mediatrix+unit+as+a+NAT-Firewall+between+the+LAN+and+the+Internet , https://documentation.media5corp.com/display/DGWLATEST/Configuring+the+Network+Firewall
- SBC rules to process VoIP calls
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Configuration
Basic SBC Configuration Concepts
Call Agents
Call Agents represent logical end-points that connect the Mediatrix unit to peers.
For security reasons, the Mediatrix unit communicates by default only with well-known and defined peers. When a request cannot be associated with a Call Agent, it is rejected. Each Call Agent is tied to a specific peer, ensuring all inbound and outbound communications with that peer. Routing rulesets are used to route SIP signaling between Call Agents, where inbound requests from a Call Agent are sent to another (or the same) Call Agent that will send an outbound request to it's related peer. Call Agents can be associated with one or several Rulesets which can be applied to the inbound or the outbound requests.
When there is an inbound request, to determine the Call Agent the inbound request will go through, the Mediatrix unit uses the destination IP address to choose the Network interface. Then the destination port of the inbound request will determine the Signaling or Media Interface used. Finally, the source address and the source port, will allow the Mediatrix unit to direct the request to the appropriate Call Agent. At this point any Rulesets associated with the Call Agent will be applied in order of priority.

When a request is sent out, i.e. there is an outbound request, the Routing Rulesets will determine which Call Agent will be used. Then the Call Agent Rulesets will be applied to the outbound request, in reverse priority order. The outbound request will then be sent through the Signaling and Media Interface associated with the Call Agent. If the public IP address is used, then the SIP request will use this address as the source IP address. The outbound request will then be sent to the peer address of the Call Agent or according to the routing Ruleset if the peer is a Network.

In addition, a Call Agent tracks REGISTER requests or monitor the peer host using SIP options. When these features are activated, the Call Agent registration state and monitoring state are updated allowing a Routing Ruleset to select another Call Agent based on the state of a primary Call Agent.
Eight default types of Call Agents were created for Mediatrix unit. This should be enough to cover all your needs. However, for advanced users, it is possible to create new Call Agents.
Seven of the eight default Call Agents allow the Mediatrix unit to communicate with seven different types of end-points.
Call Agent Name | SIP or Endpoint Peer |
---|---|
wan_ip_trunk_ca | SIP server located on the WAN. |
trunk_lines_ca | Public Switch Telephony Network (PSTN), through PRI, BRI or FXO ports. |
phone_lines_ca | Telephones, through FXS ports. |
lan_ip_pbx_ca | IP Private branch exchange (PBX) located on the LAN. |
local_users_ca | Users via SIP telephony located on the LAN. |
remote_users_ca | Users using SIP telephony located on the WAN. |
registration_ca | Used to route the registrations issued by the Registration Agent. |
secondary_ip_trunk_ca | SIP server if the wan_ip_trunk_ca Call Agent is not available. |
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phone_lines_ca Call Agent
The phone_lines_ca Call Agent is used to route calls via an FXS port of the Mediatrix unit.
Typically it will route calls to and from analog phones and faxes within the enterprise premises. This Call Agent will be used for example when a call is routed from one colleague to another using analog phones connected on the FXS ports of the Mediatrix unit. In such a scenario, the call does not go through the LAN network nor the Internet. Since by default this Call Agent uses the loop_m Media Interface to route the media, the media_relay Ruleset is usually associated with this Call Agent.

The phone_lines_ca Call Agent is associated with the phone_lines_gw gateway of the SipEp service. By default, each FXS port sends a REGISTER by the phone_lines_ca Call Agent. Therefore, the administrator must make sure to route these REGISTERs to a server, or to disable them in the SipEp service. If needed, FXS ports can be configured in the Pots service.

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trunk_lines_ca Call Agent
The trunk_lines_ca Call Agent is used to route calls to the Public Switch Telephony Network (PSTN) or a enterprise's existent PBX through the PRI, BRI or FXO ports of the unit.
Calls made outside the enterprise premises to telecommunication service providers or calls made within the enterprise through a PBX will typically be routed through this Call Agent. Since by default this Call Agent uses the loop_m Media Interface to route the media, the media_relay Ruleset is usually associated with this Call Agent.
If there are no PRI, BRI or FXO card on the unit, calls through trunk lines can also be routed by the wan_ip_trunk_ca or secondary_ip_trunk_ca Call Agent, provided there is an analog or digital gateway to convert the VoIP call to an analog or digital call, or vice versa.

Further more, the trunk_lines_ca Call Agent is associated with the trunk_lines_gw of the SipEp service. FXO/PRI/BRI ports will need to be configured in either the Pots, Isdn, R2 or Eam services for the calls to be routed properly.

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local_users_ca Call Agent
The local_users_ca Call Agent is used to route calls from and to local users, i.e. located in the LAN, via VoIP calls.
For the call to use this Call Agent, the endpoint must belong to the company and use the company's system. This Call Agent is often used as a regrouping point of all local endpoints to be routed to a PBX or Trunk lines. For instance, if you are a Media5 corporation employee, with a cell phone using the Media5 fone application and the entreprise IP-PBX, then your call will be routed through the local_users_ca Call Agent. If an internal employee is using an analog phone, the call can also be routed through this Call Agent provided the call goes first through a gateway to convert the analog call into a VoIP call.

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remote_users_ca Call Agent
The remote_users_ca Call Agent is used to route calls from users working out the office and using VoIP calls through an external network or Internet.
Calls are routed through a WAN then to the Mediatrix unit via the remote_users_ca Call Agent.

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lan_ip_pbx_ca Call Agent
The lan_ip_pbx_ca Call Agent routes calls to and from an IP PBX located in the LAN. The IP PBX manages all internal communications between different SIP clients (soft phones or SIP gateways).
This Call Agent is usually used to link a local PBX with an external trunk (IP or PSTN).

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wan_ip_trunk_ca Call Agent
The wan_ip_trunk_ca Call Agent is used to route VoIP calls.
Calls are routed from or to the main SIP server or provider located in the WAN. Typical peers for this Call Agent are the head office of the enterprise, an Internet telephony service provider (ITSP) or an IP Multimedia Core Network Subsystem (IMS).

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secondary_ip_trunk_ca Call Agent
The secondary_ip_trunk_caCall Agent is used to route VoIP calls.
This Call Agent is identical to the wan_ip_trunk_ca Call Agent. The secondary_ip_trunk_ca can be used for different purposes. The most common use for this Call Agent, is to route calls to and from the backup server in the event the primary server does not respond. However, this Call Agent can also be used, for example, to route calls directly to and from a Branch Office or route a specific type of call such as international calls. Typical peers for this Call Agent are the head office of the enterprise, an Internet telephony service provider (ITSP) or an IP Multimedia Core Network Subsystem (IMS).

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registration_ca Call Agent
The registration_ca Call Agent is used to route the registrations issued by the Registration Agent.
The registration_ca Call Agent should not be used for other purposes. The registrations issued by the Registration Agent must be routed from the registration_ca to the Call Agent facing the destination Sip registrar (typically wan_ip_trunk_ca_ca or secondary_ip_trunk_ca).
The registration is routed using the User Name, Password and Domain to build the AOR and the R-URI . The Contact is the contact header of the registration, it should contain the IP-address or FQDN of the Mediatrix unit.

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Signaling and Media Interfaces
The Signaling Interface is used for SIP signaling and the Media Interface is used for media ( RTP, UDPTL ) processing.
When configuring a Call Agent, you must select a Signaling Interface and a Media Interface. These interfaces are used whenever SIP signaling or media packets are sent to or received by the Call Agent.
It is possible to create several Signaling and Media Interfaces on the same Network Interface but for different purposes. For example, one set of Signaling and Media Interfaces for a WAN SIP Trunk and another set of Signaling and Media Interfaces for remote user calls. This means, for instance, that two Signaling Interfaces will be created on the same Network Interface, using the same IP address, but with a port range that will differ according to their intended use and to avoid conflicts.
A Media or Signaling Interface can be used by more than one Call Agent, but a specific Signaling or Media Interface can be created for a specific Call Agent. This provides the liberty to define non conflicting range of contactable interfaces on any physical network interfaces of the units for your network structure needs (such as Vlans, PPPoE interfaces, multiple Ethernet ports or multiple addresses on a link).
- lan1_m
- lan1_s
- loop_m
- loop_s
- uplink_s
- uplink_m
loop_m and loop_s interfaces are used to communicate with the internal services of the unit. For example, the loop interfaces can be used to communicate with the SipEp service to access phone ports.
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Penalty Box
The penalty box feature is enabled for a specific Call Agent to temporarily avoid contacting the peer hosts (addresses) that are expected not to answer.
Without the use of the penalty box, DNS SRV failover is delayed until the SIP transaction times out. In a DNS SRV failover, without the use of the penalty box, the Call Agent will first try to communicate with the peer host on the first server, then once the SIP transaction has timed out, it will try the second and so on, always waiting for the SIP transaction timer to expire. With the penalty box, the call Agent will not try any servers that are already in the penalty box. Remember, dynamic call routing (e.g. survivability) based on server availability requires the penalty box to be enabled.
- If the transaction timer of a communication is expired, the peer host will be considered Down and will be put in the penalty box.
- If the transaction timer of a keep-alive communication is expired, the peer host will be considered Down and will be put in the penalty box.
- If after sending a keep-alive request or any other message to a peer host, the Call
Agent receives one of the selected error codes, the peer host will be put in the penalty
box. It is the error code that will indicate that the peer host cannot be used. Note: When configuring the Call Agent it is possible to indicate the error codes that will trigger the peer host to be sent to the penalty box.( SBC/Configuration/Blacklisting Error Codes)
It is possible to configure how much time a peer host will remain in the penalty box, and the delay before which a peer host is considered down. This delay starts after the expiration of the transaction timer. It is also possible to disable the penalty box feature by using the special value 0 as the duration the peer host will remain in the penalty box.
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Keep-Alive
The Keep-Alive monitoring parameter allows the unit to periodically send messages to a server to make sure the server can still be reached.
The Keep-Alive parameter is set individually for each Call Agent. SIP options are sent periodically for each Call Agent to their corresponding server. Any response received from the server means that it can be reached. No additional processing is performed on the response. If no response is received after the retransmission timer expires, the Sbc service considers the server as unreachable. In this case, any call attempt through the Call Agent is refused and the peer host will be sent to the Penalty Box. SIP options are still sent when the server cannot be reached and as soon as it can be reached again, new calls are allowed.
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Basic SBC Configuration Tasks
Configuring a Call Agent
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Creating a Media Interface
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Creating a Signaling Interface
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Configuring the Call Agent Penalty Box
- Go to SBC/Configuration.
-
Click
next to the Call Agent you wish to configure.
- In the Configure Call Agent table, set the Keep-Alive field to 30.
- Set the Blacklisting Duration to 60.
- Click Save.

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Setting the Keep-Alive Interval
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Rulesets
Basic Ruleset Concepts
Rulesets
Rulesets define one or several rules used to filter, manipulate or route inbound or outbound requests.
- Call Agent Rulesets: they describe how inbound or outbound requests are handled by a specific Call Agent. They can also implement services or collect data.
- Routing Rulesets: they are used to globally route outbound requests, i.e. that they apply to all Call Agents.
When a request arrives at a Call Agent from a peer, the inbound rules of the Rulesets associated with the Call Agent are executed. Then, Routing Rulesets are executed until a Call Agent is selected for the destination. Last, the outbound rules of the Rulesets associated with the destination Call Agent are executed before sending the request to the peer.
Inbound rules of the Ruleset are executed in ascending Ruleset priority order. Outbound rules are executed in descending Ruleset priority order.
The Mediatrix unit is fundamentally rule driven. This means that almost all features can be activated based on certain conditions evaluated at run-time, based on parts of the signaling messages or media payload. All rules are constructed using the same pattern. They consist of a set of one or more conditions. If all conditions apply (logical conjunction), a set of one or more actions is executed. It is important to understand that rules are generally applied only on incoming or outbound requests. However, some rules have a scope that goes beyond these incoming or outbound requests. For example, header filters apply to all requests exchanged, including incoming requests.
- Factory: Read only Ruleset delivered with the application.
- Custom: User defined Ruleset.
Call Agent Rulesets have an *.crs extension and Routing Rulesets use the *.rrs extension. For more details on Ruleset conditions and descriptions, refer to the DGW Configuration Guide - Reference Guide document published on the Media5 Documentation Portal.
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Ruleset Replacement Expressions
Ruleset replacement expressions are used when the value of a parameter, a command or an action is not known in advance, i.e. the value depends on the result of the SIP message processing.
A Ruleset replacement expression is a string that represents a SIP processing status. Replacement expressions always start with the dollar (“$”) sign followed by an identifier. When the Ruleset uses a replacement expression, the replacement expression is replaced by the value of the SIP processing status representing the replacement expression.
- $aU uses the User part of the P-Asserted-Identity header
- $th uses the Host part of the To header
- sip:$aU@$th, used as the parameter of the Set R-URI action, uses the P-Asserted-Identity and To headers of the incoming request and puts them into the Request URI of the outgoing request.
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Ruleset Replacement Expression Exhaustive List
Macro | Replacements | Description |
---|---|---|
$r | Request-URI (R-URI). The expression refers to current Request-URI which may be changed during the course of request processing | |
$r. | Complete R-URI header | |
$ru | user@host[:port] part of R-URI | |
$rU | User part of the R-URI | |
$rd | R-URI domain (host:port) | |
$rh | Host part of the R-URI | |
$rp | Port number of the R-URI | |
$rP | R-URI Parameters | |
$f | From header | |
$f. | Complete From header field | |
$fu | user@host[:port] part of the From URI | |
$fU | User part of the From URI | |
$fd | From URI domain (host:port) | |
$fh | Host part of the From URI | |
$fp | Port number of the From URI | |
$fn | Display name of the From header field | |
$fP | Parameters of the From header field. Does not include the parameters of the URI. | |
$ft | The tag of the From header field | |
$fH | The name of the From header field, exactly as it is stated in the SIP request. | |
$t | To header | |
$t. | Complete To header | |
$tu | user@host[:port] part of the To URI | |
$tU | User part of the To URI | |
$td | To URI domain (host:port) | |
$th | Host part of the To URI | |
$tp | Port number of the To URI | |
$tn | Display name of the To header field | |
$tP | Parameters of the To header field. Does not include the parameters of the URI. | |
$tt | The tag of the To header | |
$tH | The name of the To header, exactly as it is stated in the SIP request. | |
$a | P-Asserted-Identity (PAI) | |
$a. | Complete PAI header. | |
$au | user@host[:port] part of the PAI URI | |
$aU | User part of the PAI URI | |
$ad | PAI URI Domain (host:port). | |
$ah | Host part of the PAI URI | |
$ap | Port number of the PAI URI | |
$aP | Parameters of the PAI header field. Does not include the parameters of the URI. | |
$at | The tag of the PAI | |
$aH | The name of the PAI header field, exactly as it is stated in the SIP request. | |
$p | P-Preferred-Identity (PPI) | |
$p. | Complete PPI header field. | |
$pu | user@host[:port] part of the PPI URI. | |
$pU | User part of the PPI URI. | |
$pd | PPI URI Domain (host:port). | |
$ph | Host part of the PPI URI. | |
$pp | Port number of the PPI URI. | |
$pP | Parameters of the PPI header field. Does not include the parameters of the URI. | |
$pt | The tag of the PPI header field. | |
$pH | The name of the PPI header, exactly as it is stated in the SIP request. | |
$c | Call-ID | |
$ci | Call-ID of the SIP request | |
$s | Source party | |
$si | Source IP address of the inbound SIP request | |
$sp | Source port number of the inbound SIP request | |
$d | Expected destination party | |
$di | Destination IP address of the outbound SIP request. This replacement expression is only available in outbound rules. | |
$dp | Destination port number of the outbound SIP request. This replacement expression is only available in outbound rules. | |
$R | Interface of the inbound SIP request | |
$Ri | IP address of the Signaling Interface on which the inbound SIP request was received | |
$Rp | Port number of the Signaling Interface on which the inbound SIP request was received | |
$Rf | ID of the Signaling Interface on which the inbound SIP request was received | |
$Rn | Name of the Signaling Interface on which the inbound SIP request was received | |
$RI | The configured address of the SignalingInterface.PublicIpAddr parameter on which the inbound SIP request was received | |
$H | Arbitrary Headers. The replacement expressions in this group mention the name of an arbitrary header between parentheses. The core headers (From, To, Call-ID, Via, Route, Record-Route, Contact) cannot be replaced. Example: $H(Server) will be replaced by the value of the Server header field. |
|
$H(headername) | Value of the 'headername' header | |
$HU(headername) | User part of the URI in the 'headername' header. | |
$Hd(headername) | URI Domain (host:port) of the 'headername' header. | |
$Hu(headername) | user@host[:port] part of the URI in the 'headername' header. | |
$Hh(headername) | Host part of the URI in the 'headername' header. | |
$Hp(headername) | Port number of the URI in the 'headername' header. | |
$Hn(headername) | Display name of the 'headername' header. | |
$HP(headername) | Parameters of the 'headername' header field. Does not include the parameters of the URI. | |
$HH(headername) | Header headername (as URI) headers | |
$m | Request method | |
$m | The method of the request. | |
$V | Call parameter | |
$V(gui.varname) | Value of the 'varname' call parameter. | |
$B | Cnum and Rnum | |
$B(cnum.rnum) | Value of the regular expression backreference.
|
|
$U | Register cache | |
$Ua | Originating AoR from the registration cache. This replacement expression can only be used after the execution of a‘Restore contact from registrar’ or a ‘Retarget R-URI from cache’ action. | |
$UA | Originating alias from the registration cache. This replacement expression can only be used after the execution of a ‘Restore contact from registrar’ or a ‘Retarget R-URI from cache’ action. | |
$_ | Operations on Values | |
$_u(value) | Changes the value to uppercase. | |
$_l(value) | Changes the value to lowercase. | |
$_s(value) | Size of the value. | |
$_5(value) | MD5 sum of the value. | |
$_r(value) | Random number from 0 to 'value' - 1. Example: $_r(5) gives 0, 1, 2, 3 or 4. | |
$# | URL-encoded | |
$#(value) | Encodes the value into a URL format with appropriate escaping for characters outside the ASCII character set. | |
$attr | Global Attributes | |
$attr(version) | Version number of the DGW firmware. The returned value matches with the %version% macro in DGW firmware. | |
$attr(profile) | Profile identification of the DGW firmware. The returned value matches with the %profile% macro in DGW firmware. | |
$attr(serial) | Serial number of the Mediatrix unit. The returned value matches with the %serial% macro in DGW firmware. | |
$attr(mac) | MAC address of the Mediatrix unit. The returned value matches with the %mac% macro in DGW firmware. | |
$attr(product) | Product name of the Mediatrix unit. The returned value matches with the %product% macro in DGW firmware. | |
$attr(productseries) | Product series name of the Mediatrix unit. The returned value matches with the %productseries% macro in DGW firmware. |
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Basic Ruleset Tasks
Creating a New Ruleset
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Cloning a Ruleset
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Modifying a Ruleset
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Adding Rules to a Ruleset
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Changing the Name and Description of a Ruleset
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Deleting a Ruleset
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Changing the Execution Priority Level of a Call Agent Ruleset
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Changing the Execution Priority Level of a Routing Ruleset
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Changing the Execution Priority Level of a Ruleset Rule
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Changing the Execution Priority Level of a Rule Action
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Changing the Execution Priority Level of a Rule Condition
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Importing Rulesets

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Associating Call Agent Rulesets to a Call Agent
- The Call Agents must be configured.
- Importing Rulesets must be completed.

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Associating Routing Rulesets to Your Configuration

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Registration
Basic SBC Registation Concepts
Registration Agent
The Registration Agent is a feature that performs REGISTERs on behalf of other users.
- When users cannot register themselves.
- To separate internal and external networks in Demarcation Point scenarios
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Registration Caching
Registration processing allows maintaining an endpoint reachable even from behind NATs.
When a call is routed by a Ruleset that includes the REGISTER throttling and Enable REGISTER caching actions, the SBC replaces the information of the Contact header with its own IP address before forwarding the call to an endpoint. Because a private IP address is used as the contact address in the Contact header field of the REGISTER messages, it becomes impossible to reach the user from the public Internet without going through the SBC since the contact's address is private.
The manipulated registration information is then registered in the registrar. When a Call is destined to the user, the call will be directed to the SBC. In order for the SBC to know which user is actually being contacted, the SBC keeps a local copy of the user's registration. The local copy includes the private IP address and the user’s SIP URI as well as the public IP address included in the IP header that was assigned to the SIP message by the NAT.
If periodic request-response traffic does not cross the NAT behind which the user is located, the NAT address binding expires and the user becomes unreachable. Therefore, the SBC forces re-registration to keep NAT bindings alive.
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Registration Throttling
Registration throttling protects SIP infrastructures against registration overloads.
Although registration overloads are often self-inflicted by the keep-alive functionality, it may also be caused by a router outage, broken client or Denial of Service attack. The SBC fends off such overloads by using high-performance in-memory registration cache that serves upstream registrations at high-rate, handles them locally, and propagates them down-stream at substantially reduced rate. That’s the case if the registrations were to create new bindings, deleting existing ones or if they were to expire downstream. The propagated registration changes become effective on the SBC only if confirmed by the downstream server. If a registration expires without being refreshed the SBC issues a warning event.
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Registration Cache Clearing
Clearing the registration cache will remove all inaccurate information that could remain if an equipment connecting to the SBC is restarted with new information such as a new IP address.
When the registration cache is cleared, all equipments connecting to the SBC need to re-register themselves before being able to receive SIP requests because without re-registration the SBC will not have the private contact address to know where to route the SIP.
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Basic SBC Registration Tasks
Configuring a Registration Agent
Configure the registration agent used to issue the registrations.

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Finding a Specific AoR in the Registration Cache
- Go to SBC/Registration
- In the Filter table, enter the AoR.
- Click Search.
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Clearing the Registration Cache
- Go to SBC/Registration.
- Click Clear Registration Cache located under the Registration Cache table.
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SBC Advanced Parameters
SBC/Configuration Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Tcp Connect Timeout
Refer to Sbc.SignalingInterface.TcpConnectTimeout .Tcp Idle Timeout
Refer to Sbc.SignalingInterface.TcpIdleTimeout.Registration Expiration
Refer to Sbc.RegistrationAgent.ExpireValue.Registration Expiration
Refer to Sbc.RegistrationAgent.RetryInterval.Min Severity
Refer to Sbc.MinSeverity.Top
SBC/Status Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration variables
Tcp Connect Timeout
Refer to Sbc.SignalingInterfaceStatus.TcpConnectTimeout.Tcp Idle Timeout
Refer to Sbc.SignalingInterfaceStatus.TcpIdleTimeoutNeed Restart Info
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POTS
Basic Concepts
Caller ID Information
The caller ID is a generic name for the service provided by telephone utilities that supply information such as the telephone number or the name of the calling party to the called subscriber at the start of a call.
In typical caller ID systems, the coded calling number information is sent from the central exchange to the called telephone. This information can be shown on a display of the subscriber telephone set. In this case, the caller ID information is usually displayed before the subscriber decides to answer the incoming call. If the line is connected to a computer, caller information can be used to search in databases and additional services can be offered.
In call waiting, the caller ID service supplies information about a second incoming caller to a subscriber already busy with a phone call. However, caller ID on call waiting is not supported by all caller ID-capable telephone displays.
The following basic caller ID features are supported:
- Date and Time
- Calling Line Identity
- Calling Party Name
- Visual Indicator (MWI)
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Caller ID Generation
Caller ID information is sent depending on the application and country-specific requirements: Caller ID generation using DTMF signalling and Caller ID generation using Frequency Shift Keying (FSK)
- DTMF signalling performed during or before ringing, depending on the country settings or
endpoint configuration. The Mediatrix unit provides the calling line identity according to
the following standards:
- Europe: ETSI 300 659-1 January 2001 (Annex B):
- Access and Terminals (AT)
- Analogue access to the Public Switched Telephone Network (PSTN)
- Subscriber line protocol over the local loop for display (and related) services
- Part 1: On-hook data transmission
- Country-specific custom DTMF variations:
- Telebras DTMF (Brasil and Argentina)
- TDK DTMF (Denmark)
- Europe: ETSI 300 659-1 January 2001 (Annex B):
- Frequency Shift Keying (FSK). Different countries use different standards to send caller
ID information. The Mediatrix unit is compatible with the following widely used
standards:
- ETSI 300 659-1
- Continuous phase binary FSK modulation is used for coding that is compatible with:
- BELL 202
- ITU-T V.23
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Caller ID Transmission
For most countries, the caller ID is transmitted after the first ring.
One notable exception is the UK, where Caller ID is sent after the dual tone alerting state tone on an inverted polarity line.
Other modes of transmission can be configured with the Caller ID Transmission parameter (under POTS/Config/General Configuration).
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Flash Hook
Flash hook can be described as quickly depressing and releasing the plunger or the actual handset-cradle to create a signal indicating a change in the current telephone session.
- call waiting
- second call
- call on hold
- conferences
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Country Override Flash Hook Detection Range
This is the range in which the hook switch must remain pressed to perform a flash hook.
When selecting a country (under Telephony/Misc/Country Selection), each country has a default minimum and maximum time value within which pressing and releasing the plunger is actually considered a flash hook. However, these values can be overridden and customised with the Country Override Flash Hook Detection Range (under POTS/FXS Configuration/Country Customisation).
- The minimal delay and maximal delay, in ms, separated by a “-”.
- The minimal value allowed is 10 ms.
- The maximum value allowed is 1200 ms.
- The space character is not allowed.
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FXO Force End-of-Call
The forced end-of-call service regroups all features permitting the unit to terminate a call. This can be required in a telephony network where the FXO loop current drops are not always detected.
- On call failure: This feature is set by setting the Force End of Call On Call Failure parameter to Enable. When a call failure happens, the call is terminated after the timeout configured with the Call Failure Timeout (sec) parameter has elapsed and an error tone is played.
- On silence detection: A call is ended when silence is detected for a delay higher than the value configured by in the Silence Detection Timeout (sec) parameter (Refer to the FXO Silence Detection). The mode is set with the Force End Of Call On Silence Detection Mode parameter.
- On tone detection: A call is ended when a selected tone is detected. The tone for this purpose depends on the detection mode specified by the Force End Of Call On Tone Detection Mode parameter which can be country specific (not available in all countries) or a custom tone. (Refer to FXO Tone Detection).
All previously mentioned parameters are available under POTS/FXO Configuration/ FXO Force End of Call.
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FXO Silence Detection
Silence detection allows the Mediatrix unit to close a line when no voice activity or silence is detected for a specified amount of time.
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FXO Tone Detection
The FXO Tone Detection feature is used to resolve scenarios in which the far-end disconnection tone cannot be detected.
- Tone Detection Custom Frequency
- Tone Detection Custom Cadence
- Detection Custom Repetition
- on is a numerical value representing the time, in milliseconds, during which the tone can be detected
- off is a numerical value representing the time, in milliseconds, during which the tone cannot be detected
- the on and off values can range from 0 to 32,767 ms.
- Specifying more than 4 pairs will only use the first 4 pairs (eight first values).
- If less than 4 pairs are specified, 0 values will be added as necessary.
- The first zero (0) found in the string signals the end of the cadence (i.e. “200, 0, 300, 400” is the same as “200”).
- If it starts with a value of zero (0) , the ring pattern is invalid.
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FXS Country Override Loop Current
When a remote end-user goes on-hook, the Mediatrix unit signals the far end disconnect by performing a current loop drop (< 1 mA) on the analog line.
This current loop drop is typically used for disconnect supervision on analog lines. If the Line Supervision Mode parameter (under POTS/FXS Configuration) is set to DropOnDisconnect then the Mediatrix unit signals the far end disconnect by performing a current loop drop on the analog line. By default, the Mediatrix unit maintains a current drop for 1000 ms,, then a busy tone is generated to indicate the user to hang up. The current loop drop duration can be configured with the Power Drop on Disconnect Duration parameter (under POTS/FXS Configuration). (For more details, refer to the FXS Line Supervision Mode parameter in the DGW Configuration Guide - Reference guide published on the Media5 Documentation Portal).
When an FXS analog line goes off hook, it causes current to flow by closing the loop. The Country Selection parameter (Telephony/Misc/Country) allows the selection of predefined country settings for the tone profiles, ring patterns, and other parameters such as input and output gains. The value of the loop current for each country is by default 30 mA but can be overridden to a value ranging from 20 mA to 32 mA with the Country Override Loop Current parameter (under POTS/FXS Configuration/Country Customisation) provided the Override Country Configuration parameter is enabled (under POTS/FXS Configuration/Country Customisation).
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Basic FXS Tasks
Selecting the Detection/Generation Method of the Caller ID
- Go to POTS/Config.
- In the General Configuration table, complete the fields are required.
- Click Apply.
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Configuring FXS Parameters
- Go to POTS/FXS Configuration.
- In the FXS Configuration table, complete the fields as required.
- Click Apply.

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Overriding FXS Default Country Parameters
- Go to POTS/FXS Configuration.
- In the Country Customisation table, complete the fields as required.
- Click Apply.

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Basic FXO Tasks
Configuring FXO Dialing Parameters
- Go to POTS/FXO Configuration.
- In the FXO Dialing Configuration table, complete the fields as required.
- Click Apply.

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Configuring FXO Answering Configuration

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Configuring the FXO Incoming Call Behavior

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Configuring FXO Line Verification
- Go to POTS/FXO Configuration.
- In the FXO Line Verification table, for each endpoint, complete the fields as required.
- Click Apply.

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Configuring FXO Force End of Call
- Go to POTS/FXO Configuration.
- In the FXO Force End of Call table, complete the fields as required.
- Click Apply.

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Configuring the Dial Tone Detection

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Configuring the Answering Delay
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Configuring the Far End Disconnect Parameters

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Disabling Dial Tone Detection
- Go to POTS/FXO Configuration.
- In the FXO Dialing Configuration table, from the Dial Tone Detection Mode dropbox, select Disable.
- Click Apply.

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Advanced POTS Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
For FXS
- specify the Calling Party Name of the caller ID (CLIP) : Pots.FxsCallerIdPrivateCallingPartyName
- to override a set of services that are activated during an emergency call: Pots.FxsEmergencyCallOverride
- To set the period before the phone starts to ring in the event where the originator of an emergency call hangs-up before the emergency call center disconnects the call: Pots.FxsEmergencyRingTimeout
- To customise a distinctive ringID: Pots.FxsDistinctiveRingId
- To customise a distinctive ring pattern: Pots.FxsDistinctivePattern
For FXO
- To override the FXO Custom Basic Parameters: Pots.FxoCustomBasicParameters.OverrideDefaultCountryParameters and Pots.FxoCustomBasicParameters.Impedance
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FXS Distinctive Ring
The FXS endpoints support four distinctive ringing for basic incoming calls.
To use the distinctive ringing with the Mediatrix unit, the received SIP INVITE message must contain the Alert-Info header field with the proper Call Property value.
The following is an example of an Alert-Info via SIP INVITE:
Alert-Info: <http://127.0.0.1/Bellcore-dr2>
The custom distinctive ring configuration allows the administrator to modify the ring pattern. These parameters can only be configured by the CLI or SNMP. Refer to the Advanced POTS Parameters . section.
- on is a numerical value representing the time, in milliseconds, during which ring tone will be active on the phone.
- off is a numerical values representing the time, in milliseconds, during which the phone will not ring.
For instance, 2000, 1000, 2000, 0 or 2000, 1000, 2000 is a cadence in which the frequency plays for 2 seconds, stops for 1 second, and plays for 2 more seconds.
- It can have up three pairs of “on,off”. If less than 3 pairs are specified, 0 values will be added as necessary. Specifying more than six will only use the six first values.
- If it starts with a value of zero (0) , the ring pattern is invalid.
- The first zero (0) found in the string signals the end of the cadence (i.e. “200, 0, 300” is the same as “200”).
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Examples of FXO Tone Detection
Configuring the Detection of an 8 second 425 Hz Continuous Tone
- Go to POTS/FXO Configuration.
- In the FXO Force End of Call table, from the Force End Of Call On Tone Detection Mode selection list, choose Custom Tone.
- In the Tone Detection Custom Frequency field, enter 425.
- In the Tone Detection Custom Cadence field, insert 8000,0 or 8000.
- Click Apply.

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Configuring an On/Off British Reorder Tone
- Go to POTS/FXO Configuration.
- In the FXO Force End of Call table, from the Force End Of Call On Tone Detection Mode selection list, choose Custom Tone.
- In the Tone Detection Custom Frequency field, enter 400.
- In the Tone Detection Custom Cadence field, enter 400, 350, 225, 525 .
- Click Apply.

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Configuring the Detection of the Special Information Tone (SIT)
- Go to POTS/FXO Configuration.
- In the FXO Force End of Call table, from the Force End Of Call On Tone Detection Mode selection list, choose Custom Tone.
- In the Tone Detection Custom Frequency field, enter 950.
- In the Tone Detection Custom Cadence field, enter 330,660 .
- Click Apply.
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Cascade for Incoming Calls
A corporate PBX uses two VoIP gateways for inbound and outbound communication through a VoIP provider.
- Two Mediatrix devices connected to a SIP Trunk
For example: Cascade for incoming calls:

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Cascade for Outgoing Calls
Corporate IP-PBX uses two VoIP gateways for inbound and outbound communication through the PSTN
- Two Mediatrix units in the LAN
For example:

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ISDN
Cabling Information
ISDN Reference Points
ISDN specifies a number of reference points that define logical interfaces between the various equipment types on an ISDN access line.
- S: The reference point between user terminals and the NT2. This is used in point-to-multipoint BRI connections.
- T: The reference point between NT1 (Modem) and NT2 (PBX) devices. This is used in point-to- point PRI/BRI connections.

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BRI S/T Connection (RJ-48)

Pin# | TE mode | NT mode |
---|---|---|
1 | Not Connected | Not Connected |
2 | Not Connected | Not Connected |
3 | Tx + | Rx + |
4 | Rx + | Tx + |
5 | Rx - | Tx - |
6 | Tx - | Rx - |
7 | Not connected | Not Connected |
8 | Not connected | Not Connected |
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PRI Connection (RJ-48)

Pin # | NT Mode | TE Mode |
---|---|---|
1 | Transmit #2 (+) | Receive #2 (+) |
2 | Transmit #1 (-) | Receive #1 (-) |
3 | Not connected | Not connected |
4 | Receive #2 (+) | Transmit #2 (+) |
5 | Receive #1 (-) | Transmit #1 (-) |
6 | Not connected | Not connected |
7 | Not connected | Not connected |
8 | Not connected | Not connected |
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Status
Basic ISDN Concepts
Preset Configuration
The ISDN Preset Configuration contains a set of values for the configuration of the parameters used by the ISDN connections.
The preset configuration files are located in the file system persistent memory. Depending on the Mediatrix unit you are using, the available ISDN Preset configuration files will differ or, it may also be possible that no preset configuration files are available depending on the Profile. Preset configuration files are provided by Media5 or can be user-defined, i.e. the current ISDN configuration was exported from a unit.
- units that do not use the default values provided by Media5 (for instance, using T1 instead of E1)
- using the same configuration on several units
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Integrated Services Digital Network (ISDN)
ISDN is a set of digital transmission protocols defined by a few international standards body for telecommunications, such as the ITU-T. One or the other of these protocols are accepted as standards by virtually every telecommunications carrier all over the world.
- The user side is implemented in ISDN terminals (phones, terminal adapters, etc.)
- The network side is implemented in the exchange switches of the network operator.
- Both sides have different signaling states and messages.
- ISDN Basic Rate Interface (BRI)
- ISDN Primary Rate Interface (PRI)
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Supported Signaling Protocols
Protocol | Description |
---|---|
DSS1 | Digital Subscriber Signaling System No.1 |
DMS100 | Digital Multiplex System 100 |
NI2 | National ISDN No.2 |
5ESS | 5 Electronic Switching System |
QSIG | ECMA's protocol for Private Integrated Services Networks |
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Auto configuration
- Endpoint Type
- Clock Mode
- Port Pinout (PRI interfaces only)
- Line Coding (PRI interfaces only)
- Line Framing (PRI interfaces only)
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ISDN Parameters Auto-Configured by Auto-Sensing
- PortPinout
- ClockMode
- LineFraming
- LineCoding
- EndpointType
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Basic ISDN Tasks
Auto-Detecting and Auto-Configuring ISDN Interfaces


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Verifying the ISDN Status
At any time, it is possible to check the status of the ISDN links.
- Go to ISDN/Status
- The Physical Link and Signaling status will be displayed for each interface.
If the ISDN cables are properly connected and the basic interface settings are correct, the Physical Link should be Up.

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Advanced ISDN Concepts
Definitions
Term | Definitions |
---|---|
Originating Side | Where the call is initiated on the ISDN network. At the originating side, the USER (TE) uni-side initiates the call by sending a SETUP message towards the NETWORK (NT). Then, the NT interface redirects the call to some other network, for example SS7 or VoIP. |
Destination Side | Where the call reaches its ISDN destination. The NT interface at the destination receives the call from another network, then sends a SETUP message over the ISDN link to one or more TE interfaces. |
ISDN Interface | A physical ISDN port, either a BRI or PRI interface. |
IsdnInterface | This is the 4th layer of the ISDN stack, referred in ITU-T Q.931 (05/98) as the Resource Management and Call Control entities. |
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Inband Tones Generation
In an ISDN network, most of the call setup tones are played locally by the TE equipments (i.e. telephone handset), although some require that the tones be played inband by the NT peer.
When interworking with other networks occurs, such as in the IsdnInterface, the need for the tones to be played inband is more likely to arise.
The IsdnInterface has configurability to enable inband tones to be played locally, on a per-interface basis. This option is present when the IsdnInterface is acting as both the NT and the TE UNI-side. However, in TE mode only, the ringback tone is played.
The Call Setup tones (dial tone, ringback, etc.) are played in the direction where the call has been initiated. The call disconnection tones are played in both directions, but of course will not arrive to the peer who disconnected the call.
When an inband tone is played, a Progress Indicator IE #8 "Inband information or appropriate pattern available" is added to the ISDN message corresponding to the call state change, and in a PROGRESS ISDN message if no state change is occurring.
On TE interfaces, as soon as the NT peer advertises that it plays inband tones through a Progress Indicator IE #8 or #1, the local inband tones generation is disabled for the rest of the call. Refer to the UseImplicitInbandInfoEnable interop parameter for special handling of Progress Indicator #1.
Whenever a tone is played inband locally or when the ISDN peer advertises that inband information is available, the CallManager is notified. The IP media path can then be opened earlier in the call, and can be closed with some delay after the call disconnection initiation. However, the configuration and associated behaviors of the higher-level entities are out of the scope of this document.
The following tables summarize the inband tones generation behaviour for both NT and TE endpoint types.
Signal IE Handling Enabled | Inband Tones Generation Enabled | Inband Tone Played |
---|---|---|
No | No | No |
No | Yes | Yes |
Yes | Don't Care | No |
Signal IE Handling Enabled | Signal IE Received | Inband Tones Generation Enabled | NT Peer Advertised Inband Tones | Inband Tone Played |
---|---|---|---|---|
No | Don't care | No | Don't Care | No |
Yes | Yes | No | Don't Care | Yes |
Yes | No | Don't Care | Don't Care | No |
No | Don't Care | Yes | Yes | No |
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Signal Handling
The Signal IE is used by the NT ISDN interface to tell its TE interfaces peers that they must generate an inband tone locally. Thus, the Signal IEs are sent by the NT only.
When the Signal IE handling is enabled on a given TE interface, the inband tones will be played towards the IP gateway when a Signal IE is received. On a NT interface, a Signal IE will be inserted in the ISDN messages sent to the TE peer when appropriate.
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Interop Parameters
Interop parameters allow the Mediatrix unit to properly work, communicate, or connect with specific ISDN devices.
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Channel Allocation Strategy
- ascending;
- descending;
- round-robin ascending;
- round-robin descending.
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Ascending
In this mode, the IsdnInterface always allocates the free channel that has the lowest number.
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Descending
The highest-numbered free channel is allocated.
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Round-Robin Ascending
Starting from the enabled channel with the lowest number, the channels are selected increasingly at each allocation.
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Round-Robin Descending
Same as round-robin ascending, except that it is exactly the opposite!
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Ressource Management
Reservation of Channels for Incoming and Outgoing Calls
Channels can be reserved for incoming calls or for outgoing calls.
The IncomingChannelRange and OutgoingChannelRange parameters are defined for this purpose.
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Supplementary Services
Supplementary Services Support
Three generic protocols are defined for the control of supplementary services, two of which are stimulus, the third being functional.
- Keypad protocol;
- Feature key management protocol;
- Functional protocol.
The FacilityServicesEnable parameter is used to control the second category, called the common information element procedure, which uses the FACILITY information element.
When the facility services are disabled and the interface receives a FACILITY message, it answers it with a STATUS. When the facility services are enabled, the interface processes the FACILITY messages.
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CLIP
In ISDN, the Calling Line Information Presentation (CLIP) is an optional service offered to the called party which provides the calling party’s ISDN number. When the service is enabled, a Calling Party Number Information Element (CPN IE) containing the caller’s IA5 digits is sent in the SETUP ISDN message.
CLIP is supplemented by privacy rules defined by CLIR and CLIR Override. Refer to the diagrams in the Interaction between CLIP, CLIR, and CLIR override section for details.
For all ISDN signaling protocols except QSIG, operation is as follows: on the originating side, the TE interface always sends the Calling Party Number IE (unless CLIP is disabled). It is up to the NT interface at the destination side to apply the appropriate privacy rules. If the originating side is NT, the Calling Party number is sent only if the Calling Number parameter is not set to 'Restricted' or if the Override flag parameter is set to 'Enabled'.
CLIP is enabled through the ClipEnable parameter, which can take the following values:
Disable | Calling Party Number IE is not sent. |
Enable | Calling Party Number IE is sent in the SETUP message. |
UserOnly | Calling Party Number IE is sent in the SETUP message only if the ISDN interface is configured as a TE. |
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CLIR
The Calling Line Information Restriction (CLIR) is a supplementary service offered to the calling party to restrict presentation of the calling party’s ISDN number to the called party.
CLIR uses the Calling Party Number (CPN) IE’s Presentation Indicator (PI) to disable presentation of the calling number to the called party. CLIR can be disabled by the CLIR override option, described later. Refer to the diagrams in the Interaction between CLIP, CLIR, and CLIR override section for details.
For all ISDN signaling protocols except QSIG, operation is as follows: when the service is enabled on a TE originating interface, the Calling Party Number IE’s Presentation Indicator field is set to "Restricted" upon transmission of an ISDN SETUP message from TE to NT. However, the TE must include the IA5 digits in the Calling Party Number.
When the service is enabled on a NT interface that receives a call, the Calling Party number IE Presentation Indicator is set to "Restricted" in the calling property returned to the call managing system.
For QSIG, when the service is enabled at the outgoing interface, the Calling Party number IE Presentation Indicator parameter is set to 'Restricted'. At the incoming side, this parameter has no effect. However, if the PI flag is set to "Restricted" in the received CPN IE, the calling party number is removed. See ECMA-148 section 8.
CLIR is enabled through the ClirEnable parameter, which can take the following values:
Disable | There is no privacy restriction applied on the CLIP service. |
Enable |
ISDN signaling protocols (except QSIG):
QSIG signaling protocol:
|
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CLIR Override
CLIR override is an option that allows the calling party number to be presented to the destination party even when the Calling Party Number (CPN) IE’s Presentation Indicator (PI) is set to "Restricted". This option is typically used for police or emergency services.
For all ISDN signaling protocols except QSIG, operation is as follows: if the CLIR Override is enabled on the NT interface at the originating side, the Calling Party Number IA5 digits is included in the Calling Party Number IEs even if the Presentation Indicator is set to "Restricted".
For QSIG, the Calling Line Information Restriction Override is a service offered at the destination interface. If the CLIR Override is not enabled and the Presentation Indicator is set to "Restricted" then the Calling Number is not presented. See ECMA-148 section 8.
Refer to the diagrams in the Interaction between CLIP, CLIR, and CLIR override section for details.
CLIR override is enabled through the ClirOverrideEnable parameter, which can take the following values:
Disable | The parameter has no effect. |
Enable |
ISDN signaling protocols (except QSIG):
QSIG signaling protocol:
|
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Interaction between CLIP, CLIR, and CLIR override
The following diagrams show how CLIP, CLIR and CLIR Override override work together to bring (or not) the calling party number from the call originator to the call destination. Refer to the ISDN Signaling Protocols (Except QSIG) and QSIG Signaling Protocol sections for the corresponding diagrams. Call flow must be read from the left (originating network side) to the right (destination network side).
These diagrams also show on which interfaces the ClipEnable, ClirEnable and ClirOverrideEnable parameters have an effect. This is where they must be configured.
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ISDN Signaling Protocols (Except QSIG)
- TE interface at the Originating Network Side;
- NT interface at the Originating Network Side;
- TE interface at the Destination Network Side;
- NT interface at the Destination Network Side.

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QSIG Signaling Protocol
To describe how CLIP/CLIR/CLIR override work together, we only need to identify the interface that sends the SETUP message and the interface that receives it.

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COLP
In ISDN, the Connected Line Identification Presentation is an optional service offered at the originating interface by the NT peer. When the service is enabled, a Connected Line Identification Presentation Element containing the connected number IA5 digits is sent under some conditions in the CONNECT ISDN message.
On the originating side, the TE interface always sends the Connected Party Number IE, it is up to the NT interface at the destination side to apply the appropriate privacy rules. If the originating side is NT, the Connected Party number is sent only if the Connected Number is not set to Restricted or if the Override flag is enabled.
For QSIG, the Connected Line Information Presentation is also an optional service offered at the outgoing and incoming interface. If available, the Connected Party Number IE containing the connected IA5 digits is included in the CONNECT ISDN message at the outgoing interface. However, the Connected Party Number is not presented at the incoming interface if the Connected Number is "Restricted" and the Override flag is not enabled see ECMA-148, section 6.
The COLP can also be affected by the uCP_ISDN_COLP_NUMBER call property in the same way that the CONP is affected by uCP_ISDN_CONP_NAME call property. See CONP section for more information.
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COLR
Generally, the Connected Line Identification Restriction is a service offered to the TE at the originating interface.
When the service is enabled on a TE originating interface, the Connected Party Number IE’s Presentation Indicator field is set to "Restricted" upon transmission of an ISDN CONNECT message from TE to NT interface. However, the TE interface must include the IA5 digits in the Connected Party Number.
For QSIG, when the service is enabled at the outgoing interface, the Connected Party number IE Presentation Indicator parameter is set to 'Restricted'. At the incoming side, this parameter has no effect. See ECMA-148 section 8.
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COLR Override
In ISDN, the Connected Line Identification Restriction Override is a service offered at the originating interface by the NT peer.
If the CLIR Override is enabled on the NT interface at the originating interface, the Connected Party Number IA5 digits are included in the Connected Party Number IEs even if the Presentation Indicator is set to "Restricted".
For QSIG, this parameter has no effect. See ECMA-148 section 8.
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CONP
The Connected Name identification Presentation (CONP) is a supplementary service which provides the name of the answering or alerting user to the calling user.
For ISDN-PBX to IP-PBX calls, if the PrivacyHeadersInResponse parameter is enabled, the uCP_ISDN_CONP_NAME call property will be set from the 180 Ringing, 183 Session Progressing, or 200 OK SIP message accordingly to the values of the P-Asserted-Identity SIP header. If the ConpEnable is enabled, the ISDN CONP called name and connected name will be set accordingly to the value of the uCP_ISDN_CONP_NAME call property respectively in the ISDN Alerting and Connect message.
The following diagram shows a detailed call from ISDN-PBX to IP-PBX with the parameters involved on both the IP and ISDN sides.
For IP-PBX to ISDN-PBX calls, if the ConpEnable parameter is enabled, the uCP_ISDN_CONP_NAME call property will be set from the ISDN Alerting, ISDN Progress, or ISDN Connect from the value of the Called or Connected Name Facility Information Element. If the PrivacyHeadersInResponse parameter is enabled, the P-Asserted-Identity SIP header friendly name will be set to the uCP_ISDN_CONP_NAME call property.
The following diagram shows a detailed call from IP-PBX to ISDN-PBX with the parameters involved on both the IP and ISDN sides.
If the number of characters in the connected/called party name exceeds 50, the gateway will truncate the excess characters.
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Facility Message Waiting Delay
Upon reception of a SETUP from the remote peer, the interface can optionally wait for a configurable amount of time for a FACILITY message before processing the call. As soon as it receives a FACILITY message or the delay expires, it goes on with normal call processing. The delay is configured via the MaximumFacilityWaitingDelay parameter.
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MSN (Multiple Subscriber Number)
The Multiple Subscriber Number is a service allowing the TE to configure up to three numbers. This service is available only for a BRI interface configured in TE Point To Multipoint. When this service is enabled in the TE, the Called Party Number (Called E.164) received from IE is matched with these numbers. If the Called Party Number is found, the call can be processed. In the case where the E.164 is not matched, the call is silently discarded.
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Notify
The NOTIFY is an ISDN service independent of the HOLD and RETRIEVE. It serves only to notify an ISDN endpoint when the remote peer, usually a SIP endpoint, holds or resumes a call. So a NOTIFY REMOTE HOLD message is sent to the ISDN endpoint when the remote peer puts the call on hold, and a NOTIFY REMOTE RETRIEVAL message is sent when the remote peer resumes the call.
If the ISDN SignalingChannelOutgoingNotifyEnable paramater is disabled, no NOTIFY message is sent.
The BRI phone can use this message to inform the user of the new call state, by displaying the remote hold or retrieval message on its LCD screen for example. Note that the BRI phone keeps the voice path opened, so the hold tone or MOH can be heard.
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Calling Party Name
- Facility information element;
- Display information element;
- User-User information element.
- Display
- Facility
- User-User
Calling Party Name is accepted in a Display Information Element only when explicitly identified as a Calling Party Name (i.e. only when "Display Type" = "Calling Party Name" in the information element).
Protocol | CallingNameDelivery | |||
---|---|---|---|---|
eFacility | eDisplay | eUserUser | eSignalingProtocol | |
DSS1 | IE User-User | IE User-User | IE User-User | IE User-User |
Dms100 | IE Facility | IE Display | IE Display | IE Display |
NI-2 | IE Facility | IE Facility | IE Facility | IE Facility |
5ESS | IE Facility | IE Facility | IE User-User | IE Facility |
QSIG | IE Facility | IE Facility | IE Facility | IE Facility |
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Call Rerouting
The Call Rerouting supplementary service allows to reroute an incoming public ISDN call (originated from PSTN) within or beyond the private ISDN network (such a PBX) as specified in the ETS30020701, section 10.5. The Rerouting data are received and relayed through a FACILITY message containing a Facility Information Element. The Rerouting data are encoded in a CallRerouteing invoke component as specified in the ETS30020701, section 7.
In a Mediatrix typical CallRerouting scenario, when the CallRerouting supplementary service is enabled (the Isdn.SignalingChannelFacilityServicesEnable parameter is enabled and the Isdn.SignalingChannelCallReroutingBehavior parameter is set to "RelayReroute" or "ProcessLocally") and a Facility Information Element containing a CallRerouteing invoke component is received via a FACILITY message on a TE endpoint (from the private network), the ISDN service parses the CallRerouting data and forward it to the CallManager via a specific CallMessage.
To prevent infinite CallRerouting loops, the ISDN service inspects the rerouteingCounter value and returns an error if a loop is detected or if the maximal rerouteingCounter value allowed by the ETS300 207 01 is reached (>5). When the CallRerouting service is not supported (Isdn.SignalingChannelFacilityServicesEnable parameter is disabled or Isdn.SignalingChannelCallReroutingBehavior set to "Unsupported"), the CallRerouting request is automatically rejected.
Upon reception of a CallMessage specifying a Rerouting request, the ISDN service inspects the CallRerouting properties set and according to the Isdn.SignalingChannelCallReroutingBehavior parameter, the services takes an action. If the parameter is set to "RelayReroute", a Facility Information Element containing a CallRerouteing invoke component is transmitted to the ISDN peer (public network side) via a FACILITY message. The ISDN service waits for an answer from the peer.
If the parameter is set to "ProcessLocally" or a negative CallRerouting answer is received (a negative answer received would mean that the public network side (PSTN) is unable to complete the call Rerouting request), the Isdn service initiates a new call to process locally the call Rerouting request. The new call is requested to the CallManager without specifying a destination interface to force the CallRouter service to select the appropriate route. If the new call is routed to an ISDN interface, the ISDN service sends a SETUP containing a DivertingLegInformation2 invoke component in the Facility IE as specified in the ETS 300 207 01, section 10.2 and section 10.4. The data related to the call diversion set in the DivertingLegInformation2 are transferred from the CallRerouting properties.
An illustration of a typical ISDN Call Rerouting scenario (Call Forward Unconditionnal) in a Mediatrix device would be as the following sequence diagram:

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Malicious Call Identification
Malicious Call Identification (MCID) is a supplementary service that enables the service provider to identify the source of malicious calls. A user who receives a malicious call from another network can notify the PSTN of the malicious nature of the call, allowing the offnet system to take action, such as notifying legal authorities.
To invoke the MCID supplementary service, the called user shall send a mCIDRequest invoke component carried by a Facility information element in a FACILITY message.
- if accepted, a mCIDRequest return result component, or
- if rejected, a mCIDRequest return error component carried by a Facility information element in a FACILITY message
To enable the MCID supplementary service, the Isdn.SignalingChannel.FacilityServicesEnable and Isdn.SignalingChannel.McidEnable parameters must both be set to Enable. Further more, the MCID feature is only available for DSS1 signaling.
An illustration of a typical ISDN MCID scenario in a Mediatrix device:
On the reception of a SIP INFO message containing the P-Call-Info: malicious
proprietary header, the associated ISDN call will send an ISDN FACILITY message indicating
that this call is tagged as malicious.
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InformationFollowing Operation
The "informationFollowing" operation is supported for NI2 signaling only.
When a SETUP message is received containing an "informationFollowing" operation, the unit immediately sends a PROCEEDING message. The unit then waits normally for a FACILITY message containing the calling party name, for a maximum time configured with the MaximumFacilityWaitingDelay parameter.
The only difference between this behavior and the usual behavior (i.e. without the "informationFollowing" operation), is the immediate sending of the PROCEEDING message before waiting for the calling party name.
Note that the "informationFollowing" operation is mutually exclusive with the configuration parameter CallProceedingDelay, which configures a delay before sending the PROCEEDING message. If the PROCEEDING message is sent due to the "informationFollowing" operation, the CallProceedingDelay parameter is ignored.
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Advice of Charge
To enable the Advice of Charge (AOC) support on the ISDN interface you must enable the FACILITY services and at least one of the following AOC support: AOC-E (End of Call) or AOC-D (During the Call).
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Default Values for Call Properties
Each ISDN interface can be configured with default values for the following parameters in the Calling Party Number IE and the Called Party Number IE.
Information Element (IE) | Parameter | Configuration Parameter |
---|---|---|
Calling Party Number | Type of Number (TON) | DefaultCallingTon |
Calling Party Number | Numbering Plan Indication (NPI) | DefaultCallingNpi |
Calling Party Number | Presentation Indicator (PI) | DefaultCallingPi |
Calling Party Number | Screening Indicator (SI) | DefaultCallingSi |
Called Party Number | Type of Number (TON) | DefaultCalledTon |
Called Party Number | Numbering Plan Indication (NPI) | DefaultCalledNpi |
These parameters provide default values that are inserted in the Calling Party Number IE and the Called Party Number IE when these values are not already provided by the call properties.
Another way to control these values is by using the "Properties Manipulation" feature of the Call Router. This method has precedence over the parameters described here.
- TON and NPI: If the value is not available from the Call Properties, the corresponding value from DefaultCallingTon, DefaultCalledTon. DefaultCallingNpi or DefaultCalledNpi parameter is used directly.
- PI: If PI is not available from the Call Properties, its value is determined by the
following two steps.
- First, it is set to the default value defined by "DefaultCallingPi".
- Second, it can be overridden by the CLIP and CLIR services: the value can be set to "Restricted" by the CLIR service and the value can be set to "NotAvailable" if there is no number to forward.
- SI: Like the other parameters, the DefaultCallingSi parameter is ignored if the SI
value is provided by the Call Properties. If SI is not provided by the call properties, it is
set to the value provided by DefaultCallingSi except for one special case:
when the DefaultCallingSi parameter is set to "Context Dependent", the unit applies
internal rules to set SI to the value that makes most sense according to context. These
internal rules are as follows:
- For all signaling protocols except QSIG:
- If interface is configured as NT (network side), SI is set to "NetworkProvided"
- If interface is configured as TE (user side), SI is set to "UserProvidedNotScreened"
- For QSIG signaling protocol:
- If the calling party number string is not empty, SI is set to "UserProvidedVerifiedAndPassed"
- If the calling party number string is empty, SI is set to "NetworkProvided"
- For all signaling protocols except QSIG:
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Advanced ISDN Tasks
Exporting a Preset Configuration File

The preset configuration file will be displayed under Management/File, in the Internal files table.
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Primary Rate Interface
PRI (E1/T1) Configuration
Important Information for North America
Mediatrix units are configured to default for E1, which is used in most countries in Europe, Middle-East, Africa and Oceania. For the T1 interface used in North America, some settings MUST be changed.
Setting | T1 (North America) | E1 (Default) |
---|---|---|
Line Coding | B8ZS | HDB3 |
Line Framing | ESF (usually), or SF(D4) | CRC4 (usually), or NO-CRC4 |
Signaling Protocol | NI2 (usually) | DSS1 (usually) |
Preferred Encoding Scheme | u-Law | a-Law |
Fallback Encoding Scheme | a-Law | u-Law |
Channel Range | 1-23 | 1-30 |
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Supported Signaling Protocols
Protocol | Description |
---|---|
DSS1 | Digital Subscriber Signaling System No.1 |
DMS100 | Digital Multiplex System 100 |
NI2 | National ISDN No.2 |
5ESS | 5 Electronic Switching System |
QSIG | ECMA's protocol for Private Integrated Services Networks |
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Important PRI Settings
- In TE (Terminal Equipment) mode, the unit is normally in slave mode and will automatically update its clock from the other (telco) side.
- In NT (Network Termination) mode, the unit is normally in master mode and will provide the clock to the other side.
- For more information on clock reference when using multiple interfaces, refer to the Mediatrix Gateways and ISDN Clock Synchronisation and Synchronising Unit Operation (TDM Sync) published on the Media5 Documentation Portal.
Refer to the Supported Signaling Protocols section.
Only valid when receiving a SETUP message. The user sending the SETUP message does not indicate an alternative bearer capability.
This is typically used for fractional T1 or E1 service.
- Channels start at 1 and make abstraction of the synchronisation and signaling timeslots.
- Channels outside of the range defined for this field are ignored. For example:
- Fractional T1 512K: Channel Range 1-8 (corresponds to B channels 1-8, D channel 24)
- Fractional E1 on ramp 10: Channel Range 1-10 (corresponds to timeslot 0 + B channels 1-10 + D channel 16)
- Fractional E1 on ramp 10: Channel Range 1-20 (corresponds to timeslot 0 + B channels 1-15 + D channel 16 + B channels 17-21)
- Bearer channels are by default usable for both incoming and outgoing calls. Use this range to reserve channels for incoming or outgoing calls.
- Channels outside of the range defined by ChannelRange parameter are ignored.
- Channels reserved in both IncomingChannelRange and OutgoingChannelRange parameters are considered usable for both incoming and outgoing calls.
- The space character is ignored and duplication is not allowed.
- Channels must be specified in low to high order.
The value for calls from SIP to ISDN is set to 34 by default, but ranges from 0 to 82.Some telephone companies do not allow customers to pass Calling Name and will drop calls if it is not set to zero.
Call properties set in the Call Router have precedence over the default values of the table. For more details on the Call Router, refer to the Call Router user guide published on the Media5 Documentation Portal.
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Using a Preset Configuration File

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Associating a PRI Port to a Line Type and Protocol
- Go to System/Hardware.
- In the PRI Ports Configuration table, from the Line Type selection list, select either E1 or T1.
- From the Signaling selection list, associate a type of signaling to the PRI port.
- Click Apply.
- Restart the unit.
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Configuring the E1T1 Interface (PRI)
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Advanced Primary Rate Interface (PRI) Tasks
Modifying Port Pinout

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Basic Rate Interface (BRI)
ADvanced BRI Tasks
Configuring the BRI Interface
It is important to take into consideration the following information:
- Endpoint Type: Values used for the Mediatrix unit must be opposite to the value used for the PBX. For instance, if the PBX is set to TE, then the Mediatrix unit must be set to NT. When the BRI interface Signaling Protocol is set to QSIG, the endpoint type is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG. NOTE: To use a specific interface as the clock reference, this parameter must be set to TE. For more information on Clock Synchronisation, refer to the Technical Bulletin - Mediatrix Gateways and ISDN Synchronisation and Synchronising Unit Operation (TDM Sync) published on the Media5 Documentation Portal.
- Preferred Encoding Scheme: Only G.711 u-Law and G.711 a-Law codecs are allowed. G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711 a-Law as preferred encoding protocol.
- Fallback Encoding Scheme: Only G.711 u-Law and G.711 a-Law codecs are supported. Only valid when receiving a SETUP message. The user sending the SETUP message does not indicate alternative bearer capability.
- Clock Mode: "Auto" should be the value to use. In a BRI configuration, setting the clock mode to slave for a NT endpoint can be set for interop usage, while setting the clock mode to master for a TE endpoint is invalid (slave mode is automatically applied in this case). For more information on Clock Synchronisation, refer to the Technical Bulletin - Mediatrix Gateways and ISDN Synchronisation and Technical Bulletin - Synchronising Unit Operation (TDM Sync) published on the Media5 Documentation Portal.
- Calling Name Max Length: The value for calls from SIP to ISDN ranges from 0 to 82.
- Exclusive B-Channel Selection: When the parameter is enabled only the requested B channel is accepted when a call is initiated; if the requested B channel is not available, the call is cleared.
- Monitor Link State Parameter: When enabled with the Ignore OPTONS on no usable endpoints also enabled under the SIP/Interop page, this will influence how the SIP options are answered.
- Connection Type: depends on the equipment to which the Mediatrix unit port is connected to and it must be the same for all interconnected pieces of equipment.
- Signaling Protocol: Must match the connected ISDN equipment or network.
- TEI Negotiation : Only applies on Point to Multipoint connections.
- Call properties set in the Call Router have precedence over the default values of the Interface Configuration table. For more details on the Call Router, refer to the DGW Configuration Guide - Call Router user guide published on the Media5 Documentation Portal.
- In strings, the space character is ignored and duplicating causes is not allowed.
- Some ISDN switches may require that the Sending Complete information element be included in the outgoing SETUP message to indicate that the entire number is included and there are no further destination digits to be sent.
- An ISDN telephone may send INFORMATION messages that contain a “Keypad Facility”. You can thus trigger a supplementary service (Hold, Conference, etc.) by sending a keypad facility. Since the keypads can be received via several INFORMATION messages, they are accumulated until they match or reset if the keypad reception timeout (second) has elapsed since the last keypad has been received. The keypad reception timeout can only be modified via SNMP. If the keypad reception timeout is set to 0, it disables the timeout, thus assuming that all keypads will be received in a single INFORMATION message
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Setting the Clock Mode
- Go to ISDN/Basic Rate Interface.
- In the Select Endpoint dropdown menu, select the endpoint you want to configure.
- In the Interface Configuration table, set the Clock Mode.

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Interop
Advanced Interop Concepts
Interop Parameters
Interop parameters allow the Mediatrix unit to properly work, communicate, or connect with specific ISDN devices.
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Advanced Interop Tasks
Configuring Interop Parameters
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Timer
Basic Timer Concepts
ISDN Clock Mode
ISDN is a synchronous network, meaning that all endpoints on the network need to synchronize on the same clock signal. Typically, one endpoint acts as the clock MASTER, generating the clock signal, and the other endpoints act as clock SLAVE, synchronizing on the clock signal received from the MASTER.
By default, a NT type endpoint acts as clock MASTER, and a TE type endpoint acts as clock SLAVE. This default behaviour can be changed by using the web interface of the Mediatrix units. The clock mode of the ISDN endpoints can be set to either clock MASTER or clock SLAVE.
- In transmission, all endpoints, regardless of their type, send a clock signal along with the data they send.
- In reception, all endpoints, regardless of endpoint type, use the clock they receive from the other end to synchronize the received data.
- In transmission, a clock SLAVE adjusts the clock it sends, based on the clock received from the other end in reception.
- In transmission, a clock MASTER sends an absolute clock signal that does not depend on the clock received from the other end.
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ISDN Clock Reference
On Mediatrix units with multiple ISDN endpoints, it is possible to select which endpoint should be used as the clock reference. When an endpoint is designated as clock reference, the other endpoints on the Mediatrix unit use this endpoint s clock as a synchronization source for generating their own clock.
In the following example, the clock signal generated by the ISDN PBX is received on the Mediatrix C740 #1 s TE Slave endpoint and taken as a reference for this unit. Therefore the clock signal generated by this unit s NT Master endpoint is synchronized on this signal. The Mediatrix C740 #2 s TE Slave endpoint receives this signal and uses it as a reference for this unit, meaning that the clock signal generated by this unit s NT Master endpoint towards the ISDN phone is synchronized on this signal, therefore on the PBX s signal.

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Current Clock Reference
- Only a clock SLAVE interface can be used as clock reference.
- If there is a configured preferred clock reference interface, this interface becomes the clock reference as soon as it is UP (and running in SLAVE mode).
- If the preferred clock reference interface is DOWN or if there is no configured preferred clock reference interface, the first clock SLAVE interface to become UP becomes the clock reference.
- If no clock SLAVE interface is UP, there is no clock reference and the unit uses its own internally-generated clock signal.
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IP Connection Between ISDN Networks
When using Mediatrix units to connect different ISDN networks through IP, each ISDN network runs on its own clock because there is no way to share the clock signal between the networks. Therefore, VoIP calls between different ISDN networks always experience periodical frame slips, which result in periodical packet losses, or measured bit error rates.

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Mediatrix Unit Clock Synchronization
An ISDN network will be reliable when all endpoint clocks are synchronized correctly. When endpoints are not synchronized, they each run on their own internally-generated clock signals (free-running). The following are acceptable clock signal deviations from the reference specified clock frequency in ppm (parts-per-million).
Recommendation | Interface | Acceptable deviation |
ITU-T I.430 | Basic Rate (BRI) | 100 ppm |
ITU-T I.431 | Primary Rate (PRI) | 50 ppm |
Mediatrix units are tested and certified against these specifications, and are guaranteed to follow the specified clock signal accuracy.
When endpoints are not synchronized, the clock signals are obviously not running at the exact same frequency, because of normal hardware deviations. The clock signals therefore gradually drift away from each other and a periodical frame slip happens
Interface | Acceptable deviation | Worst case difference between endpoints | Frame slip rate |
Basic Rate (BRI) |
100 ppm for each endpoint on a link |
200 ppm | 2E-4 (one slip every 5000 frames) |
Primary Rate (PRI) |
50 ppm for each endpoint on a link |
100 ppm | 1E-4 (one slip every 10000 frames) |
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Direct ISDN Connection Between Endpoints
It is important to configure each endpoint appropriately so the clock signal is synchronized correctly with the rest of the network. The following table shows the different endpoint clock mode combinations and the associated behaviour.
Endpoint 1 \ Endpoint 2 | Master | Slave |
---|---|---|
Master |
|
|
Slave |
|
|
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TDM Sync Feature
The TDM Sync feature provides a means to synchronise the operation of T1/E1/BRI circuits spanning across multiple units and to specify which port of a unit should be used as the reference.
All ISDN ports must be synchronised. When using one device, it is possible to specify a "source" port for the clock on which the other ports can be synchronised on. With several devices, the ports of the different units can also be synchronised. Therefore, it is very helpful to have a special TDM Sync port, with the clock, which can be shared between the devices.
- Choppy video communications
- Failure of fax transmissions
- Lost of audio frames
The Sync signal transferred between units is an 8 KHz Frame Sync Pulse.
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Services
Basic ISDN Services Concepts
Analog/Digital Link Down Call Flow

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Basic ISDN Services Tasks
Enabling ISDN Supplementary Services
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Enabling the Network Traffic Control (NTC) Service
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Other Advanced ISDN Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Interop Play Local Ringback When No MediaStream
Configure the isdn.InteropPlayLocalRingbackWhenNoMediaStream parameter to set how to play the local ringback when there is no stream. For more details, refer to the DGW Configuration Guide - Reference Guide published on the Media5 Documentation Portal.Top
Definitions
Term | Description |
---|---|
BRI |
Basic Rate Interface |
E1 |
European PRI digital signal carrier. 32 channels (30 voice channels + synchronization and signaling) |
ISDN |
Integrated Services Digital Network |
NT |
Network Termination. The endpoint on the telephone switch side. |
PRI |
Primary Rate Interface |
T1 |
North-American PRI digital signal carrier. 24 channels (23 voice + 1 signaling) |
TE |
Terminal Equipment, the endpoint on the customer side |
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BRI Configuration
Important BRI Settings
- Values used for the Mediatrix unit must be opposite to the value used for the PBX. For instance, if the PBX is set to TE, then the Mediatrix unit must be set to NT.
- When the BRI interface Signaling Protocol is set to QSIG, the endpoint type is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.
- To use a specific interface as the clock reference, this parameter must be set to TE.
- Auto should be the value to use.
- In a BRI configuration, setting the clock mode to slave for a NT endpoint can be set for interop usage, while setting the clock mode to master for a TE endpoint is invalid (slave mode is automatically applied in this case). For more information on Clock Synchronisation, refer to the Technical Bulletin -Mediatrix Gateways and ISDN Synchronisation and Technical Bulletin - Synchronising Unit Operation (TDM Sync) published on the Media5 Documentation Portal.
When enabled with the Ignore OPTONS on no usable endpoints also enabled under the SIP/Interop page, this will influence how the SIP options are answered.
Depends on the equipment to which the Mediatrix unit port is connected to and it must be the same for all interconnected pieces of equipment.
Must match the connected ISDN equipment or network.
Only G.711 u-Law and G.711 a-Law codecs are allowed. G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711 a-Law as preferred encoding protocol.
Only valid when receiving a SETUP message. The user sending the SETUP message does not indicate alternative bearer capability.
The value for calls from SIP to ISDN ranges from 0 to 82.
When the parameter is enabled only the requested B channel is accepted when a call is initiated ; if the requested B channel is not available, the call is cleared.
- Set the actual keypad string that is to be considered as a hook-flash in the Hook-Flash Keypad field.
- An ISDN telephone may send INFORMATION messages that contain a “Keypad Facility”. You can thus trigger a supplementary service (Hold, Conference, etc.) by sending a keypad facility.
- Since the keypads can be received via several INFORMATION messages, they are accumulated until they match or reset if the keypad reception timeout (second) has elapsed since the last keypad has been received. The keypad reception timeout can only be modified via SNMP. If the keypad reception timeout is set to 0, it disables the timeout, thus assuming that all keypads will be received in a single INFORMATION message.
- Setting this parameter to an empty string disables the hook-flash detection.
- The permitted keypad must be made up of IA5 characters. See ITU-T Recommendation T.50.
- The space character is not allowed.
- Causes must be specified in low to high order.
- Cause duplication is not allowed.
Only applies on Point to Multipoint connections.
Call properties set in the Call Router have precedence over the default values of the Interface Configuration table. For more details on the Call Router, refer to the DGW Configuration Guide - Call Router user guide published on the Media5 Documentation Portal.
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Configuring the BRI Interface
It is important to take into consideration the following information:
- Endpoint Type: Values used for the Mediatrix unit must be opposite to the value used for the PBX. For instance, if the PBX is set to TE, then the Mediatrix unit must be set to NT. When the BRI interface Signaling Protocol is set to QSIG, the endpoint type is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG. NOTE: To use a specific interface as the clock reference, this parameter must be set to TE. For more information on Clock Synchronisation, refer to the Technical Bulletin - Mediatrix Gateways and ISDN Synchronisation and Synchronising Unit Operation (TDM Sync) published on the Media5 Documentation Portal.
- Preferred Encoding Scheme: Only G.711 u-Law and G.711 a-Law codecs are allowed. G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711 a-Law as preferred encoding protocol.
- Fallback Encoding Scheme: Only G.711 u-Law and G.711 a-Law codecs are supported. Only valid when receiving a SETUP message. The user sending the SETUP message does not indicate alternative bearer capability.
- Clock Mode: "Auto" should be the value to use. In a BRI configuration, setting the clock mode to slave for a NT endpoint can be set for interop usage, while setting the clock mode to master for a TE endpoint is invalid (slave mode is automatically applied in this case). For more information on Clock Synchronisation, refer to the Technical Bulletin - Mediatrix Gateways and ISDN Synchronisation and Technical Bulletin - Synchronising Unit Operation (TDM Sync) published on the Media5 Documentation Portal.
- Calling Name Max Length: The value for calls from SIP to ISDN ranges from 0 to 82.
- Exclusive B-Channel Selection: When the parameter is enabled only the requested B channel is accepted when a call is initiated; if the requested B channel is not available, the call is cleared.
- Monitor Link State Parameter: When enabled with the Ignore OPTONS on no usable endpoints also enabled under the SIP/Interop page, this will influence how the SIP options are answered.
- Connection Type: depends on the equipment to which the Mediatrix unit port is connected to and it must be the same for all interconnected pieces of equipment.
- Signaling Protocol: Must match the connected ISDN equipment or network.
- TEI Negotiation : Only applies on Point to Multipoint connections.
- Call properties set in the Call Router have precedence over the default values of the Interface Configuration table. For more details on the Call Router, refer to the DGW Configuration Guide - Call Router user guide published on the Media5 Documentation Portal.
- In strings, the space character is ignored and duplicating causes is not allowed.
- Some ISDN switches may require that the Sending Complete information element be included in the outgoing SETUP message to indicate that the entire number is included and there are no further destination digits to be sent.
- An ISDN telephone may send INFORMATION messages that contain a “Keypad Facility”. You can thus trigger a supplementary service (Hold, Conference, etc.) by sending a keypad facility. Since the keypads can be received via several INFORMATION messages, they are accumulated until they match or reset if the keypad reception timeout (second) has elapsed since the last keypad has been received. The keypad reception timeout can only be modified via SNMP. If the keypad reception timeout is set to 0, it disables the timeout, thus assuming that all keypads will be received in a single INFORMATION message
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Auto-Detecting and Auto-Configuring ISDN Interfaces


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Advanced Tasks
Enabling ISDN Supplementary Services
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Configuring Interop Parameters
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Modifying Port Pinout

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Exporting a Preset Configuration File

The preset configuration file will be displayed under Management/File, in the Internal files table.
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Other Advanced ISDN Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Interop Play Local Ringback When No MediaStream
Configure the isdn.InteropPlayLocalRingbackWhenNoMediaStream parameter to set how to play the local ringback when there is no stream. For more details, refer to the DGW Configuration Guide - Reference Guide published on the Media5 Documentation Portal.Top
Cabling Information
ISDN Reference Points
ISDN specifies a number of reference points that define logical interfaces between the various equipment types on an ISDN access line.
- S: The reference point between user terminals and the NT2. This is used in point-to-multipoint BRI connections.
- T: The reference point between NT1 (Modem) and NT2 (PBX) devices. This is used in point-to- point PRI/BRI connections.

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BRI S/T Connection (RJ-48)

Pin# | TE mode | NT mode |
---|---|---|
1 | Not Connected | Not Connected |
2 | Not Connected | Not Connected |
3 | Tx + | Rx + |
4 | Rx + | Tx + |
5 | Rx - | Tx - |
6 | Tx - | Rx - |
7 | Not connected | Not Connected |
8 | Not connected | Not Connected |
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PRI Connection (RJ-48)

Pin # | NT Mode | TE Mode |
---|---|---|
1 | Transmit #2 (+) | Receive #2 (+) |
2 | Transmit #1 (-) | Receive #1 (-) |
3 | Not connected | Not connected |
4 | Receive #2 (+) | Transmit #2 (+) |
5 | Receive #1 (-) | Transmit #1 (-) |
6 | Not connected | Not connected |
7 | Not connected | Not connected |
8 | Not connected | Not connected |
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R2 CAS
Status
Basic R2 Concepts
R2 Auto-Configuration
The R2 Auto-configuration feature allows the detection and the configuration of a specific or of all R2 interfaces so that the R2 link goes up and becomes usable with a minimal user interaction.
When launching an auto-configuration process, it stops automatically when all selected interfaces have been tested. For each interface, the auto-configuration process is considered successful when the link becomes up or a failure when all combinations have been tried without having a link up.
- PortPinout : TE / NT
- ClockMode : MASTER / SLAVE
- LineCoding : AMI / HDB3
- LineFraming : CRC4 / NOCRC4
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R2 Preset Configurations
It is possible to save a set of configurations as presets for future R2 connections.
These preset files are located in the file system's persistent memory. They differ depending on the Mediatrix unit you are using. Using preset files is especially useful for units that do not use the default values provided by Media5.
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R2 Basic Tasks
Auto-Configuring R2 Interfaces
- Go to R2/Status.
- In the Automatic Configuration table, from the selection list, choose the interfaces to auto-configure.
- Click Start Sensing.

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Using an R2 Preset Configuration
Remember that the Preset configuration must first be imported in the file management system under Management/File.
- Go to R2/Status.
- In the R2 Preset Configuration table, from the Suggestion list, choose the Preset to apply.
- Click Apply.

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Saving an R2 Preset Configuration
It is a best practice to export the R2 configuration before performing a factory reset or a partial reset, as user-defined presets are deleted.
- Go to R2/Status.
- In the Preset Name field, enter the name of the R2 configuration you wish to save.
- Click Save.

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R2 Configuration
Basic R2 Configuration Concepts
Channel Associated Signaling (CAS)
The Channel Associated Signaling (CAS) is a method of signalling where each traffic channel has a dedicated signaling channel.
- Forwards is the direction from the calling party to the called party.
- Backwards is the direction from the called party to the calling party.
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Line Signals for the Digital Version of MFC/R2
The MFC/R2 digital line signals (defined in ITU-T Q.421) are the ABCD bits of CAS in timeslot 16 of an E1.
They represent the states of the line, and are similar to the states of an analog line. In general, only bits A and B are used. In most systems, bits C and D are set to fixed values and never change. There are some national variants where bit C or D may be used for metering pulses.
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Interregister Signals
The interregister, or interswitch, signals in MFC/R2 signaling (defined in ITU-T Q.441 ) are encoded as the presence of 2, and only 2, out of 6 specific tones, spaced at 120 Hz intervals.
- one for forward signals
- one for backward signals
- 10 signals for the digits 0 to 9
- additional signals available for supervisory purposes
Fro more details refer to the https://www.itu.int/rec/T-REC-Q.400-Q.490-198811-I
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Basic R2 Configuration Tasks
Configuring R2 CAS Parameters

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Advanced R2 Tasks
Overriding Country Default R2 Signaling Variants

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Overriding Country Default R2 Timers Variants

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Overriding Country Default R2 Digit Timers Variants

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Overriding Country Default Link Timers Variants

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Overriding Country Default Tone Variants

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Advanced R2 CAS Parameters
The PRI statistics are available in the Hardware.StatsInfo table. To reset the statistics, you must set "ResetStats" in the Hardware.StatsInfo.ResetStats parameter.
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E and M CAS
Status
Basic E and M Concepts
E&M Auto-Configuration
The E&M Auto-configuration feature allows the detection and the configuration of all, or a specific, E&M interfaces so that the E&M link goes up and becomes usable with a minimal user interaction.
When launching an auto-configuration process, it stops automatically when all selected interfaces have been tested. For each interface, the auto-configuration process is considered a success when the link becomes up or a failure when all combinations have been tried without having a link up.
- PortPinout : TE / NT
- ClockMode : MASTER / SLAVE
- LineCoding : B8ZS / AMI
- LineFraming : SF / ESF
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E&M Preset Configurations
It is possible to save a set of configuration as presets for future E&M connections.
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Basic E and M Tasks
Auto-Configuring E&M

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Saving an E&M Preset Configuration
- Go to E&M/Status.
- In the E&M Preset Configuration table, in the Preset Name field, indicate the name of the Preset.
- Click Save.

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Using an E&M Preset Configuration
- Go to E&M/Status.
- In the E&M Preset Configuration table, from the Local Presetselection list, choose the configuration file to apply.
- Click Apply.

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E and M Config
Basic E and M Config Concepts
Channel Associated Signaling (CAS)
When using CAS signaling, each traffic channel has a dedicated signaling channel. In other words, the signaling for a particular traffic circuit is permanently associated with that circuit.
E&M (earth & magneto, or ear & mouth) is a type of CAS signalling that defines line signaling and register signaling. It is also called Signalling System R1 and is mainly used in North America. E&M was originally developed to allow PABXs in different geographic locations to communicate over an analog private circuit. Some digital interfaces such as CAS also use versions of E&M signaling.
- Forwards is the direction from the calling party to the called party.
- Backwards is the direction from the called party to the calling party.
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Basic E and M Config Tasks
Configuring E&M CAS Parameters

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Advanced E and M Configuration Tasks
Overriding Country Default E&M Signaling Variants

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Overriding Country Default E&M Timers Variants

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Overriding Country Default E&M Digit Timers Variants

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Overriding Country Default E&M Link Timers Variants

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Advanced E&M CAS Parameters
The PRI statistics are available in the Hardware.StatsInfo table. To reset the statistics, you must set "ResetStats" in the Hardware.StatsInfo.ResetStats parameter.
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SIP
Gateways
Basic Gateway Concepts
SIP Gateways
In the scope of this document and Mediatrix products in general, a "SIP gateway" is a software entity of the DGW application that ties the SIP communications to different network interfaces and listening ports. Not to be confused with a "VoIP gateway" which usually represents the whole gateway device.
- Redirect ISDN calls to different SIP servers depending on the call.
- Hunt calls across several gateways.
- Terminate communication sessions between two or multiple parties.
There are two types of SIP gateways: trunk gateway and endpoint gateway. For more details on their difference, refer to Trunk Gateway vs Endpoint Gateway
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Important Information to Know before Using SIP Gateways
Limitations
- Adding a SIP gateway triggers a warning message if the total number of registrations configured reached the defined limit. Refer to Number of Registrations.
- The Mediatrix unit supports a maximum of 10 gateways.IMPORTANT: Downgrading from a version supporting 10 gateways (introduced in DGW v.46.0) to an older version with a limit of 5, will erase gateways in excess of 5. After downgrade, the configuration will require verification.
Naming
Gateway names only support alphanumeric characters, “-”, and “_”.Network Interfaces
- When a gateway is associated with a Network interface for signaling, it applies to all transport types ( UDP, TCP, etc.).
- The LAN interface may be used by a SIP gateway to be bound on the LAN. However, there is no SIP to SIP routing between the LAN and the Uplink interface..
Port Use
If two or more SIP gateways use the same port, only the first SIP gateway starts correctly. The others are in error and not started. The SIP gateway is also in error and not started if the port is already used.Top
Trunk Gateway vs Endpoint Gateway
Trunk Gateways | Endpoint Gateways |
---|---|
Operates like a traditional SIP gateway | Operation is inspired from the IMS (IP Multimedia Subsystem) model. |
No concept of being connected to a SIP server at the SIP level. |
|
Accepts incoming SIP requests from peers on a listening port. | The gateway has no listening port. It accepts incoming SIP requests only from the server on which it is registered. |
When the destination is an FQDN, each SIP transaction is possibly sent to a different IP address, depending on the DNS query result. The gateway assumes that all SIP servers identified by a single FQDN have a synchronised state. | The Endpoint Gateway is designed to operate with a destination specified by an
FQDN. The addresses obtained from the DNS resolution of this FQDN are used as potential “SIP connection” addresses, in ordered priorities. The endpoint gateway first attempts to register to the highest priority server. If the SIP connection (registration) fails, the next priority server is tried, and so on. A failback scheme periodically attempts to switch to the server with the highest priority available. Failover/fallback to another server requires the SIP user to register on that server prior to establishing SIP communication. |
Supports UDP, TCP, and TLS transport:
|
Supports UDP, TCP, and TLS transport
|
SIP dialogs are established independently of each other (in some conditions, the selection of the destination server may depend on the keep alive parameters configured under SIP/Servers). | SIP dialogs for a given SIP user can only be established once the user is registered to the server. |
The call router shows a single SIP source/destination for the gateway. | The call router and gateway status tables show an instance of the gateway for each user of the gateway. |
Supports endpoint, gateway, user, and unit registrations. | Supports endpoint registrations only. |
Supports NAPTR DNS queries. | NAPTR DNS queries are not supported. |
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Basic Gateway Tasks
Creating a SIP Gateway on the LAN Network Interface
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Creating a SIP Gateway on the Uplink Network Interface
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Modifying a SIP Gateway
- Go to SIP/Gateways.
- In the Gateway Configuration table, modify the fields of the gateway.
- Click Apply.
- Click restart required services.
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Deleting a SIP Gateway
- Go to SIP/Gateways.
- In the Gateway Configurationtable, click - next to the gateway to delete.
- Click Apply.
- Click restart required services, located at the top of the page.
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SIP Gateway Advanced Parameters
All SIP gateway parameters are configurable via the DGW Web Interface.
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SIP Servers
Basic SIP Server Concepts
SIP Servers
SIP servers manage all SIP calls in the network, therefore they are the main component of an IP PBX system.
Depending on the environment and the needs of the SIP-based IP-PBX phone system, there are four types of SIP servers that can be used with the Mediatrix units.
- Registrar host: receives REGISTER requests and places the information in the location service for the domain it handles
- Messaging host: receives MWI SUBSCRIBE requests and places the information in the location service for the domain it handles.
- Proxy host: An entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. It is the final destination of the SIP requests.
- Outbound proxy host: A proxy that receives requests from a client, even though it may not be the server resolved by the Request-URI. It is an intermediate step before reaching the proxy host.
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Important Information to Know Before Using SIP Servers
Servers
- The registrar and proxy SIP hosts must have one of the following formats:
- IP address
- IP address: port number, for example:192.168.0.5:5060
- IP address: 0 where "0" indicates the default port 5060 or 5061 in secure mode.
- FQDN
- If any of the SIP server parameters corresponds to a domain name that is bound to a SRV record, the corresponding port must be set to 0 for the unit to perform SRV type DNS requests (as per RFC 3263). Otherwise, the unit will not use DNS SRV requests, but will rather only use Type A requests because it does not need to be specified which port to use.
- All SIP servers identified by a single FQDN are considered by a trunk gateway as having a synchronised state.
- When the destination of a SIP server is an FQDN, each SIP transaction is possibly sent to a different IP address, depending on the DNS query result.
- The outbound SIP proxy server is enabled only if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to 0.0.0.0:0 or leaving the field empty disables the outbound proxy server.
- The outbound proxy address is never included in the SIP packets, it is only used as a physical network destination for the packets.
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Basic SIP Server Tasks
Configuring the Default Registrar Host

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Configuring the Default Messaging Server Host

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Configuring the Default Proxy Host

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Configuring the Default Outbound Proxy Host
- Go to SIP/Servers.
- In the Default Servers table, in the Proxy Host field indicate the server IP address or FQDN to use for all gateways.
- In the Outbound Proxy Host field, indicate the server IP address or FQDN to use for this gateway.
- Click Apply.
- Click restart required services, located at the top of the page.

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Assigning a Specific Registrar Host to a Gateway
You can assign a specific Registrar Server to the default gateway, or to a gateway you have created.
- Go to SIP/Servers.
- In the Registrar Servers table, from the Gateway Specific drop down located on the same row as local, select Yes.
- In the Registrar Host field, indicate the server IP address or FQDN.
- Click Apply.

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Assigning a Specific Messaging Server Host to a Gateway
You can assign a specific Messaging Server to the default gateway, or to a gateway you have created.
- Go to SIP/Servers.
- In the Messaging Servers table, from the Gateway Specific drop down select Yes to assign a specific messaging server to the gateway.
- In the Messaging Server Host field, indicate the IP address: Port number to use.
- Click Apply.
- Click restart required services located at the top of the page.

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Assigning a Specific Proxy Host to a Gateway
You can assign a specific Registrar Server to the default gateway, or to a gateway you have created.
- Go to SIP/Servers.
- In the Proxy Servers table, from the Gateway Specific drop down list located next to the gateway you wish to assign a specific Proxy SIP server, select Yes.
- In the Proxy Host field, indicate the server IP address or FQDN.
- Click Apply.

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Assigning a Specific Outbound Proxy Host to a Gateway
You can assign a specific Registrar Server to the default gateway, or to a gateway you have created.
- Go to SIP/Servers.
- In the Registrar Servers table, from the Gateway Specific drop down located on the same row as local, select Yes.
- In the Outbound Proxy Host field, indicate the server IP address or FQDN.
- Click Apply.
.
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SIP Server Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
For more details on the following advanced parameters, refer to the DGW Configuration Guide - Reference Guide published on the Media5 Documentation Portal.
Setting the default Outbound Proxy Router Type
SipEp.defaultProxyOutboundTypeSetting Gateway-Specific Outbound Proxy Router Type
SipEp.gwSpecificProxy.OutboundTypeTop
Registrations
Basic Registration Concepts
Registration
The DGW firmware handles multiple registration modes, either per endpoint, per gateway, or for the whole unit. Simultaneous registrations are supported for each endpoint or to multiple SIP gateways (a maximum of 5 is supported).
Each endpoint may have its own registration information. You can set information for each endpoint such as its telephone number and friendly name.
Unit registration is used to register a contact not directly related to endpoints. This is generally used to indicate to a registrar the IP location of the Mediatrix unit when it is used as a gateway. In SIP, a registration is valid for a period of time defined by the registrar. Once a unit is registered, the SIP protocol requires the User Agent to refresh this registration before the registration expires. Typically, this re-registration must be completed before the ongoing registration expires, so that the User Agent's registration state does not change (i.e., remains 'registered').

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Registering to all SIP Gateways vs to a specific SIP Gateway
Each endpoint can register to a specific SIP Gateway or register to all configured SIP Gateways simultaneously. DGW handles each registration separately, allowing the registration of individual endpoints to different SIP Gateways.
You can define a specific gateway to register to each endpoint of the Mediatrix unit. For instance, you could set all endpoints of the Mediatrix unit to register to all SIP Gateways simultaneously and set a specific endpoint to register to just one of the SIP Gateways. Using one or more specific parameter usually requires to enable an override parameter and set the specific configuration to apply. Refer to Advanced SIP Registration Parameters section.
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Number of Registrations
The total number of registrations is the sum of all the endpoint and gateway pairs.
For most Mediatrix devices, the total number of registrations is limited to 64. However, for the Mediatrix G7 Series, the Mediatrix S7 Series, the Sentinel 400, and the Sentinel 100, the limit is 500. If a custom DGW profile is used, the limit can be different depending on what value was defined in the profile. For example, when using a Sentinel 400, up to 500 registrations are possible, but if the profile specifies 200, then the unit will only be able to manage up to 200 registrations.
The Mediatrix unit supports a maximum of 10 gateways. An endpoint configured with All gateways generates as many registrations as the number of gateways. In a setup with 3 gateways, one endpoint configured with All as the gateway name counts for 3 in the total number of registrations. The registrations are enabled gateway by gateway until the limit is reached. Endpoint registrations are used first, then unit registrations. The remaining registrations are not registered and do not appear in the Status table. If the number of registrations exceeds the defined limit, a warning is displayed on the Web interface (as well as in the CLI and SNMP interfaces) and a syslog notify (Level Error) is sent. Adding a gateway or an endpoint triggers a warning message if the total number of registrations configured reached the defined limit
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Registration Number Example
In this example, three gateways (Default, gw1, and gw2) are used with the Mediatrix unit, as displayed in the Gateway Configuration table.
The default ports (0 = 5060 for plain SIP and 0 = 5061 for secure SIP) can be used for two gateways, as long as they are attached to different signaling networks (in the Gateway Configuration table, default/Uplink and gw1/Lan1 use the same ports 0 and 0). If you wish to define another gateway for the same signaling network, a different port should be used than the default ones (Gateway Configuration table, gw2 is attached to the Uplink signaling network, as is the default gateway. For this scope, gw2 is only functional if it uses different ports than the default gateway). When a Gateway Name for a User Name is set to All, this means that 3 registrations are used.


User Name | Gateway Name | Number of Registrations |
---|---|---|
ur1 | All | 3 |
ur2 | gw2 | 1 |
te1 | all | 3 |
te2 | all | 3 |
te3 | gw1 | 1 |
te4 | default | 1 |
TOTAL NUMBER OF REGISTRATIONS | 12 |
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SIP Registration Basic Tasks
The following technical bulletins describe the basic tasks related to registration. Keep in mind that these documents are for units configured with the default values of the parameters.
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Advanced SIP Registration Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Non specific registration parameters
- Registration Delay Value: SipEp.InteropRegistrationDelayValue
- Behaviour on Initial-Registration Reception: SipEp.BehaviorOnInitialRegistrationReception
- Unregistered Unit Behavior: SipEp.DefaultUnitRegistrationUnregisteredBehavior
- Default Registration Retry Time Value: SipEp.DefaultRegistrationRetryTime
- Default Unregistered Endpoint Behaviour: SipEp.DefaultRegistrationUnregisteredBehavior
User Agent registration parameters
- Preferred Language: SipEp.UserAgent.AcceptLanguage
- Contact Domain: SipEp.UserAgent.ContactDomain
- SIP User Agent Header: SipEp.InteropSendUaHeaderEnable
Endpoint specific registration parameters
- Specific Unregistered Endpoint Behaviour: SipEp.GwSpecificRegistration.UnregisteredBehavior
Gateway specific registration parameters
- Specific Unregistered Endpoint Behaviour: SipEp.GwSpecificRegistration.UnregisteredBehavior
- Gateway Specific Registration Retry Time: SipEp.GwSpecificRegistrationRetryTime
- Expiration Value in Registration: SipEp.GwSpecificRegistration.ExpirationValue
- Registration Refresh: SipEp.GwSpecificRegistrationRefreshTime
- Registration Expiration: SipEp.GwSpecificRegistration.ProposedExpirationValue
- Enable Configuration: SipEp.GwSpecificRegistration.EnableConfig
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SIP Authentication
Basic SIP Authentication Concepts
SIP Authentication
SIP authentication is a security feature that allows a SIP server to validate the authenticity of the sender, and to accept only the requests when they have the proper credentials.
To authenticate a SIP request, the server issues an authentication challenge to which the SIP client must respond with the proper username/password credentials. The Mediatrix unit can be configured with the authentication information needed to respond to the authentication challenges issued by SIP servers.
The authentication information is configured in the Authentication table. Each entry of the table has a Criteria and a Realm, if it is enabled, that define when it is appropriate to use this entry.
- endpoint-specific: Applies only to challenges received for SIP requests related to a specific endpoint. For instance, the registration associated with the endpoint in the user agent table or the INVITE sent to initiate a call from the endpoint.
- gateway-specific: Applies only to challenges received for SIP requests on a specific SIP gateway.
- user-name specific: Applies only to challenges for a context that uses a specific user name.
- global to the whole unit: Applies to all challenges received for SIP dialogs. The defined user names and passwords will apply to all the endpoints of the unit.
The Authentication table may have between 20 and 100 rows. If you have less than 20 rows, the Mediatrix unit automatically adds new rows up to the minimum of 20.
- The challenge needs to be for a SIP request related to the endpoint specified in the Endpoint column if the corresponding Criteria column is set to Endpoint.
- The challenge needs to be for a SIP request performed on the SIP gateway specified in the Gateway column if the corresponding Criteria column is set to Gateway.
- The challenge needs to be for a context that uses the user name specified in the
User Name field if the
corresponding Criteria column is
set to username. The
user name associated with a context is:
- the user name of the FROM if the context sent the original SIP request, or
- the user name of the request URI if the context received the original SIP request
- The challenge applies to a unit if the corresponding Criteria column is set to Unit.
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Authentication Table Entries - Order is Important
The priority of an entry in the Authentication table is important. The most specific authentication credential must be set before the more generic ones, otherwise the challenges will be responded with the generic credentials rather than the specific ones. If authentication fails with the credentials found in the Authentication table, the SIP server may issue another authentication challenge. In that case, the next entry in the table having a matching criteria is used to reply to this new challenge. This can be repeated until no more matching entry is found.

- Three gateways are defined in the Gateway table (under SIP/Gateways)
- gateway_1
- gateway_2
- gateway_3
- The unit has 4 endpoints:
- Slot4/FXS1
- Slot4/FXS2
- Slot4/FXS3
- Slot4/FXS4
- The SIP requests related to Slot4/FXS1, Slot4/FXS3, andSlot4/FXS4 are sent via gateway_2
- The SIP requests related to Slot4/FXS2 are sent via gateway_3
Step | Description |
---|---|
1 | Endpoint Slot4/FXS3 needs to register to the SIP server. |
2 | A SIP REGISTER request is sent to the SIP server. |
3 | The SIP server must authenticate the request, so it challenges the request with a 401 Unauthorized challenge response. |
4 | Upon reception of this challenge, the Mediatrix unit searches the Authentication table to find the first matching Criteria entry. |
5 | Entry #1 of the Authentication table has a Criteria that matches endpoint Slot4/FXS1, but because the REGISTER was issued for Slot4/FXS3, the match fails. |
6 | Entry #2 of the Authentication table has a Criteria that matches endpoint Slot4/FXS3 and because the REGISTER was issued for Slot4/FXS3, the match succeeds, and the credentials of this entry are used to respond to the challenge. |
7 | If the credentials configured in entry #2 are good, the SIP server accepts to handle the REGISTER request. |
Step | Description |
---|---|
1 | Endpoint Slot4/FXS4 must register to the SIP server. |
2 | A SIP REGISTER request is sent to the SIP server. |
3 | The SIP server must authenticate the request, so it challenges the request with a 401 Unauthorized challenge response. |
4 | Upon reception of this challenge, the Mediatrix unit searches the Authentication table to find the first matching Criteria entry. |
5 | Entry #1 of the Authentication table has a Criteria to match endpoint Slot4/FXS1 but because the REGISTER was issued for Slot4/FXS4, the match fails. |
6 | Entry #2 of the Authentication table has a Criteria to match endpoint Slot4/FXS3 but because the REGISTER was issued for Slot4/FXS4, the match fails. |
7 | Entry #3 of the Authentication table has a Criteria to match gateway gateway_1 but because the REGISTER was issued via gateway_2, the match fails. |
8 | Entry #4 of the Authentication table has a Criteria to match gateway_2 and because the REGISTER was issued via gateway_2, the match succeeds, and the credentials of this entry are used to respond to the challenge. |
9 | If the credentials configured in entry #4 are good, the SIP server accepts to handle the REGISTER request. |
Step | Description |
---|---|
1 | Endpoint Slot4/FXS2 must register to the SIP server. |
2 | A SIP REGISTER request is sent to the SIP server. |
3 | The SIP server must authenticate the request, so it challenges the request with a 401 Unauthorized challenge response. |
4 | Upon reception of this challenge, the Mediatrix unit searches the Authentication table to find the first matching Criteria entry. |
5 | Entry #1 of the Authentication table has a Criteria to match endpoint Slot4/FXS1, but because the REGISTER was issued for Slot4/FXS2, the match fails. |
6 | Entry #2 of the Authentication table has a Criteria to match endpoint Slot4/FXS3, but because the REGISTER was issued for Slot4/FXS2, the match fails. |
7 | Entry #3 of the Authentication table has a Criteria to match gateway gateway_1, but because the REGISTER was issued via gateway_3, the match fails. |
8 | Entry #4 of the Authentication table has a criteria to match gateway gateway_2, but because the REGISTER was issued via gateway_3, the match fails. |
9 | Entry #5 of the Authentication table has a criteria to match the whole unit, so the match succeeds, and the credentials of this entry are used to respond to the challenge. |
10 | If the credentials configured in entry #5 are good, the SIP server accepts to handle the REGISTER request. |
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Important Information To Know before Using Sip Authentication
Access Rights
The SIP /Authentication page is not accessible if you have the User or Observer access right.Naming
The SIP username (i.e. the one in the username criteria) is checked against SIP username syntax of RFC3261.Authentication
The Authentication table (under SIP /Authentication) may have between 20 and 100 rows. If you have less than 20 rows, the Mediatrix unit automatically adds new rows up to the minimum of 20.Authentication Order
The order of the tried entries in the SIP/Authentication table is from the first row to the last row. The row sequence is important. Refer to Authentication Table Entries - Order is Important .Endpoint Authentication
- Several usernames/passwords can be defined for a single Endpoint.
- Endpoint Authentication can be defined for all types of endpoints i.e. E1T1/FXS/FXO/BRI/PRI.
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Basic SIP Authentication Tasks
Defining Authentication Credentials for a Gateway
- Administrator access rights are required to access this page.
- The Mediatrix unit can support up to 5 gateways.

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Defining Authentication Credentials for an Endpoint

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Defining Authentication Credentials for the Unit
- You will not be able to access this page if you have a User or Observer access right.
- The challenge applies to a unit if the corresponding Criteria column is set to Unit.

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Defining Authentication Credentials for a Username

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Selecting the Priority Level of an Authentication
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Transport
Basic SIP Transport Concepts
SIP Transport Types
You can globally set the transport type for SIP all the endpoints of the Mediatrix unit to either UDP (User Datagram Protocol), TCP (Transmission Control Protocol), or TLS (Transport Layer Security).
Please note that RFC 3261 states the implementations must be able to handle messages up to the maximum datagram packet size. For UDP, this size is 65,535 bytes, including IP and UDP headers. However, the maximum datagram packet size the Mediatrix unit supports for a SIP request or response is 5120 bytes excluding the IP and UDP headers. This should be enough, as a packet is rarely bigger than 2500 bytes.
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Important Information to Know Before Configuring SIP Transport
- For SIP signaling, UDP and TCP are mutually exclusive with TLS. Activating TLS automatically disables these unsecure protocols.
- The TLS Persistent Connections Status table is not displayed if the TLS transport is not activated.
- Secure transport (TLS) requires to:
- Synchronise the time in the unit (Refer to Selecting the Unit's Time Zone).
- Install the security certificates used to authenticate the server to which you will connect. Refer to Technical Bulletin - Using Trusted CA and Host Certificates published on the Media5 Documentation Portal.
- Configure the unit so that a “transport=tls” parameter is added to the Contact header of your SIP requests. (SIP/Transport/General Configuration table, Add SIP Transport in Contact Header.
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Basic SIP Transport Tasks
Preparing the Unit to Use TLS for SIP
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Enabling TLS Transport for SIP
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Enabling UDP Transport for SIP
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Enabling TCP Transport for SIP
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Configuring SIP Transport Parameters
- Go to SIP/Transport.
- Enable one or both of the Add SIP Transport in Registration or Add SIP Transport in Contact Header parameters, depending on what is required.
- Apply.

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Advanced SIP Transport Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
- To set transport TLS Cipher Suite settings: refer to the DGW Configuration Guide - Transport Layer Security document published on the Media5 documentation portal.
- To set whether or not the unit uses the symmetric source port feature when Sending SIP traffic over UDP: SipEp.InteropSymmetricUdpSourcePortEnable.
- To set TLS authentication: refer to the DGW Configuration Guide - Transport Layer Security document published on the Media5 documentation portal.
- To set whether or not to force a DNS NAPTR request: SipEp.InteropForceDnsNaptrInTls.
- To set the proper SIP failover conditions: SipEp.defaultSipFailoverConditions and SipEp.GwSpecificFailover.SipFailoverConditions.
- To select the SIP gateway on which failover conditions will be applied: SipEp.gwSpecificFailover.EnableConfig.
- To set the failover conditions on a specific gateway: SipEp.gwSpecificFailover.SipFailoverConditions.
- To set the persistent port interval: SipEp.TransportPersistentPortInterval.
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Interop
Basic SIP Interop Concepts
Interoperability
Interoperability is the ability of computer systems or software to exchange information and to be able to use the information that has been exchanged.
- Device and user registration issues
- Problems when trying to transfer calls
- Increased vulnerability to VoIP cyber attacks
- Messaging delays or failure
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Basic SIP Interop Tasks
Setting the SIP INVITE for T.38 Fax Error Behavior
- Go to SIP/Interop.
- In the Behavior on T.38 INVITE Not Accepted table, set the behavior of each SIP error code.
- Click Apply.
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Configuring SIP Interoperability
- Go to SIP/Interop.
- In the SIP Interop table, set each parameter as required.
- Click Apply.

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Configuring SDP Interoperability
- Go to SIP/Interop.
- In the SDP Interop table, set each parameter as required.
- Click Apply.

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Selecting the Security Level to Validate TLS Server Certificates
- At least one certificate must be returned by the peer even if no validation is made.
- No Validation and Trusted Certificate certificate validation should only be used for lab purposes.
- The Host name must absolutely be known by the DNS server the unit is contacting.
- The certificate authority (CA) must be added to the Cert service.

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Configuring the Behavior of the T.38 INVITE Not Accepted
This task allows you to set the unit’s behaviour after receiving an error to a SIP INVITE for T.38 fax.
- Go to SIP/Interop.
- In the Behavior on T.38 INVITE Not Accepted table, from the Behavior selection list, set the required behavior for each SIP Error Code.
- Click Apply.

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Configuring Miscellaneous Interoperability Parameters
- Go to SIP/Interop.
- In the Misc Interop table, set each parameter as required.
- Click Apply.

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Selecting the SIP TLS Server Certificate Security Level
Only the setting for SIP over TLS transport is available over the Web GUI. For others, like file transfer or TR-069, settings are available with the script. And the levels of support are different. SIP over TLS security level has one more level (Trusted Certificate level) which other services do not have.
- Go to SIP/Interop.
- In the TLS Interop table, select the security level used to validate certificates.
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Troubleshooting SIP Interoperability
Problem | Solution |
---|---|
Media negotiation problems because the Mediatrix unit sends a BYE after receiving a 200 OK. | Set the Enforce Offer Answer Model value to Disable and the Allow Less Media In Response value to Enable. |
No ringing is heard for outgoing calls | The Early RTP feature was enabled (SipEp.InteropListenForEarlyRtpEnable) although the server does not support early RTP (or early media). |
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Advanced Interoperability Interface Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Direction Attributes
- Defining the direction attribute when putting a call on hold: SipEp.InteropOnHoldSdpStreamDirection
- Defining if the direction attribute is present: SipEp.InteropSdpDirectionAttributeEnable
- Enable/Disable SDP Detect Peer Direction Attribute Support: SipEp.InteropSdpDetectPeerDirectionAttributeSupportEnable
- Defining the SDP direction attribute level: SipEp.InteropSdpDirectionAttributeLevel
- Defining the behaviour with the “sendonly” direction attribute: SipEp.InteropOnHoldAnswerSdpStreamDirection
On hold
- Defining the on hold SDP connection address: SipEp.InteropOnHoldSdpConnectionAddress
Headers
- Max-Forwards Header: SipEp.InteropMaxForwardsValue Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request reaches its destination, it is rejected with a “483 (Too Many Hops)” error response. The Max-Forwards SIP header is always present and the default value is 70.
- Resolving the route header: SipEp.InteropResolveRouteHeaderEnable
- Setting whether or not to ignore the Require header: SipEp.InteropIgnoreRequireHeaderEnable
- Setting the SIP User-Agent header format: SipEp.InteropUaHeaderFormat
SIP INFO
- Set the call waiting Private Number Criteria for SIP INFO: SipEp.InteropCallWaitingSipInfoPrivateNumberCriteria
- Defining the SIP INFO Without Content Answer behaviour: SipEp.InteropSipInfoWithoutContentAnswer
Others
- Defining the local ring behaviour on provisional response: SipEp.InteropSdpDirectionAttributeLevel
- Setting the maximum length of the session ID and the session version number: SipEp.InteropSdpOriginLineSessionIdAndVersionMaxLength
- Overriding the register home domain value: SipEp.InteropRegisterHomeDomainOverride
- Enabling the DNS SRV record lock feature: SipEp.InteropLockDnsSrvRecordPerCallEnable
- Enabling the Early RTP feature:
SipEp.InteropListenForEarlyRtpEnable
Note: Do not enable this feature unless the server supports early RTP (or early media). Failing so prevents any ringing to be heard for outgoing calls.
- Setting ACK branch matching: SipEp.InteropAckBranchMatching
- Setting the reject code: SipEp.InteropRejectCodeForUnsupportedSdpOffer
- Setting the keep alive option format: SipEp.InteropKeepAliveOptionFormat
- Defining the unsupported Content-Type behaviour: SipEp.InteropUnsupportedContentType
- If the configured DTMF transport is "Out-of-band using RTP", the unit rather uses the payload type found in the answer: SipEp.InteropUseDtmfPayloadTypeFoundInAnswer
- Determine the behaviour of the device when answering a request offering more than one active media: SipEp.InteropAllowMultipleActiveMediaInAnswer
- Enabling this parameter may improve interoperability with VoLTE endpoints: SipEp.InteropSend183WithSdpBefore180WithoutSdp
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Misc
Basic Concepts
SIP Penalty Box
The penalty box feature is useful when a given host FQDN resolves to a non-responding address. When the address times out, it is put into the penalty box for a given amount of time. During that time, this address is considered as “non-responding” for all requests
This feature is useful when DNS requests return multiple or varying addresses for a host FQDN. It makes sure that, when a host is down, no SIP request is sent to it for a minimal amount of time. When enabled, this feature takes effect immediately on the next call attempt.
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SIP Penalty Box vs Transport Types
Media5 recommends to use this feature with care when supporting multiple transports or you may experience unwanted behaviours. When the Mediatrix unit must send a packet, it retrieves the destination from the packet. If the destination address does not specify a transport to use and does not have a DNS SRV entry that configures which transport to use, then the Mediatrix unit tries all transports it supports, starting with UDP. If this fails, it tries with TCP. The unit begins with UDP because all SIP implementations must support this transport, while the mandatory support of TCP was only introduced in RFC 3261.
Let’s say for instance that the Mediatrix unit supports both the UDP and TCP transports. It tries to reach endpoint “B” for which the destination address does not specify a transport and there is no DNS SRV entry to specify which transports to use in which order. It turns out that this endpoint “B” is also down. In this case, the Mediatrix unit first tries to contact endpoint “B” via UDP. After a timeout period, the UDP target is placed in the penalty box and the unit then tries to contact endpoint “B” via TCP. This fails as well and the TCP target is also placed in the penalty box.
Now, let’s assume endpoint “B” comes back to life and the Mediatrix unit tries again to contact it before UDP and TCP targets are released from the penalty box. First, the unit tries UDP, but it is currently in the penalty box and there is another transport left to try. The Mediatrix unit skips over UDP and tries the next target, which is TCP. Again, TCP is still in the penalty box, but this time, it is the last target the Mediatrix unit can try, so penalty box or not, TCP is used all the same to try to contact endpoint “B”.
There is a problem if endpoint “B” only supports UDP (RFC 2543-based implementation). Endpoint “B” is up, but the Mediatrix unit still cannot contact it: with UDP and TCP in the penalty box, the unit only tries to contact endpoint “B” via its last choice, which is TCP.
The same scenario would not have any problem if the penalty box feature was disabled. Another option is to disable TCP in the Mediatrix unit, which makes UDP the only possible choice for the unit and forces to use UDP even if it is in the penalty box.
You must fully understand the above problem before configuring this feature. Mixing endpoints that do not support the same set of transports with this feature enabled can lead to the above problems, so it is suggested to either properly configure SRV records for the hosts that can be reached or be sure that all hosts on the network support the same transport set before enabling this feature.
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Default Conversion of Standard SIP Error Codes To ISDN Q.850 Cause Codes
SIP Error Codes | Q.850 ISDN Cause Codes |
---|---|
400: Bad Request | 41 Temporary Failure |
401: Unauthorized | 21 Call rejected |
402: Payment required | 21 Call rejected |
403: Forbidden | 21 Call rejected |
404: Not found | 1 Unallocated number |
405: Method not allowed | 63 Service or option unavailable |
406: Not acceptable | 79 Service/option not implemented |
407: Proxy authentication required | 21 Call rejected |
408: Request timeout | 102 Recovery on timer expiry |
410: Gone | 22 Number changed (w/o diagnostic) |
413: Request Entity too long | 127 Interworking |
414: Request-URI too long | 127 Interworking |
415: Unsupported media type | 79 Service/option not implemented |
416: Unsupported URI Scheme | 127 Interworking |
420: Bad extension | 127 Interworking |
421: Extension Required | 127 Interworking |
423: Interval Too Brief | 127 Interworking |
480: Temporarily unavailable | 18 No user responding |
481: Call/Transaction Does not Exist | 41 Temporary Failure |
482: Loop Detected | 25 Exchange - routing error |
483: Too many hops | 25 Exchange - routing error |
484: Address incomplete | 28 Invalid Number Format |
485: Ambiguous | 1 Unallocated number |
486: Busy here | 17 User busy |
500: Server internal error | 41 Temporary failure |
501: Not implemented | 79 Not implemented, unspecified |
502: Bad gateway | 38 Network out of order |
503: Service unavailable | 41 Temporary failure |
504:Server time-out | 102 Recovery on timer expiry |
504: Version Not Supported | 127 Interworking |
513: Message Too Large | 127 Interworking |
600: Busy everywhere | 17 User busy |
603: Decline | 21 Call rejected |
604: Does not exist anywhere | 1 Unallocated number |
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Default Conversion of Standard Q.850 ISDN Cause Codes to SIP Error Codes
The following table presents the conversion of Q.850 ISDN Cause Codes to SIP Error Codes. It is possible to override these default conversions and to configure the conversions of any other Q.850 ISDN Cause Codes between 1 and 127. (Refer to the Basic tasks.)
Normal Event
Q.850 ISDN Cause Codes | SIP Error Codes |
---|---|
1: Unassigned (unallocated) number. | 404 Not Found |
2: No route to specified transit network. | 404 Not Found |
3: No route to destination. | 404 Not Found |
6: Channel unacceptable. | 500 Internal Server Error |
7: Call awarded and being delivered in an established channel. | 500 Internal Server Error |
16 normal call clearing | --- BYE or CANCEL |
17: User busy. | 486 Busy Here |
18: No user responding. | 408 Request Timeout |
19: User alerting, no answer. | 480 Temporarily unavailable |
20: Subscriber absent. | 480 Temporarily unavailable |
21: Call rejected. | 403 Forbidden |
22: Number changed (w/o diagnostic). | 410 Gone |
22: Number changed (w diagnostic). | 301 Moved Permanently |
23: Redirection to new destination. | 410 Gone |
26: Non-selected user clearing. | 404 Not Found |
27: Destination out of order. | 502 Bad Gateway |
28: Invalid number format (incomplete number) | 484 Address incomplete |
29: Facility rejected. | 501 Not implemented |
30: Response to STATUS ENQUIRY. | 500 Internal Server Error |
31 normal unspecified | 480 Temporarily unavailable |
Resource unavailable
ISUP Cause Value | SIP Response |
---|---|
34: No circuit/channel available. | 503 Service unavailable |
38: Network out of order. | 503 Service unavailable |
41: Temporary failure. | 503 Service unavailable |
42: Switching equipment congestion. . | 503 Service unavailable |
43: Access information discarded. | 500 Internal Server Error |
44: Requested circuit/channel not available. | 500 Internal Server Error |
47: Resource unavailable, unspecified | 503 Service unavailable |
Service or option not available
ISUP Cause Value | SIP Response |
---|---|
55: Incoming calls barred within CUG. | 403 Forbidden |
57: Bearer capability not authorized. | 403 Forbidden |
58: Bearer capability not presently available. | 503 Service unavailable |
63: Service or option not available, unspecified. | 500 Internal Server Error |
Service or option not implemented
ISUP Cause Value | SIP Response |
---|---|
65: Bearer capability not implemented. | 488 Not Acceptable Here |
66: Channel type not implemented. | 500 Internal Server Error |
69: Requested facility not implemented. | 500 Internal Server Error |
70: Only restricted digital information bearer. . | 488 Not Acceptable Here |
79: Service or option not implemented, unspecified | 501 Not Implemented |
Invalid Message
ISUP Cause Value | SIP Response |
---|---|
81: Invalid call reference value. | 500: Internal Server Error |
82: Identified channel does not exist. | 500 Internal Server Error |
83: A suspended call exists, but this call identity does not. | 500 Internal Server Error |
84: Call identity in use. | 500 Internal Server Error |
85: No call suspended. | 500 Internal Server Error |
86: Call having the requested call identity has been cleared. | 500 Internal Server Error |
87: user not member of CUG. | 403 Forbidden |
88: Incompatible destination.. | 503 Service unavailable |
91: Invalid transit network selection. | 500 Internal Server Error |
95: Invalid message, unspecified | 500 Internal Server Error |
Protocol error
ISUP Cause Value | SIP Response |
---|---|
96: Mandatory information element is missing. | 500: Internal Server Error |
97: Message type non-existent or not implemented. | 500: Internal Server Error |
98: Message not compatible with call state or message type non-existent or not implemented. | 500: Internal Server Error |
99: Information element non-existent or not implemented. | 500: Internal Server Error |
100: Invalid information element contents. | 500: Internal Server Error |
101: Message not compatible with call state. | 500: Internal Server Error |
102: Recovery on time expiry. | 504 Gateway timeout |
111: Protocol error, unspecified. | 500 Server internal error |
Interworking
ISUP Cause Value | SIP Response |
---|---|
127: Interworking, unspecified | 500 Server internal error |
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PRACK
Reliable provisional responses (PRACK) is supported as per RFC 3262.
- user agent client
- user agent server
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Session Timer Extension
The session timer extension allows detecting the premature end of a call caused by a network problem or a peer’s failure by resending a refresh request periodically.
This refresh request sent by the Mediatrix unit is either a reINVITE or an UPDATE, according to the configuration of the Session Refresh Request Method parameter.
A successful response (200 OK) to this refresh request indicates that the peer is still alive and reachable. A timeout to this refresh request may mean that there are problems in the signalling path or that the peer is no longer available. In that case, the call is shut down by using normal SIP means.
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SDP in Session Timer reINVITEs or UPDATEs
The reINVITE is sent with the last SDP that was negotiated. Receiving a session timer reINVITE should not modify the connection characteristics. If the reINVITE method is used, it is sent with the last SDP that was negotiated. Reception of a session timer reINVITE should not modify the connection characteristics. If the UPDATE method is used, it is sent without any SDP offer.
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Relation Between Minimum and Maximum Values
A user agent that receives a Session-Expires header whose value is smaller than the minimum it is willing to accept replies a “422 Timer too low” to the INVITE and terminates the call. The phone does not ring.
It is up to the caller to decide what to do when it receives a 422 to its INVITE. The Mediatrix unit will automatically retry the INVITE, with a Session-Expires value equal to the minimum value that the user agent server was ready to accept (located in the Min-SE header). This means that the maximum value as set in the Mediatrix unit might not be followed. This has the advantage of establishing the call even if the two endpoints have conflicting values. The Mediatrix unit will also keep retrying as long as it gets 422 answers with different Min-SE values.
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Session Refresh
Sending a session timer reINVITE or UPDATE is referred to as refreshing the session.
Normally, the user agent server that receives the INVITE has the last word on who refreshes. The Mediatrix unit always lets the user agent client (caller) perform the refreshes if the caller supports session timers. In the case where the caller does not support session timers, the Mediatrix unit assumes the role of the refresher.
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Event Handling
The Mediatrix unit supports receiving event handling Notifications to start a remote reboot or a sync of configuration for specific endpoint(s).
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Basic Tasks
Configuring the SIP Penalty Box
- Go to SIP/Misc.
- In the Penalty Box table, from the Penalty Box Activationlist, choose Enable.
- In the Penalty Box Times field, enter the duration during which the SIP target will remain in the SIP penalty box.

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Overriding the Default Mapping for SIP Error Code to ISDN Cause
- Go to SIP/Misc.
-
In the SIP to Cause Error Mapping
table, click
.
- In the Configure New SIP to Cause Error Mapping table, from the Suggestion list, choose a SIP code and cause.
- Click Apply.

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Overriding the Default Mapping for ISDN Cause to SIP Error Codes
- Go to SIP/Misc.
-
In the Cause to SIP Error
Mapping
table, click
.
- In the Configure New Cause to SIP Error Mapping table, from the Suggestion list, choose a SIP code and cause.
- Click Apply.

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Choosing to Use Additional Headers
- Go to SIP/Misc.
- In the Additional Headers table, set the Reason Support, the Referred-By Support and the Privacy Headers In Response fields as required.
- Click Apply.

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Defining the Type of PRACK Support
- Go to SIP/Misc.
- In the Prack table, set the UAC PRACK Support (RFC 3262) and UAS PRACK Support (RFC 3262) fields as required.
- Click Apply.

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Setting the Session Refresh Information

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Overriding the SIP Domain Used
- Go to SIP/Misc.
- In the Gateway Configuration table, in the SIP Domain Override field, enter the SIP domain name that should be used instead of the home domain proxy.
- Click Apply.

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Setting the Blind Transfer Method
- Go to SIP/Misc.
- In the SIP Transfer table, from the Blind Transfer Method choose the required method.
- Click Apply.

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Configuring Call Diversion
- Go to SIP/Misc.
- In the Diversion table, from the selection list, choose the required Method.
- Click Apply.

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Configuring Supported DNS Queries
- Go to SIP/Misc.
- In the DNS table, from the Supported DNS Queries selection , select the type of DNS queries that the SipEp service supports.
- Click Apply.

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Configuring Event Handling
- Go to SIP/Misc.
- In the Event Handling table, for each gateway, from the Reboot selection list, choose how the gateway handles the reboot SIP NOTIFY messages.
- From the Check Sync selection list, choose how the gateway handles check-sync SIP NOTIFY messages.
- Click Apply.

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Enabling Messaging Subscription
- Go to SIP/Misc.
- In the Messaging Subscription table, from the selection list, choose Enable.
- Click Apply.

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Configuring Advice of Charge (AOC)
- Go to SIP/Misc.
- In the AOC table, for each gateway, choose from the AOC-D Support and AOCE Support fields how the AOC-D and AOC-E messages are sent.
- Click Apply.

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Enabling the Media Security Agreement Parameter
- Go to SIP/Misc.
- In the Security Mechanism Agreement table, from the Media Security Agreement selection list, choose Enable.
- Click Apply.

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Advanced Misc SIP Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
- To set the forked provisional responses behaviour:: SipEp.interopForkedProvisionalResponsesBehavior
- To set the DNS failure concealment parameter:
Sip.DnsFailureConcealment
Note: This parameter applies only to Endpoint Gateway types; it has no effect on Trunk Gateways. The behavior on Trunk Gateways always matches the "none" value.
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MEDIA
Codec
Basic Codec Concepts
Codec Configuration
This document describes the steps required to enable or disable the codecs of the Mediatrix unit, as well as access the codec-specific parameters.
- Default configurations that apply to all endpoints of the Mediatrix unit.
- Specific configurations that override the default configurations. Specific configurations can be set for each endpoint in the Mediatrix unit.
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Codec Vs Bearer Capabilities
The Codec Vs. Bearer Capabilities Mapping section allows you to select the codec to prioritise or select in the outgoing INVITE when the incoming SETUP ITC (Information Transfer Capability) matches the configured one. It is also possible to select the ITC value to set in the outgoing SETUP bearer capabilities when the incoming INVITE codec matches the configured one.
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Generic Voice Activity Detection (VAD)
VAD defines how the Mediatrix sends information pertaining to silence. This allows the unit to detect when the user talks, thus avoiding to send silent RTP packets.
- Default configurations that apply to all endpoints of the Mediatrix unit.
- Specific configurations that override the default configurations. Specific configurations can be set for each endpoint in the Mediatrix unit.
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Data Codec Selection Procedure Flow
The Mediatrix unit follows a procedure when selecting data codec. This procedure is the Mediatrix unit default behaviour. Some interop parameter may modify this procedure. Tones are detected on the analog ports only.

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Basic Codec Tasks
Mipt Default vs Specific Endpoint Configuration Examples
Global>Mipt.EpSpecificCodec
_____________________________________________________________
| EpId | EnableConfig | GenericVoiceActivityDetection |
|__________|_______________|__________________________________|
| FXO1 | Disable | Conservative |
| FXO2 | Disable | Conservative |
| FXO3 | Disable | Conservative |
| FXO4 | Disable | Conservative |
| FXS1 | Disable | Conservative |
| FXS2 | Disable | Conservative |
| FXS3 | Disable | Conservative |
| FXS4 | Disable | Conservative |
|__________|_______________|__________________________________|
Global>Mipt.epSpecificCodecG711Mulaw
___________________________________________________________________
| EpId | EnableConfig | VoiceEnable | VoicePriority | DataEnable |
|________|_______________|_____________|_______________|____________|
| FXO1 | Disable | Enable | 0 | Enable |
| FXO2 | Disable | Enable | 0 | Enable |
| FXO3 | Disable | Enable | 0 | Enable |
| FXO4 | Disable | Enable | 0 | Enable |
| FXS1 | Disable | Enable | 0 | Enable |
| FXS2 | Disable | Enable | 0 | Enable |
| FXS3 | Disable | Enable | 0 | Enable |
| FXS4 | Disable | Enable | 0 | Enable |
|________|_______________|_____________|_______________|____________|
_______________________________________________
| EpId | DataPriority | MinPtime | MaxPtime |
|________|______________|__________|____________|
| FXO1 | 0 | 30 | 30 |
| FXO2 | 0 | 30 | 30 |
| FXO3 | 0 | 30 | 30 |
| FXO4 | 0 | 30 | 30 |
| FXS1 | 0 | 30 | 30 |
| FXS2 | 0 | 30 | 30 |
| FXS3 | 0 | 30 | 30 |
| FXS4 | 0 | 30 | 30 |
|________|______________|__________|____________|
Global>Mipt.epSpecificCodecG711Mulaw[EpId=FXS1].DataPriority=10
Global>mipt.Restart
Global>Mipt.epSpecificCodecG711Mulaw
____________________________________________________________________
| EpId | EnableConfig | VoiceEnable | VoicePriority | DataEnable |
|________|_______________|_____________|_______________|____________|
| FXO1 | Disable | Enable | 0 | Enable |
| FXO2 | Disable | Enable | 0 | Enable |
| FXO3 | Disable | Enable | 0 | Enable |
| FXO4 | Disable | Enable | 0 | Enable |
| FXS1 | Disable | Enable | 0 | Enable |
| FXS2 | Disable | Enable | 0 | Enable |
| FXS3 | Disable | Enable | 0 | Enable |
| FXS4 | Disable | Enable | 0 | Enable |
|________|_______________|_____________|_______________|____________|
________________________________________________
| EpId | DataPriority | MinPtime | MaxPtime |
|________|_______________|___________|__________|
| FXO1 | 0 | 30 | 30 |
| FXO2 | 0 | 30 | 30 |
| FXO3 | 0 | 30 | 30 |
| FXO4 | 0 | 30 | 30 |
| FXS1 | 10 | 30 | 30 |
| FXS2 | 0 | 30 | 30 |
| FXS3 | 0 | 30 | 30 |
| FXS4 | 0 | 30 | 30 |
|________|_______________|___________|__________|
Global>Mipt.EpSpecificCodec[EpId=FXS1].EnableConfig=Enable
Global>Mipt.EpSpecificCodec[EpId=FXS1].GenericVoiceActivityDetection=Transparent
Global>Mipt.EpSpecificCodec
____________________________________________________________
| EpId | EnableConfig | GenericVoiceActivityDetection |
|________|________________|__________________________________|
| FXO1 | Disable | Conservative |
| FXO2 | Disable | Conservative |
| FXO3 | Disable | Conservative |
| FXO4 | Disable | Conservative |
| FXS1 | Enable | Transparent |
| FXS2 | Disable | Conservative |
| FXS3 | Disable | Conservative |
| FXS4 | Disable | Conservative |
|________|________________|__________________________________|
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Enabling Default Codecs

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Enabling Codecs For Specific Endpoints

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Configuring Codec Vs. Bearer Capabilities Mapping
- Go to Media/Codecs.
- In the Codecs vs. Bearer Capabilities Mapping table, select Enable from the Enable drop-down list.
- For each enabled codec, complete the fields are required.
- Click Apply.

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Configuring Generic Voice Activity Detection (VAD)

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Configuring Generic Voice Activity Detection (VAD) For Specific Endpoint
- Go to Media/Codecs.
- From the Select Endpoint drop-down list select the port to configure.
- In the Generic Voice Activity Detection (VAD) table, from the Endpoint Specific drop-down list, select Yes.
- Enable the G.711 and G.726 Voice Activity Detection (VAD) by selecting the proper setting in the drop-down list.
- Click Apply.

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Basic Fax Concepts
T.38 Fax
The Mediatrix unit can send faxes in T.38 mode over UDP. T.38 is used for fax if both units are T.38 capable; otherwise, transmission in clear channel over G.711 as defined is used (if G.711 μ-law and/or G.711 A-law are enabled). If no clear channel codecs are enabled and the other endpoint is not T.38 capable, the fax transmission fails.
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Fax Parameters
The Mediatrix unit handles G3 fax transmissions at speeds up to 14.4 kbps. Automatic fax mode detection is standard on all endpoints. Real-Time Fax Over UDP with the T.38 protocol stack is also available.
- The fax codec may be re-negotiated by using a reINVITE.
- The goal of the reINVITE is to allow both user agents to agree on a fax codec, which is
either:
- Clear channel (G.711 or G.726) without Echo Cancellation nor Silence Suppression (automatically disabled).
- T.38.
- Upon fax termination, if the call is not BYE, the previous voice codec is recovered with another reINVITE.
- For fax speeds higher than 14.4 kbps, the Clear channel codec is recommended.
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Fax Information Required for Troubleshooting
Information | Type | |
---|---|---|
Fax Transmission Protocol | Clear Mode | T.38 |
Fax Transmission Speed | High speed | Low speed |
Fax mode | Automatic | Manual |
Receiving fax | Model | Make |
Sending fax | Model | Make |
Fax mode | ECM | non- ECM |
Firewall | Yes | No |
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T.38 Fax Transmission
T.38 is an ITU recommendation for allowing transmission of fax over IP networks in real time.
PROs
- Allows for redundancy, therefore increases the reliability of the transmissions.
- Faxes in T.38 are not as sensitive to network imperfection like packet loss or jitter as faxes in Clear Channel.
CONs
- The number of redundancy packets will increase the bandwidth used
- The higher the speed, the more bandwidth is used.
- May take more bandwidth than a fax in Clear channel.
Requirements
- The Mediatrix must be able to detect a CNG , v21 preamble or T.38 packet to switch to T.38
- Reasonable delay, 1 second round trip is acceptable however 2 seconds could cause timeout or collision
Configuration

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Fax Transmission through a Firewall
Using T.38 with a Firewall
Requirements
- The communication channel must remain open for a fax to go through a firewall i.e. the unit sends "no signal" packets to fill the dead air intervals that could occur during a fax transmission and cause the closure of the firewall.
Configuration
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Clear Channel Fax Transmission
Modulated Fax information from the PSTN is passed in-band end-to-end over a voice speech path in a IP network.
- The configured voice codec is used for fax transmission. This technique works only when the configured codec is G711 with no VAD and no echo cancellation (EC) or when the configured codec is a clear channel codec or G.726/32. Low bit-rate codecs cannot be used for fax transmission.
- Gateway dynamically changes the codec from the codec configured for voice to G.711 with no VAD and no EC for the duration of the fax session. This method is referred to as "codec up speed" or "fax pass-through with up speed".
PROs
- Less intrusive, does not modify the packets
- Does not allows for redundancy
- Takes less bandwidth than a T.38 fax transmission
- The bandwidth usage is practically constant.
- Bandwidth only affected by the P-Time.
CONs
- Sensitive to network imperfection like packet loss or jitter
Configuration

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Clear Channel Fax Flow
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FAX Detection Flow
IF | AND | THEN |
---|---|---|
If T.38 is enabled |
|
Then the Mediatrix switches to T.38 |
|
Then the Mediatrix unit switches to Clear mode. | |
If a CED Note: Note that the CED tone can be detected on both the IP side (egress
side) or on the analog side (ingress side). is detected before a CNG |
The Mediatrix unit will first switch to Clear Channel and if T.38 is enabled, it will then switch to T.38. |
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Basic Fax Tasks
Enabling T.38 Fax Transmission
- Go to Media/Codecs.
-
In the Codec section,
click
located on the same line as T.38.
- Set the Enable filed to Enable.
- Click Apply.

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Setting the Redundancy Level

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Configuring the T.38 No Signal

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Configuring the T.38 No Signal Timeout

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Configuring the Clear Channel Fax Transmission
- Go to Media/Codecs.
-
In the Codec section,
click
located on the same line as the Codec you wish to use for Fax Transmission.
- In the selected Codec table, set the fields as required.
- Make sure to enable the Data Transmission.
- Click Apply.

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Security
Basic Security Concepts
Secure Real-Time Transport Protocol
Secure Real-time Transport Protocol (SRTP) is a profile of Real-time Transport Protocol (RTP) that provides encryption, message authentication, and replay attack protection.
SRTP can be enabled on all Mediatrix unit endpoints, or on one or more specific endpoints. However, SRTP encryption and authentication requires more processing therefore, if SRTP is enabled, the number of calls that the Mediatrix unit can handle simultaneously may be reduced, depending of the enabled codecs . (For more details on resources limitations with SRTP and conferences, refer to the DGW Configuration Guide - Limitations of DGW Platforms document published on the Media5 Documentation Portal).
To reduce the impact on the number of simultaneous calls a Mediatrix unit can handle, is it possible to disable all voice or data codec, including the T.38 protocol, and keep only the G.711 voice codec enabled.
The Mediatrix unit supports the MIKEY protocol using pre-shared keys (MIKEY-PS as per RFC 3830) or the SDES protocol for negotiating SRTP keys
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Basic Tasks
Enabling Secure Media (SRTP) on All Endpoints
All new SIP exchanges will contain RTP/SAVP negotiation elements.

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Enabling Secure Media (SRTP) on a Specific Endpoint
All new SIP exchanges going through the specified endpoint will contain RTP/SAVP negotiation elements.

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Allowing Unsecure T.38 with Secure RTP
The T.38 protocol must be enabled under Media/Codec.
This procedure is required only if SRTP is used and is available provided the Select Endpoint selection list is set to Default.- Go to Media/Security.
- In the Security table, under the RTP section, set the Mode selection list to Secure with fallback.
- Under the T.38 section, set the Allow Unsecure T.38 with Secure RTP selection list to Yes.
- Click Apply.

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Modifying the SRTP Basic Port
- Go to Media/Misc.
- In the Base Ports table,in the filed next to SRTP, indicate in the field to first port to use in SRTP.
The first port to be used in SRTP will be the one specified.

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Advanced RTP Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
For more details, refer to the DGW Configuration Guide - Reference Guide published on the Media5 Documentation Portal.
- Mipt.EnforceSymmetricRtpEnable: to enforce that incoming RTP packets are from the same source as the destination of outgoing RTP packets.
- Mipt.InteropDtmfRtpInitialPacketQty: to specify the quantity of packets sent at the beginning and at the ending of an Out-of-Band DTMF using RTP.
- Mipt.InteropPacketReceptionMode: to select the mode that control the range of packetisation times (ptime) applied at the reception of RTP packets.
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RTP Statistics Configuration
Basic RTP Statistics Concepts
RTP Packet Statistics
The Mediatrix unit collects valuable RTP statistics that can be read via the DGW Web interface, SNMP, and the CLI.
The statistics are sent as Mipt notification messages, therefore the syslog information must be configured properly before setting the statistics (under System/ Event Log/Notification Events). (For more details, refer to the DGW Configuration Guide - Generating Logs published on the Media5 Documentation Portal).
- statistics for the last 10 connections (under Media/RTP Statistics/Connection Statistics)
- statistics for the last 10 collection periods (under Media/RTP Statistics/Collection Period Statistics)
Statistic | Connection Statistics | Collection Period Statistics |
---|---|---|
Octets Tx | Number of octets transmitted during the connection. | Number of octets transmitted during the collection period. This value is obtained by cumulating the octets transmitted in all connections that were active during the collection period. |
Octets Rx | Number of octets received during the connection. | Number of octets received during the collection period. This value is obtained by cumulating the octets received in all connections that were active during the collection period. |
Packets Tx | Number of packets transmitted during the connection. | Number of packets transmitted during the collection period. This value is obtained by cumulating the packets transmitted in all connections that were active during the collection period. |
Packets Rx | Number of packets received during the connection. | Number of packets received during the collection period. This value is obtained by cumulating the packets received in all connections that were active during the collection period. |
Packets Lost | Number of packets lost during the connection. This value is obtained by subtracting the expected number of packets based on the sequence number from the number of packets received. | Number of packets lost during the collection period. This value is obtained by cumulating the packets lost in all connections that were active during the collection period. |
Min. Jitter | Minimum interarrival time, in ms, during the connection. All RTP packets belonging to the connection and received at the RTP level are considered in the calculation. | Minimum interarrival time, in ms, during the collection period. This value is the lowest interarrival jitter for all connections that were active during the collection period. |
Max. Jitter | Maximum interarrival time, in ms, during the connection. All RTP packets belonging to the connection and received at the RTP level are considered in the calculation. | Maximum interarrival time, in ms, during the collection period. This value is the highest interarrival jitter for all connections that were active during the collection period. |
Avg. Jitter | Average interarrival time, in ms, during the connection. All RTP packets belonging to the connection and received at the RTP level are considered in the calculation. | Average interarrival time, in ms, during the collection period. This value is the weighted average of the interarrival jitter for all connections that were active during the collection period. For each connection, the total jitter of packets received during the collection period and the total number of packets received during the collection period are used in the weighted average calculation. |
Min. Latency | Minimum latency, in ms, during the connection. The latency value is computed as one half of the round-trip time, as measured through RTCP. | Minimum latency, in ms, during the collection period. This value is the lowest latency for all connections that were active during the collection period. |
Max. Latency | Maximum latency, in ms, during the connection. The latency value is computed as one half of the round-trip time, as measured through RTCP. | Maximum latency, in ms, during the collection period. This value is the highest latency for all connections that were active during the collection period |
Avg. Latency | Average latency, in ms, during the connection. The latency value is computed as one half of the round-trip time, as measured through RTCP. | Average latency, in ms, during the collection period. This value is the weighted average of the latency for all connections that were active during the collection period. For each connection, the total latency of packets received during the collection period and the total number of packets received during the collection period are used in the weighted average calculation. |
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Channel Statistics
The Mediatrix unit collects valuable RTP statistics that can be read via SNMP and the CLI. Channel statistics are cumulated RTP statistics for all calls using a specific channel of a telephony interface. Channel statistics are updated at the end of each call.
The statistics are associated with the channel in use at the end of the call. In some cases, such as in hold/resume scenarios, the channel assignment may change during a call. This can result in discrepancies between the RTP statistics and the actual usage of the telephony interface.
Parameter | Description |
---|---|
PacketsSent | Number of packets transmitted on the channel since service start. This value is obtained by cumulating the packets transmitted in all the connections that ended during the collection period. |
PacketsReceived | Number of packets received on the channel since service start. This value is obtained by cumulating the packets received in all the connections that ended during the collection period. |
BytesSent | Number of bytes transmitted on the channel since service start. This value is obtained by cumulating the bytes transmitted in all the connections that ended during the collection period. |
BytesReceived | Number of bytes received on the channel since service start. This value is obtained by cumulating the bytes received in all the connections that ended during the collection period. |
AverageReceiveInterarrivalJitter | Average interarrival time, in microseconds, for the channel since service start. This value is based on the average interarrival jitter of each call ended during the collection period. The value is weighted by the duration of the calls. |
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Basic Tasks
Configuring RTP Statistic Collection
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Advanced RTP Statistic Parameters
- using a MIB browser
- using the CLI
- Mipt.ChannelStatistics:Table with Statistics per endpoint/channel since last reboot or statistics reset.
- Mipt.ChannelStatistics.Reset: To reset channel statistics values of a specific endpoint to zero: Mipt.ChannelStatistics[EpChannelId=<endpoint>].Reset=Reset For example: Mipt.ChannelStatistics[EpChannelId=FXS1].Reset=Reset
- Mipt.StatsCollectionPeriodDuration: Specifies the collection period duration (in minutes).
- Mipt.StatsPerConnectionNotificationEnable: Enables the generation of connection end statistics notification.
- Mipt.StatsPerPeriodNotificationEnable: Enables the generation of period statistics notification.
- Mipt.LastConnectionsStats: Table with Last 10 connections statistics.
- Mipt.LastPeriodsStats:Table with Last 10 periods statistics.
- Mipt.MinSeverity: To set the minimal severity to issue a notification message incoming from this service.
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Misc
Basic Media Misc Concepts
Jitter Buffer
Because of network congestion, timing drift, or route changes, data packets can arrive at different intervals and pace. These variations in packet arrival time are called jitter which can be the cause of sound distortion. To insure a clear connection, incoming voice data packets are intentionally delayed by a jitter buffer and then sent in evenly spaced intervals to the voice processor.
- If the voice is scattered, try to increase the maximum jitter buffer value.
- If the delay in the voice path (end to end) is too long, you can lower the target jitter value, ONLY if the end-to-end delay measured matches the target jitter value
For instance, if the target jitter value is 50 ms, the maximum jitter is 300 ms and the delay measured is 260 ms, it would serve nothing to reduce the target jitter.
However, if the target jitter value is 100 ms and the measured delay is between 100 ms and 110 ms, then you can lower the target jitter from 100 ms to 30 ms.
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Dual Tone Multi Frequency (DTMF)
DTMF (dual tone multi frequency) is the signal that is generated from a touch key of a phone and that is sent to the phone company.
Key digit | Low frequency (Hz) | High Frequency (Hz) |
---|---|---|
1 | 697 | 1209 |
2 | 697 | 1336 |
3 | 697 | 1477 |
4 | 770 | 1209 |
5 | 770 | 1336 |
6 | 770 | 1477 |
7 | 852 | 1209 |
8 | 852 | 1336 |
9 | 852 | 1477 |
0 | 941 | 1209 |
* | 941 | 1336 |
# | 941 | 1477 |
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DTMF Out-of-Band Transport Method
When using the Out-of-Band transport method, the DTMFs and the voice are transmitted in two different streams where the voice is sent over RTP, but DTMF is sent either in RTP or SIP depending on the chosen transport method (Out-of-Band using RTP or Out-of-Band using SIP. However, the Out-of-Band transport method can only be used if the SIP peer supports the method, otherwise the DTMF transport method falls back to In-band.
Because some compression codecs such as G.723.1 and G.729 effectively distort voice because they lose information from the incoming voice stream during the compression and decompression phases. For normal speech this is insignificant and becomes unimportant. In the case of pure tones (such as DTMF) this distortion means the receiver may no longer recognize the tones. The solution is to send this information as a separate packet to the other endpoint, which then plays the DTMF sequence back by regenerating the true tones. Such a mechanism is known as Out-of-Band DTMF. The Mediatrix unit receives and sends Out-of-Band DTMFs as per ITU Q.24. DTMFs supported are 0-9, A-D, *, #.
The DTMF Out-of-Band (using either SIP or RTP) transport method is configurable by endpoint, or can be selected for all the endpoints of the unit. ISDN endpoints are normally configured to use an Out-of-Band transport method for DTMF transmission.
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DTMF In-band Transport Method
This is the most used transport method for DTMFs transmission. In this case, the DTMFs and the voice are transmitted together in the RTP stream.
This transport method is only reliable with G.711 or G.729 codecs. The DTMF In-band transport method is configurable by endpoint, or the same method can be selected for all the endpoints of the unit. In general, FXS, FXO, R2, and E&M endpoints are configured to use the In-band transport method for DTMF transmission.
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Basic Media Misc Tasks
Configuring the Jitter Buffer of all Endpoints
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose Default.
-
In the Jitter Buffer
table, from the Level
selection list choose one of the following
- Normal
- Optimize Quality
- Optimize Latency
- Click Apply.

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Configuring the Jitter Buffer of a Specific Endpoint
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose the endpoint for which the jitter buffer need to be configured.
- In the Jitter Buffer table, from the Endpoint Specific selection list, choose Enable.
-
From the Level
selection list choose one of the following:
- Normal
- Optimize Quality
- Optimize Latency
- Click Apply.
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Configuring the Jitter Buffer of all Endpoints for Fax/Modem
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose Default.
- In the Jitter Buffer table, from the Level selection list, choose Fax/Modem.
- Click Apply.

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Configuring the Jitter Buffer of a Specific Endpoint for Fax/Modem
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose the endpoint you wish to configure for fax or modem.
- In the Jitter Buffer table, from the Endpoint Specific selection list, choose Enable.
- From the Level selection list, choose Fax/Modem.
- Click Apply.

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Customising the Jitter Buffer of all Endpoints
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose Default.
- In the Jitter Buffer table, from the Level selection list, choose Custom.
- Complete the fields of the Voice Call and Data Call sections.
- Click Apply.

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Customising the Jitter Buffer of a Specific Endpoint
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose an endpoint.
- In the Jitter Buffer table, from the Endpoint Specific selection list, choose Enable.
- From the Level selection list, choose Custom.
- Complete the fields of the Voice Call and Data Call sections.
- Click Apply.

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Configuring Machine Detection
- Go to Media/Misc.
- Set the CNG Tone Detection selection list to Enable.
- Set the CED Tone Detection selection list to Enable.
- Set the V.21 Modulation Detection selection list to Enable.
- Set the Behavior On CED Tone Detection selection list to Fax Mode.
- Click Apply.

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Configuring DTMF Transport for all Endpoints
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose Default.
-
From the Transport Method drop down
list, choose the transport method set in the VoIP server.
- In-band
- Out-of-Band using RTP, (RFC2833). This is usually the preferred method. Remember to set the payload type if different (e.g.: 96, 101 or 110 in some cases)
- Out-of-Band using SIP. For Cisco or Avaya systems, from the SIP Transport Method field use Info DTMF Relay. For legacy Nortel and others, leave Draft Choudhuri SIP Info Digit 00.
- Signaling Protocol Dependent choose this method if unsure. It will try to use the method negotiated by the VoIP server.
- Click Apply.

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Configuring DTMF Transport for a Specific Endpoint
- Go to Media/Misc.
- From the Select Endpoint drop down list, choose the endpoint for which you wish to configure DTMF transport.
- In the DTMF Transport table, from the Endpoint Specific drop down list, choose Enable.
-
From the Transport Method drop down
list, choose the transport method set in the VoIP server.
- In-band
- Out-of-Band using RTP, (RFC2833). This is usually the preferred method. Remember to set the payload type if different (e.g.: 96, 101 or 110 in some cases)
- Out-of-Band using SIP. For Cisco or Avaya systems, from the SIP Transport Method field use Info DTMF Relay. For legacy Nortel and others, leave Draft Choudhuri SIP Info Digit 00.
- Signaling Protocol Dependent choose this method if unsure. It will try to use the method negotiated by the VoIP server.
- Click Apply.

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Selecting the Base Port for T.38 Fax Calls

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Advanced Jitter Buffer Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
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Advanced DTMF Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
DTMF Detection
- To set the Rise Time criteria: Telif.InteropDtmfDetectionRiseTimeCriteria
- To set the Detection Positive Twist: TelIf.InteropDtmfDetectio.PositiveTwist
- To set the Detection Negative Twist: TelIf.InteropDtmfDetection.NegativeTwist
- To set the Max Power Threshold: TelIf.InteropDtmfDetection.MaxPowerThreshold
- To set the Min Power Threshold: TelIf.InteropDtmfDetection.MinPowerThreshold
- To set the Detection Break Power Threshold: TelIf.InteropDtmfDetection.BreakPowerThreshold
Using the Payload Type Found in the Answer
- To use the payload type found in the answer:SipEp.InteropUseDtmfPayloadTypeFoundInAnswer
Initial quantity of RTP packets, only available when using the Out-of-Band using RTP transport method.
- To set the initial quantity of RTP packets: Mipt.InteropDtmfRtpInitialPacketQty
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Telephony
DTMF Maps
Basic DTMF Maps Concepts
DTMF Maps
A DTMF map (also called digit map or dial map) allows you to compare the number users just dialed to a pattern string. Depending on the DTMF maps, the call can be accepted or rejected.
- If the dialed number matches an entry of the Allowed DTMF Map, the call is accepted.
- If the dialed number matches an entry of the Refused DTMF Map table, then the call is refused and an error tone i.e. reorder tone, network congestion, or fast busy tone is played.
- If the dialed number matches no entry of either the Allowed DTMF Map or the Refused DTMF Map, the call is refused and an error tone i.e. reorder tone, network congestion, or fast busy tone is played.
- If the dialed number matches an entry of both the Allowed DTMF Map and the Refused DTMF Map, the call is refused and
an error tone i.e. reorder tone, network congestion, or fast busy tone is played.Note: The Refused DTMF Map table has precedence over the Allowed DTMF Map table.
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DTMF Maps Order
The order in which the DTMF maps are used is very important. You must use specific DTMF maps first and then more generic ones. Otherwise you may end up with all calls being directed the same way. This is true when creating the DTMF maps in the Allowed DTMF Map table and in the DTMF Map field of this table.
- all employees have a 3-digit extension all starting with 1, 2 or 3.
- All managers have an extension finishing with 9
[1-3]x9
[1-3]xx
calls
made to a managers, will always be redirected to the Manager's assistant extension. [1-3]xx
[1-3]x9
managers will always
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DTMF Map vs DTMF Map Transformation
When creating DTMF maps to allow or refuse calls, the DTMF maps must be written and a transformation to be applied to the dialed number may be required.
The DTMF map is used to recognise the dialed numbers and the transformation modifies the dialed numbers before sending the call. A transformation is not mandatory, it depends if the dialed number must be changed or not before the call is made.
For example, when the dialed number is the extension of a colleague, no transformation is required. The DTMF map will recognise that the dialed numbers are the allowed extension of a colleague, and the call will be made with the dialed number.
However, in the example where calls made to management extensions are redirected to the management's assistant, a transformation is required. For example, if the DTMF is xx9 and the transformation 123, this means that whenever an extension finishing by 9 is dialed, the call is sent to extension 123.
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DTMF Maps Syntax
The permitted DTMF map syntax is taken from the core MGCP specification, RFC 2705, section 3.4:
DigitMap = DigitString / '(' DigitStringList ')'
DigitStringList = DigitString 0*( '|' DigitString )
DigitString = 1*(DigitStringElement)
DigitStringElement = DigitPosition ['.']
DigitPosition = DigitMapLetter / DigitMapRange
DigitMapLetter = DIGIT / '#' / '*' / 'A' / 'B' / 'C' / 'D' / 'T'
DigitMapRange = 'x' / '[' 1*DigitLetter ']'
DigitLetter ::= *((DIGIT '-' DIGIT ) / DigitMapLetter)
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DTMF Maps Special Characters
DTMF maps use specific characters and digits in a particular syntax
Character | Use |
---|---|
Digits (0, 1, 2... 9) | Indicates specific digits in a telephone number expression. |
T | 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. |
x | Matches any digit, excluding “#” and “*”. |
| | Indicates a choice of matching expressions (OR). Note: Enclose the DTMF map in
parenthesis when using the “|” option. |
. | Matches an arbitrary number of occurrences of the preceding digit, including 0. |
[ | Indicates the start of a range of characters. |
] | Indicates the end of a range of characters. |
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Basic DTMF Maps Tasks
Configuring Endpoint-Specific DTMF Timers
- Go to Telephony/DTMF Maps.
- In the General Configuration table, click Edit Endpoints.
- In the Endpoint Specific table, for each Endpoint to configure, from the Override selection list, choose Enable.
- For each enabled endpoint, complete the timeout fields as required.
- Click Apply
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Creating a DTMF Map that Applies to all Endpoints
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Creating a DTMF Map that Applies to Specific Endpoints
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DTMF Maps Examples
Default DTMF Maps

- x.# indicates to dial, right away, a number that finishes by a #. The transformation indicates to not dial the #.
- x.T indicates to dial the number if not digit is dialed after 3 seconds.
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Configuring a DTMF Map to Use a Timer to Make a Call

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Configuring a DTMF Map to make a Call with an Extension
- Go to Telephony/DTMF Maps.
- Make sure the DTMF Map selection list, is set to Allowed.
- In the Allowed DTMF Map table, from the Enable selection list, choose, Enable.
- From the Apply to selection list, choose Endpoint.
- From the Suggestion selection list, choose the endpoints for which you wish to associate a DTMF map.
- in the DTMF Map field enter [2-4]xx
- Click Apply
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Configuring a DTMF Map to make an Internal (extension) or External Call
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Configuring a DTMF Map to Force the Use of the # OR * to Dial a Number

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Configuring a DTMF Map to Make a Call Outside the Country

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Basic DTMF Transformations
DTMF Transformations will modify the dialed number. For example, let's say that the dialed number is 18195551111#.
Action | Transformation | Result |
---|---|---|
Add the prefix “0” to the dialed number | 0x | 018195551111# |
Remove the suffix “#” from the dialed number | x{#} | 018195551111 |
Remove the first four DTMFs from the dialed number | (4)x | 5551111# |
Remove the international code and termination and replace the area code by another one | (1){819}514x{#} | 5145551111 |
Replace the signalled DTMFs by a extension "123" | 123 | 123 |
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Call Forward
Basic Call Forward Tasks
Configuring Call Forward on Busy
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Configuring Call Forward on No Answer
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Configuring Call Forward Unconditional
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Telephony Services
Basic Telephony Services Concepts
Default vs Specific Endpoint Configuration
It is possible to configure all endpoints to the default value, or an endpoint to a specific value, or a mix of both.
- Default configuration
- Mix of default and specific endpoint configurations
- All endpoints enabled with specific configurations
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Automatic Call
The automatic call feature allows you to define a telephone number that is automatically dialed when taking the handset off hook. When this service is enabled, the second line service is disabled but the call waiting feature is still functional. The user can still accept incoming calls.
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Call Completion
The call completion service allows you to configure the Completion of Calls on No Reply (CCNR) and Completion of Calls to Busy Subscriber (CCBS) features. CCBS allows a caller to establish a call with a “busy” callee as soon as this callee is available to take the call. It is implemented by monitoring the activity of a UA and look for the busy-to-idle state transition pattern. CCNR allows a caller to establish a call with an “idle” callee right after this callee uses his phone. It is implemented by monitoring the activity of a UA and look for the idle-busy-idle state transition pattern. The information about the call completion is not kept after a restart of the EpServ service. This includes the call completion activation in the Pots service and the call completion monitoring in the SipEp service.
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Call Waiting
The call waiting tone indicates to an active call that a new call is waiting on the second line. Users can activate/deactivate the call waiting tone for their current call. This is especially useful when transmitting faxes. The user that is about to send a fax can thus deactivate the call waiting tone to ensure that the fax transmission will not be disrupted by an unwanted second call. When the fax transmission is completed and the line is on-hook, the call waiting tone is automatically reactivated.
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Basic Telephony Services Tasks
TelIf Default vs Specific Endpoint Configuration Examples
Here are some examples from the command line interface CLI for default parameters, followed by specific endpoint configuration.
Global>TelIf.DefaultCountryCustomizationUserGainInputOffset
0
Global>TelIf.DefaultCountryCustomizationUserGainOutputOffset
0
Global>TelIf.SpecificCountryCustomizationUserGain
_____________________________________________________________
| InterfaceId | EnableConfig | InputOffset | OutputOffset |
|______________|_______________|______________|_______________|
| Slot1/E1T1 | Disable | 0 | 0 |
| Slot2/E1T1 | Disable | 0 | 0 |
| Slot3/E1T1 | Disable | 0 | 0 |
| Slot4/E1T1 | Disable | 0 | 0 |
| Slot5/FXS1 | Disable | 0 | 0 |
| Slot5/FXS2 | Disable | 0 | 0 |
| Slot5/FXS3 | Disable | 0 | 0 |
| Slot5/FXS4 | Disable | 0 | 0 |
| Slot6/FXO1 | Disable | 0 | 0 |
| Slot6/FXO2 | Disable | 0 | 0 |
| Slot6/FXO3 | Disable | 0 | 0 |
| Slot6/FXO4 | Disable | 0 | 0 |
| Slot7/FXS1 | Disable | 0 | 0 |
| Slot7/FXS2 | Disable | 0 | 0 |
| Slot7/FXS3 | Disable | 0 | 0 |
| Slot7/FXS4 | Disable | 0 | 0 |
| Slot8/FXO1 | Disable | 0 | 0 |
| Slot8/FXO2 | Disable | 0 | 0 |
| Slot8/FXO3 | Disable | 0 | 0 |
| Slot8/FXO4 | Disable | 0 | 0 |
|______________|_______________|______________|_______________|
Command examples with their results.
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot1/E1T1].EnableConfig = Enable
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot1/E1T1].InputOffset = -6
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot1/E1T1].OutputOffset = -8
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot5/FXS3].EnableConfig = Enable
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot5/FXS3].InputOffset = -3
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot5/FXS3].OutputOffset = -3
Global>TelIf.SpecificCountryCustomizationUserGain
_____________________________________________________________
| InterfaceId | EnableConfig | InputOffset | OutputOffset |
|______________|_______________|______________|_______________|
| Slot1/E1T1 | Enable | -6 | -8 |
| Slot2/E1T1 | Disable | 0 | 0 |
| Slot3/E1T1 | Disable | 0 | 0 |
| Slot4/E1T1 | Disable | 0 | 0 |
| Slot5/FXS1 | Disable | 0 | 0 |
| Slot5/FXS2 | Disable | 0 | 0 |
| Slot5/FXS3 | Enable | -3 | -3 |
| Slot5/FXS4 | Disable | 0 | 0 |
| Slot6/FXO1 | Disable | 0 | 0 |
| Slot6/FXO2 | Disable | 0 | 0 |
| Slot6/FXO3 | Disable | 0 | 0 |
| Slot6/FXO4 | Disable | 0 | 0 |
| Slot7/FXS1 | Disable | 0 | 0 |
| Slot7/FXS2 | Disable | 0 | 0 |
| Slot7/FXS3 | Disable | 0 | 0 |
| Slot7/FXS4 | Disable | 0 | 0 |
| Slot8/FXO1 | Disable | 0 | 0 |
| Slot8/FXO2 | Disable | 0 | 0 |
| Slot8/FXO3 | Disable | 0 | 0 |
| Slot8/FXO4 | Disable | 0 | 0 |
|______________|_______________|______________|_______________|
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot3/E1T1].InputOffset = -1
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot3/E1T1].OutputOffset = +1
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot7/FXS3].InputOffset = +2
TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot7/FXS3].OutputOffset = -4
Global>TelIf.SpecificCountryCustomizationUserGain
_____________________________________________________________
| InterfaceId | EnableConfig | InputOffset | OutputOffset |
|______________|_______________|______________|_______________|
| Slot1/E1T1 | Enable | -6 | -8 |
| Slot2/E1T1 | Disable | 0 | 0 |
| Slot3/E1T1 | Disable | -1 | 1 |
| Slot4/E1T1 | Disable | 0 | 0 |
| Slot5/FXS1 | Disable | 0 | 0 |
| Slot5/FXS2 | Disable | 0 | 0 |
| Slot5/FXS3 | Enable | -3 | -3 |
| Slot5/FXS4 | Disable | 0 | 0 |
| Slot6/FXO1 | Disable | 0 | 0 |
| Slot6/FXO2 | Disable | 0 | 0 |
| Slot6/FXO3 | Disable | 0 | 0 |
| Slot6/FXO4 | Disable | 0 | 0 |
| Slot7/FXS1 | Disable | 0 | 0 |
| Slot7/FXS2 | Disable | 0 | 0 |
| Slot7/FXS3 | Disable | 2 | -4 |
| Slot7/FXS4 | Disable | 0 | 0 |
| Slot8/FXO1 | Disable | 0 | 0 |
| Slot8/FXO2 | Disable | 0 | 0 |
| Slot8/FXO3 | Disable | 0 | 0 |
| Slot8/FXO4 | Disable | 0 | 0 |
|______________|_______________|______________|_______________|
Global>TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot3/E1T1].EnableConfig
Disable
Global>TelIf.SpecificCountryCustomizationUserGain[InterfaceId=Slot7/FXS3].EnableConfig
Disable
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Configuring Telephony Services for all Endpoints
- Go to Telephony/Services.
- From the Select Endpoint drop-down menu, select Default.
- Complete the fields as required.
- Click Apply.
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Configuring Telephony Services for a Specific Endpoint
- Go to Telephony/Services.
- From the Select Endpoint drop-down menu, select the endpoint to configure.
- From the Endpoint Specific drop-down menu, select Yes.
- Complete the remaining fields as required.
- Click Apply.
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Configuring Hook Flash Processing
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Configuring Automatic Calls
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Configuring Call Completion
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Configuring Call Transfer
Top
Configuring Call Waiting
Top
Configuring Conference
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Configuring Delayed Hotline
Top
Configuring Direct IP Address Call
Top
Configuring Hold
Top
Configuring Second Call
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Tone Customisation
Basic Tone Customisation Tasks
Identifying the Tone Pattern
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Overriding the Pattern of a Tone
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Music on Hold
Basic Music on Hold Concepts
MP3 File Upload
You can only upload MP3 music files. Before you start, you must choose to upload the MP3 file either from the Server (this file will be loaded every time the unit restarts) or from the Web Browser (loaded once).
To load the music file from a server, refer to Importing an MP3 File from a Server. To load the music on your unit, refer to Importing an MP3 File from a Web Browser.
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MP3 File Requirements
The MP3 file must be encoded following the MPEG 2.5 Audio Layer III specifications:
- Sample Rate: 8kHz
- Mono
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Basic Music on Hold Tasks
Enabling Music on Hold
- Go to Telephony/Music on Hold.
- From the Music on Hold Configuration table, from the Streaming drop-down menu, select Enable.
- Click Apply.
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Importing an MP3 File from a Server
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Misc
Basic Misc Telephony Concepts
Call Detail Record (CDR)
Call detail record (CDR) in VoIP contains information about recent system usage such as:
- the identities of sources (points of origin)
- the identities of destinations (endpoints)
- the duration of each call
- the total usage time in the billing period
- etc.
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Call Detail Record Example

Syslog message: LOCAL0.INFO: 1;1141399338;1140175242;37;16;200;2017;07;07;15;00;20
- 1:CDR ID (internal)
- 1141399338: calling number (before being transformed by the Call Router)
- 1140175242: called number (before being transformed by the Call Router)
- 37: call duration (seconds)
- 16: reason for ISDN disconnection
- 200: reason for SIP disconnection
- 2017: year
- 07: month
- 07: day
- 15: hour
- 00: minute
- 20: second
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Formal Protocol Syntax Description
- Macro=%[Width]|[.Precision]|MacroId
- Width=DIGIT
- Precision=DIGIT
- CDR Log : %sipid --> CDR Log : SipUser001
- CDR Log : %15sipid --> CDR Log : SipUser001
- CDR Log : %15.5sipid --> CDR Log : SipUs
- CDR Log : %.5sipid --> CDR Log : SipUs
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Call Detail Record Macros and Control Characters
- %% : %
- \n : Split message
- %id : CDR ID. The CDR ID is unique. The ID is incremented by one each time it is represented in a CDR record.
- %sipid : SIP call ID. Blank if no SIP interface was used during the call.
- %ocgnum : Original calling number. Calling number as received by the unit.
- %cgnum : Calling number. Calling number after manipulation by the call router.
- %ocdnum : Original called number. Called number as received by the unit.
- %cdnum : Called number. Called number after manipulation by the call router.
- %oiname : Origin interface name. Interface on which the call was received. Ex. isdn-Slot2/Pri1.
- %diname : Destination interface name. Interface on which the call was relayed. Ex. SIP-Default.
- %chan : Channel number. Blank if no PRI/BRI interface was used during the call. If 2 PRI/BRI interfaces were involved, display the originating interface.
- %sipla : SIP local IP address.
- %sipra : SIP remote IP address or FQDN (next hop).
- %siprp : SIP remote port (next hop).
- %mra : Media remote IP address. Source IP address of the incoming media stream. If the stream was modified during the call, display the last stream.
- %mrsp : Media remote source port. Source port of the incoming media stream. If the stream was modified during the call, display the last stream.
- %mrp : Media remote port. Destination port of the outgoing media stream. If the stream was modified during the call, display the last stream.
- %tz : Local time zone
- %cd : Call duration (in seconds) (connect/disconnect).
- %sd : Call duration (in seconds) (setup/connect).
- %pdd : Post dial delay (in seconds) (setup/progress).
- %css : Call setup second (local time)
- %csm : Call setup minute (local time)
- %csh : Call setup hour (local time)
- %csd : Call setup day (local time)
- %csmm : Call setup month (local time)
- %csy : Call setup year (local time)
- %ccs : Call connect second (local time)
- %ccm : Call connect minute (local time)
- %cch : Call connect hour (local time)
- %ccd : Call connect day (local time)
- %ccmm : Call connect month (local time)
- %ccy : Call connect year (local time)
- %cds : Call disconnect second (local time)
- %cdm : Call disconnect minute (local time)
- %cdh : Call disconnect hour (local time)
- %cdd : Call disconnect day (local time)
- %cdmm : Call disconnect month (local time)
- %cdy : Call disconnect year (local time)
- %miptxc : IP Media last transmitted codec
- %miptxp : IP Media last transmitted p-time
- %dr : Call disconnect reason, expressed as a Q.850 cause. The Q.850 codes are used to represent the disconnect cause no matter what type of interface initiated the disconnect (SIP, FXS, ISDN, ...). A value of 0 means that no cause code is available.
- %rxp : Received media packets. Excluding T.38.
- %txp : Transmitted media packets. Excluding T.38.
- %rxpl : Received media packets lost. Excluding T.38.
- %rxmd : Received packets mean playout delay (ms, 2 decimals). Excluding T.38.
- %rxaj : Received packets average jitter (ms, 2 decimals). Excluding T.38.
- %sipdr : SIP status code of the received SIP response that caused the disconnect or rejection. A value of 0 means that no status code is available.
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Basic Misc Telephony Setting Tasks
Configuring the Country for a Specific Endpoint
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Configuring the Country for all Endpoints
- Go to Telephony/Misc.
- Make sure the Select Endpoint drop-down list is set to Default.
- In the Country table, select your country from the drop-down list.
- Click Apply.

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Configuring the User Gain for all Endpoints
Top
Configuring the User Gain for Specific Endpoints
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Starting the CDR Service
- Go to System/Services.
-
From the User Service
table, on the Call Detail Record (CDR)
line, click
.
- Click Apply.
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Configuring the CDR
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Advanced Telephony Parameters
User Gain Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration variables
Parameters |
---|
TelIf.DefaultCountryCustomizationUserGainInputOffset |
TelIf.DefaultCountryCustomizationUserGainOutputOffset |
TelIf.SpecificCountryCustomizationUserGain[].EnableConfig |
TelIf.SpecificCountryCustomizationUserGain[].InputOffset |
TelIf.SpecificCountryCustomizationUserGain[].InterfaceId |
TelIf.SpecificCountryCustomizationUserGain[].OutputOffset |
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Advanced Dialing Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration variables
Parameters |
---|
TelIf.DefaultCountryCustomizationDialingDtmfDuration |
TelIf.DefaultCountryCustomizationDialingInterDtmfDialDelay |
TelIf.DefaultCountryCustomizationDialingInterMfR1DialDelay |
TelIf.DefaultCountryCustomizationDialingMfR1Duration |
TelIf.DefaultCountryCustomizationDialingOverride |
TelIf.SpecificCountryCustomizationDialing[].DtmfDuration |
TelIf.SpecificCountryCustomizationDialing[].EnableConfig |
TelIf.SpecificCountryCustomizationDialing[].InterDtmfDialDelay |
TelIf.SpecificCountryCustomizationDialing[].InterMfR1DialDelay |
TelIf.SpecificCountryCustomizationDialing[].MfR1Duration |
TelIf.SpecificCountryCustomizationDialing[].Override |
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Fax Calling Tone Detection Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration variables
Parameters |
---|
TelIf.DefaultMachineDetectionBehaviorOnCedToneDetection |
TelIf.DefaultMachineDetectionCedToneDetection |
TelIf.DefaultMachineDetectionCngToneDetection |
TelIf.DefaultMachineDetectionV21ModulationDetection |
TelIf.SpecificMachineDetection[].BehaviorOnCedToneDetection |
TelIf.SpecificMachineDetection[].CedToneDetection |
TelIf.SpecificMachineDetection[].CngToneDetection |
TelIf.SpecificMachineDetection[].EnableConfig |
TelIf.SpecificMachineDetection[].InterfaceId |
TelIf.SpecificMachineDetection[].V21ModulationDetection |
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Testing Telephony Services
Testing the Hold Service
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- A calls B.
- Hook flash from A.
- Verify that B is on hold.
- Hook flash again from A.
- Verify that voice path is back.
- Release the call.
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Testing the Call Waiting Service
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- A calls B.
- Verify the voice path.
- C calls A.
- Verify that A hears the stutter dial tone.
- Hang up all endpoints.
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Testing the Second Call Service
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- Place a call between A and B.
- Hook flash on A.
- A initiates a « Second Call » to C.
- Verify that the second voice path (between A and C) is established.
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Testing the Blind Transfer Service
- A calls B.
- When the call is established, hook flash on B.
- Wait for the second dial tone on B.
- B calls C.
- Wait for the ring back in B and then hang up B.
- C should ring in the mean time.
- You should now hear a ring back in A.
- C answers the call.
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Testing the Attended Transfer
- A calls B.
- When the call is established, hook flash on B.
- Wait for the second dial tone on B.
- B calls C.
- Answer C when it is ringing.
- Verify voice path between B and C.
- Hang up B.
- The call should be transferred.
- Call is established between A and C.
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Testing the Conference Call Service
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- Place a call between A and B.
- Hook flash on A.
- A initiates a « Second Call » to C .
- Once the voice path between A and C is established, hook flash again on A to start the conference between A, B and C.
- Verify that a 3 way voice path is established.
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Testing the Automatic Call Service
- Go to Telephony/Services.
- In the Services Configuration table, enable the Automatic Call Activation on endpoint A.
- Set the Automatic Call Target to 456 (endpoint B).
- Click Apply.
- Pick up telephone A.
- A call between A and B should be automatically established.

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Testing the Delayed Hotline Call Service
- Go to Telephony/Services.
- In the Services Configuration table, under the Delayed Hotline section, enable the Delayed Hotline Activation on endpoint A.
- Set the Delayed Hotline Condition to FirstDTMFTimeout.
- Set the Delayed Hotline Target to 456 (endpoint B).
- Click Apply.
- Pick up phone A and wait for the First DTMF timer expiration.

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Configuring the Call Forward Service for Testing
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- Go to Telephony/Services
- In the Call Forward Unconditional section, set the Allow Activation via Handset to Enable on endpoint A.
- Set the DTMF Map Activation field to *54.
- Set the DTMF Map Deactivation field to *55.
- Set the Activationfield to Active.
- Set the Forwarding Address to 456 (endpoint B).
- Click Apply.
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Testing the Call Forward Service
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Configuring the CCBS Call Completion Service for Testing
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- Go to Telephony/Services.
- In the Services Configuration table, under theCall Completion section, set the Allow CCBS Activation via Handset to Enable on endpoint A.
- Set the CCBS DTMF Map Activation field to *98.
- Set the CCBS DTMF Map Deactivation field to *99.
- Click Apply.
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Testing the CCBS Call Completion Service
- Endpoint A: 123
- Endpoint B: 456
- Endpoint C: 789
- Make a call from B to C.
- Once the call is established between B and C, call B from A.
- A should hear the busy tone.
- Dial *98 to activate the CCBS service.
- Hang up A.
- Hang up B and C.
- Wait a few seconds.
- A will hear a special ring.
- Pick up A and the call will be completed with B.
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Testing the Direct IP Call Service
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Default Ports Used by DGW
Connection Type | Default Port Number | Transport Protocol |
---|---|---|
Debug Signaling Log Host | 6000 | UDP |
DHCP | 68 | UDP |
FTP | 21 | TCP |
HTTP | 80 | TCP |
HTTPS | 443 | TCP |
Persistent TLS Base Port | 16000 | TCP |
Radius default port for accounting | 1813 | TCP |
Radius default port for authentication | 1812 | TCP |
RTP (including RTCP) | Range starting from 5004 1 | UDP |
Secure SIP | 5061 | TCP |
SIP | 5060 | UDP |
SNMP Listening | 161 | UDP |
SNMP Trap | 162 | UDP |
SNTP | 123 | UDP |
SRTP (including SRTCP) | Range starting from 5004 1 | UDP |
SSH | 22 | TCP |
Syslog | 514 | UDP |
T.38 | 6004 | UDP |
Telnet | 23 | TCP |
TFTP | 69 | UDP |
|
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Calculating the RTP Port Range of a Mediatrix Unit
To calculate the port range of a Mediatrix unit, use the Amount of Ports to Reserve and the Default Ports Used by DGW tables.
RTP (including RTCP) | (Base RTP port) to (Base RTP port + amount of ports to reserve for all the telephony ports of the unit) |
SRTP (including SRTCP) | (Base SRTP port) to (Base SRTP port + amount of ports to reserve for all the telephony ports of the unit) |
T.38 | (Base T.38 port) to (Base T.38 port + amount of ports to reserve for all the telephony ports of the unit) |
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Amount of Ports to Reserve
Telephony ports | Amount of Ports to Reserve for | |
---|---|---|
RTP/SRTP | T.38 | |
2 FXS | 16 | 2 |
4 FXS | 24 | 4 |
8 FXS | 40 | 8 |
16 FXS | 80 | 16 |
24 FXS | 112 | 24 |
28 FXS | 128 | 28 |
2 FXO | 16 | 2 |
4 FXO | 24 | 4 |
8 FXO | 40 | 8 |
16 FXO | 80 | 16 |
24 FXO | 112 | 24 |
28 FXO | 128 | 28 |
1 E1 | 136 | 30 |
2 E1 | 272 | 60 |
3 E1 | 408 | 90 |
4 E1 | 544 | 120 |
5 E1 | 680 | 150 |
6 E1 | 816 | 180 |
7 E1 | 952 | 210 |
8 E1 | 1088 | 240 |
1 T1 | 108 | 23 |
2 T1 | 216 | 46 |
3 T1 | 324 | 69 |
4 T1 | 432 | 92 |
5 T1 | 540 | 115 |
6 T1 | 648 | 138 |
7 T1 | 756 | 161 |
8 T1 | 864 | 184 |
1 BRI | 16 | 2 |
2 BRI | 24 | 4 |
4 BRI | 40 | 8 |
8 BRI | 80 | 16 |
16 BRI | 144 | 32 |
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Examples of Port Range Calculations
Mediatrix G7 with 2 E1 and 4 FXS ports
Voice in RTP/SRTP | T.38 | |
---|---|---|
Ports reserved for 2 E1 | 272 | 60 |
Ports reserved for 4 FXS | 24 | 4 |
Base port | 5004 | 6004 |
Upper port | 5004 + 272 + 24 = 5300 | 6004 + 60 + 4 = 6068 |
Port Range | 5004 to 5300 | 6004 to 6068 |
Sentinel 400 with 2 T1, 4 FXS , and 4 FXO ports
Voice in RTP/SRTP | T.38 | |
---|---|---|
Ports reserved for 2 T1 | 216 | 46 |
Ports reserved for 4 FXS | 24 | 4 |
Ports reserved for 4 FXO | 24 | 4 |
Base port | 5004 | 6004 |
Upper port | 5004 + 216 + 24 + 24 = 5268 | 6004 + 46 + 4 + 4 = 6058 |
Port Range | 5004 to 5268 | 6004 to 6058 |
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Call Router
Call Router Basic Concepts
Call Router Configuration Limitations
Configuration Parameter | Maximum Number |
---|---|
Route | 40 |
Signaling Property | 40 |
Transformations | 40 |
Transformation Rules | 100 |
SIP Header Translations | 100 |
Call Property Translation Override | 100 |
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Call Router
The DGW Call Router uses the Call Routing service (CRout) to route calls between a SIP gateway and an endpoint (FXS, FXO, PRI, or BRI).
In some specific cases, the Call Router can also route a call from an endpoint to another endpoint.
- ISDN to ISDN (TDM hairpinning)
- ISDN to SIP
- E&M to E&M (TDM hairpinning)
- E&M to SIP
- R2 to R2 (TDM hairpinning)
- R2 to SIP
- FXS to SIP
- FXO to SIP
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General Call Router Workflow

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Call Router Specific Naming Rules
- a route with "route-", for instance: route-isdn_sip.
- a SIP interface with "sip-", for instance: sip-default.
- an ISDN interface with "isdn-", for instance: isdn-default.
- a hunt with "hunt-", for instance: hunt-trunkLines.
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Routing
Routing is the operation that consists in finding which route to use for a call, and applying the various transformations and translations defined within the route.
- The inbound call leg, for the incoming call that entered the Call Router.
- The outbound call leg, for the outgoing call made to the callee after the processing made by the Call Router.

Step | Description |
---|---|
Incoming Call | A call coming from a telephony interface (FXO, FXS, BRI, or PRI) or from SIP. |
Read Call Properties |
The Call Router reads the call properties of the call's signaling data. Call properties are common to all telephony technologies and contain information that the Call Router will use to route the call. For more details on call properties, refer to the DGW Configuration Guides -Call Properties document published on the Media5 Documentation Portal. |
Find Route | The Call Router sequentially selects, from several routes defined in the
Routes table, the proper
route to use for the call. The route is chosen if the destination is valid and
provided the value of the following criteria are matched with the call properties of
the call:
|
Transform Call Properties (optional) | Transformations applied to call properties are defined in the Transformations and Transformation Rules tables.
Specific, user-defined transformations are applied to a call property criteria.
Transformations can include, for example, to:
For more details on call properties, refer to the DGW Configuration Guides - Call Properties document published on the Media5 Documentation Portal. |
Translate Signalling Properties (optional) | Translations applied to the signaling properties are defined in the Signaling Properties table.
Translations can include, for example, to:
|
Send Call to Destination | When the Call Router has completed its processing, the call is sent to its destination which is the exit point (SIP gateway, PRI interface, FXS interface, etc.) on which the outgoing call leg is created to forward the call to the callee. |
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Source and Destination
A call always has a source and a destination.
A call can go through several destinations before reaching its final destination.
A source and a destination can be:
- an endpoint (FXS, FXO, PRI, or BRI)
- a SIP gateway
- a route
- a hunt
- an endpoint (FXS, FXO, PRI, or BRI)
- a SIP gateway
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Signaling Properties
Signaling properties specify how to set up a call to the destination i.e. either a Mediatrix unit or a third-party equipment.
Signaling properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol level. Signaling properties are applied after the call has been validated against the transformations, if there is a transformation specified in the route. As for the routes, the call is validated against the call signaling properties one after the other until the call matches one signaling property. At this point, the transformation is executed by modifying the behaviour of the call. Up to 40 signalling properties can be added.
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SIP Signaling Properties
SIP signaling properties are signaling properties used on the SIP leg of routing to control several SIP features.
SIP signaling properties are used by the mechanism through which the Call Router controls several SIP features of the calls during the SIP leg of the call. As for call properties, SIP signaling properties are used by the Call Router to apply routing decisions, but on the SIP leg.
SIP signalling properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol level. Once a Route has been chosen, transformations are applied and then SIP signaling properties are applied.
A SIP signaling property transformation overrides the default value of the SIP signaling properties of an incoming SIP message. It modifies the properties before the call is sent to its destination. When a Route is chosen, SIP signaling properties are applied if possible. At this point, the transformation of the SIP signaling properties is performed.
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Routes
Routes are used by the Call Router to send calls from one endpoint to another.
- the source the call must come from for the route to be applied
- the call properties the call must match for the route to be applied i.e. the property criteria
- the criteria rule, i.e. the regular expression applied to call property
- the transformations that will be applied to the call properties i.e. which transformation are used
- the call signaling properties
- the destination of the call
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Route Ordering - Very Important
The order of the routes in the Routes table is important.
The most exclusive routes should be first and the most inclusive routes last, otherwise all calls will be routed by the same route, hence routed to the same destination.
For example, one of the first routes could be for a specific username for which the call would systematically be sent to a specific destination. The last route could route the unmatched calls to the receptionist's extension. If a call cannot be matched to any route, then the call is cancelled and an error message is issued.
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Call Property Transformations
A Call Property Transformation is a mechanism through which the Call Router transforms call properties on the basis of specific criteria. Because this mechanism is applied when routing is processed by the Call Router, it provides a fine level of granularity to act on the signaling data of the call, allowing the user to route calls just about anywhere or to choose what signaling data will be displayed in the Caller Id.
- Defining on which criteria the transformation is based on and which criteria the transformation is applied to, both criteria being a call property, either identical or not. If the value of the Criteria Based On and Criteria Rule fields matches, then the transformation rule is applied to the le call property defined in the Transformation Applies To field.
- Transforming the call property specified in the transformation criteria.
Call property transformations can be used, for example, to:
- Modify or block caller IDs
- Add or remove prefixes to the called number
- Block outgoing international calls
- Send all calls to a specific extension
Call property transformations thus influence routing and/or the setup message leaving the call router. They are specifically called within a route. As for the Routes table, the Transformations table finds the first matching entry. It then executes it by transforming a call property. A transformation always examines one call property and changes another property. The call router executes all transformations that match by following the Transformation Rules table rows as they are entered. If you want the Call Router to try to match one row before another one, you must put that row first.
- calling party call properties
- called party call properties
- generic properties used for call properties that apply to both calling and called parties
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Call Property Transformation Workflow

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Route/Transformation/Next Transformation Recursiveness Example
|-Route 1
| |-Transformation 1
| | |-Next Transformation 1
| | | |-Next Transformation 2
| | | | |-Next Transformation 3
| | | | | |-Next Transformation 4
| | | | | | |-Next Transformation 5
| |-Transformation 2
| | |-Next Transformation 1
| | | |-Next Transformation 2
| | | | |-Next Transformation 3
| |-Transformation x
| |-Transformation 40
|-Route 2
| |-Transformation 1
| |-Transformation 2
| |-Transformation 3
| | |-Next Transformation 1
| | | |-Next Transformation 2
| | | | |-Next Transformation 3
|-Route 3
|-Route x
|-Route 40
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Call Properties
Although telephony technologies (SIP, FXS, ISDN, R2, CAS, etc.) all have the same purpose, i.e. make calls, they all have their own signaling, their own set of conventions, and their own vocabulary. Call properties are the common representation of the basic information pieces (signaling data) that all telephony technologies carry to route calls. The Call Router uses the call properties to apply routing decisions.
Using this single common representation simplifies the way we create routes in the Call Router because we can make abstraction, to a certain point, of the individual telephony technologies.
- Undefined: No value is assigned to the call property
- Defined: A value is assigned to the call property
- Empty: The call property is defined, but its value is empty (e.g. an empty string)
For example, when a call comes from ISDN, a number of call properties are set with the information extracted from the ISDN signaling: Calling Name, Called E164, ... The other Call Properties that are not relevant to ISDN are left undefined: for instance Calling Host and Calling URI, which makes sense only for calls coming from SIP.
For more details on call properties, refer to Call Properties - Incoming Calls and Call Properties - Outgoing Calls sections .
The transformation applied to the call properties can be configured in the Transformations and Transformation Rules tables of the DGW 2.0 Web interface.
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Calling vs Called Call Properties
Many Call Properties exist in a Calling and Called version, with names such as CallingXxx or CalledXxx. This indicates if the Call Property is related to the Calling (incoming) or the Called (outgoing) side of the call.
For example, in a SIP to ISDN call from 987654321@voipprovider.com to 8191234567, the CallingE164 and CalledE164 Call Properties are assigned the 987654321 and 8191234567 values, respectively.
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SIP Call Properties - Source
Call Property | SIP Information |
---|---|
Called URI | Use the URL of the 'To' field as the 'Called URI' Call Property. |
Calling URI | Use the URL of the 'From' field as the 'Calling URI' Call Property. |
Called Name | Use the 'friendly name' Call Property of the 'To' field as the 'Called Name'
Call Property. If there is no 'friendly name', the 'Called Name' Call Property is undefined. |
Calling Name | Use the 'friendly name' Call Property of the 'From' field as the 'Calling Name'
Call Property If there is no 'friendly name' Call Property , the 'Calling Name' Call Property is undefined. |
Called E164 | If the 'username' Call Property of the 'Request-Uri' field is E164 compatible,
then the 'username' Call Property of the 'Request-Uri' field is used as the 'Called
E164' Call Property. The “+” prefix and “-” separator are removed. If there is no 'username' Call Property or if the 'username' Call property is not E164 compatible, then the 'Called E164' Call Property remains empty. |
Calling E164 | If the 'username' Call Property of the 'From' field is E.164 compatible, then
the 'username' Call Property of the 'From' field is used as the 'Calling E164' Call
Property. The “+” prefix and the “-” separator are removed. If there is no 'username' Call Property or if the 'username' Call Property is not E.164 compatible, then the 'Calling E164' Call Property is undefined. |
Called Host | Use the 'Host' Call Property of the 'To' field as the 'Called Host' Call property. |
Calling Host | Use the 'Host' Call Property of the 'Contact' field as the 'Calling Host' Call Property. |
Called TON | If the 'username' Call Property of the 'To' field is E.164 compatible and has
the + sign in front, the 'Called TON' Call Property is 'international'. Otherwise the 'Called TON' Call Property is undefined. |
Calling TON | If the 'user name' Call Property of the 'From' field is E.164 compatible and
has a + sign in front, the 'Calling TON' Call Property is 'international'.
Otherwise the 'Calling TON' Call Property is undefined. |
Called Phone Context | If the 'username' Call Property of the 'phone-context' Call Property of the
'To' field is E.164 compatible, then the 'Called Phone Context' Call Property is the
'phone-context' Call Property of the 'username' Call property of the 'To' field.
Otherwise the 'Called Phone Context' Call Property is undefined. |
Calling Phone Context | If the 'username' Call Property of the 'phone-context' Call Property of the 'From' field is E.164 compatible, then the 'Calling Phone Context' Call Property is the 'phone-context' of the 'username' of the 'To' field. Otherwise the 'Calling Phone Context' Call Property is undefined. |
Called SIP Username | Use the 'username' Call Property of the 'Request-Uri' field as the 'Called SIP
Username Call Property'. Note that this does not include the 'username' Call Properties such as the 'phone-context' . |
Calling SIP Username | Use the 'username' Call Property of the 'From' field as the 'Calling SIP
Username' Call Property Note that this does not include the 'username' Call Property such as the 'phone-context' Call Property. |
Last Diverting Reason | If the INVITE field contains at least one 'Diversion' header, then the value of
the 'Last Diverting Reason' is the 'reason' field of the first 'Diversion' header:
The 'Reason' field is not case sensitive. |
Original Diverting Reason | If the INVITE field contains more than one 'Diversion' header, the 'Original
Diverting Reason' Call Property is the 'Reason' field of the last 'Diversion'
header:
The 'reason' field is not case sensitive. |
Last Diverting E.164 | If the INVITE field contains at least one 'Diversion' header, the value of the
'Last Diverting E.164' Call Property is the 'username' Call Property of the URI (can
be a SIP URI, SIPS URI, or TEL URI) of the first 'Diversion' header converted into
an E.164. If the 'username' Call Property of the URI has no value, or if it is not E.164 compatible, then the 'Last Diverting E.164' Call Property is undefined. |
Original Diverting E.164 | If the INVITE field contains more than one 'Diversion' header, the 'Original
Diverting E.164' is the 'username' Call Property of the URI (can be a SIP URI, SIPS
URI, or TEL URI) of the last 'Diversion' header converted into an E.164. If the 'username' Call Property has no value or is not E.164 compatible, the 'Original Diverting E.164' Call Property is undefined. |
Diverting Counter | If the INVITE field contains at least one 'Diversion' header, the 'Diverting Counter' is the sum of the value of the 'counter' field of all 'Diversion' headers. If a 'Diversion' header does not contain the 'counter' field, the value is assumed to be one for the header. |
All others | The property is undefined. |
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SIP Call Properties - Destination
SIP Field | Description |
---|---|
To |
Use the value of the 'Calling URI' Call Property as the value of the 'To' field. If the 'Calling URI' Call Property has no value, the 'To' field is built with the
following elements:
|
From | If the 'Called URI' Call Property has a value, it is used as the value of the
'From' field. Otherwise, the 'From' field is built with the following elements:
|
Request URI |
Use the value of the 'Calling URI' Call Property as the value of the 'Request URI' field. If the 'Calling URI' Call Property has no value, the 'Request URI' field is built
with the following elements:
|
Contact | If the 'Called URI' Call Property has a value, it is used as the value of the
'Contact' field. Otherwise, the 'Contact' field is built with the following
elements:
|
Diversion | If the 'Last Diverting E.164' Call property has a value, the value of the
'Diversion' header is added. The 'Diversion' header is built with the following
elements:
|
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SIP Header Translations Override
A SIP Header Translation overrides the default value of SIP headers in an outgoing SIP message.
It modifies the SIP headers before the call is sent to its destination. As for the Routes table, the SIP Header Translation Overrides table finds the first matching entry. It then executes it by modifying the behaviour of the call.
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Hunt
A hunt consists in a group of destinations that can be associated with a route.
A hunt is used to make sure that if a destination is not available, that other destinations will be tried to route the call. For example, if the hunt specifies 4 different port destinations, then each destination will be checked for availability to route the call. If the first port is not available, the second port will be checked, and so on.
To use a hunt, the destination of a route must designate the hunt (i.e. the name of an entry in the Hunt table). The hunt name must be unique in the Hunt table.
The behavior of a hunt is defined by:
- A list of destinations to try
- An algorithm defining the sequence in which the hunt is performed. The Hunt configuration must specify which algorithm to use
- A timeout value
- A list of rejection causes that causes the hunt to continue
A maximum of 40 hunts can be created and used.
The execution order of the hunt destinations is defined by the sequential order of the Hunt table, separated by commas. The first destination defined is the first one used.
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Auto-Routing
When the auto-routing feature is enabled, the Call Router can automatically create routes from a source to a destination.
- one directing incoming calls from the associated auto-routing SIP gateway to the endpoint
- one sending outgoing calls from the endpoint to the associated auto-routing SIP gateway
- User: the route has been manually entered
- Auto: this is an auto-routing route.
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TDM Hairpinning and SIP Loopback
TDM Hairpinning is defined as a standard inbound telephony call that is simply routed back out using the same Information Layer 1 Protocol. TDM Hairpinning is only supported between ISDN to ISDN, E&M to E&M or R2 to R2 endpoints and needs both calls to use the same codec.
However, a different approach, SIP loopback, is possible between two telephony interfaces for cases where TDM hairpinning is not supported.
- creating a route from the source telephony interface to a different SIP gateway which points to the unit itself
- overriding the SIP destination by using a call property, for example Called E164
- creating a route from the SIP interface to the destination telephony interface

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SIP Redirects
The SIP redirect feature allows a user to redirect a SIP call.
The SIP redirection can be used as a Route destination. When the route source is a SIP interface, incoming SIP INVITEs are replied with a 302 “Moved Temporarily” SIP response. This type of destination is only valid when the source of the route is a SIP interface. When a route is configured with a SIP redirect destination, incoming SIP INVITEs are replied with a 302 "Moved Temporarily" SIP response.


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Basic Call Router Tasks
Creating a Basic Route
Several configuration steps are required to be able to route a call from one endpoint to another.
- Create a route to determine the destinations the calls should be sent to.
- Create the transformation rules to be applied to selected call properties by creating a transformation.
- Associate the call property transformations to a specific route.
- Apply changes.
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Examples of Call Property Transformations
Replace 1xxx by 102
Any destination with prefix 1 will be replaced by destination 102 and send the call through the ISDN-BRI1 interface
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Replace the Prefix 9 by 1 800 in an Outgoing Phone Number
For every called number coming from the SIP interface and beginning with 9, replace the 9 by 1 800, then send the call through the ISDN-BRI1 interface.

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Add the 1 450 Prefix to an Outgoing Phone Number
For every called number coming from the SIP interface add 1 450 in front of the number and then send the call through the ISDN-BRI1 interface
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Send last 3 Digits
For every called number coming from a SIP interface, only keep the 3 last digits of the called number and send the call through the ISDN-BRI1 interface.
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Replacing the Entire Source Phone Number by 1800 123 4567

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Multiple Called URI Transformations
For all outgoing calls, send the 3 last digits of the called URI in the INVITE, display the U037098000 number in the From field, and forward to the SIP server all the 2-digit numbers beginning with 2.

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Setting called numbers NPI and TON ISDN properties
Setting called numbers NPI and TON ISDN properties on SIP to ISDN calls.

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SIP to Two Destinations
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Send Specific Host to FXS phone1
On a Mediatrix 4102S, send all calls coming from the somewhere.com host to FXS phone 1.
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Allow Specific Number Range
Allow outgoing Sip calls only for users with an extension number ranging from 76610 to 76619, 76650 to 76659, and 76660 to 76669.

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Remove the Prefix 9
For every called number coming from the SIP interface and beginning with 9, remove the 9 from the number and send the call through the ISDN-BRI1interface.
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Add CLIR Feature
Add the CLIR feature to every outgoing call coming from a SIP interface going through an FXS interface.
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Block Incoming International Call
Block all incoming international calls (i.e. numbers starting with 00) by sending the calls to a dead-end destination.

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Block Outgoing International Call
Block all outgoing international calls (i.e. numbers starting with 00) by sending the call to a dead-end destination.

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Anonymous Call Rejection
Reject any anonymous calls from a Sip interface to a Sip interface.
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Calling E164 Transformations
Modify the calling E164 number with the user ID and for incoming calls send to the PBX the user ID present in the ''To'' string of the INVITE.
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Routes
Creating a Route from a SIP Gateway to a Destination
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Creating a Route from a Physical Interface to a SIP Destination

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Modifying a Route
When the route is used, the modifications will be applied.

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Deleting a Route
- Go to Call Router/Route Config.
-
In the Routes table,
click
located on the same row as the route you wish to delete.
- Click Save.
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Modifying the Execution Priority Level of a Route
- Go to Call Router/Route Config.
-
In the Routes table,
click
or
located on the same row as the route you wish to prioritise.
- Click Save.
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Call Property Transformations
Creating a Call Property Transformation

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Modifying a Call Property Transformation
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Modifying a Call Property Transformation Rule
- Go to Call Router/Route Config.
-
In the Transformation Rules table,
click
located on the same row as the transformation you wish to modify.
- In the Configure Transformation Rule table, modify the fields as required.
- Click Save.
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Deleting a Call Property Transformation
- Go to Call Router/Route Config.
-
In the Transformations
table, click
located on the same row as the transformation you wish to delete.
- Click Save.
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Deleting a Call Property Transformation Rule
- Go to Call Router/Route Config.
-
In the Transformation Rules table,
click
located on the same row as the call property transformation rule you wish to delete.
- Click Save.
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Modifying the Execution Priority Level of a Call Property Transformation Rule
- Go to Call Router/Route Config.
-
In the Transformation Rules table,
click
or
located on the same row as the call property transformation rule you wish to move.
- Click Save.
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Associating a Call Property Transformation to a Route
-
In the Routes table,
click
located on the same line as the route you wish to associate a call property transformation to.
- From the selection list of the Transformations field, select the transformation you wish to associate to the route.
- Click Save.

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Signaling Properties
Creating a Signaling Property
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Modifying a Signaling Property
- Go to Call Router /Route Config.
-
In the Signaling Properties table,
click
located on the same row as the signaling property you wish to modify.
- In the Configure Signaling Property table, modify the fields as required.
- Click Save.
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Deleting a Signaling Property
- Go to Call Router/Route Config.
-
In the Signaling Properties table,
click
located on the same row as the signaling property you wish to delete.
- Click Save.
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Modifying the Execution Priority Level of a Signaling Property
- Go to Call Router/Route Config.
-
In the Signaling Properties table,
click
or
located on the same row as the signaling property you wish to move.
- Click Save.
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SIP Header Translations
Creating a SIP Header Translation Override

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Modifying a SIP Header Translation Override
- Go to Call Router/Route Config.
-
In the SIP Header Translation Overrides
table, click
located on the same row as the route you wish to modify.
- In the Configure SIP Header Translation Override table, modify the fields as required.
- Click Save.
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Deleting a SIP Header Translation Override
- Go to Call Router/Route Config.
-
In the SIP Header Translation Overrides
table, click
located on the same row as the signaling property you wish to delete.
- Click Save.
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Call Property Translations Override
Creating a Call Property Translation Override
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Modifying a Call Property Translation Override
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Deleting a Call Property Translation Override
- Go to Call Router/Route Config.
-
In the Call Property Translation
Override
table, click
located on the same row as the call property translation override you wish to delete.
- Click Save.
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SIP Redirects
Creating SIP Redirection
- Go to Call Router/Route Config.
-
In the SIP Redirects
table, click
.
- click
located on the same row as an existing SIP redirection to add a new SIP redirection above or,
- click
located at the bottom of the table to add a SIP redirection at the end of the table.
- click
- Set the Name field.
- Set the Destination Host field.
- Click Save.

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Modifying a SIP Redirect
- Go to Call Router /Route Config.
-
In the SIP Redirects
table, click
located on the same row as the SIP redirect you wish to modify.
- In the Configure SIP Redirects table, modify the fields as required.
- Click Save.
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Deleting a SIP Redirect
- Go to Call Router/Route Config.
-
In the SIP Redirects
table, click
located on the same row as the SIP redirect you wish to delete.
- Click Save.
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Hunt
Creating a Hunt
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Modifying a Hunt
- Go to Call Router/Route Config.
-
In the Hunt table,
click
located on the same row as the hunt group you wish to modify.
- In the Configure Hunt table, modify the fields as required.
- Click Save.
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Deleting a Hunt
- Go to Call Router/Route Config.
-
In the Hunt table,
click
located on the same row as the Hunt group you wish to delete.
- Click Save.
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Auto-Routing
Enabling Auto-Routing
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Linking an Endpoint to Several SIP Gateways
This procedure is for outgoing calls only. For incoming calls, only one destination is used, however, it is possible to set a hunt to multiple SIP gateways.
- Go to Call Router/Auto-Routing.
-
From the Endpoints auto-routing table,
click
located on the row of the endpoint you wish to link to several SIP gateways.
- From the Configure Auto-Routing section, complete the fields as required.
- From the Apply To The Following Endpoints table, select the endpoints you wish to link to the gateways selected at previous step.
- Click Apply.
- Click Apply, again.
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Routing Examples
This is an example on how a call is routed to it's final destination.
Routes | Source | Criteria Property | Criteria Rule | Transformations | Signaling properties | Destination |
---|---|---|---|---|---|---|
1 | sip-default | fxs-Slot3/FXS1 | ||||
2 | fxs-Slot3/FXS1 | Called E164 | 911 | map911 | sip-default | |
3 | fxs-Slot3/FXS1 | normalize | privacyId | sip-default |
- Route 1 does not match because "fxs-Slot3/FXS1 does not match the "sip-default"
- Route 2 does not match because although "fxs-Slot3/FXS1" matches the fxs-Slot3/FXS1" source, the "911" criteria rule does not match 4505552222.
- Route 3 is chosen because "fxs-Slot3/FXS1" matches the source "fxs-Slot3/FXS1" source and the criteria rule included all criteria rules
The SIP-default destination is associated with the call. Therefore the call will be routed to the network using the SIP default gateway.
Transformation | Name | Criteria | Transformation |
---|---|---|---|
1 | map911 | Called E164 | |
2 | normalize | Called E164 | Called E164 |
Apply transformations
- Transformation 1 is not used, because "map911" name does not match.
- Transformation 2 is applied because both name and criteria match.
Name | Criteria Rule |
---|---|
map911 | |
normalize | 819.......$ |
normalize | 514.......$ |
normalize | 450.......$ |
- Criteria rule 1 is not applied because the name does not match.
- Criteria rule 1 does not apply because the criteria does not match (called number does not start by 819) even if the name matches.
- Criteria rule 3 does not apply because the criteria does not match (called number does not start by 514) even if the name matches.
- Criteria rule 4 is applied because the name and the criteria match.
Signaling properties | Name | ... | Privacy | ... |
---|---|---|---|---|
1 | privacyId | Id |
- Signaling property 1 is chosen because the name matches (no criteria to validate) therefore the "privacy=ID" property is added to the call.
For more Route examples, refer to DGW Configuration Guide - Call Router Basic Routes document published on the Media5 Documentation Portal.
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Regular Expressions
A regular expression is a string used to find and replace strings in other large strings.
- The expression cannot begin by “^”, it is implicit in the expression.
- The matching criterion implicitly matches from the beginning of the string, but not necessarily up to the end. For instance, 123 will match the criterion 1, but it will not match the criterion 2.
- To match the whole string, you must end the criterion with “$”. For instance, 123 will not match the criterion 1$ and will match the criterion 123$.
- Use the “local_ip_port“ macro to replace the properties by the local IP address and port of the listening network of the SIP gateway used to send the INVITE.
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Regular Expression Wildcards
Character | Description |
---|---|
. | Single-digit place holder. For instance, 555 .... matches any dialed number beginning with 555, plus at least four additional digits. Note that the number may be longer and still match. |
* | Repeats the previous digit 0, 1, or more times. For instance, in the pattern: 1888*1 the pattern matches: 1881, 18881, 188881, 1888881 Note: If you are trying to handle the asterisk (*) as part of a dialed number, you must use \*. |
[ ] | Range of digits.
|
( ) | Indicates a group (also called pattern), for instance, 555(2525). It is used when replacing a number in a mapping. For more details, refer to Regular Expression Groups. |
? | Matches 0 or 1 occurrence of the previous item. For instance, 123?4 matches both 124 and 1234.. |
+ | Repeats the previous digit one or more time. For instance 12+345 matches 12345, 122345, etc. (but not 1345). If you use the + at the end of a number, it repeats the last number one or more times. For instance: 12345+ matches, 12345, 123455, 1234555, etc. |
| | Indicates a choice of matching expressions (OR). |
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Regular Expression Groups
Groups can be used in Transformation Rules to reuse parts of the Call Property that was used in the Criteria.
A group is placed within parenthesis. You can use up to nine groups defined by “\1” to “\9” and matching is not case sensitive. “\0” represents the whole string. Lets say for instance you have the following string: 9(123(45)6)
Replacement | Result |
---|---|
\0 | 9123456 |
\1 | 123456 |
\2 | 45 |
\3 |
- Calling/Called E.164
- Calling/Called Name
- Calling/Called Host
- Calling/Called URI
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Information on Regular Expressions
For more information on Regular Expressions (RegEx) refer to:
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Management
Configuration Scripts
Basic Configuration Scripts Concepts
Configuration Scripts Import and Export
Importing and exporting configuration scripts allows you to modify in whole or in part the configuration script used on your unit.
Configuration scripts are files containing textual commands that are sent over the network to a Mediatrix unit. Upon receiving the file, the unit executes each command line in sequence. Script commands can assign values to configuration variables, or execute configuration commands.
A configuration script can be used on any firmware version, regardless of the firmware version it was exported from. It is possible to import a complete configuration script, a subset of the configuration script or even a few lines of a configuration script.
- Change one or several script commands
- Add new commands
- Change parameter values
- Add parameters
- Replace the complete configuration script
Scripts are written by the system administrator and can be used to accomplish various tasks, such as automating recurrent configuration tasks or batch-applying configuration settings to multiple devices. Scripts can be executed once or periodically at a specified interval. They can also be scheduled to be executed when the Mediatrix unit starts.
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Configuration Scripts vs. Backup and Restore
- Configuration scripts are files containing textual commands that are sent over the network
to a Mediatrix unit. Upon receiving the file, the Mediatrix unit executes each command line in
sequence. Script commands can assign values to configuration variables, or execute
configuration commands.
Scripts are written by the system administrator and can be used to execute several operations, such as automating recurrent configuration tasks or batch-applying configuration settings to multiple devices. Scripts can be executed once or periodically at a specified interval. They can also be scheduled to execute when the Mediatrix unit restarts.
- The backup/restore feature is used to backup a specific Mediatrix unit's configuration for
safety purposes. When needed, the configuration image file that is generated by a backup
operation can be restored to put the unit back into the exact configuration it was when the
backup was taken.
Configuration image files contain a Mediatrix unit's configuration information. They are not intended to be edited and must not be confused with configuration scripts. When restoring a configuration image, the whole Mediatrix unit s current configuration is replaced with the configuration found in the configuration image file. Restoring a configuration image is therefore an operation that is completely different from executing a configuration script.
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How to Write a Configuration Script
Configuration scripts are text files containing command lines that are interpreted by the Mediatrix unit. Most commands contained in a script assign values to configuration variables. Script commands can also execute configuration commands.
Writing configuration scripts requires a bit of knowledge about the Mediatrix unit’s configuration variables tree structure. Each parameter that is accessed via the Mediatrix unit’s web interface maps to a variable in the configuration tree.
The following sub-sections give sample commands performing common tasks.
Assigning Scalar Values
The following is a sample script command assigning a value to a scalar configuration variable:
Assigning Table Cell Values
The following is a sample script command assigning a value to a configuration table cell:
Executing Commands
Configuration commands are used to make the Mediatrix unit perform actions such as restarting the unit, restarting a service, refreshing its SIP registration, etc.
The following is a sample script command executing a configuration command:
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DHCPv4 Auto-Provisioning
The Mediatrix unit can be configured to automatically download new configuration scripts upon receiving options 66 (tftp-server) or 67 (bootfile), or vendor-specific option 43 using sub-options 66 and 67 in a DHCPv4 answer
A DHCP server answer includes both Bound and Renew. The contents of option 66, 67 or 43 defines which script to download. The unit's configuration is not used to download the script. This allows the unit, for instance, to download a script from a server after a factory reset and to reconfigure itself without a specific profile. If the imported configuration script is identical to the last executed script, it will not be run again. The script retry mechanism is not enabled for the DHCPv4 triggered scripts. If options 66, 67 and 43 are received, all scripts are executed independently. The script defined by the tftp-server (option 66) option is executed first. If you are using HTTPS to transfer scripts, you must have a time server SNTP that is accessible and properly configured.
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Basic Configuring Scripts Tasks
Configuring a Privacy Key
- Go to Management > Configuration Scripts.
- In the Execute Scripts table, set a privacy key of your choosing in the Privacy Key field.
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Importing a Configuration Script Using a File Server

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Executing a Configuration Script from the Unit File Management System

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Importing a Configuration Script from Your PC

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Importing a Configuration Script to the Unit File Management System

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Exporting a Configuration Script Using a File Server

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Exporting a Configuration Script to Your PC

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Executing Configuration Scripts from a File Server Periodically
- Configuring the FTP Server
- Configuring the TFTP Server
- Configuring the HTTP Server
- Configuring the HTTPS Server
Mediatrix units do not all include a real time clock allowing them to maintain accurate time when they are shutdown. You must have a time server SNTP that is accessible and properly configured or the automatic configuration update feature may not work properly. Refer to Configuring the Mediatrix Unit to Use an SNTP Server.

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Executing Configuration Scripts from the Unit File Management System Periodically
You must have a time server SNTP that is accessible and properly configured or the automatic configuration update feature may not work properly. Refer to Configuring the Mediatrix Unit to Use an SNTP Server. Configuration scripts files must be available in the unit's file management system. Refer to Importing a Configuration Script to the Unit File Management System.
A configuration script must have been imported to the unit's file management system. Refer to Importing a Configuration Script to the Unit File Management System.
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Executing Configuration Scripts from a File Server Each Time the Unit is Started

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Executing Configuration Scripts from the Unit File Management System Each Time the Unit is Started
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Configuring the DHCP to Trigger Configuration Script Execution
The Mediatrix unit can be configured to automatically import new configuration scripts upon receiving options 66 (tftp-server) or 67 (bootfile), or vendor-specific option 43 using sub-options 66 and 67 in a DHCPv4 answer. A DHCP answer includes both Bound and Renew.
- Configuring the FTP Server
- Configuring the TFTP Server
- Configuring the HTTP Server
- Configuring the HTTPS Server
Mediatrix units do not all include a real time clock allowing them to maintain accurate time when they are shutdown. If you are using HTTPS, you must have an SNTP server that is accessible and properly configured or the automatic configuration update feature may not work properly. Refer to Configuring the Mediatrix Unit to Use an SNTP Server.
For more details on DHCPv4 Auto-Provisionning, refer to DHCPv4 Auto-Provisioning
- Go to Management/Configuration Scripts/Execute.
- In the Automatic Script Execution section, from the Allow DHCP to Trigger Scripts Execution selection list, choose Enable.
- Click Apply.
The instructions sent FROM the DHCP server can be in different formats and will be understood by the Mediatrix unit according to what was chosen for the ScriptsDhcpOptionsFormat MIB parameter (not accessible via Web page). Possible values with their respective formats are:
- Fully Qualified: Script=[protocol]://[username] :[password]@[server]/[path]/[file].
- Url: [protocol]:// [username] :[password]@[server]/[path]/[file]
- ServerHost: Allow one DHCP option to specify the IP address or FQDN of a file server. Uses the path and filename specified in the ScriptLocation and ScriptGenericFileName parameters, use the transfer protocol, username and password specified in ScriptTransferProtocol, ScriptTransferUsername and ScriptTransferPassword parameters.
- AutoDetect: A value beginning with "Script=" is considered as "FullyQualified", A value beginning with "[protocol]://" is considered as a URL. A value that looks like an IPv4/IPv6 address or domain name is considered as a "ServerHost". (default value)
When the unit starts, it will receive the location of the config script from the DHCP response, as per the format defined by the ScriptsDhcpOptionsFormat parameter. The unit will then import and execute the configuration scripts from the specified location. Any changes to the script will be applied to the running configuration. The unit configuration is only updated if at least one parameter value defined in the imported configuration scripts is different from the actual unit configuration.

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Advanced Configuration Scripts Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Scripts Transfer Certificate Validation
Refer to Conf. ScriptsTransferCertificateValidation.Scripts Transfer Certificate Trust Level
Refer to Conf. ScriptsTransferCertificateTrustLevel.Scripts Transfer Cipher Suite
Refer to Conf. ScriptsTransferCipherSuite.Scripts Transfer Tls Version
Refer to Conf. ScriptsTransferTlsVersion .Scripts Dhcp Options Format
Refer to Conf. ScriptsDhcpOptionsFormat.Scripts Transfer Retries Number
Refer to Conf. ScriptsTransferRetriesNumber.Top
Backup/Restore
Basic Backup and Restore Concepts
Configuration Backup and Restore
Performing a configuration backup allows you to have a copy of the entire configuration at the time the backup was performed.
- to revert the running configuration to a valid configuration when the running configuration generates error messages or no longer works;
- to deploy a valid configuration on other units;
- to revert to a known configuration when too many changes were made to the running configuration;
- to deploy a backup configuration on a replacement unit.
- the status and configuration parameters;
- the certificates;
- the Rulesets.
But the configuration backup does not include the File service files, except for the Rulesets.
Backup made on firmware version vX | Unit firmware version | Possible |
---|---|---|
Backup.v3 | Firmware version 3 | YES |
Backup.v1 | Firmware version 3 | YES |
Backup.v1 | Firmware version 1 | YES |
Backup.v3 | Firmware version 1 | NO |
The Configuration Back and Restore performed on Dgw v2.0.31 includes all Rulesets. However, starting on Dgw v.2.0.32 only the Rulesets modified by the user will be included in the configuration backup and restore. If a configuration backup is performed on Dgw v2.0.31 and restored on a newer version of DGW, all Rulesets existing in v2.0.31 will be copied as custom Rulesets i.e the system will not use the factory Rulesets of the newer version even if they were not modified.
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Basic Backup Tasks
Performing a Configuration Backup to a File Server

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Performing a Configuration Backup to the Unit File Management System

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Basic Restore Tasks
Restoring a Configuration From a File Server

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Restoring a Configuration from the Unit File Management System

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Restoring a Configuration from Your PC

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Advanced Backup and Restore Configuration Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
Image Transfer Cipher Suite
For more details on the following parameters refer to DGW Configuration Guide - Configuration Reference Guide published on the Media5 Documentation Portal. Refer to Conf.ImageTransferCipherSuite.Image Transfer Tls Version
Refer to Conf.ImageTransferTlsVersion.Top
Firmware
Basic Firmware Concepts
Single File Upgrade
The single file used to upgrade DGW is a binary file (.bin) that contains the modules and the features to install on your Mediatrix unit when a new release is available.
The single file is uploaded to the Mediatrix using the transfer protocols HTTP, HTTPS, FTP, and TFTP.
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Basic Upgrade Tasks
Adding the Single File to the Server
- Copy the Single File .bin to your server. (FTP, HTP, HTTPS, or TFTP)
- You may rename the file with a shorter name. This will make it easier to type the URL in the Mfp Url field in the Installing Another Firmware Version Using a Single File step.
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Installing Another Firmware Version Using a Single File
Depending on the type of protocol you will use, the following procedures must be completed.

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Basic Downgrade Tasks
Reverting to the Previous Firmware Version from DGW v.44.1 or newer
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Advanced Firmware Upgrade Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
- Fpu. MfpTransferTlsVersion
- Fpu. MfpTransferCipherSuite
- Fpu. MfpTransferCertificateValidation
- Fpu. MfpTransferCertificateTrustLevel
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Certificates
Certificates Concepts
Certificates
The Mediatrix unit uses digital certificates, which are a collection of data used to verify the identity of individuals, computers, and other entities on a network.
- the certificate's name
- the issuer and issued to names
- the validity period (the certificate is not valid before or after this period)
- the use of certificates such as:
- TlsClient: The certificate identifies a TLS client. A host authenticated by this kind of certificate can act as a client in a SIP over TLS connection when mutual authentication is required by the server.
- TlsServer: The certificate identifies a TLS server. A host authenticated by this kind of certificate can serve files or web pages using the HTTPS protocol or can act as a server in a SIP over TLS connection.
- whether or not the certificate is owned by a Certification Authority (CA)
Although certificates are factory-installed new ones can also be added. Since TLS certificates are validated in terms of time (certificate validation/expiration date, etc.), the use of NTP (Network Time Protocol) is mandatory when using the security features.
- Host Certificates: used to certify the unit (e.g.: a web server with HTTPS requires a host certificate).
- Others: Any other certificate including trusted CA certificates used to certify peers (e.g.: a SIP server with TLS).
- Conf.ScriptsTransferCertificateValidation
- Cwmp.TransportCertificateValidation
- Eth.Eap.CertificateValidation
- Fpu.MfpTransferCertificateValidation
- Nlm.PCaptureTransferCertificateValidation
- Sbc.CertificateValidation
- SipEp.InteropTlsCertificateValidation (also available in the DGW Web page under SIP/Interop)
The certificates must be uploaded to the Mediatrix units. They define how a Mediatrix unit will certify the remote host in order to mark it as secure and suitable for a TLS connection. If the Mediatrix unit does not trust the remote certificate (i.e. does not authenticate it with either one of the 3 methods: HostName, trustedCertificate, DnsSrv), then the Mediatrix unit will not establish the connection.
- for testing purpose,
- if one cannot identify the required CA cert, or
- the CA cert has mismatched Common Name/Subject Alternate Name. (In this case there is no fallback, it will fail if the name does not match)
- SIP
- Configuration Web pages
- File transfers (scripts, firmwares, etc.) with HTTPS
- Configuration using TR-069
- Wired Ethernet Authentication with EAP (802.1x)
One common use of the host certificate is to allow HTTPS Web access to the unit (which in this case, the device is the TLS server). For more details refer to the Technical Bulletins - Creating a Media5 Host Certificate with Open SSL document on the Media5 Documentation Portal.
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Certificates Tasks
Importing a Trusted CA or SIP Server Certificate through the Web Page

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Trusted CA Certificate Content Example
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
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Generating a Private Key
- a length of 2048 bits
- encryption with a 256 bit AES algorithm.
The output filename is your_device.key.
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Creating a Certificate Signing Request (CSR) from a Private Key
A CSR is generated from the private key created in the Generating a Private Key procedure with a SHA256 signature algorithm. This is a result example.
[root@localhost mycert]# openssl req -new -key 192.168.1.31.key -out 192.168.1.31.csr -sha256
Enter pass phrase for 192.168.1.31.key:
You are about to be asked to enter information that will be incorporatedinto your certificate request.
What you are about to enter is what is called a Distinguished Name or a DN.
There are quite a few fields but you can leave some blankFor some fields there will be a defaultvalue,
If you enter '.', the field will be left blank.
-----
Country Name (2 letter code)[XX]:CA
State or Province Name (full name[]:Quebec
Locality Name (eg, city) [Default City]:Montreal
Organization Name (eg, company) [Default Company Ltd]:Media5
Organizational Unit Name (eg,section)[]:TAC
Common Name (eg, your name or your server's hostname)[]:192.168.1.31
Email Address[]:tac@media5corp.com
Please enter the following 'extra'attributes
to be sent with your certificate request
A challenge password []
:An optional company name []:
[root@localhost mycert]#
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Signing the CSR file by Your Own Certificate Authority (CA)
Use this procedure if your certificates are signed by a Certificate Authority you have access to.
- CA.key is the private key of your CA
- CA.crt is the CA’s public certificate
- CA.srl is the serial number file
- 3652 days is the validity period of the certificate
- host_ext.cnf defines the usage of the
certificate. It
contains:
[ host_ext ] basicConstraints = CA:false keyUsage = digitalSignature, keyEncipherment, dataEncipherment extendedKeyUsage = serverAuth, clientAuth
[root@localhost mycert]# openssl x509 -req -extfile host_ext.cnf -extensions host_ext -sha256 -days
3652 -in 192.168.1.31.csr -CA CA.crt -CAkey CA.key -CAserial CA.srl -out 192.168.1.31.crt
Signature ok
subject=/C=CA/ST=Quebec/L=Montreal/O=Media5/OU=TAC/CN=192.168.1.31/emailAddress=tac@media5corp.com
Getting CA Private Key
Enter pass phrase for CA.key:
root@localhost mycert]#
When
the certificate will be imported to the Mediatrix unit, the information defined for the
keyUSage of the host_ext.cnf file will be displayed in Management>Certificates/Host Certificates table, under the
Usage column.Top
Signing the CSR by a Third Party Certificate Authority (CA)
Use this procedure if your certificates are signed by a Certificate Authority you do not have access to.
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Self-signing the CSR File
Use this procedure if your certificates are self-signed, i.e. security is not an issue.
host_ext.cnf is a file containing the following which defines the usage of the certificate:
[ host_ext ]
basicConstraints = CA:false
keyUsage = digitalSignature, keyEncipherment,dataEncipherment
extendedKeyUsage = serverAuth, clientAuth
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Combining the Private Key and the Signed Certificate
The host certificate required by the Mediatrix contains two parts: the private key and the signed certificate.
[root@localhost mycert]# cat 192.168.1.31.key 192.168.1.31.crt > 192.168.1.31.pem
[root@localhost mycert]# more 192.168.1.31.pem
-----BEGIN RSA PRIVATE KEY-----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-----END RSA PRIVATE KEY-----
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
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Importing a Host Certificate through the Web Page

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SNMP Configuration
SNMP Basic Concepts
Simple Network Management Protocol (SNMP)
The Simple Network Management Protocol (SNMP) can be used to configure all the parameters available in the Mediatrix CPE, to perform firmware updates, to import a configuration and to monitor the Mediatrix CPE.
To configure the Mediatrix CPE parameters with the SNMP, a secure SNMPv3 or a non-secure SNMPv1 connexion can be used. The CPE does not grant an SNMPv3 access without authentication and privacy. Because the connexion is initiated by the Management Server, the communication is usually unable to go through the NAT Firewall.
Unit monitoring is possible with SNMP because it provides access to all the status parameters of the CPE. Furthermore, the CPE can send notifications, called traps, to the Management Server, that will allow the administrator to monitor specific events. Because it is the CPE that sends the notifications, the communication is usually able to go through the NAT Firewall however the SNMP protocol, based on UDP, does not insure reliable delivery of notifications.
- GetRequest
- SetRequest
- GetResponse
- SetResponse
- Trap
- GetWalk
- UMN
- HP Openview
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SNMP Basic Tasks
Securing SNMP Interface
- Go to Management/SNMP.
-
In the SNMP Configuration table, set
the following parameters:
- Set Enable SNMP V1 to Disable.
- Set Enable SNMP V2 to Disable.
- Set the Privacy Protocol.
- In the Privacy Password field, enter a password of your choosing.
- Click Apply.

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CWMP
Basic CWMP Concepts
Data Model
The Data model is a hierarchical structure of supported device parameters.
Broadband Forum publishes its data model standards, with complete information about each parameter value, the type and the meaning. For more details refer to https://www.broadband-forum.org/standards-and-software/technical-specifications/tr-069-files-tools. Each manufacturer uses parts of this standard in his implementation, but can include proprietary parameters.
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Device Profile
The Cwmp service supports the data model template for device (TR-106) if the RootElement is set to ' Device'


A TR-106 device is a CPE device residing in the home network. This device has a single network interface connected on the LAN side of the home router or directly on the Internet. The Tr106LanNetworkInterface parameter identifies that network interface.
- Supported data model for the Device:1 object:
- Baseline:1
- LAN:1
- Time:1
- VoiceService:1 (Partial Support)
- GatewayInfo:1
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Services Parameters
Unless otherwise specified, {i} = 1
Name | Access | Notification | |
---|---|---|---|
VoiceService.{i}.3 | Present | N/A | |
The top-level object for a CPE with voice capabilities. {i} is limited to '1' in current implementation. |
|||
VoiceServiceNumberOfEntries | Read-only | None | |
The number of entries in the VoiceService table.
Returns '1'. |
|||
VoiceService.{i}.Capabilities. | Present | N/A | |
The overall capabilities of the VoIP CPE.
{i} is limited to 1 in current implementation. |
|||
MaxProfileCount | Read-only | None | |
Maximum supported number of distinct profiles.
Returns '1'. |
|||
MaxLineCount | Read-only | None | |
Maximum supported number of lines for all
profiles. Returns the number of ports on the CPE device. |
|||
SignalingProtocols | Read-only | None | |
List of supported signaling protocols.
Returns 'SIP/2.0' value. |
|||
VoiceService.{i}.Capabilities.SIP. | Present | N/A | |
SIP-specific capabilities.
{i} is limited to '1' in current implementation. |
|||
Role | Read-only | None | |
The role of this VoIP CPE.
Returns 'UserAgent' value. |
|||
Transports | Read-only | None | |
List of supported SIP transport protocols.
Returns 'UDP, TCP, TLS' value. |
|||
URISchemes | Read-only | None | |
List of supported URI schemes beyond the URI
schemes required by the SIP specification.
Returns 'sip' value. |
|||
VoiceService.{i}.VoiceProfile.{i} | Present | N/A | |
Object associated with a collection of voice lines with common characteristics. Both {i}s are limited to 1 in current implementation. |
|||
Reset | Read/Write | None | |
When the value is set to 'true', all lines of the profile are forced to be reset, causing the unit to be reinitiatlised and all start-up actions, such as registration, to be performed. This restarts SipEp service. Always returns false, i.e '0'. |
|||
VoiceService.{i}.VoiceProfile.{i}.SIP. | Present | N/A | |
Voice profile parameters that are specific to SIP user agents. Both {i}s are limited to '1' in current implementation. |
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ProxyServer | Read/Write | None | |
Host name or IP address of the SIP proxy server. An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is passed on to another entity that can further process the request. Proxies are also useful for enforcing policy and for firewall traversal. A proxy interprets, and, if necessary, rewrites parts of a request message before forwarding it. Returns and sets the host part of the SipEp.DefaultStaticProxyHomeDomainHost parameter. Setting a new value to this parameter triggers a non-graceful service restart. |
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ProxyServerPort | Read/Write | None | |
Destination port to be used when connecting to the SIP server. Returns and sets the port of the SipEp.DefaultStaticProxyHomeDomainHost parameter. Setting a new value to this parameter triggers a non-graceful service restart. |
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RegistrarServer | Read/Write | None | |
Host name or IP address of the SIP registrar server. A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles. Returns and sets the host part of the SipEp.DefaultStaticRegistrarServerHost parameter. Setting a new value to this parameter triggers a non-graceful service restart. |
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RegistrarServerPort | Read/Write | None | |
Destination port to be used in connecting to the SIP registrar server. An empty string or '0' indicates that the CPE will use the default value, i.e '5060'. Returns and sets the port of the SipEp.DefaultStaticRegistrarServerHost parameter. Setting a new value to this parameter triggers a non-graceful service restart. |
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UserAgentDomain | Read/Write | None | |
CPE domain string. This value is saved in an internal database. The value is initially set to an empty string and reset to an empty string on a factory default reset. If an empty string is specified, the SIP listening IP address is used. This value is set to the SipEp.UserAgent.ContactDomain parameter when no specific URI domain is specified by the ACS (see .VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.URI Returns the last value set by the ACS. Setting a new value to the UserAgentDomain parameter triggers a registration refresh. |
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UserAgentPort | Read/Write | None | |
Port used for incoming call control signaling.
Returns the value of the SipEp.Gateway[name=default].Port parameter. Sets the value of the SipEp.Gateway[name=default].Port and SipEp.Gateway[name=default].SecurePort parameters Setting a new value to the UserAgentPort parameter triggers a non-graceful service restart. |
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UserAgentTransport | Read/Write | None | |
Returns the enable transports in this priority: 'TLS', 'UDP' and 'TCP'.
When set, disables all transports except the one being set. If 'TCP' is set, TLS and UDP will be disabled and TCP will be enabled. Maps to the following parameters:
Setting a new value to the UserAgentTransport parameter triggers a non-graceful service restart. |
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OutboundProxy | Read/Write | None | |
Host name or IP address of the outbound proxy. To disable the outbound proxy, use '0.0.0.0'. From RFC 3261: A proxy that receives requests from a client, even though it may not be the server resolved by the Request-URI. Typically, a UA is manually configured with an outbound proxy, or can learn about one through auto-configuration protocols. When enabled, the initial route for all SIP requests will contain the outbound proxy address, suffixed with the loose routing parameter 'lr'. The Request-URI still contains the home domain proxy address. Requests are directed to the first route (the outbound proxy). Returns and sets the host part of the SipEp.DefaultStaticProxyOutboundHost parameter. Setting a new value to the OutboundProxy parameter triggers a non-graceful service restart. |
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OutboundProxyPort | Read/Write | None | |
Destination port to be used in connecting to the outbound proxy. This parameter is ignored unless the value of the OutboundProxy parameter is not empty. Returns and sets the port of the SipEp.DefaultStaticProxyOutboundHost parameter. Setting a new value to the OutboundProxyPort parameter triggers a non-graceful service restart. |
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ConferenceCallURI | Read/Write | Passive | |
URI used in the request-URI of the INVITE sent to the conference server. This parameter only has an effect when the conference type is 'ConferenceServer'. See the EpServ.DefaultConferenceType parameter. Please refer to the documentation shipped with your device for more details regarding this parameter's semantics. If a specific configuration is set in the GwSpecificConference.ServerHost parameter and the GwSpecificConference.EnableConfig parameter is set to 'Enable', then it overrides the current default configuration. Returns and sets the SipEp.DefaultStaticConferenceServerUri parameter. |
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VoiceService.{i}.VoiceProfile.{i}.Line. | |||
Object associated with a distinct voice line. At most, one enabled entry in this table can exist with a given value for the DirectoryNumber parameter. Both {i}s are limited to 1 in current implementation. |
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VoiceService.{i}.VoiceProfile.{i}.Line.{i}. | Present | N/A | |
First two {i}s are limited to 1 in current implementation. The Line.{i} starts from 1 up to the number of ports on the device, also represented by the VoiceService.{i}.Capabilities.MaxLineCount parameter. If the CPE device has different types of lines, the numerical index follows this order: FXS, FXO, E1T1, and BRI lines. |
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Enable | Read/Write | Passive | |
Enables or disables this line, or places it into a quiescent state. Returns the value of the EpAdm[i].InitialAdminStateConfig parameter:
Set the Enable parameter has the following effect:
|
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DirectoryNumber | Read/Write | Passive | |
Directory number associated with this line.
Returns and sets the value of the SipEp.UserAgent.FriendlyName parameter. Setting a new value to the DirectoryNumber parameter triggers a registration refresh. |
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Status | Read-only | Passive | |
Indicates the status of this line. Returned
value is based on the following parameters:
Detailed mapping is as follows:
|
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VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP. | Present | N/A | |
Voice line parameters that are specific to SIP
call signaling. First two {i}s are limited to 1 in current implementation. The last {i} takes all values from 1 to the value of the VoiceService.{i}.Capabilities.MaxLineCount parameter. |
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AuthUserName | Read/Write | Passive | |
Username used to authenticate the connection to the server.
Returns and sets the value of the SipEp.Authentication.Username parameter. Also sets the value of the SipEp.UserAgent.Username parameter if the value of the VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.URI parameter is empty Configures the SipEp.Authentication[].CriteriaSelection and SipEp.Authentication[].Endpoint parameters. Setting a new value to the AuthUserName parameter triggers a registration refresh. Mapping between index i of Line{i} and the index of the SipEp.Authentication table behaves differently from the other internal tables. The numerical value of i is directly used as the index of the SipEp.Authentication table. If i = 2, index of the SipEp.Authentication table is 2. Therefore, any modification to this table made by other means must follow this convention. |
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AuthPassword | Write | None | |
Password used to authenticate the connection to
the server. Set value of proper columnar SipEp.Authentication.Password parameter. Configures the SipEp.Authentication[].CriteriaSelection and SipEp.Authentication[].Endpoint parameter. Setting a new value to the AuthPassword parameter triggers a registration refresh. Mapping between index i of Line{i} and the index of the SipEp.Authentication table works the same as for the AuthUserName parameter. Returns an empty string. |
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URI | Read/Write | Active | |
URI by which the user agent will identify
itself for this line. Returns the SIP URI used by the device (formed with the SipEp.UserAgent.Username@SipEp.UserAgent.ContactDomain parameter). This value is saved in an internal database. The value is initially set to an empty string and reset to an empty string on a factory default reset. Sets the value of the SipEp.UserAgent.Username parameter and, if an optional host part is given, sets the value of the SipEp.UserAgent.ContactDomain parameter. Setting a new value to the URI parameter triggers a registration refresh. |
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VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Stats. | Present | N/A | |
Statistics for this voice line instance.
First two {i}s are limited to 1 in current implementation. The last {i} takes all values from 1 to the value of VoiceService.{i}.Capabilities.MaxLineCount parameter Note: These statistics are from the IP network standpoint. If a call is directly
established between two telephony interfaces (between two local FXS lines for
instance), statistics are not cumulated.
Note: Unless otherwise stated, these statistics are updated each time the
processing of a call terminates.
Note: These statistics are cleared when the service they originate from is started
or restarted.
Note: These statistics are cumulative for all the calls on the line. They are
updated at the end of each call.
Note: The Line statistics are obtained from the EpServ and Mipt services. Lines in
TR-104 are mapped to endpoints of the device. The CWMP service builds an
internal map to convert between the numerical index {i} used as the 'Line'
number and the textual endpoint indexes used in the tables of the EpServ and
Mipt services. When statistics are reported for multi-channel endpoints, they
cover all the channels of the endpoint. For instance, the number of received
packets for a 'Line' is the number of packets received for all the channels of
the related endpoint.
Note: The media statistics are only available for FXS and FXO lines. The call
statistics are only available for FXS lines.
|
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ResetStatistics | Write | None | |
When set to one, resets the statistics for this voice line. Always returns 'false'. Maps to the Mipt.EndpointStatistics[i].Reset and EpServ.CallStatistics[i].Reset parameters. |
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PacketsSent | Read-only | Passive | |
Returns the total number of RTP packets sent for this line. Maps to the Mipt.EndpointStatistics[i].PacketsSent parameter. |
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PacketsReceived | Read-only | Passive | |
Returns the total number of RTP packets received for this line. Maps to the Mipt.EndpointStatistics[i].PacketsReceived parameter. |
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BytesSent | Read-only | Passive | |
Returns the total number of RTP payload bytes sent for this line. Maps to the Mipt.EndpointStatistics[i].BytesSent parameter. |
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BytesReceived | Read-only | Passive | |
Returns the total number of RTP payload bytes received for this line. Maps to the Mipt.EndpointStatistics[i].BytesReceived parameter. |
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IncomingCallsReceived | Read-only | Passive | |
Returns the total number of incoming calls received. Maps to the EpServ.EndpointStatistics[i].IncomingCallsReceived parameter. |
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IncomingCallsAnswered | Read-only | Passive | |
Returns the total number of incoming calls answered by the local user. Maps to the EpServ.EndpointStatistics[i].IncomingCallsAnswered parameter. |
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IncomingCallsConnected | Read-only | Passive | |
Returns the total number of incoming calls that successfully completed call setup signaling. Maps to the EpServ.EndpointStatistics[i].IncomingCallsConnected parameter. |
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IncomingCallsFailed | Read-only | Passive | |
Returns the total number of incoming calls that failed to successfully complete call setup signaling. Maps to the EpServ.EndpointStatistics[i].IncomingCallsFailed parameter. |
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OutgoingCallsAttempted | Read-only | Passive | |
Returns the total number of outgoing calls attempted. Maps to the EpServ.EndpointStatistics[i].OutgoingCallsAttempted parameter. |
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OutgoingCallsAnswered | Read-only | Passive | |
Returns the total number of outgoing calls answered by the called party. Maps to the EpServ.EndpointStatistics[i].OutgoingCallsAnswered parameter. |
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OutgoingCallsConnected | Read-only | Passive | |
Returns the total number of outgoing calls that successfully completed call setup signaling. Maps to the EpServ.EndpointStatistics[i].OutgoingCallsConnected parameter. |
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OutgoingCallsFailed | Read-only | Passive | |
Returns the total number of outgoing calls that failed to successfully complete call setup signaling. Maps to the EpServ.EndpointStatistics[i].OutgoingCallsFailed parameter. |
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CallsDropped | Read-only | Passive | |
Returns the total number of calls that were successfully connected (incoming or outgoing), but dropped unexpectedly while in progress without explicit user termination. Maps to the EpServ.EndpointStatistics[i].CallsDropped parameter. |
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TotalCallTime | Read-only | Passive | |
Returns the cumulative duration of all IP calls on the endpoint since service start, in seconds. Maps to the EpServ.EndpointStatistics[i].TotalCallTime parameter. |
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AverageReceiveInterarrivalJitter | Read-only | Passive | |
Average received interarrival jitter, in microseconds. This value is based on the average interarrival jitter of each call ended during the collection period. The value is weighted by the duration of the calls. Maps to the Mipt.EndpointStatistics[i].AverageReceiveInterarrivalJitter parameter. |
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Gateway Profile
The Cwmp service supports the data model template for internet gateway device (TR-98) if the RootElement is set to ' InternetGatewayDevice'.

The Internet Gateway Device is a CPE device at the frontier of the home network and the public Internet. It normally implements at least 2 network interfaces, one for the WAN and one for the LAN.
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TR-069 or CPE WAN Management Protocol (CWMP)
The Technical Report 069 (TR-069), also known as CWMP, is a Broadband Forum technical specification. This protocol can be used to monitor and update the Mediatrix unit configurations and firmware. In other words, when using TR-069, the Mediatrix unit can get in contact with an Auto Configuration Server (ACS) to initiate a configuration script transfer/execution and a firmware upgrade.
The first time the Mediatrix unit is connected to the network, it will attempt to contact the Auto Configuration Server (ACS), which is the entry point for the administrator. The Mediatrix unit will obtain the URL of the ACS using either the DHCP server with option 43 or by retrieving the information directly from the Customer's Profile. Therefore, upon start-up, the Mediatrix unit will contact the ACS, which in return will send the required configuration files and initiate, if necessary, a firmware update. This automated sequence is what is referred to as zero-touch, as the Mediatrix unit is automatically configured by the ACS according to the instructions given by the administrator without manual intervention on the unit.
- verify if new configurations are available,
- verify if a new firmware update is available and
- send notifications for monitoring purposes.
- Passive: the information is sent according to the schedule.
- Active: the information is sent immediately when a parameter status changes, regardless of the periodic schedule.
Furthermore, the administrator can initiate a connection to the Mediatrix unit to perform immediate maintenance or monitoring. This will only be possible if the NAT firewall has been configured to allow communications initiated by the ACS.
The TR-069 protocol can be activated on units that are already deployed with a licence key (For more details on licences refer to theTechnical Bulletin - How to activate a licence on a Mediatrix unit published on the Media5 Documentation Portal). However, it can be enabled/disabled for a specific configuration via the Management interface.
- SetParameterValues
- GetParameterValues
- AddObject
- DeleteObject
- Download
- Reboot
- Upload
- FactoryReset
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TR-104
The Technical Report 104 (TR-104) describes the data model for provisioning of a voice-over-IP (VoIP) CPE device by an Auto Configuration Server (ACS) using the mechanism defined in TR-069. Configuring the TR-104 parameters in DGW allows you to enable or disable the provisioning parameters for VoIP CPE, such as calls statistics.
- The Media5 implementation of TR-104 is limited to the SIPEndpoint profile.
- Active notifications are not supported for all objects except when explicitly mentioned. (for more details request the Supported TR-069 Methods and Parameters document from your representative).
- Only FXS lines can be managed by the TR-104 profile.
- BRI, PRI or FXO lines are not supported by the TR-104 profile.
- A single voice profile is supported, it is instantiated by default. No creation or deletion of voice profile is allowed.
- Only a subset of parameters is currently supported.
- When TR-104 is used, it is highly recommended not to use other means of configuration (since TR-104 assumes some configuration).
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TR-106
Technical Report 106 (TR-106) is the Data Model Template for TR-069-Enabled Devices. The configuration of the TR-106 Network Interface parameter in DGW allows you to specify the network interface to be used for the TR-106 LAN profile to report network statistics.
- If no network interface is configured for TR-106, i.e. the field is left empty under Management/Misc/CWMP, the Mediatrix unit will use the network interface configured under Management/Misc/System Management/Network Interface.
- If the network interface configured for TR-106 is set to All, the Mediatrix unit will use the
network interface configured for contacting the ACS in the TR-106 Configuration section of the
CWMP Configuration table. The
TR-106 Data Model template can only be used if Root Element is set to
Device. Refer to Configuring the Report Network Statistics
(Through the TR-106 Data Model).Note: This parameter can also be set via Cwmp.RootElement.
The TR-106 Data Model template can only be used if the Cwmp.RootElement parameter value is Device.
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TR-111
Technical Report 111 (TR-111) allows the remote management of home networking devices through NAT traversal mechanisms, as defined in TR-069 Annex G (formerly in TR-111).
The TR-111 parameters defined in DGW allows the activation and configuration of a STUN server, so devices behind a NAT can be reached .
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Special Consideration for TR-104 Parameters
Some parameters under VoiceService tree requires some parameters to be configured in a defined way to work. The default values of the Mediatrix profiles interfere with TR-104 functionality. Here is a table that summarizes what needs to be configured for each parameter.
Device.Services.VoiceService.{i}.4 | {i} is limited to '1' in current implementation. |
Device.Services.VoiceService.{i}.VoiceProfile.{i}. | Both {i}s are limited to 1 in current implementation. |
Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}. | The Line.{i} starts from 1 up to the number of ports on the device, also represented by the VoiceService.{i}.Capabilities.MaxLineCount parameter. |
Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP. | |
AuthUserName / AuthPassword | The realm must be manually set or disabled. |
URI |
Setting the CRout.AutoRoutingCriteriaType to 'SipUsername' for auto routing might be required to work with the proper SIP URIs received in INVITE. SipEp.UserAgent[].Register MUST be set to 'Enable' if registration is required. This can not be enabled by TR-104 parameters (but can be by a configuration script). |
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Basic CWMP Tasks
Configuring the Access to the Auto Configuration Server (ACS)
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Configuring the CWMP Establishment (TR-069)
- Go to Management/CWMP.
-
In the CWMP Configuration table, set
the following parameters:

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Configuring VoIP Provisioning (TR-104 Data Model)
- Go to Management/CWMP.
- In the CWMP Configuration table, under the TR-104 Configuration section, from the drop down list, select Enable.
- Click Apply.
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Configuring the Access to the STUN Server TR-111
- Go to Management/CWMP.
- In the CWMP Configuration table, under the TR-111 Configuration section, complete the fields as required.
- Click Apply.
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Configuring the Report Network Statistics (Through the TR-106 Data Model)
The TR-106 Data Model template can only be used if the Cwmp.RootElement parameter value is Device.
- Go to Management/CWMP.
-
In the CWMP Configuration table,
under the TR-106 Configurationsection,
in the Network Interface field indicate the network interface referred by the TR-106 LAN
profile.
- If left empty, use the network interface configured in the Hoc.ManagementInterface parameter.
- If the Hoc.ManagementInterface parameter is set to All, use the network interface used for contacting the ACS.
- Click Apply.
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Advanced CWMP Parameters
Advanced CPE WAN Management Protocol (CWMP) Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
For ACS
- To set the parameter type validation when the ACS assigns a value to a parameter Cwmp.interopParameterTypeValidation
- To set the access mode to the local log table from the ACS: Cwmp.NlmLocalLogLogEnable
- To define the username to authenticate an ACS making a connection request to the CPE: Cwmp.ConnectionRequestUsername
- To define the password to authenticate an ACS making a connection request to the CPE: Cwmp.ConnectionRequestPassword
For TR-069 (CWMP) establishment
- To set the HTTPS transport cipher suite configuration for TR-069 (CWMP): Cwmp.TransportHttpsCipherSuite .
- To set the HTTPS Transport Tls Version configuration for TR-069 (CWMP): Cwmp.TransportHTTPSTlsVersion.
- To set the level of security to use when validating the server's certificate when connecting to the ACS using HTTPS: Cwmp.TransportCertificateValidation
For TR-111
- To set the period range, in seconds, at which STUN Binding Requests must be sent by the unit for the purpose of maintaining the STUN connection: Cwmp.tr111StunKeepAlivePeriod. The current implementation does not allow a range. The minimum and maximum values must be the same.
- To set the value of the STUN username attribute to be used in Binding Requests: Cwmp.tr111StunUsername
- To set the level of security to use when validating the server's certificate when connecting to the ACS using HTTPS: Cwmp.TransportCertificateValidation
For the MAC address
- To set the MAC address format: Cwmp.InteropMacAddressFormat.
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Supported Parameters for the GatewayDevice Object
Name | Access | Notification |
---|---|---|
InternetGatewayDevice. | Present | N/A |
The top-level object for an Internet Gateway Device. | ||
DeviceSummary | Read-only | Passive |
Explicit summary of the top-level data model of the device, including version and profile information. Returns the 'InternetGatewayDevice:1.0[](Baseline:1)' value. |
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LANDeviceNumberOfEntries | Read-only | Not supported |
Number of instances of the LANDevice object.
Returns '0'. |
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WANDeviceNumberOfEntries | Read-only | Not supported |
Number of instances of the WANDevice object.
Returns '0'. |
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InternetGatewayDevice.DeviceInfo. | Present | N/A |
This object contains general device information. | ||
Manufacturer | Read-only | Passive |
The manufacturer of the CPE. Returns the value of the HiddenManufacturer hidden parameter. |
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ManufacturerOUI | Read-only | Passive |
Organisation unique identifier of the device manufacturer. Represented as a six hexadecimal digit value, using all upper-case, and including any leading zeros. Returns the '0090F8' string. |
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ModelName | Read-only | Passive |
Model name of the CPE. Returns the value of the Dcm.UnitInfoProductName parameter. |
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Description | Read-only | Passive |
A full description of the CPE device. Returns the value of the concatenation (separated by spaces) of the following parameters:
This is the same value as the system description in SNMP. |
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ProductClass | Read-only | Passive |
Identifies the class of product for which the serial number applies to. Returns the value of the Dcm.UnitInfoProductName parameter. |
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SerialNumber | Read-only | Passive |
Serial number of the CPE. Returns the value of the Dcm.UnitInfoSerialNumber parameter. |
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HardwareVersion | Read-only | Always Passive |
Identifies the particular CPE model and version. Returns the value of the concatenation (separated by a space) of the following parameters:
|
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SoftwareVersion | Read-only | Always Active |
Identifies the software version currently installed on the CPE. Returns the value of the Fpu.MfpInstalledInfo.MfpVersion parameter (first row). |
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SpecVersion | Read-only | Passive |
Represents the version of the specification implemented by the device. Returns '1.0'. |
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ProvisioningCode | Read/Write | Active |
Identifies the primary service provider and other provisioning information. Returns the last value set by the ACS. This value is saved in an internal database. The value is initially set to an empty string and also reset to an empty string on a factory default reset. |
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UpTime | Read-only | Passive |
Time in seconds since the CPE was last restarted. Returns the system uptime in seconds. Uses the same time source as the SNMP and Web system uptime. |
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VendorLogFileNumberOfEntries | Read-only | Passive |
Returns the number of entries of the DeviceInfo.VendorLogFile.{i}. table. | ||
InternetGatewayDevice.DeviceInfo.VendorLogFile.{i}. | Present | N/A |
Table of log files. This table is informational only and does not allow the ACS to operate on these files in any way. Returns the vendor log files table, each table entry represents a Vendor log file. |
||
Name | Read-only | None |
Returns the Vendor log file name. | ||
MaximumSize | Read-only | None |
The maximum size of the log file in bytes. When the maximum file size is not defined, the value is '0'. | ||
Persistent | Read-only | None |
When the value is 'true', the log file contents are preserved when the device is restarted. When the value is 'false', the log file contents are deleted when the device is restarted. |
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InternetGatewayDevice.ManagementServer. | Present | N/A |
This object contains parameters relating to the CPE's association with an ACS. | ||
URL | Read/Write | Passive |
URL used to connect the CPE to the ACS using the CPE WAN Management Protocol. This parameter supports only a valid HTTP or HTTPS URL. Returns the value of the AcsStaticUrl parameter if the value of the AcsUrlConfigSource parameter is set to 'Static'. Otherwise, returns an empty string. When receiving a SetParameterValues command:
|
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Username | Read/Write | Passive |
Username used to authenticate the CPE when making a connection to the ACS using the CPE WAN Management Protocol. This username is only used for HTTP-based authentication of the CPE. Returns and sets the value of the Username parameter. |
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Password | Write | Passive |
Password used to authenticate the CPE when making a connection to the ACS using the CPE WAN Management Protocol. This password is only used for HTTP-based authentication of the CPE. Returns an empty string. Sets the value of the Password parameter. |
||
PeriodicInformEnable | Read/Write | Passive |
Whether or not the CPE must periodically send CPE information to the ACS by calling the Inform method. Returns and sets the value of the PeriodicInformEnable parameter. |
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PeriodicInformInterval | Read/Write | Passive |
The duration in seconds of the interval for which the CPE will attempt to connect with the ACS and call the Inform method. Applicable only if the value of the PeriodicInformEnable parameter is 'Enable'. Returns and sets the value of the PeriodicInformInterval parameter. |
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PeriodicInformTime | Read/Write | Passive |
An absolute time reference in UTC to determine when the CPE will initiate the Inform method. When the Inform method is called, it occurs at this reference time plus or minus an integer multiple of the PeriodicInformInterval parameter. The Unknown Time value indicates that no particular time reference is specified. That is, the unit locally chooses the time reference and only needs to follow the specified PeriodicInformInterval parameter. If absolute time is not available to the unit, its periodic Inform behavior is the same as if the PeriodicInformTime parameter was set to the Unknown Time value, i.e '0001-01-01T00:00:00Z'. Returns and sets the value of the PeriodicInformTime parameter. |
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ParameterKey | Read-only | Always |
Returns the last value set by the ACS in the last SetParameterValues, AddObject, or DeleteObject method call. This value is saved in an internal database. The value of the ParameterKey parameter is initially set to an empty string and reset to an empty string on a factory default reset. |
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ConnectionRequestURL | Read-only | Always Active |
The HTTP URL used by the ACS to make a Connection Request notification to the CPE. Returns the 'http://[host]:[port]' value where:
|
||
ConnectionRequestUsername | Read/Write | Active |
Username used to authenticate an ACS making a Connection Request to the CPE. Returns the value of the Cwmp.ConnectionRequestUsername parameter. |
||
ConnectionRequestPassword | Write | Passive |
Password used to authenticate an ACS making a Connection Request to the CPE. Returns an empty string. Sets the value of the Cwmp.ConnectionRequestPassword parameter. |
||
UpgradesManaged | Read/Write | Passive |
Indicates whether or not the ACS will manage upgrades for the CPE. The parameter is supported, but the functionality is not implemented. Returns the last value set by the ACS. This value is saved in an internal database. The value is initially set to 'false' and reset to 'false' on a factory default reset. |
||
UDPConnectionRequestAddress | Read-only | Active |
The public IP address and port of the unit to use for external UDP connection requests. This address is found in the STUN binding response. | ||
STUNEnable | Read/Write | Active |
Enables or disables the use of STUN by the CPE. This applies only to the use of STUN in association with the ACS to allow UDP Connection Requests. Returns the value of the Cwmp.Tr111StunEnable parameter. |
||
STUNServerAddress | Read/Write | Active |
Host name or IP address of the STUN server used by the CPE to send Binding Requests if STUN is enabled via the STUNEnable parameter. If the value of the STUNServerAddress parameter is empty and the value of the STUNEnable parameter is 'Enable', then the CPE uses the address of the ACS extracted from the host portion of the ACS URL. Otherwise, the value of the STUNServerAddress parameter is used by the CPE to send Binding Requests. Returns the address part of the Cwmp.Tr111StunServerHost parameter. |
||
STUNServerPort | Read/Write | Active |
Port number of the STUN server used by the CPE to send Binding Requests if STUN is enabled via the STUNEnable parameter. By default, this is 3478, i.e. the default STUN port. Returns the port part of the Cwmp.Tr111StunServerHost parameter. |
||
NATDetected | Read-only | Active |
When STUN is enabled, this parameter indicates whether or not the CPE has detected the address and/or the port mapping in use. A 'True' value indicates that the received MAPPED-ADDRESS in the most recent Binding Response differs from the CPE's source address and port. Returns 'True' if the value of the Cwmp.Tr111NatDetected parameter is set to 'yes'. Returns 'False' otherwise or when the STUN is disabled. |
||
STUNMinimumKeepAlivePeriod | Read/Write | Active |
Configures the first session retry wait interval, in seconds. If STUN is enabled, the minimum period, in seconds, during which the STUN Binding Requests may be sent by the CPE for the purpose of maintaining the STUN connection. Returns the minimal value of the Cwmp.Tr111StunKeepAlivePeriod parameter. |
||
STUNUsername | Read/Write | Active |
The value of the STUN USERNAME attribute to be used in Binding Requests when STUN is enabled. If the value of the STUNUsername parameter is empty, the CPE sends the STUN Binding Requests without the STUN USERNAME attribute. Returns and sets the value of the Cwmp.Tr111StunUsername parameter. |
||
InternetGatewayDevice.Time. | Present | N/A |
This object contains parameters relating to the NTP or SNTP time client on the CPE. | ||
NTPServer1 | Read/Write | Active |
First NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=1].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=1].HostName parameter and sets the value of the Hoc.SntpConfigSource parameter to 'Static'. |
||
NTPServer2 | Read/Write | Active |
Second NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=2].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=2].HostName parameter. |
||
NTPServer3 | Read/Write | Active |
Third NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=3].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=3].HostName parameter. |
||
NTPServer4 | Read/Write | Active |
Fourth NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=4].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=4].HostName parameter. |
||
CurrentLocalTime | Read-only | Passive |
The current date and time in the CPE's local time zone. Returns the value of the Hoc.SystemTime parameter |
||
InternetGatewayDevice.Services. | Present | N/A |
Refer to the Services Parameters table. |
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Supported Parameters for the GatewayDevice Object
Name | Access | Notification |
---|---|---|
Device. | Present | N/A |
The top-level object for a Device. | ||
DeviceSummary | Read-only | Always Passive |
Explicit summary of the top-level data model of the device, including version and profile information. Returns the 'Device:1.0[](Baseline:1)' value. |
||
Device.DeviceInfo. | Present | N/A |
Manufacturer | Read-only | Passive |
The manufacturer of the CPE. Returns the value of the HiddenManufacturer hidden parameter. |
||
ManufacturerOUI | Read-only | Passive |
Organisation unique identifier of the device manufacturer. Represented as a six hexadecimal digit value, using all upper-case, and including any leading zeros. Returns the '0090F8' string. |
||
ModelName | Read-only | Passive |
Model name of the CPE. Returns the value of the Dcm.UnitInfoProductName parameter. |
||
Description | Read-only | Passive |
A full description of the CPE device. Returns the value of the concatenation (separated by spaces) of the following parameters:
This is the same value as the system description in SNMP. |
||
ProductClass | Read-only | Passive |
Identifies the class of product for which the serial number applies to. Returns the value of the Dcm.UnitInfoProductName parameter. |
||
SerialNumber | Read-only | Passive |
Serial number of the CPE. Returns the value of the Dcm.UnitInfoSerialNumber parameter. |
||
HardwareVersion | Read-only | Always Passive |
Identifies the particular CPE model and version. Returns the value of the concatenation (separated by a space) of the following parameters:
|
||
SoftwareVersion | Read-only | Always Active |
Identifies the software version currently installed on the CPE. Returns the value of the Fpu.MfpInstalledInfo.MfpVersion parameter (first row). |
||
DeviceStatus | Read-only | Passive |
Current operational status of the device.
Returns the 'UP' string when the device is up. |
||
UpTime | Read-only | Passive |
Time in seconds since the CPE was last restarted. Returns the system uptime in seconds. Uses the same time source as the SNMP and Web system uptime. |
||
VendorLogFileNumberOfEntries | Read-only | Passive |
Returns the number of entries of the DeviceInfo.VendorLogFile.{i}. table. | ||
Device.DeviceInfo.VendorLogFile.{i}. | Present | N/A |
Table of log files. This table is informational only and does not allow the ACS to operate on these files in any way. Returns the vendor log files table, each table entry represents a Vendor log file. |
||
Name | Read-only | None |
Returns the Vendor log file name. | ||
MaximumSize | Read-only | None |
The maximum size of the log file in bytes. When the maximum file size is not defined, the value is '0'. | ||
Persistent | Read-only | None |
When the value is 'true', the log file contents are preserved when the device is restarted. When the value is 'false', the log file contents are deleted when the device is restarted. |
||
Device.GatewayInfo. 5 | Present | N/A |
This object contains information associated with a connected Internet Gateway Device. |
||
ManufacturerOUI | Read-only | Active |
Organizationally unique identifier of the associated Internet Gateway Device. An empty string indicates that there is no associated Internet Gateway Device that has been detected. Returns the GatewayManufacturerOui value received via the suboption 4 of the 'Vendor-Specific Information'. Returns an empty value if the contents of the suboption is not present or incorrect. |
||
ProductClass | Read-only | Active |
Identifier of the product class of the associated Internet Gateway Device. An empty string indicates either that there is no associated Internet Gateway Device that has been detected, or the Internet Gateway Device does not support the use of the ProductClass parameter. Returns the GatewayProductClass value received via the suboption 6 of the 'Vendor-Specific Information'. Returns an empty value if the contents of the suboption is not present or incorrect. |
||
SerialNumber | Read-only | Active |
Serial number of the associated Internet Gateway Device. An empty string indicates that there is no associated Internet Gateway Device that has been detected. Returns the value of GatewaySerialNumber parameter received via the suboption 5 of the 'Vendor-Specific Information'. Returns an empty value if the contents of the suboption is not present or incorrect. |
||
Device.Time. | Present | N/A |
This object contains the parameters related to the NTP or SNTP time client on the CPE. | ||
NTPServer1 | Read/Write | Active |
First NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=1].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=1].HostName parameter and sets the value of the Hoc.SntpConfigSource parameter to 'Static'. |
||
NTPServer2 | Read/Write | Active |
Second NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=2].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=2].HostName parameter. |
||
NTPServer3 | Read/Write | Active |
Third NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=3].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=3].HostName parameter. |
||
NTPServer4 | Read/Write | Active |
Fourth NTP timeserver. Either a host name or IP address. Returns the value of the Hoc.SntpServersInfo[Priority=4].HostName parameter. Sets the value of the Hoc.StaticSntpServers[Priority=4].HostName parameter. |
||
CurrentLocalTime | Read-only | Passive |
The current date and time in the CPE's local time zone. Returns the value of the Hoc.SystemTime parameter. |
||
LocalTimeZone | Read/Write | Active |
The local time zone definition.
Returns and sets the value of the Hoc.StaticTimeZone parameter. |
||
Device.LAN. | Present | N/A |
This object contains parameters related to the IP-based LAN connectivity of a device. Returns the informations of the network interface used by the device for connecting to the ACS.
|
||
AddressingType | Read-only | Passive |
The method used to assign an address to this interface. Returns 'DHCP' if the value of the Bni.NetworkInterface.ConnectionType parameter is 'ipDhcp'. Returns 'Static' if the value of the Bni.NetworkInterface.ConnectionType parameter is 'ipStatic'. Returns an empty string if the Bni.NetworkInterface.ConnectionType parameter contains any other value. |
||
IPAddress | Read-only | Always Active |
The current IP address set on this interface.
Returns the IP address from the Bni.NetworkInterfacesStatus.IpAddr parameter. |
||
SubnetMask | Read-only | Passive |
The current subnet mask set to this interface.
Returns the IP address mask from the Bni.NetworkInterfacesStatus.IpAddr parameter. |
||
DefaultGateway | Read-only | Passive |
The IP address of the current default gateway for this interface.
Returns the default gateway from the Hoc.DefaultRouterInfo parameter. |
||
DNSServers | Read-only | Passive |
List of DNS server IP addresses for this interface.
Returns the list of DNS server IP addresses, separated by a coma, defined in the Hoc.DnsServersInfo parameter. |
||
MACAddress | Read-only | Always Active |
The physical address of this interface.
Returns the value of the Dcm.unitInfoMacAddress parameter. |
||
Device.LAN.Stats. | Present | N/A |
This object contains statistics for the network interface used for connecting the device to the ACS. | ||
ConnectionUpTime | Read-only | Passive |
The time in seconds during which this network interface has been connected.
Returns the value of the Bni.NetworkInterfacesStatus.ConnectionUptime parameter. |
||
TotalBytesSent | Read-only | Passive |
Total number of IP payload bytes sent over this interface since the device was last restarted, as specified in the DeviceInfo.UpTime parameter. Returns the value of the Bni.InterfaceStatistics.TxBytes parameter. |
||
TotalBytesReceived | Read-only | Passive |
Total number of IP payload bytes received over this interface since the device was last restarted, as specified in the DeviceInfo.UpTime parameter. Returns the value of the Bni.InterfaceStatistics.RxBytes parameter. |
||
TotalPacketsSent | Read-only | Passive |
Total number of IP packets sent over this interface since the device was last restarted, as specified in the DeviceInfo.UpTime parameter. Returns the value of the Bni.InterfaceStatistics.TxPackets parameter. |
||
TotalPacketsReceived | Read-only | Passive |
Total number of IP packets received over this interface since the device was last restarted, as specified in the DeviceInfo.UpTime parameter. Returns the value of the Bni.InterfaceStatistics.RxPackets parameter. |
||
Device.ManagementServer. | Present | N/A |
This object contains parameters relating to the CPE's association with an ACS. | ||
URL | Read/Write | Passive |
URL used to connect the CPE to the ACS using the CPE WAN Management Protocol. This parameter supports only a valid HTTP or HTTPS URL. Returns the value of the AcsStaticUrl parameter if the value of the AcsUrlConfigSource parameter is set to 'Static'. Otherwise, returns an empty string. When receiving a SetParameterValues command:
|
||
Username | Read/Write | Passive |
Username used to authenticate the CPE when making a connection to the ACS using the CPE WAN Management Protocol. This username is only used for HTTP-based authentication of the CPE. Returns and sets the value of the Username parameter. |
||
Password | Write | Passive |
Password used to authenticate the CPE when making a connection to the ACS using the CPE WAN Management Protocol. This password is only used for HTTP-based authentication of the CPE. Returns an empty string. Sets the value of the Password parameter. |
||
PeriodicInformEnable | Read/Write | Passive |
Whether or not the CPE must periodically send CPE information to the ACS by calling the Inform method. Returns and sets the value of the PeriodicInformEnable parameter. |
||
PeriodicInformInterval | Read/Write | Passive |
The duration in seconds of the interval for which the CPE will attempt to connect with the ACS and call the Inform method. Applicable only if the value of the PeriodicInformEnable parameter is 'Enable'. Returns and sets the value of the PeriodicInformInterval parameter. |
||
PeriodicInformTime | Read/Write | Passive |
An absolute time reference in UTC to determine when the CPE will initiate the Inform method. When the Inform method is called, it occurs at this reference time plus or minus an integer multiple of the PeriodicInformInterval parameter. The Unknown Time value indicates that no particular time reference is specified. That is, the unit locally chooses the time reference and only needs to follow the specified PeriodicInformInterval parameter. If absolute time is not available to the unit, its periodic Inform behavior is the same as if the PeriodicInformTime parameter was set to the Unknown Time value, i.e '0001-01-01T00:00:00Z'. Returns and sets the value of the PeriodicInformTime parameter. |
||
ParameterKey | Read-only | Always |
Returns the last value set by the ACS in the last SetParameterValues, AddObject, or DeleteObject method call. This value is saved in an internal database. The value of the ParameterKey parameter is initially set to an empty string and reset to an empty string on a factory default reset. |
||
ConnectionRequestURL | Read-only | Always Active |
The HTTP URL used by the ACS to make a Connection Request notification to the CPE. Returns the 'http://[host]:[port]' value where:
|
||
ConnectionRequestUsername | Read/Write | Active |
Username used to authenticate an ACS making a Connection Request to the CPE. Returns the value of the Cwmp.ConnectionRequestUsername parameter. |
||
ConnectionRequestPassword | Write | Passive |
Password used to authenticate an ACS making a Connection Request to the CPE. Returns an empty string. Sets the value of the Cwmp.ConnectionRequestPassword parameter. |
||
UpgradesManaged | Read/Write | Passive |
Indicates whether or not the ACS will manage upgrades for the CPE. The parameter is supported, but the functionality is not implemented. Returns the last value set by the ACS. This value is saved in an internal database. The value is initially set to 'false' and reset to 'false' on a factory default reset. |
||
UDPConnectionRequestAddress | Read-only | Active |
The public IP address and port of the unit to use for external UDP connection requests. This address is found in the STUN binding response. | ||
STUNEnable | Read/Write | Active |
Enables or disables the use of STUN by the CPE. This applies only to the use of STUN in association with the ACS to allow UDP Connection Requests. Returns the value of the Cwmp.Tr111StunEnable parameter. |
||
STUNServerAddress | Read/Write | Active |
Host name or IP address of the STUN server used by the CPE to send Binding Requests if STUN is enabled via the STUNEnable parameter. If the value of the STUNServerAddress parameter is empty and the value of the STUNEnable parameter is 'Enable', then the CPE uses the address of the ACS extracted from the host portion of the ACS URL. Otherwise, the value of the STUNServerAddress parameter is used by the CPE to send Binding Requests. Returns the address part of the Cwmp.Tr111StunServerHost parameter. |
||
STUNServerPort | Read/Write | Active |
Port number of the STUN server used by the CPE to send Binding Requests if STUN is enabled via the STUNEnable parameter. By default, this is 3478, i.e. the default STUN port. Returns the port part of the Cwmp.Tr111StunServerHost parameter. |
||
NATDetected | Read-only | Active |
When STUN is enabled, this parameter indicates whether or not the CPE has detected the address and/or the port mapping in use. A 'True' value indicates that the received MAPPED-ADDRESS in the most recent Binding Response differs from the CPE's source address and port. Returns 'True' if the value of the Cwmp.Tr111NatDetected parameter is set to 'yes'. Returns 'False' otherwise or when the STUN is disabled. |
||
STUNMinimumKeepAlivePeriod | Read/Write | Active |
Configures the first session retry wait interval, in seconds. If STUN is enabled, the minimum period, in seconds, during which the STUN Binding Requests may be sent by the CPE for the purpose of maintaining the STUN connection. Returns the minimal value of the Cwmp.Tr111StunKeepAlivePeriod parameter. |
||
STUNUsername | Read/Write | Active |
The value of the STUN USERNAME attribute to be used in Binding Requests when STUN is enabled. If the value of the STUNUsername parameter is empty, the CPE sends the STUN Binding Requests without the STUN USERNAME attribute. Returns and sets the value of the Cwmp.Tr111StunUsername parameter. |
||
Device.Services. | Present | N/A |
Refer to the Services Parameters table. |
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Transport Layer Security (TLS) Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
For certificate transfert
- To set the HTTPS transfer cipher suite for certificate transfer: Cert.TransferHttpsCipherSuite
- To set the HTTPS transfer Tls Version for certificate transfer:: Cert.TransferHttpsTlsVersion
- To set the level of security to use when validating the server's certificate when connecting to the ACS using HTTPS: Cwmp.TransportCertificateValidation
For file transfer
- To set the HTTPS transfer cipher suite for file transfer: File.TransferHttpsCipherSuite
- To set the HTTPS transfer Tls Version configuration for file transfer: File.TransferHttpsTlsVersion
For DGW Web access
- To set the Https Cipher Suite for secure DGW Web access: Web.HttpsCipherSuite.
- To set the Http Mode used for DGW Web access: Web.HttpMode
- To select the Secure Server Port used to access the DGW Web interface: Web.SecureServerPort
- To set the HTTPS Cipher Suite for secure DGW Web access: Web.HttpsCipherSuite
- To set the Tls Version used for secure DGW Web access: Web.TlsVersion
For SIP TLS transport
- To set the TLS transport cipher suite used for secure SIP transport: SipEp.TransportTlsCipherSuite
- To set Transport Tls Version used for secure SIP transport: SipEp.TransportTlsVersion
- To set TLS client authentication: SipEp.InteropTlsClientAuthenticationEnable
For TR-069 (CWMP) establishment
- To set the HTTPS transport cipher suite configuration for TR-069 (CWMP): Cwmp.TransportHttpsCipherSuite
- To set the HTTPS Transport Tls Version configuration for TR-069 (CWMP): Cwmp.TransportHTTPSTlsVersion
- To set the level of security to use when validating the server's certificate when connecting to the ACS using HTTPS: Cwmp.TransportCertificateValidation
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Access Control
Basic Access Control Concepts
Important Information
- The Management/Access Control page is only accessible if you have admin Access Rights.
- A maximum of 10 users can be added in the Users table.
- When a partial reset is triggered,
- the default accounts are restored, with their default values and access rights.
- the Radius authentication is disabled.
- The password is case sensitive. All characters are allowed.
- The username is case sensitive.
- The Mediatrix unit’s Radius server settings do not support IPv6.
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Access Right Permissions
Access Right | Observer | User | Admin |
---|---|---|---|
Read Configuration Parameters | √ | √ | √ |
Modify Configuration Parameters | √ | √ | |
Read/Write Passwords, Secrets, Secret Keys | √ | ||
Change Access Rights | √ | ||
Execute Configuration Scripts | √ | √ | |
Export Configuration | √ | √ | |
Backup/Restore Configuration | √ | ||
Firmware Updates and Rollback | √ |
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Protection Against Brute Force Login Attempts
Mediatrix units have a protection against brute force login attempts.
When this protection is enabled, a user account is temporarily locked after repetitive login failures. The protection is enabled by default. The maximum number of login attempts before locking the user's account and the duration of the lock are configurable.
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Service Access Control Type
It is possible to define the type of authentication and accounting to use for the CLI, SNMP, and Web services. In other words, it is possible to decide if authentication or accounting requests are sent to a RADIUS server or validated against the username and password stored locally in the Users table of DGW.
- Authentication provides a way of identifying a user, typically by having the user enter a valid user name and valid password before access is granted.
- Accounting measures the resources a user consumes during access. This can include the amount of system time or the amount of data a user has sent and/or received during a session.
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Radius Servers used For Authentication
A radius server can be used for Authentication for the CLI and the Web services
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Radius Servers used For Accounting
A radius server can be used for Accounting Requests for the CLI, Snmp, and Web services
It is possible to use up to three Accounting Radius servers for each of the CLI, the Snmp, and Web services.
- the Priority #1 Radius server. If the accounting fails or the request reaches the timeout, the accounting request is sent to:
- the Priority #2 Radius server. If the accounting request fails or the request reaches the timeout, the accounting request is sent to:
- Priority #3 Radius server. If the accounting request fails or the request reaches the timeout, the accounting request is dropped.
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Advanced Access Control Parameters
- using a MIB browser
- using the CLI
- creating a configuration script containing the configuration parameters
- To set the maximum login attempts allowed before locking the account: Aaa.LoginLockedMaxRetry
- To set how much time the account will remain locked: Aaa.LoginLockedTimeout
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Basic Access Control Tasks
Adding a User to DGW

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Changing the Rights of a DGW User
- Go to Management/Access Control.
- In the Users table, from the Access Rights selection list, choose the appropriate rights.
- Click Apply.
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Changing the Login Password
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Deleting a User from DGW

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Protecting User Accounts Against Brute Force Attacks
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Setting the Access Control Type of the CLI Service

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Setting the Access Control Type of the Snmp Service

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Setting the Access Control Type of the Web Service

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File
File Basic Concepts
File Manager
The DGW file manager allows for the importation of data files used by DGW.
When a DGW Mediatrix unit is purchased, several preset files are already present in the Internal files table (Management/File). these can be the default rulesets, a configuration script, etc.
Path | Type of File | Platforms |
---|---|---|
moh/ | MP3 files for Music On Hold (MOH) | All except the Mediatrix 4102S |
conf/ | Configuration scripts and backup files | All except the Mediatrix 4102S |
sbc/rulesets/ | Session Border Controller rulesets | Sentinel 100 and 400 |
vm/drives/ | Virtual Machine images | Sentinel 400 |
vm/images/ | Bootable ISO images | Sentinel 400 |
Path | Type of File | Platforms |
---|---|---|
nlm/diag | Log files generated by System/Diagnostic (Nlm service) | All except the Mediatrix 4102S |
nlm/logs | Notification files generated by System/Event Log (Nlm service) | All except the Mediatrix 4102S and Mediatrix C7 Series |
nlm/pcaptures | Network capture files generated by System/Packet Capture (Nlm service) | All except the Mediatrix 4102S |
sbc/logs | SIP/RTP traffic logged by the Log received traffic action in Call Agent Rulesets (Sbc service) | Sentinel 100 and 400 |
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Large Files - Important Information
- Closing the unit Web page will stop the transfer.
- Web Browsers do not always display a progress indicator, be cautious not to accidentally abort the file transfer.
- Transfer speed depends on connection speed, therefore transfer can take several minutes.
- Free disk space in computer must be at least equivalent to the size of the imported file.
- Anti-virus programs sometimes abort large file transfers; consider closing it.
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File Basic Tasks
Importing an ISO image to the Unit File Management System
If the ISO Image is larger than 10mb, use the Importing an ISO Image Larger than 10mb to the Unit File Management System.

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Importing an ISO Image Larger than 10mb to the Unit File Management System

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Deleting a File from DGW
- In the Web interface of DGW, go to Management/File.
-
Click
located on the same line of the file you wish to remove.
- Click OK.
- A message will be displayed asking: Are you sure you want to delete this file?, click OK.

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Importing an MP3 File from a Web Browser
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Importing a Virtual Machine Image to the Unit File Management System
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Downloading a Local Capture File From the Mediatrix Unit to Your PC
- Go to Management/File.
- In the Internal files table, click the name of the file you have given to your capture.
- Save your capture file.
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Misc
Disabling the Rescue Interface

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Online Help
If you are not familiar with the meaning of the fields and buttons, click Show Help, located at the upper right corner of the Web page. When activated, the fields and buttons that offer online help will change to green and if you hover over them, the description will bedisplayed.
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Acronyms
3rd Generation cellular data network
4th Generation cellular data network
Authentication, Authorization and Accounting
Access Concentrator
Auto Configuration Server
Asymmetric Digital Subscriber Line
Advanced Encryption Standard
Automatic Gain Control
Automatic Number Identification
Advice of Charge End-of-Call
Basic Encoding Rules
Basic Network Interfaces
Basic Rate Interface
Certification Authority
Channel Associated Signalling
Country Code
Completion of Call to Subscriber
Completion of Calls on No Reply
Call Detail Record
Call Detail Record
Certificate Manager
Challenge Handshake Authentication
Command Line Interface
Calling Line Information Presentation
Calling Line Information Restriction
Comfort Noise Generator
Calling Name Identity Presentation
Connected Line Identification Presentation
Connected Line Identification Restriction
Configuration Manager
Call Routing
Conjugate Structure-Algebraic Excited Linear Prediction
CPE Wan Management Protocol
Device Control Manager
Distinguished Encoding Rules
Dynamic Host Configuration Protocol Server
Dynamic Host Configuration Protocol
Dynamic Host Configuration Protocol
Dialed Number Identification Service
Domain Name Server
Differentiated Services Code Point
Digital Subscriber Lines
Digital Subscriber Signalling System No.1
Daylight Saving Time
Dual Tone Multi-Frequency
European PRI digital signal carrier. 32 channels (30 voice channels + synchronization and signaling)
E and M Channel Associated Signaling
Element Management System
Endpoint Administration
Endpoint Services
Ethernet Manager
Federal Communication Commission
File Manager
Firmware Pack Updater
Fully Qualified Domain Name
Frequency Shift Keying
Foreign Exchange Office
Foreign Exchange Service/Station
Greenwich Mean Time
High-bit-rate Digital Subscriber Line
Host Configuration
Hyper Text Markup Language
Hyper Text Transfer Protocol
HTTP over the Transport Layer Security
Hertz
Internet Control Message Protocol
Institute of Electrical and Electronics Engineers
Internet Engineering Task Force
IP Multimedia Core Network Subsystem
Internet Protocol
Internet Protocol Private Branch eXchange
IP routing
IP Synchronisation
Internet Protocol version 4
Internet Protocol version 6
Integrated Services Digital Network
Integrated Services Digital Network
Information Transfer Capability
Internet Service Provider
International Telecommunication Union
KiloBits Per Second
Local Area Network
Light Emitting Diode
Local Firewall
Local Number Portability
Link Layer Discovery Protocol-Media Endpoint Discovery
Link Layer Discovery Protocol
Link Layer Discovery Protocol
Local Quality of Service
Media Access Control
Multi-Frequency Code
Management Information Base
Multimedia Internet KEYing
Media IP Transport
Music on Hold
Multiple Subscriber Number
Maximum Trasnmission Unit
Message Waiting Indicator
Network Termination. The endpoint on the telephone switch side.
Network Address Translation
Network Address Translation
Network Firewall
Notifications and Logging Manager
Network Traffic Control
Network Time Protocol
Online Certificate Status Protocol
Password Authentication Protocol
Private Branch eXchange
Pulse Code Modulation
Process Control Manager
Privacy Enhanced Mail
Presentation Indicator
Private Integrated Services Network
Portable Operating System Interface
Plain Old Telephony System
Plain Old Telephony System Line
Point-to-Point Protocol
Point-to-Point Protocol over Ethernet
Provisional Response Acknowledgement
Primary Rate Interface
Public Switched Telephony Network
Quality of Service
R2 Channel Associated Signaling
Remote Authentication Dial-In User Service
Rate-adaptive Digital Subscriber Line
Request For Comment
Realtime Control Protocol
Real Time Transport Protocol
Session Border Controller
Session Border Controller
Service Controller Manager
Switched Circuit Network
Secure Description
Session Description Protocol
Secure Hash Algorithm
Screening Indicator
Session Initiation Protocol
SIP Endpoint
Service Level Agreement
Simple Network Management Protocol
Simple Network Management Protocol
Simple Network Time Protocol
Secure Real-Time Transport Control Protocol
Secure Real-Time Transport Protocol
Secure Socket Shell
Secure Socket Layer
Standard Saving Time
Spanning Tree Protocol
Terminal Balance Return Loss
Transmission Control Protocol
Transmission Control Protocol/Internet Protocol
Time-division multiplexing
Terminal Equipment, the endpoint on the customer side
Terminal Endpoint Identifier
Telephony Interface
Trivial File Transfer Protocol
Transport Layer Security
Type of Number
User Datagram Protocol
Unit Manager Network
User-Network Interface
Uniform Resource Identifier
Universal Time Coordinated
Voice Activity Detector
Virtual Local Area Network
Voice Over IP
Virtual Private Network
Wide Area Network
Wireless IEEE802.11 a, b or g network
Web
Wireless Local Area Network
Windows Internet Name Service
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Glossary
3rd Generation cellular data network. A technology used for mobile devices and mobile telecommunications use services and networks. It is based on a set of standards that comply with the International Mobile Telecommunications-2000 (IMT-2000) specifications by the International Telecommunication Union.
4th Generation cellular data network. A technology used for mobile devices and mobile telecommunications use services and networks. It is based on a set of standards that comply with the International Mobile Telecommunications-2000 (IMT-2000) specifications by the International Telecommunication Union.
An Ethernet local area network that works on twisted pair wiring. 10 indicates the transmission speed of 10 Mbps, while base refers to basehand signaling i.e. that it only carries Ethernet signals. T refers to the twisted pair of cables this technology uses.
A version of Ethernet that operates at 10 times the speed of a 10 BaseT Ethernet. 100 indicates the transmission speed of 100 Mbps, while base refers to basehand signaling i.e. that it only carries Ethernet signals. T refers to the twisted pair of cables this technology uses.
A version of Ethernet that operates at 10 times the speed of a 100 BaseT Ethernet. 1000 indicates the transmission speed of 1,000 Mbps, while base refers to basehand signaling i.e. that it only carries Ethernet signals. T refers to the twisted pair of cables this technology uses.
System service that authenticates a user and grants rights to perform specific tasks on the system.
Device capable of sending or receiving data over a data communications channel.
Accounting measures the resources a user consumes during access. This can include the amount of system time or the amount of data a user has sent and/or received during a session. Accounting is carried out by logging session statistics and usage information and is used for authorization control, billing, trend analysis, resource utilisation, and capacity planning activities.
The ITU-T companding standard used in the conversion between analog and digital signals in PCM (Pulse Code Modulation) systems. A-law is used primarily in European telephone networks and contrasts with the North American mu (μ)-law standard. See also mu (μ)-law.
In Channel Associated Signaling, the sending of the calling numbers is known as Automatic Number Identification.
In ISDN signaling, an Advice Of Charge (AOC-D) message is sent to advise of the current charge (D)uring a call or an AOC-E message is sent to advise of the total charge at the (E)nd of a call.
The preliminary digits that a user must dial to be connected to a particular outgoing trunk group or line.
Authentication provides a way of identifying a user, typically by having the user enter a valid user name and valid password before access is granted. The process of authentication is based on each user having a unique set of criteria for gaining access. The AAA server compares a user's authentication credentials with other user credentials stored in a database. If the credentials match, the user is granted access to the network. If the credentials do not match, authentication fails and network access is denied.
Basic Rate Interface or Basic Rate Access is an Integrated Services Digital Network (ISDN) configuration defined in the physical layer standard I.430 produced by the ITU. This configuration consists of two 64 kbit/s “bearer” channels (B channels) and one 16 kbit/s “data” channel (D channel). The B channels are used for voice or user data, and the D channel is used for any combination of: data, control/signalling and X.25 packet networking. The two B channels can be bonded together giving a total data rate of 128 kbit/s. BRI is the kind of ISDN interface most likely to be found in a residential service.
User service managing the layer 3 network interfaces.
User service allowing the administrator to generate custom call notifications with information such as endpoints, point of origin, duration, etc.
Calls through the unit can be routed based on a set of routing criteria.
User service manipulating properties and routing calls between the telephony interfaces and the SIP endpoints.
System service that manages the security certificates used for the authentication of the unit and its peers before establishing a secure connection.
With this method of signaling, each traffic channel has a dedicated signaling channel. In other words the signaling for a particular traffic circuit is permanently associated with that circuit. Channel-associated call control is still widely used today mostly in South America, Africa, Australia and in Europe.
User service allowing the administrator to manage the unit using the SSH or TELNET protocols.
System service executing configuration scripts as well as performing backup/restore of the unit's configuration.
In international direct telephone dialing, a code that consists of 1-, 2-, or 3-digit numbers in which the first digit designates the region and succeeding digits, if any, designate the country.
User service allowing the administrator to manage the unit using the TR-069 protocol.
System service managing the auto-detection and identification of unit hardware components as well as the licence activation keys.
User service managing a DHCP server on each network interface.
DNIS is a telephone service that identifies for the receiver of a call the number that the caller dialed. It's a common feature of 800 and 900 lines. If you have multiple 800 or 900 numbers to the same destination, DNIS tells which number was called. DNIS works by passing the touch tone digits (dual tone multi frequency or MF digits) to the destination where a special facility can read and display them or make them available for call center programming.
A technology for bringing high-bandwidth information to homes and small businesses over ordinary copper telephone lines. xDSL refers to different variations of DSL, such as ADSL, HDSL, and RADSL.
DER for ASN.1, as defined in ITU-T Recommendation X.509, is a more restrictive encoding standard than the alternative BER (Basic Encoding Rules) for ASN.1, as defined in ITU-T Recommendation X.209, upon which DER is based. Both BER and DER provide a platform-independent method of encoding objects such as certificates and messages for transmission between devices and applications
Internet service that translates domain names into IP addresses. For instance, the domain name www.example.com might translate to 198.105.232.4.
In telephone systems, multi-frequency signaling in which a standard set combinations of two specific voice band frequencies, one from a group of four low frequencies and the other from a group of four higher frequencies, are used. Although some military telephones have 16 keys, telephones using DTMF usually have 12 keys. Each key corresponds to a different pair of frequencies. Each pair of frequencies corresponds to one of the ten decimal digits, or to the symbol “#” or “*”, the “*” being reserved for special purposes.
TCP/IP protocol that enables PCs and workstations to get temporary or permanent IP addresses (out of a pool) from centrally-administered servers.
European PRI digital signal carrier. 32 channels (30 voice channels + synchronization and signaling).
Technique that allows for the isolation and filtering of unwanted signals caused by echoes from the main transmitted signal.
Service managing the E and M CAS telephony interfaces.
User service allowing for high-level management of telephony endpoints.
User service managing the telephony services of each endpoint.
System service managing the unit's Ethernet link interfaces.
The restoration of the original state of a system after failing.
An automatic switch to a secondary system on failure of the primary system, used to insure the availability of critical resources, involving a parallel backup system running at all times so that, upon the detection of primary system failure, processing is automatically shifted to the backup.
Refers to methods for detecting that a remote party has hung up. This is also known as Hangup Supervision. There are several methods that may be used by a PBX/ACD/CO to signal that the remote party has hung up, including clear down tone, or a wink.
U.S. Government regulatory body for radio, television, interstate telecommunications services, and international services originating in the United States.
System service allowing the administrator to manage the files stored on the unit.
A firewall in a networked environment blocks some communications forbidden by the security policy. It has the basic task of controlling traffic between different zones of trust. Typical zones of trust include the Internet (a zone with no trust) and an internal network (a zone with high trust).
A network-provided service in which a telephone in a given local exchange area is connected, via a private line, to a central office in another, i.e., “foreign”, exchange, rather than the local exchange area’s central office. This is the office end of an FX circuit (frequently a PBX).
A network-provided service in which a telephone in a given local exchange area is connected, via a private line, to a central office in another, i.e., “foreign”, exchange, rather than the local exchange area’s central office. This is the station (telephone) end of an FX circuit. An FXS port will provide dial tone and ring voltage.
System service managing firmware upgrade, downgrade and rollback operations.
Refers to a transmission using two separate channels for transmission and reception and that can transmit in both ways at the same time. See also Half Duplex Connection .
ITU-T recommendation for the physical and electrical characteristics of hierarchical digital interfaces at rates up to 140Mbit/s.
ITU-T recommendation for synchronous frame structures on G.703 interfaces up to 45Mbit/s. The conventional use of G.704 on a 2Mbit/s primary rate circuit provides 30 discrete 64kbit/s channels, with a further 64kbit/s channel available for common channel signaling
Algorithm designed to transmit and receive A-law PCM (Pulse Code Modulation) voice at digital bit rates of 48 kbps, 56 kbps, and 64 kbps. It is used for digital telephone sets on digital PBX and ISDN channels.
A codec that provides the greatest compression, 5.3 kbps or 6.3 kbps; typically specified for multimedia applications such as H.323 videoconferencing.
An implementation of ITU-T G.726 standard for conversion linear or A-law or μ-law PCM to and from a 40, 32, 24 or 16 kbit/s channel.
A codec that provides near toll quality at a low delay which uses compression to 8 kbps (8:1 compression rate).
A device linking two different types of networks that use different protocols (for example, between the packet network and the Public Switched Telephone Network).
Refers to a transmission using the same channel for both transmission and reception therefore it can't transmit and receive at the same time. See also Full Duplex Connection.
System service managing the IP host parameters and other system settings.
The hunt group hunts an incoming call to multiple interfaces. It accepts a call routed to it by a routing table or directly from an interface and creates another call that is offered to one of the configured destination interfaces. If this destination cannot be reached, the hunt group tries another destination until one of the configured destinations accepts the call. When an interface accepts a call, the interface hunting is complete and the hunt group service merges the original call with the new call to the interface that accepted the call.
Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.
A request to the network exchange equipment to ask if a particular type of encoding is allowed. It is also called ISDN bearer capability or ISDN service.
A set of digital transmission protocols defined by the international standards body for telecommunications, the ITU-T (formerly called the CCITT). These protocols are accepted as standards by virtually every telecommunications carrier all over the world. ISDN complements the traditional telephone system so that a single pair of telephone wires is capable of carrying voice and data simultaneously. It is a fully digital network where all devices and applications present themselves in a digital form.
User service managing the ISDN parameters for BRI and PRI telephony interfaces.
Organization based in Geneva, Switzerland, that is the most important telecom standards-setting body in the world.
Internet-Drafts are working documents of the IETF, its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts.
A standard describing software that keeps track of the Internet’s addresses for different nodes, routes outgoing messages, and recognizes incoming messages.
Allows the packet to be forwarded to a specific network based on the packet’s criteria (source IP address and source Ethernet link).
User service managing the unit's IP routing table.
User service controlling the IP media synchronization using clock reference signals sent over IP.
IPv4 (Internet Protocol version 4) is a 32-bit address internet protocol.
IPv6 (Internet Protocol version 6) is the successor to the common Internet Protocol (IPv4). IPv6’s is a 128-bit address Internet protocol.
A distortion caused by the variation of a signal from its references which can cause data transmission errors, particularly at high speeds.
A semiconductor diode that emits light when a current is passed through it.
User service managing the IEEE 802.1ab protocol used for advertising the unit's capabilities on the network.
Data-only communications network confined to a limited geographic area, with moderate to high data rates. See also WAN .
Allows you to dynamically create and configure rules to filter incoming packets with the unit as destination. The traffic is analysed and filtered by all the configured rules.
User service allowing the administrator to filter incoming packets with the unit as final destination.
System service managing the QOS parameters applicable to the unit.
Specifications containing definitions of management information so that networked systems can be remotely monitored, configured and controlled.
A layer 2 address, 6 bytes long, associated with a particular network device; used to identify devices in a network; also called hardware or physical address.
The Media Interface is used for media ( RTP, UDPTL) processing.
User service managing the voice and data encodings over the IP network.
The PCM (Pulse Code Modulation) voice coding and companding standard used in Japan and North America. See also A-Law.
User service managing the option to play an audio file when a telephony endpoint is on hold.
A group of computers, terminals, and other devices and the hardware and software that enable them to exchange data and share resources over short or long distances. A network can consist of any combination of local area networks (LAN) or wide area networks (WAN).
- Source rules: They are applied on the source address of outgoing packets.
- Destination rules: They are applied on the destination address of incoming packets.
User service allowing the administrator to change the source or the destination IP address of a packet.
Allows dynamically creating and configuring rules to filter packets forwarded by the unit. Since this is a network firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and filtered by all the rules configured.
User service allowing the administrator to filter traffic that is routed between networks.
User service allowing the administrator to perform traffic shaping on the network interfaces.
User service managing the routing and filtering of the unit's event notification messages.
A line condition caused when a telephone handset is removed from its cradle.
A line condition caused when a telephone handset is resting in its cradle.
Includes three principal elements: control information (such as destination, origin, length of packet), data to be transmitted, and error detection. The structure of a packet depends on the protocol.
Standard telephone service used by most residential locations; basic service supplying standard single line telephones, telephone lines, and access to the public switched network.
User service managing the FXS and FXO analog telephony interfaces.
A proposal specifying how a host personal computer interacts with a broadband modem (i.e., DSL, cable, wireless, etc.) to access the growing number of Highspeed data networks. Relying on two widely accepted standards, Ethernet and the point-to-point protocol (PPP), the PPPoE implementation requires virtually no more knowledge on the part of the end user other than that required for standard Dial up Internet access. In addition, PPPoE requires no major changes in the operational model for Internet Service Providers (ISPs) and carriers. The base protocol is defined in RFC 2516.
Network access point, the identifier used to distinguish among multiple simultaneous connections to a host.
POSIX is a set of standard operating system interfaces based on the UNIX operating system. The need for standardization arose because enterprises using computers wanted to be able to develop programs that could be moved among different manufacturer's computer systems without having to be recoded.
A telecommunications standard for carrying multiple DS0 voice and data transmissions between two physical locations. All data and voice channels are (ISDN) and operate at 64 kbit/s. North America and Japan use a T1 of 23 B channels and one D channel which corresponds to a T1 line. Europe, Australia and most of the rest of the world use the slightly higher capacity E1, which is composed of 31 B channels and one D channel. Fewer active B channels (also called user channels) can be used for a fractional T1. More channels can be used with more T1's, or with a fractional or full T3 or E3.
An information element (IE) field that determines whether a caller’s CLI can be displayed on a Caller ID device or otherwise presented to the called party.
A small to medium sized telephone system and switch that provides communications between onsite telephones and exterior communications networks.
System service managing the start-up and shutdown sequences of the system.
A formal set of rules developed by international standards bodies, LAN equipment vendors, or groups governing the format, control, and timing of network communications. A set of conventions dealing with transmissions between two systems. Typically defines how to implement a group of services in one or two layers of the OSI reference model. Protocols can describe low-level details of machine-to-machine interfaces or high-level exchanges between allocation programs.
An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.
The local telephone company network that carries voice data over analog telephone lines.
QSIG is an ISDN based signaling protocol for signaling between private branch exchanges (PBXs) in a Private Integrated Services Network (PISN). It makes use of the connection-level Q.931 protocol and the application level ROSE protocol. ISDN "proper" functions as the physical link layer.
Measure of the telephone service quality provided to a subscriber. This could be, for example, the longest time someone should wait after picking up the handset before they receive dial tone (three seconds in most U.S. states).
User service managing the E1 CAS telephony interfaces.
RTCP is the control protocol designed to work in conjunction with RTP. It is standardised in RFC 1889 and 1890. In an RTP session, participants periodically send RTCP packets to convey feedback on quality of data delivery and information of membership.
An IETF standard for streaming real-time multimedia over IP in packets. Supports transport of real-time data like interactive voice and video over packet switched networks.
A server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and MAY offer location services.
A formal document from the IIETF that is the result of committee drafting and subsequent review by interested parties. Some RFCs are informational in nature. Of those that are intended to become Internet standards, the final version of the RFC becomes the standard and no further comments or changes are permitted. Change can occur, however, through subsequent RFCs that supersede or elaborate on all or parts of previous RFCs.
An SBC session is a SIP call established between two endpoints not including the SBC. A session usually has 2 call legs, one incoming and one outgoing of the SBC.
A service provided by ISDN that can be used to test the trustworthiness of the calling party’s number. This signalling-related information element is found in octet 3a of the ISDN SETUP message.
System service allowing the administrator to enable or disable services.
User service allowing the administrator to perform SIP to SIP normalization, call routing, NAT traversal and survivability.
A Session Border Controller used in Voice over Internet Protocol (VoIP) networks to control the signaling and media streams involved in establishing, conducting and analysing telephone calls or other interactive media communications.
Describes multimedia sessions for the purpose of session announcement, session invitation and other forms of multimedia session initiation. SDP communicates the existence of a session and conveys sufficient information to enable participation in the session. SDP is described in RFC 2327.
A protocol for transporting call setup, routing, authentication, and other feature messages to endpoints within the IP domain, whether those messages originate from outside the IP cloud over SCN resources or within the cloud.
The Signaling Interface is used for SIP signaling.
User service allowing the administrator to manage the unit using the SNMP protocol.
A standard of network management that uses a common software agent to manage local and wide area network equipments from different vendors; part of the Transmission Control Protocol / Internet Protocol (TCP/ IP) suite and defined in RFC 1157.
SNTP, which is an adaptation of the Network Time Protocol (NTP), is widely used to synchronize computer clocks in the global Internet. It provides comprehensive mechanisms to access national time and frequency dissemination services, organize the time-synchronization subnet and adjust the local clock in each participating subnet peer. In most places of the Internet of today, NTP provides accuracies of 1-50 ms, depending on the characteristics of the synchronisation source and network paths.
User service allowing the administrator to associate telephony endpoints with SIP user agents.
Session Traversal Utilities for NAT is a standardized set of methods and a network protocol to allow an end host to discover its public IP address if it is located behind a Network Address Translation (NAT)
An efficient means of splitting packets into two fields to separate packets for local destinations from packets for remote destinations in TCP/IP networks.
A SCN (Switched Circuit Network) is a general term to designate a communication network in which any user may be connected to any other user through the use of message, circuit, or packet switching and control devices. The Public Switched Telephone Network (PSTN) or a Private Branch eXchange (PBX) are examples of SCNs.
North-American PRI digital signal carrier. 24 channels (23 voice + 1 signaling)
An ITU-T Recommendation for Real-time fax over IP. T.38 addresses IP fax transmissions for IP-enabled fax devices and fax gateways, defining the translation of T.30 fax signals and Internet Fax Protocols (IFP) packets.
Method of transmitting and receiving independent signals over a common signal path by means of synchronized switches at each end of the transmission line so that each signal appears on the line only a fraction of time in an alternating pattern
The science of translating sound into electrical signals, transmitting them, and then converting them back into sound.
User service managing tone generation and detection on the telephony interfaces.
The TR-069 also known as CWMP, is a Broadband Forum technical specification. This protocol can be used for monitoring and updating CPE configurations and firmware.
The TR-104 is a part of CWMP, a Broadband Forum technical specification. This specification defines the data model for provisioning a Voice over Internet Protocol (VoIP) CPE device by an Auto-Configuration Server (ACS) using the mechanism defined in TR-069.
TR-106 specifies data model guidelines to be followed by all TR-069-enabled devices.
This specification extends the mechanism defined in TR-069 for remote management of customer premises equipment to allow a management system to more easily access and manage devices connected via LAN through an Internet gateway.
The basic communication language or protocol of the Internet. It can also be used as a communications protocol in a private network (either an intranet or an extranet).
A simplified version of FTP that transfers files but does not provide password protection, directory capability, or allow transmission of multiple files with one command.
An efficient but unreliable, connectionless protocol that is layered over IP, as is TCP. Application programs are needed to supplement the protocol to provide error processing and retransmission of data. UDP is an OSI layer 4 protocol.
A network of computers that behave as if they are connected to the same wire even though they may actually be physically located on different segments of a LAN. One of the biggest advantages of VLANs is that when a computer is physically moved to another location, it can stay on the same VLAN without any hardware reconfiguration.
A private communications network usually used within a company, or by several different companies or organizations, to communicate over a public network. VPN message traffic is carried on public networking infrastructure (e.g. the Internet) using standard (often insecure) protocols, or over a service provider's network providing VPN service guarded by well defined Service Level Agreement (SLA) between the VPN customer and the VPN service provider.
The technology used to transmit voice conversations over a data network using the Internet Protocol. Such data network may be the Internet or a corporate Intranet.
A large (geographically dispersed) network, usually constructed with serial lines, that covers a large geographic area. A WAN connects LANs using transmission lines provided by a common carrier.
User service allowing the administrator to manage the unit using HTTP(S) web pages.
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DGW Documentation
Mediatrix devices are supplied with an exhaustive set of documentation.
Mediatrix user documentation is available on the Media5 Documentation Portal.
- Release notes: Generated at each GA release, this document includes the known and solved issues of the software. It also outlines the changes and the new features the release includes.
- Configuration notes: These documents are created to facilitate the configuration of a specific use case. They address a configuration aspect we consider that most users will need to perform. However, in some cases, a configuration note is created after receiving a question from a customer. They provide standard step-by-step procedures detailing the values of the parameters to use. They provide a means of validation and present some conceptual information. The configuration notes are specifically created to guide the user through an aspect of the configuration.
- Technical bulletins: These documents are created to facilitate the configuration of a specific technical action, such as performing a firmware upgrade.
- Hardware installation guide: They provide the detailed procedure on how to safely and adequately install the unit. It provides information on card installation, cable connections, and how to access for the first time the Management interface.
- User guide: The user guide explains how to customise to your needs the configuration of the unit. Although this document is task oriented, it provides conceptual information to help the user understand the purpose and impact of each task. The User Guide will provide information such as where and how TR-069 can be configured in the Management Interface, how to set firewalls, or how to use the CLI to configure parameters that are not available in the Management Interface.
- Reference guide: This exhaustive document has been created for advanced users. It includes a description of all the parameters used by all the services of the Mediatrix units. You will find, for example, scripts to configure a specific parameter, notification messages sent by a service, or an action description used to create Rulesets. This document includes reference information such as a dictionary, and it does not include any step-by-step procedures.
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Copyright Notice
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