Configuring Web User Accounts ................64 4.3.1 Basic User Accounts Configuration ................. 66 4.3.2 Advanced User Accounts Configuration ..............68 Displaying Login Information upon Login ............... 71 Configuring Web Security Settings ................ 72 Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Web Login Authentication using Smart Cards ............72 Configuring Web and Telnet Access List ............... 73 Configuring RADIUS Settings ................74 CLI-Based Management ..................75 Enabling CLI using Telnet ..................75 Enabling CLI using SSH and RSA Public Key ............75 Establishing a CLI Session ..................
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34.3.1.1 Basic Streaming Play ................476 34.3.1.2 Supported File Formats ................. 476 34.3.1.3 Play from Offset ..................476 34.3.1.4 Remote File Systems ................476 34.3.1.5 Using Proprietary Scripts............... 476 34.3.1.6 Dynamic HTTP URLs ................476 Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 34.3.1.7 Play LBR Audio File ................477 34.3.1.8 Basic Record ..................477 34.3.1.9 Remove DTMF Digits at End of Recording ........... 477 34.3.1.10 Record Files Using LBR ................ 477 34.3.1.11 Modifying Streaming Levels Timers ............478 34.3.2 Using File Coders with Different Channel Coders ..........478...
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40.4.4 TFTP Configuration using DHCP Option 66 ............533 40.4.5 HTTP-based Automatic Updates ................533 40.4.6 Configuration using FTP or NFS ................534 40.4.7 Configuration using AudioCodes EMS ..............534 40.5 Loading Files Securely (Disabling TFTP) ............. 534 40.6 Remotely Triggering Auto Update using SIP NOTIFY ......... 535 41 Restoring Factory Defaults ................
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47 Syslog and Debug Recordings ..............579 47.1 Syslog Message Format ..................579 47.1.1 Event Representation in Syslog Messages ............580 47.1.2 Identifying AudioCodes Syslog Messages using Facility Levels ......582 47.1.3 Syslog Fields for Automatic Machine Detection ............582 47.1.4 SNMP Alarms in Syslog Messages ...............583 47.2 Configuring Syslog Settings .................
User's Manual Notices Notice This document describes the AudioCodes Mediant 600 and Mediant 1000 Voice-over-IP (VoIP) SIP media gateways. Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions.
This device includes cryptographic software written by Eric Young (eay@cryptsoft.com). Documentation Feedback AudioCodes continually strives to produce high quality documentation. If you have any comments (suggestions or errors) regarding this document, please fill out the Documentation Feedback form on our Web site at http://www.audiocodes.com/downloads.
User's Manual 1. Overview Overview This section provides an overview of the Mediant 1000 and Mediant 600 media gateways. Mediant 600 The Mediant 600 (hereafter referred to as device) is a cost-effective, wireline Voice-over-IP (VoIP) Session Initiation Protocol (SIP)-based media gateway. It is designed to interface between Time-Division Multiplexing (TDM) and IP networks in enterprises, small and medium businesses (SMB), and CPE application service providers.
Mediant 600 & Mediant 1000 Network Management Protocol (SNMP). The user-friendly, Web interface provides remote configuration using a Web browser (such as Microsoft™ Internet Explorer™). Mediant 1000 The Mediant 1000 (hereafter referred to as device) is a best-of-breed Voice-over-IP (VoIP) Session Initiation Protocol (SIP) Media Gateway, using field-proven, market-leading technology, implementing analog and digital cutting-edge technology.
SIP also provides a registration function that enables users to upload their current locations for use by Proxy servers. SIP implemented in the gateway, complies with the Internet Engineering Task Force (IETF) RFC 3261 (refer to http://www.ietf.org). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 The SIP call flow, shown in the figure below, describes SIP messages exchanged between two devices during a basic call. In this call flow example, device 10.8.201.158 with phone number 6000, dials device 10.8.201.161 with phone number 2000.
Disconnect the network cables (if connected) from the device. Connect one of the Ethernet ports located on the CPU module directly to the network interface of your computer, using a straight-through Ethernet cable. Figure 2-1: Cabling to PC Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Change the IP address and subnet mask of your computer to correspond with the default IP address and subnet mask of the device. Access the Web interface: On your computer, start a Web browser and in the URL address field, enter the default IP address of the device;...
BootP/TFTP Server You can assign an IP address to the device using BootP/TFTP protocols. This can be done using the AudioCodes AcBootP utility (supplied) or any standard compatible BootP server. Note: You can also use the AcBootP utility to load the software file (.cmp) and configuration file (.ini).
Mediant 600 & Mediant 1000 The procedure below describes how to assign an OAMP IP address, using CLI. To assign an OAMP IP address using CLI: Connect the RS-232 port of the device to the serial communication port on your computer.
Press the # key. Dial the new Default Gateway address (e.g., 192*168*0*1), and then press # to finish. Review the new Default Gateway address, and then press 1 to save. Hang up (on-hook) the handset. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Alternatively, initial configuration may be performed using an HTTP server, as discussed in the 'Automatic Update' on page 527. The Voice Menu may be used to specify the configuration URL. To set a configuration URL: Obtain the IP address of the configuration HTTP server (e.g., 36.44.0.6).
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DHCP enable / disable. Configuration server IP address. Configuration file name pattern. Voice menu password (initially 12345). Note: The voice menu password can also be changed using the Web interface or ini file parameter VoiceMenuPassword. Version 6.6 Mediant 600 & Mediant 1000...
Embedded HTTP/S-based Web server - see 'Web-based Management' on page Command Line Interface (CLI) - see 'CLI-Based Management' on page AudioCodes Element Management System - see EMS-Based Management on page Simple Network Management Protocol (SNMP) browser software - see 'SNMP-Based Management' on page ...
• Mozilla Firefox ® (Versions 5 through 9.0) Recommended screen resolutions: 1024 x 768 pixels, or 1280 x 1024 pixels Note: Your Web browser must be JavaScript-enabled to access the Web interface. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 4.1.2 Accessing the Web Interface The procedure below describes how to access the Web interface. To access the Web interface: Open a standard Web browser (see 'Computer Requirements' on page 39). In the Web browser, specify the IP address of the device (e.g., http://10.1.10.10); the...
For more information, see 'Navigation Tree' on page 43. Work pane, displaying the configuration page of the selected menu in the Navigation tree. This is where configuration is done. For more information, see 'Working with Configuration Pages' on page 45. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 4.1.4 Toolbar Description The toolbar provides frequently required command buttons, described in the table below: Description of Toolbar Buttons Icon Button Description Name Submit Applies parameter settings to the device (see 'Saving Configuration' on page 494).
The expanded (Full) view displays all the menus pertaining to the selected configuration tab; the reduced (Basic) view displays only commonly used menus. This is relevant when using the configuration tabs Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 (i.e., Configuration, Maintenance, and Status & Diagnostics) on the Navigation bar. The advantage of the Basic view is that it prevents "cluttering" of the Navigation tree with menus that may not be required. To toggle between Full and Basic view: ...
Certain pages may not be accessible or may be read-only, depending on the access level of your Web user account (see 'Configuring Web User Accounts' on page 64). If a page is read-only, "Read-Only Mode" is displayed at the bottom of the page. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 4.1.6.2 Viewing Parameters Some pages allow you to view a reduced or expanded display of parameters. The Web interface provides two methods for displaying page parameters: Displaying "basic" and "advanced" parameters - see 'Displaying Basic and Advanced Parameters' on page ...
Parameters displayed on the page with the lightning symbol take effect only after a device reset. For resetting the device, see 'Resetting the Device' on page 491. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Note: Parameters saved to the volatile memory (by clicking Submit), revert to their previous settings after a hardware or software reset, or if the device is powered down. Therefore, to ensure parameter changes (whether on-the-fly or not) are retained, save ('burn') them to the device's non-volatile memory, i.e., flash (see 'Saving Configuration' on page 494).
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Edit Edits the selected row. Delete Removes the selected row from the table. When you click this button, a confirmation box appears requesting you to confirm deletion. Click Delete to accept deletion. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Item # Button Toggles between displaying and hiding the full configuration of a Show/Hide selected row. This configuration is displayed below the table (see Item #6) and is useful for large tables that cannot display all its columns in the work pane.
Work pane and the searched parameter is highlighted in the page for easy identification, as shown in the figure below: Figure 4-15: Searched Result Screen Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Search Description Item # Description Search field for entering search key and Search button for activating the search process. Search results listed in Navigation pane. Found parameter, highlighted on relevant Web page 4.1.8 Working with Scenarios The Web interface allows you to create your own menu (Scenario) of up to 20 pages, selected from the menus in the Navigation tree (i.e., pertaining to the Configuration,...
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Save & Finish button to save your Scenario. Selected parameter(s) that you want added to a Scenario Step. Next button to add the current Step to the Scenario and enables you to add additional Steps. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Notes: • You can add up to 20 Steps per Scenario, where each Step can contain up to 25 parameters. • When in Scenario mode, the Navigation tree is in 'Full' display (i.e., all menus are displayed in the Navigation tree) and the configuration pages are in 'Advanced Parameter List' display (i.e., all parameters are shown...
In the Navigation tree, select the required Step; the corresponding page opens in the Work pane. To add parameters, select the check boxes corresponding to the desired parameters. To remove parameters, clear the check boxes corresponding to the desired parameters. Click Next. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • Edit Step Name: In the Navigation tree, select the required Step. In the 'Step Name' field, modify the Step name. On the page, click Next. • Edit Scenario Name: In the 'Scenario Name' field, edit the Scenario name.
ScenarioFileName (see Web and Telnet Parameters on page 619). The Scenario file must be located in the same folder as the ini file. For information on using AudioCodes AcBootP utility, refer to AcBootP Utility User's Guide.
Mediant 600 & Mediant 1000 Note: You can also delete a Scenario using the following alternative methods: • Loading an empty dat file (see 'Loading a Scenario to the Device' on page 57). • Loading an ini file with the ScenarioFileName parameter set to no value (i.e., ScenarioFileName = "").
Figure 4-23: Help Topic for Current Page To view a description of a parameter, click the plus sign to expand the parameter. To collapse the description, click the minus sign. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To close the Help topic, click the close button located on the top-right corner of the Help topic window or simply click the Help button. Note: Instead of clicking the Help button for each page you open, you can open it once for a page and then simply leave it open.
To access the Home page: On the toolbar, click the Home icon. Figure 4-25: Home Page of Mediant 600 Figure 4-26: Home Page of Mediant 1000 Note: The displayed number and type of telephony interfaces depends on the ordered hardware configuration.
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Mediant 600 & Mediant 1000 Gateway Operational State: • "LOCKED": device is locked (i.e. no new calls are accepted) • "UNLOCKED": device is not locked • "SHUTTING DOWN": device is currently shutting down To perform these operations, see 'Basic Maintenance' on page 491.
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(red): Power supply failure or no power supply unit installed Note: This is applicable only to Mediant 1000. Power Supply Unit 2 status indicator. See Item #11 for a description. Note: This is applicable only to Mediant 1000. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 4.2.1 Assigning a Port Name The Home page allows you to assign an arbitrary name or a brief description to each port. This description appears as a tooltip when you move your mouse over the port.
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In the 'Deny Authentication Timer' field, enter the interval (in seconds) that the user needs to wait before a new login attempt from the same IP address can be done after reaching the number of failed login attempts (defined in the previous step). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Notes: • For security, it's recommended that you change the default username and password. • The Security Administrator user can change all attributes of all Web user accounts. Web users with access levels other than Security Administrator can change only their password and username.
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To change the access level of the optional, second account: Under the Account Data for User: User group, from the 'Access Level' drop- down list, select a new access level user. Click Change Access Level; the new access level is applied immediately. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 4.3.2 Advanced User Accounts Configuration This section describes advanced Web user account configuration. This is relevant if you need the following management scheme: Enhanced security settings per Web user (e.g., limit session duration) ...
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At least two letters that are lower case (e.g., "aa") At least two numbers At least two signs (e.g., the dollar "$" sign) No spaces in the string At least four characters different to the previous password Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Status Defines the status of the Web user. New = (Default) User is required to change its password on the next login. When the user logs in to the Web interface, the user is immediately prompted to change the current password.
From the 'Display Login Information' drop-down list, select Yes. Click Submit to apply your changes. Once enabled, the Login Information window is displayed upon a successful login, as shown in the example below: Figure 4-32: Login Information Window Version 6.6 Mediant 600 & Mediant 1000...
- what the user has (i.e., the physical card) and what the user knows (i.e., the login password). This feature is enabled using the EnableMgmtTwoFactorAuthentication parameter. Note: For specific integration requirements for implementing a third-party smart card for Web login authentication, contact your AudioCodes representative. User's Manual Document #: LTRT-83313...
Web & Telnet Access List table. Figure 4-35: Web & Telnet Access List Table To delete authorized IP addresses, select the Delete Row check boxes corresponding Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 to the IP addresses that you want to delete, and then click Delete Selected Addresses; the IP addresses are removed from the table and these IP addresses can no longer access the Web and Telnet interfaces.
TCP, providing methods for key exchange, authentication, encryption, and authorization. SSH requires appropriate client software for the management PC. Most Linux distributions have OpenSSH pre-installed; Windows-based PCs require an SSH client software such as Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 PuTTY, which downloaded from http://www.chiark.greenend.org.uk/~sgtatham/putty/. By default, SSH uses the same username and password as the Telnet and Web server. SSH supports 1024/2048-bit RSA public keys, providing carrier-grade security. Follow the instructions below to configure the device with an administrator RSA key as a means of strong authentication.
Log in to the session using the username and password assigned to the Admin user of the Web interface. At the login prompt, type the username, and then press Enter: login: Admin At the password prompt, type the password, and then press Enter: Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 password: Admin After logging in, the current directory (root), available commands, available subdirectories, and a welcome message are displayed at the CLI prompt: login: Admin password: ready. Type "exit" to close the connection. SIP/ SECurity/ PStn/ DebugRecording/ MGmt/ ControlProtocol/ CONFiguration/...
| set src | set cid | set DTMFs | display | connect Table 5: Sub-Commands of Test (TC) Command Sub-Command Arguments Description set dest <number> Sets the Destination Number for the test call. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Sub-Command Arguments Description set src <number> Sets the Source Number for the test call. set cid <display string> Sets the Display Name for the test call. set DTMFs <DTMF pattern> Sets the pattern of DTMFs that is played to the PSTN side after the test call is connected.
3 = LOCAL_ALL_CHANNELS_LOOP (whole trunk only) PstnSendAlarm PS/PH/PSA <TrunkId> Sends an alarm signal at the Tx interface <AlarmSendCode> or on a specific Trunk ID. The valid value options for AlarmSendCode include the Version 6.6 Mediant 600 & Mediant 1000...
Management submenu > SNMP submenu > SNMP Community String). Configure the SNMP community strings parameters according to the table below. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 494. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To delete a community string, select the Delete check box corresponding to the community string that you want to delete, and then click Submit. SNMP Community String Parameters Description Parameter Description Read Only [SNMPReadOnlyCommunityString_x]: Up to five Community String read-only community strings (up to 19 characters each).
SNMP agent accepts and processes SNMP requests. Notes: The SNMP Trusted Managers table can also be configured using the table ini file parameter, SNMPTrustedMgr_x (see 'SNMP Parameters' on page 623). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure SNMP Trusted Managers: Open the SNMP Trusted Managers page (Configuration tab > System menu > Management submenu > SNMP submenu > SNMP Trusted Managers). Figure 6-2: SNMP Trusted Managers Select the check box corresponding to the SNMP Trusted Manager that you want to enable and for whom you want to define an IP address.
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Group The group with which the SNMP v3 user is associated. [SNMPUsers_Group] [0] Read-Only (default) [1] Read-Write [2] Trap Note: All groups can be used to send traps. Version 6.6 Mediant 600 & Mediant 1000...
User's Manual 7. EMS-Based Management EMS-Based Management AudioCodes Element Management System (EMS) is an advanced solution for standards- based management of gateways within VoP networks, covering all areas vital for the efficient operation, administration, management and provisioning (OAM&P) of AudioCodes' families of gateways.
The table ini file parameter is composed of the following elements: Title of the table: The name of the table in square brackets, e.g., [MY_TABLE_NAME]. Format line: Specifies the columns of the table (by their string names) that are to be Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 configured. • The first word of the Format line must be "FORMAT", followed by the Index field name and then an equal "=" sign. After the equal sign, the names of the columns are listed.
• Load Auxiliary Files - see 'Loading Auxiliary Files' on page AudioCodes AcBootP utility, which uses Bootstrap Protocol (BootP) and acts as a TFTP server. For information on using the AcBootP utility, refer to AcBootP Utility User's Guide. ...
The file may be loaded to the device using TFTP or HTTP. These protocols are not secure and are vulnerable to potential hackers. To overcome this security threat, the AudioCodes DConvert utility allows you to binary-encode (encrypt) the ini file before loading it to the device.
Fill in the rest of the request fields according to your security provider's instructions. Click the Create CSR button; a textual certificate signing request is displayed in the area below the button: Figure 9-1: Certificate Signing Request Group Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Copy the text and send it to your security provider. The security provider, also known as Certification Authority or CA, signs this request and then sends you a server certificate for the device. Save the certificate to a file (e.g., cert.txt). Ensure that the file is a plain-text file containing the"‘BEGIN CERTIFICATE"...
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If the device was originally operating in HTTPS mode and you disabled it in Step 2, then enable it by setting the 'Secured Web Connection (HTTPS)' field to HTTPS Only. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Mutual TLS Authentication By default, servers using TLS provide one-way authentication. The client is certain that the identity of the server is authentic. When an organizational PKI is used, two-way authentication may be desired - both client and server should be authenticated using X.509 certificates.
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The device does not query OCSP for its own certificate. • Some PKIs do not support OCSP but generate Certificate Revocation Lists (CRLs). For such cases, set up an OCSP server such as OCSPD. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Self-Signed Certificates The device is shipped with an operational, self-signed server certificate. The subject name for this default certificate is 'ACL_nnnnnnn', where nnnnnnn denotes the serial number of the device. However, this subject name may not be appropriate for production and can be changed while still using self-signed certificates.
The update is performed in such a way as to be transparent to the end users. For instance, the response of the server may indicate that the clock is running too fast on the Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 client. The client slowly robs bits from the clock counter to update the clock to the correct time. If the clock is running too slow, then in an effort to catch the clock up, bits are added to the counter, causing the clock to update quicker and catch up to the correct time.
MIIRedundancyEnable. By default, this feature is disabled. When Ethernet redundancy is implemented, the two Ethernet ports can be connected to the same switch (segment / hub). In this setup, one Ethernet port is active and the other is Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 redundant. If an Ethernet connection failure is detected, the CPU module switches over to the redundant Ethernet port. The CPU issues a Major alarm notifying of the failed physical port. If the first Ethernet port connection is restored, the Major alarm is cleared. The first physical port now becomes the redundant Ethernet port in case of failure with the active physical port (which is currently the second physical port).
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IP network interfaces, see 'Configuring Firewall Settings' on page 137. • The Multiple Interface table can also be configured using the table ini file parameter, InterfaceTable (see 'Networking Parameters' on page 609). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure IP network interfaces: Open the IP Settings page (Configuration tab > VoIP menu > Network submenu > IP Settings). Figure 11-2: IP Settings Page (Single Network Interface) Note: The IP Settings page appears only in the following circumstances: •...
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The CIDR-style presentation is the latest method for interpretation of IP addresses. Specifically, instead of using eight-bit address blocks, it uses the variable- length subnet masking technique to allow allocation on arbitrary-length prefixes. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description The prefix length for IPv4 can range from 0 to 30. Note: For valid configuration, see 'Multiple Interface Table Configuration Rules' on page 115. Web/EMS: Gateway Defines the IP address of the default gateway for the interface. When [InterfaceTable_Gate traffic is sent from this interface to an unknown destination (i.e., not in the...
Default Gateway must be in the same subnet as the associated interface. Additional static routing rules can be configured in the IP Routing table. The interface name must be configured (mandatory) and unique for each interface, Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 and can include up to 16 characters. For IPv4 addresses, the 'Interface Mode' column must be set to IPv4 Manual (numeric value 10). Each network interface must be assigned a unique VLAN ID.
Example of VoIP Interfaces per Application Type in Multiple Interface Table Application Prefix Index IP Address Gateway VLAN ID Interface Name Type Length OAMP 192.168.85.14 0.0.0.0 ManagementIF Control 200.200.85.14 200.200.85.1 myControlIF Media 211.211.85.14 211.211.85.1 myMediaIF Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 VLANs are required and the Native VLAN ID is the same VLAN ID as the Management interface (configured for Index 0): • 'VLAN Mode' is set to Enable. • 'Native VLAN ID' field is set to "1".
If such an entry is found, the device sends the packet to the indicated router. If no explicit entry is found, the packet is sent to the default gateway. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure static IP routing: Open the IP Routing Table page (Configuration tab > VoIP menu > Network submenu > IP Routing Table). Figure 11-4: IP Routing Table Page In the Add a new table entry table, add a new static routing rule according to the parameters described in the table below.
Applicati Prefix Index IP Address Gateway VLAN ID Interface Name on Type Length OAMP 192.168.0.2 192.168.0.1 Media & 10.32.174.50 10.32.174.1 2012 MediaCntrl Control Media 10.33.174.50 10.33.174.1 2013 Media1 Control 10.34.174.50 10.34.174.1 2014 Cntrl1 Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Configure static IP Routing rules in the IP Routing table, as shown below: Configured Static IP Routing Rules in IP Routing Table Destination IP Gateway IP Prefix Length Metric Interface Name Address Address 10.31.174.0 192.168.11.1 174.96.151.15...
SIP over TLS (SIPS) Control Premium control Syslog Management Bronze ICMP Management Determined by the initiator of the request ARP listener Determined by the initiator of the Network request Management Bronze SNMP Traps Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Application Traffic / Network Types Class-of-Service (Priority) DNS client Varies according to DNS settings: Depends on traffic type: OAMP Control: Premium Control Control Management: Bronze Varies according to the interface Depends on traffic type: type associated with NTP (see ...
IP network interface, configured in the Multiple Interface table (see 'Configuring IP Network Interfaces' on page 110). • You can also configure the DNS table using the table ini file parameter, DNS2IP (see 'DNS Parameters' on page 616). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure the internal DNS table: Open the Internal DNS Table page (Configuration tab > VoIP menu > Network submenu > DNS submenu > Internal DNS Table). Click Add; the following dialog box appears: Figure 11-6: Internal DNS Table - Add Record Dialog Box Configure the DNS rule, as required.
Click Add; the following dialog box appears: Figure 11-7: Internal SRV Table Page Configure the SRV rule, as required. For a description of the parameters, see the table below. Click Submit; the SRV rule is added to the table. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Internal SRV Table Parameter Description Parameter Description Domain Name Defines the host name to be translated. [Srv2Ip_InternalDomain] The valid value is a string of up to 31 characters. Transport Type Defines the transport type. [Srv2Ip_TransportType] ...
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Path to the root of the remote file system in the format: /[path]. For [NFSServers_RootPath] example, '/audio'. NFS Version NFS version used to access the remote file system. [NFSServers_NfsVersion [2] NFS Version 2 Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description [3] NFS Version 3 (default) Authentication Type Authentication method used for accessing the remote file system. [NFSServers_AuthType] [0] Null [1] Unix (default) User ID User ID used in authentication when using Unix.
STUN can’t be used when the device is located behind a symmetric NAT. • Use either the STUN server IP address (STUNServerPrimaryIP) or domain name (STUNServerDomainName) method, with priority to the first one. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To enable STUN: Open the Application Settings page (Configuration tab > System menu > Application Settings). Figure 11-10: STUN Parameters in Application Settings Page From the 'Enable STUN' (EnableSTUN) drop-down list, select Enable to enable the STUN feature.
To save the changes to flash memory, see 'Saving Configuration' on page 494. NAT Translation Table Parameters Parameter Description Index Defines the table index entry. This table can include up to 32 entries. [NATTranslation_Ind Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Source Interface Defines the name of the IP interface, as appears in the Multiple Interface Name table. [NATTranslation_So Note: If the Multiple Interface table is not configured, the default Source IP urceIPInterfaceNam Interface Name is "All". This represents the single IP interface for OAMP, Control, and Media (defined by the LocalOAMIPAddress, LocalOAMSubnetMask, and LocalOAMDefaultGW parameters).
Parameters' on page 609). The default payload type is 120. T.38 No-Op: T.38 No-Op packets are sent only while a T.38 session is activated. Sent packets are a duplication of the previously sent frame (including duplication of the Version 6.6 Mediant 600 & Mediant 1000...
Receipt of No-Op packets is always supported. 11.11 Robust Receipt of Media Streams The “robust-media” mechanism is an AudioCodes proprietary mechanism to filter out unwanted media (i.e., RTP, RTCP, and T.38) streams that are sent to the same port number on the device. In practice, the media RTP/RTCP ports may receive additional multiple unwanted media streams as result of traces of previous calls, call control errors, or deliberate attacks.
- Start Port - End Port: 0-65535 - Protocol: Any - Action Upon Match: Block • You can also configure the firewall settings using the table ini file parameter, AccessList (see 'Security Parameters' on page 635). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To add firewall rules: Open the Firewall Settings page (Configuration tab > VoIP menu > Security submenu > Firewall Settings). Click the Add button; the following dialog box appears: Figure 12-1: Firewall Settings Page - Add Record Configure the firewall parameters, as required.
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Defines the destination UDP/TCP end port (on this device) to where [AccessList_End_Port] packets are sent. The valid range is 0 to 65535. Note: When the protocol type isn't TCP or UDP, the entire range must be provided. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Protocol Defines the protocol type (e.g., UDP, TCP, ICMP, ESP or 'Any') or [AccessList_Protocol] the IANA protocol number in the range of 0 (Any) to 255. Note: This field also accepts the abbreviated strings 'SIP' and 'HTTP'.
IPSec to encrypt the IP stream. IKE negotiation comprises the following two phases: Main Mode (creates a secured channel for the Quick mode by obtaining a "master" encryption key, without any prior keys, and authenticates the peers to each other): Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • SA negotiation: The peers negotiate their capabilities using up to four proposals. Each proposal includes the Encryption method, Authentication algorithm, and the Diffie-Hellman (DH) group. The master key’s lifetime is also negotiated. • Key exchange (DH): The DH protocol creates the master key. DH requires both peers to agree on certain mathematical parameters, known as the "group".
Defines the length of the key created by the DH protocol for up to four [IPsecProposalTable_DH proposals. For the ini file parameter, X denotes the proposal number Group] (0 to 3). [0] Group 1 (768 Bits) = DH-786-Bit Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Name Description [1] Group 2 (1024 Bits) (default) = DH-1024-Bit If no proposals are defined, the default settings (shown in the following table) are applied. Default IPSec/IKE Proposals Proposal Encryption Authentication DH Group...
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[0] Pre-shared Key (default) ationMethod] [1] RSA Signature = in X.509 certificate Note: For RSA-based authentication, both peers must be provisioned with certificates signed by a common CA. For more information on Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Name Description certificates, see 'Replacing the Device's Certificate' on page 99. Shared Key Defines the pre-shared key (in textual format). Both peers must use [IPsecSATable_SharedK the same pre-shared key for the authentication process to succeed.
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Tunnel. Interface Name Assigns a network interface to this IPSec rule. The network interfaces [IPsecSATable_Interface are defined in the Multiple Interface table ('Interface Name' column) in Name] 'Configuring IP Network Interfaces' on page Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 12.4 Intrusion Detection System The device can be configured to detect malicious attacks on its system and send SNMP traps if malicious activity is identified. The Intrusion Detection System (IDS) is an important feature for Enterprises to ensure legitimate calls are not being adversely affected by attacks and to prevent Theft of Service and unauthorized access.
Figure 12-6: IDS Policy Table with Default Rules Add a Policy name: Click Add. Figure 12-7: IDS Policy Table - Add Record Configure the parameters as described in the following table, and then click Submit. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Table 12-1: IDS Policy Table Parameters Parameter Description Index Defines the table row number for the policy. CLI: policy [IDSPolicy_Index] Name Defines a name for the policy. CLI: rule The valid value is a string of up to 20 characters.
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The counter is [IDSRule_ThresholdWindo automatically reset at the end of the interval. The valid range is 1 to 1,000,000. The default is 1. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Minor-Alarm Threshold Defines the threshold that if crossed a minor severity alarm is sent. CLI: minor-alrm-thr The valid range is 1 to 1,000,000. A value of 0 or -1 means not [IDSRule_MinorAlarmThre defined.
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"!" appears before each subnet. 10.1.0.0/16 & !10.1.1.1: includes subnet 10.1.0.0/16, except IP address 10.1.1.1. Policy Selects the IDS policy, configured in 'Configuring IDS Policies' on CLI: policy page 149. [IDSMatch_Policy] Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 12.4.4 Viewing IDS Alarms The device uses SNMP (and Syslog) to notify the detection of malicious attacks. The trap displays the IDS Policy and Rule, and the Policy-Match index. The device sends the SNMP alarm, acIDSPolicyAlarm whenever a threshold of a specific IDS Policy rule is crossed.
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Remote rejects (prior to SIP 180 response) establish-remote-reject Requests and responses without a matching flow-no-match-tu Abnormal Flow transaction user (except ACK requests) flow-no-match- Requests and responses without a matching transaction transaction (except ACK requests) Version 6.6 Mediant 600 & Mediant 1000...
Open the Voice Settings page (Configuration tab > VoIP menu > Media submenu > Voice Settings). Figure 13-1: Voice Volume Parameters in Voice Settings Page Configure the following parameters: • 'Voice Volume' (VoiceVolume) - Defines the voice gain control (in decibels) for IP- to-Tel Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • 'Input Gain' (InputGain) - Defines the PCM input gain control (in decibels) for Tel- to-IP Click Submit to apply your settings. 13.1.2 Silence Suppression (Compression) Silence suppression (compression) is a method for conserving bandwidth on VoIP calls by not sending packets when silence is detected.
'Configuration Parameters Reference' on page 609. To access the fax and modem parameters: Open the Fax/Modem/CID Settings page (Configuration tab > VoIP menu > Media submenu > Fax/Modem/CID Settings). Figure 13-4: Fax/Modem/CID Settings Page Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Configure the parameters, as required. Click Submit to apply your changes. 13.2.1 Fax/Modem Operating Modes The device supports two modes of operation: Fax/modem negotiation that is not performed during the establishment of the call.
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In the Fax/Modem/CID Settings page, set the 'Fax Transport Mode' parameter to RelayEnable (FaxTransportMode = 1). Configure the following optional parameters: • 'Fax Relay Redundancy Depth' (FaxRelayRedundancyDepth) • 'Fax Relay Enhanced Redundancy Depth' (FaxRelayEnhancedRedundancyDepth) Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • 'Fax Relay ECM Enable' (FaxRelayECMEnable) • 'Fax Relay Max Rate' (FaxRelayMaxRate) 13.2.2.2 G.711 Fax / Modem Transport Mode In this mode, when the terminating device detects fax or modem signals (CED or AnsAM), it sends a Re-INVITE message to the originating device, requesting it to re-open the channel in G.711 VBD with the following adaptations:...
INVITE messages are sent. The voice channel is optimized for fax/modem transmission (same as for usual bypass mode). The parameters defining payload type for AudioCodes proprietary Bypass mode -- 'Fax Bypass Payload Type' (RTP/RTCP Settings page) and ModemBypassPayloadType (ini file) -- are not used with NSE Bypass.
Fax (IsFaxUsed = 0). In the Fax/Modem/CID Settings page, do the following: Set the 'Fax Transport Mode' parameter to Disable (FaxTransportMode = 0). Set the 'V.21 Modem Transport Type' parameter to Disable (V21ModemTransportType = 0). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Set the 'V.22 Modem Transport Type' parameter to Disable (V22ModemTransportType = 0). Set the 'V.23 Modem Transport Type' parameter to Disable (V23ModemTransportType = 0). Set the 'V.32 Modem Transport Type' parameter to Disable (V32ModemTransportType = 0).
Set the 'V.23 Modem Transport Type' parameter to Disable (V23ModemTransportType = 0). Set the 'V.32 Modem Transport Type' parameter to Disable (V32ModemTransportType = 0). Set the 'V.34 Modem Transport Type' parameter to Disable (V34ModemTransportType = 0). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 13.2.4 V.152 Support The device supports the ITU-T recommendation V.152 (Procedures for Supporting Voice- Band Data over IP Networks). Voice-band data (VBD) is the transport of modem, facsimile, and text telephony signals over a voice channel of a packet network with a codec appropriate for such signals.
Factor should be set to 13. In this special mode the clock drift correction is performed less frequently - only when the Jitter Buffer is completely empty or completely full. When such condition occurs, the correction is performed by dropping several voice packets Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 simultaneously or by adding several BFI packets simultaneously, so that the Jitter Buffer returns to its normal condition. The procedure below describes how to configure the jitter buffer using the Web interface. To configure jitter buffer using the Web interface: Open the RTP/RTCP Settings page (Configuration tab >...
Set the 'Declare RFC 2833 in SDP' parameter to No (RxDTMFOption = 0). Set the '1st Tx DTMF Option' parameter to Not Supported (TxDTMFOption = 0). Set the ini file parameter, DTMFTransportType to 2 (i.e., transparent). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Using INFO message according to Korea mode: DTMF digits are sent to the remote side in INFO messages. To enable this mode, define the following: Set the 'Declare RFC 2833 in SDP' parameter to No (RxDTMFOption = 0).
The device's RTP Multiplexing (ThroughPacket™) feature is AudioCodes proprietary method for aggregating RTP streams from several channels when the device operates with another AudioCodes device. This feature reduces the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP headers and reduces the packet/data transmission rate.
Mediant 600 & Mediant 1000 Click Submit. Reset the device for the settings to take effect. 13.3.5 Configuring RTP Base UDP Port You can configure the range of UDP ports for RTP, RTCP, and T.38. The UDP port range can be configured using media realms in the Media Realm table, allowing you to assign different port ranges (media realms) to different interfaces.
To enable the AMD beep detection, the X-Detect header in the received INVITE message must include “Request=AMD”, and the AMDBeepDetectionMode parameter must be set to 1 or 2. If set to 1, the beep is detected only after Answering Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Machine detection. If set to 2, the beep is detected even if the Answering Machine was not detected. Using the Call Progress Tone detector: To enable this detection mode, the X- Detect header in the received INVITE message must include “Request=CPT”, and one or several beep tones (Tone Type #46) must be configured in the regular CPT file.
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The device's AMD feature is based on voice detection for North American English. If you want to implement AMD in a different language or region, you must provide AudioCodes with a database of recorded voices in the language on which the device's AMD mechanism can base its voice detector algorithms for detecting these voices.
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Mediant 600 & Mediant 1000 The SIP call flows below show an example of implementing the device's AMD feature. This scenario example allows a third-party Application server to play a recorded voice message to an answering machine. Upon detection by the device of the answering machine, the device sends a SIP INFO message to the Application server: INFO sip:sipp@172.22.2.9:5060 SIP/2.0...
> IPMedia Settings). The AGC parameters are shown in the figure below: Figure 13-11: AGC Parameters in IPMedia Settings Page Configure the following parameters: • 'Enable AGC' (EnableAGC) - Enables the AGC mechanism. • 'AGC Slope' (AGCGainSlope) - Determines the AGC convergence rate. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • 'AGC Redirection' (AGCRedirection) - Determines the AGC direction. • 'AGC Target Energy' - Defines the signal energy value (dBm) that the AGC attempts to attain. Click Submit to apply your settings. Note: Below are additional AGC parameters: •...
DSP channels are required for the following: IP-to-IP application: Each IP-to-IP call session includes two legs, requiring two DSP resources. Note: In some scenarios, the IP-to-IP application does not require DSP resources. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To support these capabilities and to allow optimal management of the required DSP resources, you need to ensure suitable hardware and software configuration, as described below: Hardware Configuration: The device can obtain DSP resources using one of the following hardware configurations: •...
Once you have configured a Media Realm, you can configure it with the following: Quality of Experience parameters for reporting to AudioCodes SEM server used for monitoring the quality of calls (see Configuring Quality of Experience Parameters per Media Realm on page 185) ...
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Mediant 600 & Mediant 1000 To define a Media Realm: Open the Media Realm Table page (Configuration tab > VoIP menu > Media submenu > Media Realm Configuration). Click the Add button; the following appears: Figure 13-14: Media Realm Page - Add Record Dialog Box Configure the parameters as required.
At any given time during a call, each of these parameters can be in one of the following states according to its value in the last RTCP / RTCP XR packet: Gray - indicates that the value is unknown Green - indicates good call quality Version 6.6 Mediant 600 & Mediant 1000...
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The QoE feature is available only if the device is installed with the relevant Software License Key. • To configure the address of the AudioCodes Session Experience Manager (SEM) server to where the device reports the QoE, see 'Configuring SEM Server for Media Quality of Experience' on page 190.
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[1] Notify = (Default) Device sends a report to the SEM server. [2] Activate 2198 = RTP redundancy packets are sent to the relevant call leg. Note: This field is applicable only if the monitored parameter is remote packet loss. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Green Yellow Operation Details Note: This field is currently not supported. [QOERules_GreenYellowOpera Defines the desired RTP redundancy depth. The actual tionDetails] redundancy depth on the relevant call leg is the minimum between the desired depth and the maximum supported depth on that call leg.
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[0] High Threshold Rule [1] Critical Threshold Rule Rule Action Defines the action that the device performs when the configured [BWManagement_RuleAction] threshold is exceeded: [0] Report Only (default) [1] No more calls Version 6.6 Mediant 600 & Mediant 1000...
13.9 Configuring Server for Media Quality of Experience The device can be configured to report voice (media) quality of experience to AudioCodes Session Experience Manager (SEM) server, a plug-in for AudioCodes EMS. The reports include real-time metrics of the quality of the actual call experience and processed by the SEM.
Notes: • For a detailed description of the SRTP parameters, see SRTP Parameters on page 637. • When SRTP is used, the channel capacity may be reduced. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure media security: Open the Media Security page (Configuration tab > VoIP menu > Media submenu > Media Security). Configure the parameters as required. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 494.
The LDAP Settings page is used for configuring the LDAP server parameters. For a full description of these parameters, see 'Configuration Parameters Reference' on page 609. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure the LDAP server parameters: Open the LDAP Settings page (Configuration tab > VoIP menu > Services submenu > LDAP Settings). Figure 14-1: LDAP Settings Page The read-only 'LDAP Server Status' field displays one of the following possibilities: •...
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LDAP server again. • For a full description of the cache parameters, see 'Configuration Parameters Reference' on page 609. Version 6.6 Mediant 600 & Mediant 1000...
If the query is found: The AD returns up to four attributes - Lync / OCS, PBX / IP PBX, private (only if it equals Primary or Secondary key), and mobile. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 The device adds unique prefix keywords to the query results in order to identify the query type (i.e., IP domain). These prefixes are used as the prefix destination number value in the Outbound IP Routing table to denote the IP domains: •...
Open the Advanced Parameters page (Configuration tab > VoIP menu > SIP Definitions submenu > Advanced Parameters). Figure 14-5: LDAP Parameters for Microsoft Lync Server 2010 Configure the LDAP attribute names as desired. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Configure AD-based Tel-to-IP routing rules: Open the Outbound IP Routing Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing > Tel to IP Routing). For more information, see Configuring Outbound IP Routing Table on page 323.
The Calling Name Manipulation Table for Tel -> IP Calls table uses the numbers before manipulation, as inputs. • The LDAP query uses the calling number after source number manipulation, as the search key value. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 14.2 Least Cost Routing This section provides a description of the device's least cost routing (LCR) feature and how to configure it. 14.2.1 Overview The LCR feature enables the device to choose the outbound IP destination routing rule based on lowest call cost.
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Index 3 - no Cost Group is assigned, but as the Default Cost parameter is set to Min, it is selected as the cheapest route • Index 4 - Cost Group "B" is only second-matched rule (Index 1 is the first) Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Example 3: This example shows how the cost of a call is calculated if the call spans over multiple time bands: Assume a Cost Group, "CG Local" is configured with two time bands, as shown below:...
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Therefore, a call of 1 minute cost 7 units. Therefore, for calls under one minute, "Weekend A" carries the lower cost. However, if the average call duration is more than one minute, then "Weekend B" carries the lower cost. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Default Cost Determines whether routing rules in the Outbound IP Routing table [RoutingRuleGroups_L without an assigned Cost Group are considered a higher cost or lower CRDefaultCost] cost route compared to other matched routing rules that are assigned Cost Groups.
Select a Cost Group for which you want to assign Time Bands, and then click the Time Band link located below the table; the Time Band table for the selected Cost Group appears. Click the Add button; the Add Record dialog box appears: Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Configure the parameters as required. For a description of the parameters, see the table below. Click Submit; the entry is added to the Time Band table for the relevant Cost Group. Time Band Table Description...
Applications Enabling submenu > Applications Enabling). From the relevant application drop-down list, select Enable. Save (burn) the changes to the device's flash memory with a device reset (see 'Saving Configuration' on page 494). Version 6.6 Mediant 600 & Mediant 1000...
The SRD table can also be configured using the table ini file parameter, SRD. To configure SRDs: Open the SRD Settings page (Configuration tab > VoIP menu > Control Network submenu > SRD Table). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 From the 'SRD Index' drop-down list, select an index for the SRD, and then configure it according to the table below. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 494.
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Each SIP Interface must have a unique signaling port (i.e., no two SIP Interfaces can share the same port - no port overlapping). TLS Port Defines the listening TLS port. [SIPInterface_TLSPort] The valid range is 1 to 65534. The default is 5061. Notes: Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description This port must be outside of the RTP port range. Each SIP Interface must have a unique signaling port (i.e., no two SIP Interfaces can share the same port - no port overlapping).
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Click the Add button: the following dialog box appears: Configure the IP Group parameters according to the table below. Click Submit. To save the changes to flash memory, see 'Saving Configuration' on page 494. IP Group Parameters Parameter Description Common Parameters Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Type Defines the type of IP Group: [IPGroup_Type] [0] Server = Used when the destination address, configured by the Proxy Set, of the IP Group (e.g., ITSP, Proxy, IP-PBX, or Application server) is known.
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Defines the Request-URI host name in outgoing INVITE messages. [IPGroup_AlwaysUseRo [0] No (default). uteTable] [1] Yes = The device uses the IP address (or domain name) defined in the Outbound IP Routing Table (see Configuring the Outbound IP Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Routing Table on page 323) as the Request-URI host name in outgoing INVITE messages, instead of the value configured in the 'SIP Group Name' field. Note: This parameter is applicable only to Server-type IP Groups.
Group) of the call (i.e., both directions). For example, one Proxy Set for the Internet Telephony Service provider (ITSP) interfacing with one 'leg' of the device and another Proxy Set for the second SIP entity (e.g., ITSP) interfacing with the other 'leg' of the device. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Notes: • Proxy Sets can be assigned only to Server-type IP Groups. • The Proxy Set table can also be configured using two complementary tables: - Proxy Set ID with IP addresses: Table ini file parameter, ProxyIP.
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[2] TLS [-1] = Undefined Note: If no transport type is selected, the value of the global parameter SIPTransportType is used. Web/EMS: Enable Enables the Keep-Alive mechanism with the Proxy server(s). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Proxy Keep Alive [0] Disable (default). [ProxySet_EnablePr [1] Using Options = Enables Keep-Alive with Proxy using SIP OPTIONS oxyKeepAlive] messages. [2] Using Register = Enables Keep-Alive with Proxy using SIP REGISTER messages.
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Defines the SRD associated with the Proxy Set ID. [ProxySet_ProxySet_ The default is SRD 0. SRD] Notes: For this parameter to take effect, a device reset is required. To configure SRDs, see Configuring SRD Table on page 211. Version 6.6 Mediant 600 & Mediant 1000...
The Account table can also be configured using the table ini file parameter, Account. To configure Accounts: Open the Account Table page (Configuration tab > VoIP menu > SIP Definitions submenu > Account Table). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 In the 'Add' field, enter the desired table row index, and then click Add. A new row appears. Configure the Account parameters according to the table below. Click the Apply button to save your changes.
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If registration fails, then the user part in the INVITE Contact header contains the source party number. Application Type Defines the application type: [Account_Applicat [0] GW/IP2IP = (Default) Gateway and IP-to-IP application. ionType] Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 17.3 Configuring Proxy and Registration Parameters The Proxy & Registration page allows you to configure the Proxy server and registration parameters. For a description of the parameters appearing on this page, see 'Configuration Parameters Reference' on page 609.
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• The password from the ini file is "AudioCodes". • The equation to be evaluated is "122:audiocodes.com:AudioCodes". According to the RFC, this part is called A1. • The MD5 algorithm is run on this equation and stored for future usage.
IP Group. If this parameter is not configured, see below. According to the settings of the "global" parameter, GWOutboundManipulationSet. If this parameter is not configured, no manipulation is done. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 The figure below illustrates a SIP message manipulation example: Figure 17-1: SIP Header Manipulation Example Notes: • For a detailed description of the syntax for configuring SIP message manipulation rules, refer to SIP Message Manipulations Quick Reference Guide.
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Note: Each rule must be configured with a unique index. Manipulation Set ID Defines a Manipulation Set ID for the rule. You can define the same CLI: manipulation-set-id Manipulation Set ID for multiple rules to create a group of rules. The Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [MessageManipulation Manipulation Set ID is used to assign the manipulation rules to an IP s_ManSetID] Group (in the IP Group table) for inbound and/or outbound messages. The valid value is 0 to 19. The default is 0.
Blacklist and whitelist for defined SIP methods (e.g., INVITE) Blacklist and whitelist for defined SIP bodies Note: The Message Policy table can also be configured using the table ini file parameter, MessagePolicy. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure SIP message policy rules: Open the Message Policy Table page (Configuration tab > VoIP menu > SIP Definitions > Msg Policy & Manipulation > Message Policy Table). Click the Add button; the Add Record dialog box appears:...
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[0] Policy Blacklist =The specified SIP body (in the 'Body List' field) is istType] rejected by the policy. [1] Policy Whitelist = (Default) The specified SIP body (in the 'Body List' field) is allowed by the policy. Version 6.6 Mediant 600 & Mediant 1000...
For information on V.152 and implementation of T.38 and VBD coders, see 'Supporting V.152 Implementation' on page 168. • The Coders table can also be configured using the table ini file parameter, CodersGroup. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure the device's coders: Open the Coders page (Configuration tab > VoIP menu > Coders and Profiles submenu > Coders). Figure 18-1: Coders Table Page From the 'Coder Name' drop-down list, select the required coder.
Mediant 600 & Mediant 1000 18.2 Configuring Coder Groups The Coder Group Settings page allows you to define up to 10 groups of coders (termed Coder Groups). For each Coder Group, you can define up to 10 coders configured with packetization time (ptime), rate, payload type, and silence suppression.
The procedure below describes how to configure Tel Profiles using the Web interface. Note: Tel Profiles can also be configured using the table ini file parameter, TelProfile (see 'Configuration Parameters Reference' on page 609) Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure Tel Profiles: Open the Tel Profile Settings page (Configuration tab > VoIP menu > Coders and Profiles submenu > Tel Profile Settings). From the 'Profile ID' drop-down list, select the Tel Profile index.
[0] don’t declare RFC 2833; [1] declare RFC 2833 payload type is SDP. • You can also configure IP Profiles using the table ini file parameter, IPProfile (see Configuration Parameters Reference on page 609). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure IP Profiles: Open the IP Profile Settings page (Configuration tab > VoIP menu > Coders and Profiles submenu > IP Profile Settings). From the 'Profile ID' drop-down list, select the IP Profile index.
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For a description, see the global parameter [IpProfile_MKISize] SRTPTxPacketMKISize. Gateway Parameters Web: Fax Signaling Method For a description, see the global parameter IsFaxUsed. [IpProfile_IsFaxUsed] Web: Play Ringback Tone to IP For a description, see the global parameter PlayRBTone2IP. [IpProfile_PlayRBTone2IP] Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Enable Early Media For a description, see the global parameter EnableEarlyMedia. [IpProfile_EnableEarlyMedia] Web: Copy Destination Number to For a description, see the global parameter Redirect Number CopyDest2RedirectNumber. [IpProfile_CopyDest2RedirectN umber] Web: Media Security Behavior...
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For a description, see the global parameter [IpProfile_EnableQSIGTunnelin EnableQSIGTunneling. Web: Enable Hold For a description, see the global parameter EnableHold. [IpProfile_EnableHold] [IpProfile_EnableEarly183] For a description, see the global parameter EnableEarly183. [IpProfile_EarlyAnswerTimeout] For a description, see the global parameter EarlyAnswerTimeout. Version 6.6 Mediant 600 & Mediant 1000...
(i.e., the CO or the PBX) and connects to analog telephones, dial-up modems, and fax machines. The FXS is designed to supply line voltage and ringing current to these telephone devices. An FXS VoIP device interfaces between the analog telephone devices and the Internet. Version 6.6 Mediant 600 & Mediant 1000...
180\182\183\200\4xx uses the same logic and same limitations, in some cases the result may be a different response code. OPTIONS: terminated at each leg independently. INFO: only specific INFO’s (such as DTMF) are handled; other types are omitted. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 UPDATE: terminated at each leg independently and may cause only changes in the RTP flow - Hold\Retrieve are the only exceptions that traverse the two legs. Re-INVITE: terminated at each leg independently and may cause only changes in the RTP flow - Hold\Retrieve are the only exceptions that traverse the two legs.
AOR or Contact. Once an entry is found, the IP destination is obtained and a SIP request is then sent to the user. Figure 20-2: IP-to-IP Routing/Registration/Authentication of Remote IP-PBX Users (Example) Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 The device also supports the IP-to-IP call routing Survivability mode feature (see the figure below) for User-type IP Groups. The device stores in its database REGISTER messages sent by the clients of the User-type IP Group. If communication with the Serving IP Group (e.g., IP-PBX) fails, the User-type IP Group enters into Survivability mode in which the...
PBX via the device. All dialed calls from the IP-PBX consisting of four digits starting with digit "4" are routed to the remote IP-PBX users. Using SIP trunks, the IP-PBX connects (via the device) to two different ITSP's: • ITSP-A: Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 ♦ Implements Proxy servers with fully qualified domain names (FQDN): "Proxy1.ITSP-A" and "Proxy2.ITSP-B", using TLS. ♦ Allocates a range of PSTN numbers beginning with +1919, which is assigned to a range of IP-PBX users. ♦...
The number of media channels represents the number of digital signaling processors (DSP) channels that the device allocates to IP-to-IP calls. The remaining DSP channels can be used for PSTN calls. Two IP media channels are used per IP-to-IP call. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure the number of media channels: Open the IP Media Settings page (Configuration tab > VoIP menu > IP Media > IP Media Settings). Figure 20-6: Defining Required Media Channels In the 'Number of Media Channels' field, enter the required number of media channels (in the example above, "120"...
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In the 'Proxy Address' column, enter the IP addresses of the ITSP-B SIP trunk (e.g., "216.182.224.202" and "216.182.225.202"). From the 'Transport Type' drop-down list corresponding to the IP address entered above, select UDP. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 In the 'Enable Proxy Keep Alive' drop-down list, select "Using Options", and then in the 'Proxy Load Balancing Method' drop-down list, select Round Robin. Figure 20-8: Proxy Set ID #2 for ITSP-B Configure Proxy Set ID #3 for the IP-PBX: From the 'Proxy Set ID' drop-down list, select 3.
In the 'SIP Group Name' field, enter the host name that is used internal in the device's database for this IP Group (e.g., RemoteIPPBXusers). From the 'Serving IP Group ID' drop-down list, select 3 (i.e. the IP Group for the IP-PBX). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Note: No Serving IP Groups are defined for ITSP-A and ITSP-B. Instead, the Outbound IP Routing table (see 'Step 9: Configure Outbound IP Routing' on page 270) is used to configure outbound IP call routing for calls originating from these ITSP IP Groups.
Profiles > IP Profile Settings). Configure Profile ID #1 for the IP-PBX (as shown below): From the 'Profile ID' drop-down list, select 1. From the 'Coder Group' drop-down list, select Coder Group 1. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Click Submit. Figure 20-13: Defining IP Profile ID 1 Configure Profile ID #2 for the ITSP's: From the 'Profile ID' drop-down list, select 2. From the 'Coder Group' drop-down list, select Coder Group 2. Click Submit.
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Trunk connected to the device and interfacing with the local PSTN. • 'Source IP Group ID': enter "-1" to indicate that these calls are not assigned to any source IP Group. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 20.2.9 Step 9: Configure Outbound IP Routing This step defines how to configure the device for routing outbound (i.e., sent) IP-to-IP calls. In our example scenario, calls from both ITSP's must be routed to the IP-PBX, while outgoing calls from IP-PBX users must be routed according to destination.
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'Dest IP Group ID': select 4 to indicate the destination IP Group to where the calls must be sent, i.e., to remote IP-PBX users. • 'IP Profile ID': enter "1" to indicate the IP Profile configured for G.711. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 20.2.10 Step 10: Configure Destination Phone Number Manipulation This step defines how to manipulate the destination phone number. The IP-PBX users in our example scenario use a 4-digit extension number. The incoming calls from the ITSP's have different prefixes and different lengths.
BRI trunks can operate with E1 or T1 trunks. • If the protocol type is CAS, you can assign or modify a dial plan (in the 'Dial Plan' field) and perform this without stopping the trunk. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure trunks: Open the Trunk Settings page (Configuration tab > VoIP menu > PSTN submenu > Trunk Settings). On the top of the page, a bar with Trunk number icons displays the status of each trunk, according to the following color codes: •...
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Apply Trunk Settings and the ‘Trunk Configuration State’ displays 'Active'. To save the changes to flash memory, see 'Saving Configuration' on page 494. To reset the device, see 'Resetting the Device' on page 491. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 21.2 TDM and Timing This section describes the configuration of the TDM and clock timing parameters. 21.2.1 Configuring TDM Bus Settings The TDM page allows you to configure the device's Time-Division Multiplexing (TDM) bus settings. For a description of these parameters, see 'PSTN Parameters' on page 747.
(no compilation is required). The change doesn't affect the state machine itself, but rather the configuration. The CAS table used can be chosen in two ways (using the parameter CasChannelIndex): Single CAS table per trunk Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Different CAS table per group of B-channels in a trunk To modify the CAS state machine parameters: Open the CAS State Machine page (Configuration tab > VoIP menu > PSTN submenu > CAS State Machines).
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Defines which Signaling System to use in both directions [CasStateMachineDigitSign (detection\generation). alingSystem] [0] DTMF = Uses DTMF signaling. [1] MF = Uses MF signaling (default). [-1] Default = Default value - use value from CAS state machine. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 21.4 Configuring Digital Gateway Parameters The Digital Gateway Parameters page allows you to configure miscellaneous digital parameters. For a description of these parameters, see 'Configuration Parameters Reference' on page 609. To configure the digital gateway parameters: Open the Digital Gateway Parameters page (Configuration tab >...
‘Transparent’ coder to transport data (e.g., for D-channel). You can also use Profiles to assign ToS (for DiffServ) per source - a timeslot carrying data or signaling is assigned a higher priority value than a timeslot carrying voice. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 For tunneling of E1/T1 CAS trunks, set the protocol type to 'Raw CAS' (ProtocolType = 3 / 9) and enable RFC 2833 CAS relay mode ('CAS Transport Type' parameter is set to 'CAS RFC2833 Relay').
Mediant 600 & Mediant 1000 TelProfile_2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$ [\TelProfile] 21.5.1.1 DSP Pattern Detector For TDM tunneling applications, you can use the DSP pattern detector feature to initiate the echo canceller at call start. The device can be configured to support detection of a specific one-byte idle data pattern transmitted over digital E1/T1 timeslots.
The parameter ‘DchConfig_x = Trunk_type’ defines the type of NFAS trunk. Trunk_type is set to 0 for the primary trunk, to 1 for the backup trunk, and to 2 for an ordinary NFAS Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 trunk. ‘x’ denotes the physical trunk ID (0 to the maximum number of trunks). You can also use the Web interface (see 'Configuring Trunk Settings' on page 273). For example, to assign the first four T1 trunks to NFAS group #1, in which trunk #0 is the...
All trunks in the group must be configured with the same values for trunk parameters TerminationSide, ProtocolType, FramingMethod, and LineCode. • After stopping or deleting the backup trunk, delete the group and then reconfigure it. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 21.6.4 Performing Manual D-Channel Switchover in NFAS Group If an NFAS group is configured with two D-channels (Primary and Backup), you can do a manual switchover between these D-channels. To manually switchover from active to standby D-channel: Open the NFAS Group &...
Mediant 600 & Mediant 1000 21.8 Redirect Number and Calling Name (Display) The following tables define the device's redirect number and calling name (Display) support for various ISDN variants according to NT (Network Termination) / TE (Termination Equipment) interface direction:...
To register the Trunk Groups, click the Register button. To unregister the Trunk Groups, click Unregister. The registration method for each Trunk Group is according to the 'Registration Mode' parameter in the Trunk Group Settings page (see Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 'Configuring Trunk Group Settings' on page 293). Trunk Group Table Parameters Parameter Description Module Defines the module (i.e., FXS, FXO, PRI, or BRI) for which you want [TrunkGroup_Module] to define the Trunk Group. From Trunk Defines the starting physical Trunk number in the Trunk Group.
If the channel is unavailable (e.g., busy), the call is put on call waiting (if call waiting is enabled and no other call is on call waiting); otherwise, the call is released. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [1] Cyclic Ascending = The next available channel in the Trunk Group, in ascending cyclic order is selected. After the device reaches the highest channel number in the Trunk Group, it selects the lowest channel number in the Trunk Group, and then starts ascending again.
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Group table (see 'Configuring IP Groups' on page 214). Notes: If this parameter is not configured, the registration is performed according to the global registration parameter, ChannelSelectMode. To enable Trunk Group registration, set the global parameter, Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description IsRegisterNeeded to 1. This is unnecessary for 'Per Account' registration mode. If the device is configured globally to register Per Endpoint and an channel group includes four channels to register Per Gateway, the device registers all channels except the first four channels.
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NOTIFY messages are sent to the IP Group defined by the NotificationIPGroupID parameter. Note: This parameter appears in the table only if the VoiceMailInterface parameter is set to 3 (QSIG). Configuring Voice Mail on page 388. Version 6.6 Mediant 600 & Mediant 1000...
For example, if you enter 551 in Index 1 and 55 in Index 2, the device applies rule 1 to numbers that start with 551 and applies rule 2 to numbers that start with 550, 552, 553, Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 and so on until 559. However, if you enter 55 in Index 1 and 551 in Index 2, the device applies rule 1 to all numbers that start with 55, including numbers that start with 551.
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Rule 4: When the source number has prefix from 30 to 40 and a digit (e.g., 3122), it is changed to 2312. Rule 5: When the destination number has the prefix 6, 7, or 8 (e.g., 85262146), source number prefix 2001, it is changed to 3146. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Rule 1 Rule 2 Rule 3 Rule 4 Rule 5 Source IP Group Destination [6,7,8] Prefix Source Prefix 1001 123451001# [30-40]x 2001 Stripped Digits from Left Stripped Digits from Right Prefix to Add Suffix to Add...
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Digits [RemoveFromLeft] Web: Stripped Digits From Defines the number of digits to remove from the right of the telephone Right number prefix. For example, if you enter 3 and the phone number is Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description EMS: Number Of Stripped 5551234, the new phone number is 5551. Digits [RemoveFromRight] Web: Prefix to Add Defines the number or string that you want added to the front of the EMS: Prefix/Suffix To Add telephone number.
"7" in the 'Stripped Digits From Left' field. Example of Configured Rule for Manipulating Prefix using Special Notation Parameter Rule 1 Destination +5492028888888 Prefix Source Prefix Source IP Address Stripped Digits from Left Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Rule 1 Prefix to Add 0[5,3]15 In this configuration example, the following manipulation process occurs: The prefix is calculated as 020215. The first seven digits from the left are removed from the original number, thereby changing the number to 8888888.
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Plan Notation for Routing and Manipulation Tables' on page 607. Web: Source Trunk Group Defines the source Trunk Group ID for Tel-to-IP calls. To denote all Trunk Groups, leave this field empty. Notes: Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description This parameter is applicable only to the Calling Name Manipulations Tel2IP table. The value -1 indicates that this field is ignored in the rule. This parameter is applicable only to Tel-to-IP calls.
The Redirect Prefix parameter is used before it is manipulated. • The redirect number manipulation tables can also be configured using the ini file: - Redirect Number IP to Tel table: RedirectNumberMapIp2Tel (ini) - Redirect Number Tel to IP table: RedirectNumberMapTel2Ip (ini) Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure redirect number manipulation rules: Open the required redirect number manipulation table (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations > Redirect Number Tel > IP or Redirect Number IP >...
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Defines the number or string that you want added to the end of the [Suffix2Add] redirect number. For example, if you enter 00 and the redirect number is 1234, the new number is 123400. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Web: Presentation Enables caller ID. EMS: Is Presentation Not Configured = Privacy is determined according to the Caller ID Restricted table (see Configuring Caller Display Information on page 396). [IsPresentationRestricted [0] Allowed = Sends Caller ID information when a call is made using these destination / source prefixes.
SIP INVITE message. The same mapping occurs when an INVITE with a 'phone-context' parameter is received. For example, for a Tel-to-IP call with NPI/TON set as E164 National (values 1/2), the device sends the following SIP INVITE URI: Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 sip:12365432;phone-context= na.e.164.nt.com This is configured for entry 3 in the figure below. In the opposite direction (IP-to-Tel call), if the incoming INVITE contains this 'phone-context' (e.g. "phone-context= na.e.164.nt.com"), the NPI/TON of the called number in the outgoing Setup message is changed to E164 National.
The release cause mapping tables can also be configured using the ini file: 1) Release Cause Mapping from ISDN to SIP table: CauseMapISDN2SIP (ini). 2) Release Cause Mapping from SIP to ISDN table:CauseMapSIP2ISDN (ini). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure Release Cause mapping: Open the Release Cause Mapping page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations > Release Cause Mapping). Figure 23-6: Release Cause Mapping Page In the 'Release Cause Mapping from ISDN to SIP' group, map different Q.850 Release...
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Does not exist anywhere Unallocated number 606* Not acceptable Network out of order * Messages and responses were created because the ‘ISUP to SIP Mapping’ draft does not specify their cause code mapping. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 23.8.2 Fixed Mapping of ISDN Release Reason to SIP Response The following table describes the mapping of ISDN release reason to SIP response. Mapping of ISDN Release Reason to SIP Response ISDN Release Description Description...
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Recovery of timer expiry Request timeout Protocol error Server internal error Interworking unspecified Server internal error * Messages and responses were created because the ‘ISUP to SIP Mapping’ draft doesn’t specify their cause code mapping. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 23.8.3 Reason Header The device supports the SIP Reason header according to RFC 3326. The Reason header conveys information describing the disconnection cause of a call: Sending Reason header: If a call is disconnected from the Tel side (ISDN), the Reason header is set to the received Q.850 cause in the appropriate message...
1/1 - International number in ISDN/Telephony numbering plan 1/2 - National number in ISDN/Telephony numbering plan 1/4 - Subscriber (local) number in ISDN/Telephony numbering plan 9/4 - Subscriber (local) number in Private numbering plan Version 6.6 Mediant 600 & Mediant 1000...
Source and destination Request-URI host name prefix • Source Trunk Group (from where the call is received) • Source (calling) and destination (called) telephone number prefix and suffix • Source and destination Request-URI host name prefix Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Destination: If the call matches the configured characteristics, the device routes the call to an IP destination. If no characteristics match is found in the table, the call is rejected. The destination can be any of the following: •...
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IP address, or use an FQDN that resolves into two IP addresses. For more information on alternative routing, see 'Alternative Routing for Tel-to-IP Calls' on page 336. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Notes: • Outbound IP routing can be performed before or after number manipulation. This is configured using the RouteModeTel2IP parameter, as described below. • The Outbound IP Routing Table can also be configured using the table ini file parameter, Prefix.
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[0] Route calls before manipulation = Calls are routed before the number manipulation rules are applied (default). [1] Route calls after manipulation = Calls are routed after the number manipulation rules are applied. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Notes: This parameter is not applicable if outbound proxy routing is used. For number manipulation, see 'Configuring Source/Destination Number Manipulation' on page 299. Web: Src. IP Group ID Defines the IP Group from where the incoming IP call is received.
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10.8.8.* denotes all addresses between 10.8.8.0 and 10.8.8.255. Web: Port Defines the destination port to where you want to route the call. EMS: Destination Port [PREFIX_DestPort] Web/EMS: Transport Defines the transport layer type for sending the IP call: Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Type [-1] Not Configured [PREFIX_TransportTyp [0] UDP [1] TCP [2] TLS Note: When set to Not Configured (-1), the transport type defined by the SIPTransportType parameter is used. Web: Dest IP Group ID Defines the IP Group to where you want to route the call.
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You can implement Forking Groups when the destination is an LDAP server or a domain name using DNS. In such scenarios, the INVITE is sent to all the queried LDAP or resolved IP addresses Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description respectively. You can also use LDAP routing rules with standard routing rules for Forking Groups. When the UseDifferentRTPportAfterHold parameter is enabled, every forked call is sent with a different RTP port. Thus, ensure that the device has available RTP ports for these forked calls.
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Defines the Request-URI host name prefix of the incoming SIP INVITE [DestPrefix] message. If this routing rule is not required, leave the field empty. Note: The asterisk (*) wildcard can be used to depict any prefix. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Source Host Prefix Defines the From URI host name prefix of the incoming SIP INVITE [SrcHostPrefix] message. If this routing rule is not required, leave the field empty. Notes: The asterisk (*) wildcard can be used to depict any prefix.
If the destination does not respond to the OPTIONS message, then it is considered unavailable. You can configure the time interval for sending these OPTIONS messages, using the 'Alt Routing Tel to IP Keep Alive Time' parameter. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 These parameters are configured in the Routing General Parameters page (Configuration tab > VoIP menu > GW and IP to IP > Routing > General Parameters), as shown below: Figure 24-4: IP Connectivity Method in Routing General Parameters Page ...
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In the Routing General Parameters page (Configuration tab > VoIP menu > GW and IP to IP > Routing > General Parameters), do the following: Enable alternative routing based on IP connectivity, by setting the 'Enable Alt Routing Tel to IP AltRouting' (Tel2IPEnable) parameter to Enable. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Configure the IP connectivity reason for triggering alternative routing, by setting the 'Alt Routing Tel to IP Mode' parameter (AltRoutingTel2IPMode) to one of the following: ♦ Ping or SIP OPTIONS failure ♦ Poor QoS ♦...
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Figure 24-6: Tel to IP Reasons - Reasons for Alternative Routing Page Under the 'Tel to IP Reasons' group, select up to five different SIP response codes (call failure reasons) that invoke alternative Tel-to-IP routing. Click Submit. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 24.5.3 PSTN Fallback The PSTN Fallback feature enables the device to re-route a Tel-to-IP call to the legacy PSTN using one of its trunks if the IP destination is unavailable. For example, if poor voice quality is detected over the IP network, the device attempts to re-route the call to the PSTN.
(i.e., user@host). For example, the below configuration forwards IP-to- Tel calls to destination user “112” at host IP address 10.13.4.12, port 5060, using transport protocol TCP, if Trunk Group ID 2 is unavailable: Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 ForwardOnBusyTrunkDest 1 = 2, 112@10.13.4.12:5060;transport=tcp; When configured with user@host, the original destination number is replaced by the user part. The device forwards calls using this table only if no alternative IP-to-Tel routing rule has been configured in the Inbound IP Routing table or alternative routing fails and one of the following reasons in the SIP Diversion header of 3xx messages exists: ...
> DTMF & Supplementary submenu > DTMF & Dialing). Configure the parameters as required. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 494. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 25.1 Dialing Plan Features This section describes various dialing plan features supported by the device. 25.1.1 Digit Mapping Digit map pattern rules are used for Tel-to-IP ISDN overlap dialing (by setting the ISDNRxOverlap parameter to 1) to reduce the dialing period (for digital interface). For more information on digit maps for ISDN overlapping, see ISDN Overlap Dialing on page 288.
Dial Plan. 25.1.2 External Dial Plan File The device can be loaded with a Dial Plan file with user-defined dialing plans. For more information, see 'Dial Plan File' on page 508. Version 6.6 Mediant 600 & Mediant 1000...
'Configuration Parameters Reference' on page 609. To configure supplementary services parameters: Open the Supplementary Services page (Configuration tab > VoIP menu > GW and IP to IP submenu > DTMF & Supplementary submenu > Supplementary Services). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Configure the parameters as required. Click Submit to apply your changes, or click the Subscribe to MWI or Unsubscribe to MWI buttons to save your changes and to subscribe / unsubscribe to the MWI server.
Mediant 600 & Mediant 1000 The flowchart above describes the following "double" call-hold scenario: A calls B and establishes a voice path. A places B on hold; A hears a dial tone and B hears a held tone. A calls C and establishes a voice path.
The transfer can be initiated at any of the following stages of the call between A and C: Just after completing dialing C phone number - transfer from setup While hearing ringback – transfer from alert While speaking to C - transfer from active Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Note: For FXS interfaces, the device can also handle call transfers using SIP INVITE and re-INVITE messages, instead of REFER messages. This is useful when communicating with SIP UAs that do not support the receipt of REFER messages.
Served party: party configured to forward the call (FXS device). Originating party: party that initiates the first call (FXS or FXO device). Diverted party: new destination of the forwarded call (FXS or FXO device). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 The served party (FXS interface) can be configured through the Web interface (see Configuring Call Forward on page 397) or ini file to activate one of the call forward modes. These modes are configurable per endpoint.
When the call forwarding service is de-activated, for example, by dialing #21# and sending an INVITE with this number, the softswitch sends another SIP NOTIFY message with the following Alert-Info header: Alert-Info: <http://127.0.0.1/ Tono-Normal-Invitacion>; Aviso = Desvió-Inmediato Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 From this point on, the device plays a normal dial tone to the FXS phone when it goes off- hook. 26.6.4 BRI Call Forwarding The device supports call forwarding (CF) services initiated by ISDN Basic BRI phones connected to it.
For more information on IP voice mail configuration, refer to the IP Voice Mail CPE Configuration Guide. • For creating a CPT file using AudioCodes CPTWizard utility, refer to CPTWizard Utility User's Guide. To configure MWI, use the following parameters: ...
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Mediant 600 & Mediant 1000 ETSIVMWITypeOneStandard BellcoreVMWITypeOneStandard VoiceMailInterface EnableVMURI The device supports the following MWI features for its digital PSTN interfaces: For BRI interfaces: This feature provides support for MWI on BRI phones connected to the device and using the Euro ISDN BRI variant.
To debug a Caller ID detection on an FXO interface: Verify that the parameter EnableCallerID is set to 1. Verify that the caller ID standard (and substandard) of the device matches the standard (using parameters CallerIDType, Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 BellcoreCallerIDTypeOneSubStandard, and ETSICallerIDTypeOneSubStandard). Define the number of rings before the device starts the detection of caller ID (using the parameter RingsBeforeCallerID). Verify that the correct FXO coefficient type is selected (using the parameter CountryCoefficients), as the device is unable to recognize caller ID signals that are distorted.
The device supports the following conference modes (configured by the parameter 3WayConferenceMode): Conferencing managed by an external, AudioCodes Conference (media) server: The Conference-initiating INVITE sent by the device uses the ConferenceID concatenated with a unique identifier as the Request-URI. This same Request-URI is set as the Refer-To header value in the REFER messages that are sent to the two remote parties.
Mediant 600 & Mediant 1000 Therefore, when the MPM modules are installed in chassis slots #4, #5, and #6, the device supports up to 100 DSP resources, thereby supporting up to 100 conference participants, which is 33 three-way conferences. Notes: •...
When the call is answered by the PSAP operator, the PSAP sends a SIP 200 OK to the FXS device, and the FXS device then generates a polarity reversal signal to the E911 switch. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 After the call is disconnected by the PSAP, the PSAP sends a SIP BYE to the FXS device, and the FXS device reverses the polarity of the line toward the tandem switch. The following parameters need to be configured: ...
E911 tandem switch (E911 Selective Router) and the PSAP, using analog loop-start lines. The FXO device can be implemented instead of an E911 switch, by connecting directly to the PSAP DID loop-start lines. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Figure 26-4: FXO Device Interfacing between E911 Switch and PSAP When an IP phone subscriber dials 911, the device receives the SIP INVITE message and makes a call to the PSAP as follows: The FXO device seizes the line.
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The MF duration of all digits, except for the KP digit is 60 msec. The MF duration of the KP digit is 120 msec. The gap duration is 60 msec between any two MF digits. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Notes: • Manipulation rules can be configured for the calling (ANI) and called number (but not on the "display" string), for example, to strip 00 from the ANI "00INXXYYYY". • The called number, received as userpart of the Request URI ("301" in the example below), can be used to route incoming SIP calls to FXO specific ports, using the TrunkGroup and PSTNPrefix parameters.
911, enables emergency operators to pinpoint the location (granular location) of callers who dial the 9-1-1 emergency telephone number. Today, most enterprises implement an IP-based infrastructure providing a VoIP network with fixed and nomadic users, allowing connectivity anywhere with any device. This, Version 6.6 Mediant 600 & Mediant 1000...
E9-1-1 due to the difficulty in accurately locating the E9-1-1 caller. This section describes the E9-1-1 solution provided by Microsoft Lync Server 2010 (hereafter referred to as Lync Server 2010), and the deployed AudioCodes ELIN Gateway which provides the ISDN (or CAMA) connectivity to the PSTN-based E9-1-1 emergency providers.
(for example, a WAP) with a physical location in the Enterprise (for example, Floor 2, Wing A, and the Enterprise's street address). For more information on populating the LIS database, see 'Adding ELINs to the Location Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Information Server' on page 373. The Administrator validates addresses with the Emergency Services provider's MSAG –a companion database to the ALI database. This ensures that the civic address is valid as an official address (e.g., correct address spelling).
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Emergency Response Locations (ERLs) and assign a dedicated Emergency Location Identification Number (ELIN) to each ERL (or zone). When Lync Server 2010 sends a SIP INVITE message with the PIDF-LO to the PSTN Gateway, it Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 can parse the content and translate the calling number to an appropriate ELIN. The PSTN Gateway then sends the call to the PSTN with the ELIN number as the calling number. This ELIN number is sent to the Emergency Services provider, which sends it on to the appropriate PSAP according to the ELIN address match in the ALI database lookup.
PSAP, based on ELIN-address match lookup in the Emergency Services provider's ALI database. The figure below illustrates an AudioCodes ELIN Gateway deployed in the Lync Server 2010 environment for handling E9-1-1 calls between the Enterprise and the PSTN.
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ELIN is disregarded and no longer used (for PSAP callback). Therefore, table entries of only the most recently terminated E9-1-1 callers are considered in the ELIN table. The maximum entries in the ELIN table depend on the AudioCodes ELIN Gateway deployed in the Lync Server 2010 environment: •...
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E9-1-1 call. This feature is initiated only if the received SIP INVITE message contains a Priority header set to "emergency", as shown below: PRIORITY: emergency Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 26.11.4.3.3 PSAP Callback to Lync 2010 Clients for Dropped E9-1-1 Calls As the E9-1-1 service automatically provides all the contact information of the E9-1-1 caller to the PSAP, the PSAP operator can call back the E9-1-1 caller. This is especially useful in cases where the caller disconnects prematurely.
26.11.4.4 Configuring AudioCodes ELIN Gateway This section describes E9-1-1 configuration of the AudioCodes ELIN Gateway deployed in the Lync Server 2010 environment. 26.11.4.4.1 Enabling the E9-1-1 Feature By default, the E9-1-1 feature in the ELIN Gateway for Lync Server 2010 is disabled.
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Mediant 600 & Mediant 1000 entity (for example, another ELIN Gateway), instead of retrying the call or returning the release call to the user. To support this requirement, the ELIN Gateway can be configured to send the 503 "Service Unavailable" release cause code instead of SIP 4xx if an emergency call cannot be established.
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Mediant 600 & Mediant 1000 MLPP Precedence Level Precedence Level in DSCP Configuration Parameter Resource-Priority SIP Header 9 (highest) flash-override-override MLPPFlashOverOverRTPDSCP The device automatically interworks the network identity digits (NI) in the ISDN Q.931 Precedence Information Element (IE) to the network domain subfield of the INVITE's Resource-Priority header, and vice versa.
The device sends a Q.931 DISCONNECT over the ISDN MLPP PRI to the partner switch to preempt the remote end instrument. The device sends a 488 (Not Acceptable Here) response with this Reason cause code. Reason: preemption; cause=5; text=”Network Preemption” Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 This Reason cause code indicates preempted events in the network. Within the Defense Switched Network (DSN) network, the following SIP request messages and response codes for specific call scenarios have been identified for signaling this preemption cause: •...
Progress Indicator in the Setup Ack bit (0x10000=65536). Therefore, the recommended value is 0x10000 + 0 x1000 = 65536 + 4096 = 69632 (i.e., set the ISDNInCallsBehavior parameter to 69632). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure BRI supplementary services: Open the ISDN Supp Services Table page (Configuration tab > VoIP menu > GW and IP to IP > Digital Gateway > ISDN Supp Services). Figure 26-9: ISDN Supp Services Table Page Configure the parameters as described in the table below.
'Pulses On Answer' - number of charging units in first generated AOC-D Facility message Assign the charge code index to the desired routing rule in the Outbound IP Routing table (see 'Configuring Outbound IP Routing Table' on page 323). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 26.16 Configuring Voice Mail The Voice Mail Settings page allows you to configure the voice mail parameters. For a description of these parameters, see 'Configuration Parameters Reference' on page 609. Notes: • The Voice Mail Settings page is available only for FXO and CAS interfaces.
The activation of each feature remains in effect until it is deactivated (i.e., not deactivated after a call). • For a description of the keypad parameters, see 'Telephone Keypad Sequence Parameters' on page 793. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure the keypad features Open the Keypad Features page (Configuration tab > VoIP menu > GW and IP to IP > Analog Gateway > Keypad Features). Figure 27-1: Keypad Features Page Configure the keypad features as required.
(AoC) services for Euro ISDN trunks (see Advice of Charge Services for Euro ISDN on page 387). • The Charge Codes table can also be configured using the table ini file parameter, ChargeCode. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure the Charge Codes: Open the Charge Codes Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Charge Codes). Alternatively, you can access this page from the Metering Tones page (see 'Configuring Metering Tones' on page 390).
After you click Submit, the password is displayed as an asterisk (*). • The Authentication table can also be configured using the table ini file parameter, Authentication (see 'Configuration Parameters Reference' on page 609). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure authentication credentials per port: Set the parameter 'Registration Mode' (AuthenticationMode) to Per Endpoint. This can be configured in any of the following pages: • Proxy & Registration page (see 'Configuring Proxy and Registration Parameters' on page 228).
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FXO line, plays a dial tone, and then waits for DTMF digits. If no digits are detected within a user-defined time, the configured destination number is automatically dialed by sending a SIP INVITE message with this number. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Hotline Dial Tone Defines the duration (in seconds) after which the destination phone Duration number is automatically dialed. This is applicable only if the port has [TargetOfChannel_Hot been configured for Hotline (i.e., 'Auto Dial Status' is set to Hotline).
(Configuration tab > VoIP menu > GW and IP to IP > DTMF and Supplementary > Supplementary Services). • The Call Forward table can also be configured using the table ini file parameter, FwdInfo. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure Call Forward per port: Open the Call Forward Table page (Configuration tab > VoIP menu > GW and IP to IP > Analog Gateway > Call Forward). Configure the table as required. For descriptions of the parameters, see the table below.
Click Submit to apply your changes. Caller ID Permissions Table Parameter Description Parameter Description Caller ID Enables Caller ID generation (FXS) or detection (FXO) per port. [EnableCallerId_IsEnabl [0] Disable [1] Enable Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 27.10 Configuring Call Waiting The Call Waiting table allows you to enable or disable call waiting per FXS port. Notes: • This page is applicable only to FXS interfaces. • You can enable or disable call waiting for all the device's ports using the global parameter, 'Enable Call Waiting' in the Supplementary Services page (Configuration tab >...
The figure above shows a configuration example for using distinctive ringing and call waiting tones of Index #9 ('Priority Index' 1) in the CPT file for FXS endpoints 1 to 4 when a call is received from a source number with prefix 2. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Configure the table as required. For a description of the parameters, see the table below. Click Submit to apply your changes. Tone index Table Parameter Description Parameter Description Index Defines the table index entry. Up to 50 entries can be defined.
One-stage dialing is when the FXO device receives an IP-to-Tel call, off-hooks the PBX line connected to the telephone, and then immediately dials the destination telephone number. In other words, the IP caller doesn't dial the PSTN number upon hearing a dial tone. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 One-stage dialing incorporates the following FXO functionality: Waiting for Dial Tone: Enables the device to dial the digits to the Tel side only after detecting a dial tone from the PBX line. The ini file parameter IsWaitForDialTone is used to configure this operation.
DID wink enables the originating end to seize the line by going off-hook. It waits for acknowledgement from the other end before sending digits. This serves as an integrity check that identifies a malfunctioning trunk and allows the network to send a re-order tone to the calling party. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 The "start dial" signal is a wink from the PBX to the FXO device. The FXO then sends the last four to five DTMF digits of the called number. The PBX uses these digits to complete the routing directly to an internal station (telephone or equivalent).
Hold / Transfer toward the IP side: The FXO device doesn't initiate hold / transfer as a response to input from the Tel side. If the FXO receives a REFER request (with or without replaces), it generates a new INVITE according to the Refer-To header. Version 6.6 Mediant 600 & Mediant 1000...
PBX / CO must be recorded and its frequencies analyzed. Adding a reorder tone to the CPT file can be done using AudioCodes CPTWizard utility (refer to the CPTWizard Utility User's Guide). This method is slightly less reliable than the previous one.
FXS device. The routing is transparent as if the telephone connected to the FXS device is directly connected to the PBX. The following is required: FXO interfaces with ports connected directly to the PBX lines (shown in the figure Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 below) FXS interfaces for the 'remote PBX extension' Analog phones (POTS) PBX (one or more PBX loop start lines) LAN network 27.15.1 Dialing from Remote Extension (Phone at FXS) The procedure below describes how to dial from the 'remote PBX extension' (i.e., phone connected to the FXS interface).
ID to the relevant port for display. The remote extension connected to the FXS device can toggle between calls using the Hook Flash button. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 27.15.5 FXS Gateway Configuration The procedure below describes how to configure the FXS interface (at the 'remote PBX extension'). To configure the FXS interface: In the Trunk Group Table page (see Configuring Trunk Group Table on page 291, assign the phone numbers 100 to 104 to the device's endpoints.
IP address of the FXS device (10.1.10.3) in the field ‘IP Address’. Figure 27-15: FXO Tel-to-IP Routing Configuration In the FXO Settings page (see 'Configuring FXO Parameters' on page 392), set the parameter ‘Dialing Mode’ to Two Stages (IsTwoStageDial = 1). Version 6.6 Mediant 600 & Mediant 1000...
SIP messages that are sent directly from the UAs. For more information, see 'SAS Redundant Mode' on page 419. Note: It is recommended to implement the SAS outbound mode. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 28.1.1 SAS Outbound Mode This section describes the SAS outbound mode, which includes the following states: Normal state (see 'Normal State' on page 418) Emergency state (see 'Emergency State' on page 418) 28.1.1.1 Normal State In normal state, SAS receives REGISTER requests from the enterprise's UAs and forwards them to the external proxy (i.e., outbound proxy).
UAs, you can configure SAS to ignore messages received from UAs in normal state (the 'SAS Survivability Mode' parameter must be set to 'Always Emergency' / 2) and thereby, “force” the UAs to switch back to their primary proxy. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 28.1.2.1 Normal State In normal state, the UAs register and operate directly with the external proxy. Figure 28-3: SAS Redundant Mode in Normal State (Example) 28.1.2.2 Emergency State If the UAs detect that their primary (external) proxy does not respond, they immediately register to SAS and start routing calls to it.
28.2.1 SAS Routing in Normal State The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from UAs: Figure 28-5: Flowchart of INVITE from UA's in SAS Normal State Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the external proxy: Figure 28-6: Flowchart of INVITE from Primary Proxy in SAS Normal State User's Manual Document #: LTRT-83313...
User's Manual 28. Overview 28.2.2 SAS Routing in Emergency State The flowchart below shows the routing logic for SAS in emergency state: Figure 28-7: Flowchart for SAS Emergency State Version 6.6 Mediant 600 & Mediant 1000...
Open the Applications Enabling page (Configuration tab > VoIP menu > Applications Enabling > Applications Enabling). From the 'SAS Application' drop-down list, select Enable. Click Submit. Save the changes to the flash memory with a device reset. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 29.1.2 Configuring Common SAS Parameters The procedure below describes how to configure SAS settings that are common to all SAS modes. This includes various SAS parameters as well as configuring the Proxy Set for the SAS proxy (if required).
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'SAS Proxy Set' field in the 'SAS Configuration page (see Step 6). • Do not use Proxy Set ID 0. In the 'Proxy Address' field, enter the IP address of the external proxy server. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 From the 'Enable Proxy Keep Alive' drop-down list, select Using Options. This instructs the device to send SIP OPTIONS messages to the proxy for the keep- alive mechanism. Figure 29-2: Defining SAS Proxy Server Click Submit to apply your settings.
The Gateway application must use the same SAS operation mode as the SIP UAs. For example, if the UAs use the SAS application as a redundant proxy (i.e., SAS redundancy mode), then the Gateway application must do the same. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 29.4.1 Gateway with SAS Outbound Mode The procedure below describes how to configure the Gateway application with SAS outbound mode. To configure Gateway application with SAS outbound mode: Define the proxy server address for the Gateway application: Open the Proxy &...
Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control Network submenu > Proxy Sets Table). From the 'Proxy Set ID' drop-down list, select 0. In the first 'Proxy Address' field, enter the IP address of the external proxy server. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 In the second 'Proxy Address' field, enter the IP address and port of the device (in the format x.x.x.x:port). This is the same port as defined in the 'SAS Local UDP/TCP/TLS Port' field (see 'Configuring Common SAS Parameters' on page 426).
Associated AOR: 3434@10.33.4.226 (after manipulation, in which only the four digits from the right of the URI user part are retained) Contact: 976653434@10.10.10.10 The procedure below describes how to configure the above manipulation example. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To manipulate incoming Request-URI user part of REGISTER message: Open the SAS Configuration page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability). Under the SAS Registration Manipulation group, in the 'Leave From Right' field, enter the number of digits (e.g., "4") to leave from the right side of the user part.
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For any prefix, enter the asterisk "*" symbol (default). Note: The prefix can be a single digit or a range of digits. For available notations, see 'Dialing Plan Notation for Routing and Manipulation' on Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description page 607. Source Host Defines the source SIP URI host name - full name (usually in the From [IPInboundManipulatio header). For any host name, enter the asterisk "*" symbol (default). n_SrcHost] Destination Username...
If no routing rule is located in the database, the device sends the request according to the Request-URI header. Note: The IP-to-IP Routing table can also be configured using the table ini file parameter, IP2IPRouting (see 'Configuration Parameters Reference' on page 609). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To configure the IP-to-IP Routing table for SAS: In the SAS Configuration page, click the SAS Routing Table button; the IP-to-IP Routing Table page appears. Click Add; the Add Record dialog box appears: Figure 29-9: Add Record Dialog Box of SAS IP2IP Routing Page Configure the rule according to the table below.
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[2] Request URI = The SIP dialog is sent to the address indicated in the incoming Request-URI. If the fields 'Destination Port' and 'Destination Transport Type' are configured, the incoming Request- URI parameters are overridden and these fields take precedence. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [3] ENUM = An ENUM query is sent to include the destination address. If the fields 'Destination Port' and 'Destination Transport Type' are configured, the incoming Request-URI parameters are overridden and these fields take precedence.
'SAS Default Gateway IP' field, and this is the device itself. The device then sends the call directly to the PSTN. This feature is applicable to SAS in normal and emergency states. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure SAS emergency numbers: Open the SAS Configuration page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability). In the ‘SAS Default Gateway IP' field, define the IP address and port (in the format x.x.x.x:port) of the device (Gateway application).
[1]: Enable - SAS changes the Contact header so that it points to the SAS host and therefore, the top-most Via header and the Contact header point to the same host. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 29.6 Viewing Registered SAS Users You can view all the users that are registered in the SAS registration database. This is displayed in the 'SAS/SBC Registered Users page, as described in 'Viewing Registered Users' on page 555.
SAS Routing table. In this example, a call is routed from SAS Gateway (A) user to a user on SAS Gateway (B). Figure 30-1: SAS Cascading Using SAS Routing Table - Example Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 SAS Redundancy mode: If users cannot be distinguished (i.e., associated to a specific SAS gateway), then the SAS Redundancy feature is used to configure SAS Cascading. This mode routes the call in a loop fashion, from one SAS gateway to the next, until the user is located.
Some of the IP media functionalities are configured in the IP Media settings page. Note: This page is applicable only to Mediant 1000. This page appears only if your device is installed with the relevant Software License Key (see 'Loading Software License Key' on page 522). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 To access the IP Media Settings page: Open the IP Media Settings page (Configuration tab > VoIP menu > IP Media submenu > IP Media Settings). Figure 31-1: IP Media Settings Page User's Manual...
It is assumed that the device is controlled by a third-party, Application server (or any SIP user agent) that instructs the device to start an IP transcoding call by sending two SIP INVITE messages with SIP URI that includes the transcoding identifier name. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 For example: Invite sip:trans123@audiocodes.com SIP/2.0 The left part of the SIP URI includes the transcoding ID (’the default string is ‘trans’) and is terminated by a unique number (123). The device immediately sends a 200 OK message in response to each INVITE.
Advanced, according to MSCML (see 'Advanced Conferencing (MSCML)' on page 455) Note: The conference application is a special order option. 33.1 Simple Conferencing (NetAnn) 33.1.1 SIP Call Flow A SIP call flow for simple conferencing is shown below: Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 33.1.2 Creating a Conference The device creates a conference call when the first user joins the conference. To create a conference, the Application Server sends a regular SIP INVITE message to the device. The User Part of that Request-URI includes both the Conference Service Identifier (indicating that the requested Media Service is a Conference) and a Unique Conference Identifier (identifying a specific instance of a conference).
The Application Server can change the value of reserveconfmedia during an existing conference. By default, reserveconfmedia is set to Yes. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 33.2.2 Joining a Conference To join an existing conference, the Application Server sends a SIP INVITE message with the same Request-URI as the one that created the conference. The INVITE message may include a <configure_leg> MSCML request body. If not included, defaults are used for that leg attributes.
When issuing a Media Service on the Control Leg, it affects all Participant Legs in the conference, e.g., play an announcement. When issuing a Media Service on a Participant Leg, it affects the specific leg only. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 33.2.5 Active Speaker Notification After an advanced conference is established, the Application Server can subscribe to the device to receive notifications of the current set of active speakers in a conference at any given moment. This feature is referred to as Active Speaker Notification (ASN) and is designed according to the MSCML standard.
33.3 Conference Call Flow Example The call flow, shown in the following figure, describes SIP messages exchanged between the device (10.8.58.4) and three conference participants (10.8.29.1, 10.8.58.6 and 10.8.58.8). Version 6.6 Mediant 600 & Mediant 1000...
Sending a BYE request terminates the SIP session and stops the playing of the announcement. If the played announcement reaches its end, the device initiates a BYE message to notify the Application Server that the session is ended. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Notes: • A 200 OK message is sent only after the HTTP connection is successfully established and the requested file is found. If the file isn’t found, a 404 Not Found response is sent. •...
Remote Storage: An HTTP server that contains less-frequently used voice prompts for playback and to which voice stream recording is performed. IP Phones / MediaPack: Client applications. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 34.2.1 Operation The Application server communicates with the device using MSCML Requests (sent by the Application server), as shown in the example below: <?xml version="1.0" encoding="utf-8"?> <MediaServerControl version="1.0"> <request> ... request body ... </request> </MediaServerControl>...
(in milliseconds) the user does not enter any digits after the first DTMF digit is received, after which a response is sent indicating timeout. extradigittimer: used to enable the following: • Detection of command keys (ReturnKey and EscapeKey). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • Not report the shortest match. MGCP Digitmap searches for the shortest possible match. This means that if a digitmap of (123 | 1234) is defined, once the user enters 123, a match is found and a response is sent. If ExtraDigitTimer is defined, the match can also be 1234 because the device waits for the next digits.
<Stop> request. The device immediately terminates the request in progress and sends a response. An example is shown below of an MSCML <Stop> Request: <?xml version="1.0" encoding="utf-8"?> <MediaServerControl version="1.0"> <request> <stop id="123"> </stop> </request> </MediaServerControl> Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 34.2.6 Relevant Parameters The following parameters (described in IP Media Parameters) are used to configure the MSCML: AmsProfile = 1 (default and mandatory setting) AASPackagesProfile = 3 (default and mandatory setting) VoiceStreamUploadMethod = 1 (mandatory) ...
Mediant 600 & Mediant 1000 34.3 Voice Streaming The voice streaming layer provides you with the ability to play and record different types of files while using an NFS or HTTP server. 34.3.1 Voice Streaming Features The following subsections summarizes the Voice Streaming features supported on HTTP and NFS servers, unless stated otherwise.
Note: This feature is relevant for both NFS and HTTP. 34.3.1.10 Record Files Using LBR You may record a file using low bit rate coders for .wav and .raw files. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Notes: This feature is relevant for both NFS and HTTP. 34.3.1.11 Modifying Streaming Levels Timers Several parameters enable the user to control streaming level timers for NFS and HTTP and also the number of data retransmission when using NFS as the application layer protocol: ...
Channel Coder Channel Coder PCMA PCMU LB PCMA PCMU PCMA PCMU PCMA PCMU No No 34.3.3 Maximum Concurrent Playing and Recording For details on maximum concurrent playing and recording, refer to the Release Notes. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 34.3.4 LBR Coders Support The following table describes the different low bit rate (LBR) coders and their support for .wav, .au, and .raw files. Note: Coder support depends on the specific DSP template version installed on the device.
• For further details, see 'Configuring the NFS Settings' on page 128. 34.3.7 Supported HTTP Servers The following is a list of HTTP servers that are known to be compatible with AudioCodes voice streaming under Linux™: Apache: cgi scripts are used for recording and supporting dynamic URLs.
MinSpareThreads MaxSpareThreads ThreadsPerChild MaxRequestsPerChild 16384 </IfModule> 34.3.8 Supporting NFS Servers The table below lists the NFS servers that are known to be compatible with AudioCodes Voice Streaming functionality. Compatible NFS Servers Operating System Server Versions Solaris™ 5.8 and 5.9 nfsd 2, 3 Fedora™...
If the systems administrator wishes to use a default other than AUTH_SYS in the nfssec.conf file, then you should add "sec=sys" to each line in the dfstab file that is to be shared with an AudioCodes system. For example: > cat /etc/dfs/dfstab...
34.3.8.2 Linux-Based NFS Servers The AudioCodes device uses local UDP ports that are outside of the range of 0..IPPORT_RESERVED(1024). Therefore, when configuring a remote file system to be accessed by an AudioCodes device, use the insecure option in the /etc/exports file. The insecure option allows the nfs daemon to accept mount requests from ports outside of this range.
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Syslogs: server is full. the file system. 'NFS request aborted, reqid=209 cid=- 1 error=writeReplyError reqtype=writeFile state=writeWait [File:NfsStateMachine.cpp …]' 'NFS request aborted, reqid=186 cid=- 1 error=writeReplyError reqtype=writeFile state=writeWait [File:NfsStateMachine.cpp …]' Version 6.6 Mediant 600 & Mediant 1000...
Announcement Call Flow Example The call flow, shown in the following figure, describes SIP messages exchanged between the device (10.33.24.1) and a SIP client (10.33.2.40) requesting to play local announcement #1 (10.8.25.17) using AudioCodes proprietary method. SIP MESSAGE 1: 10.33.2.40:5060 -> 10.33.24.1:5060 INVITE sip:annc@10.33.24.1;play=http://10.3.0.2/hello.wav;repeat=2...
'Toolbar Description' on page 42) to indicate that a device reset is required. • After you reset the device, the Web GUI is displayed in Basic view (see 'Displaying Navigation Tree in Basic and Full View' on page 43). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To reset the device: Open the Maintenance Actions page (see 'Basic Maintenance' on page 491). Under the 'Reset Configuration' group, from the 'Burn To FLASH' drop-down list, select one of the following options: •...
Under the Misc Parameters group, set the 'SIP Remote Rest' parameter to Enable. Click Submit. Note: This SIP Event header value is proprietary to AudioCodes. 35.3 Locking and Unlocking the Device The Lock and Unlock option allows you to lock the device so that it doesn't accept any new calls and maintains only the current calls.
Mediant 600 & Mediant 1000 Otherwise, the lock process begins immediately. The Current Admin State' field displays the current state - "LOCKED" or "UNLOCKED". To unlock the device: Open the Maintenance Actions page (see 'Maintenance Actions' on page 491).
In the Home page, click the title of the module that you want to replace; the Remove Module button appears: Figure 36-1: Remove Module Button Click the Remove Module button; a message box appears requesting you to confirm module removal: Figure 36-2: Module Removal Confirmation Message Box Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Click OK to confirm removal; after a few seconds, the module is software- removed, the module status icon turns to grey, and the name of the module is suffixed with the word 'Reserved': Physically remove the module (refer to the Installation Manual).
B-channel is restarted. • B-channel restart can only be done if the D-channel of the trunk to which it belongs is synchronized (Layer 2). • B-channel restart does not affect the B-channel's configuration. Version 6.6 Mediant 600 & Mediant 1000...
IP address for outbound IP routing. For more information, see 'Dial Plan File' on page 508. VXML Voice Extensible Markup Language (VXML) script file. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 File Description User Info The User Information file maps PBX extensions to IP numbers. This file can be used to represent PBX extensions as IP phones in the global 'IP world'. For more information, see 'User Information File' on page 514.
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Browse button. Click the Load File button corresponding to the file you want to load. Repeat steps 2 through 3 for each file you want to load. Version 6.6 Mediant 600 & Mediant 1000...
AudioCodes DConvert utility. For a description on converting a CPT ini file into a binary dat file, refer to the DConvert Utility User's Guide.
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Can be omitted if there isn't a second cadence. • Second Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the second cadence on-off cycle. Can be omitted if there isn't a second cadence. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • Third Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the third cadence on-off cycle. Can be omitted if there isn't a third cadence. • Third Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the third cadence on-off cycle.
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First Burst Ring On Time [10msec]=30 First Burst Ring Off Time [10msec]=30 Second Burst Ring On Time [10msec]=30 Second Burst Ring Off Time [10msec]=30 Third Burst Ring On Time [10msec]=30 Third Burst Ring Off Time [10msec]=30 Version 6.6 Mediant 600 & Mediant 1000...
Channels: mono Once created, the PRT file can then be loaded to the device using AudioCodes' AcBootP utility or the Web interface (see 'Loading Auxiliary Files' on page 499). The prerecorded tones are played repeatedly. This allows you to record only part of the tone and then play the tone for the full duration.
The CAS files can be loaded to the device using the Web interface or ini file (see 'Loading Auxiliary Files' on page 499). Note: All CAS files loaded together must belong to the same Trunk Type (i.e., either E1 or T1). Version 6.6 Mediant 600 & Mediant 1000...
Plans, as required. Save the file with the ini file extension name (e.g., mydialplanfile.ini). Convert the ini file to a dat binary file, using AudioCodes DConvert utility. For more information, refer to DConvert Utility User's Guide. Install the converted file on the device, as described in 'Loading Auxiliary Files' on page 499.
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Notes: • To use the Dial Plan file, you must also use a special CAS .dat file that supports this feature. For more information, contact your AudioCodes sales representative. • For E1 CAS MFC-R2 variants, which don't support terminating digit for the called party number, usually I-15, the Dial Plan file and the DigitMapping parameter are ignored.
Plans, as required. Save the file with the ini file extension name (e.g., mydialplans.ini). Convert the ini file to a dat binary file, using AudioCodes DConvert utility. For more information, refer to DConvert Utility User's Guide. Install the converted file on the device, as described in 'Loading Auxiliary Files' on page 499.
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“tag” (in our example, "LOCL" and "LONG"): • The Destination Prefix' field is set to the value "LOCL" and the 'Stripped Digits From Left' field is set to '4'. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 • The Destination Prefix' field is set to the value "LONG" and the 'Stripped Digits From Left' field is set to '4'. Figure 38-2: Configuring Manipulation for Removing Label 38.1.5.4 Obtaining IP Destination from Dial Plan File You can use a Dial Plan index listed in a loaded Dial Plan file for determining the IP destination of Tel-to-IP /IP-to-IP calls.
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Dial Plan file, if the parameter 'Tel To IP Routing Mode' is set to 'Route calls after manipulation'. • Source number Tel-to-IP manipulation is performed on the modified source number as defined in the Dial Plan file. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 38.1.6 User Information File This section describes the various uses of the User Info file. You can load the User Info file using any of the following methods: Web interface (see 'Loading Auxiliary Files' on page 499) ...
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401 , 638001 , Mike , miked , 1234 402 , 638002 , Lee , leep, 4321 403 , 638003 , Sue , suer, 8790 404 , 638004 , John , johnd, 7694 Version 6.6 Mediant 600 & Mediant 1000...
The file is created in .xml format and installed on the device as a binary file (with a .dat extension). The XML to binary file format is processed by AudioCodes DConvert utility. For more information, refer to DConvert Utility User's Guide.
Before you can install a new Software License Key, you need to obtain a Software License Key file for your device with the required features from your AudioCodes representative. The Software License Key is an encrypted key in string format that is associated with the device's serial number ("S/N") and supplied in a text-based file.
AudioCodes AcBootP utility - see Installing Software License Key using AcBootP on page AudioCodes EMS - refer to the EMS User’s Manual or EMS Product Description Note: When you install a new Software License Key, it is loaded to the device's non-volatile flash memory and overwrites the previously installed Software License Key.
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Open the Software License Key file and check that the "S/N" line appears. If it does not appear, contact AudioCodes. Verify that you have loaded the correct file. Open the file and ensure that the first line displays "[LicenseKeys]".
Mediant 600 & Mediant 1000 39.2.2 Loading via BootP/TFTP The procedure below describes how to install a Software License Key using AudioCodes AcBootP utility. Notes: • When loading the Software License Key file, a cmp file must also be loaded during this BootP process.
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User's Manual 39. Software License Key Notes: • You can get the latest software files from AudioCodes Web site at http://www.audiocodes.com/downloads. • Before upgrading the device, it is recommended that you save a copy of the device's configuration settings (i.e., ini file) to your computer. If an upgrade failure occurs, you can then restore your configuration settings by uploading the backup file to the device.
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Mediant 600 & Mediant 1000 Click the Start Software Upgrade button; the wizard starts, requesting you to browses to a .cmp file for uploading. Note: At this stage, you can quit the Software Update Wizard, by clicking Cancel , without requiring a device reset. However, once you start uploading a cmp file, the process must be completed with a device reset.
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Enter your login user name and password, and then click OK; a message box appears informing you of the new .cmp file. Click OK; the Web interface becomes active, reflecting the upgraded device. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 39.4 Backing Up and Loading Configuration File You can save a copy/backup of the device's current configuration settings as an ini file to a folder on your computer, using the Configuration File page. The saved ini file includes only parameters that were modified and parameters with other than default values.
If there is no reply from a BootP server and if DHCP is enabled (DHCPEnable = 1), the device initiates a standard DHCP procedure to configure its network parameters. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 After changing the network parameters, the device attempts to load the device's firmware file (cmp) and various configuration files from the TFTP server’s IP address received from the BootP/DHCP server. If a TFTP server’s IP address isn’t received, the device attempts to load the cmp file and / or configuration files from a preconfigured TFTP server.
To configure the Automatic Update feature (ini file example): Setup a Web server (e.g., http://www.corp.com) and place all the required configuration files on this server. For each device, preconfigure the following parameter (DHCP / DNS are assumed): IniFileURL = 'http://www.corp.com/master_configuration.ini' Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Create a file named master_configuration.ini with the following text: # Common configuration for all devices # ------------------------------------ CptFileURL = 'http://www.corp.com/call_progress.dat' # Check for updates every 60 minutes AutoUpdateFrequency = 60 # Additional configuration per device...
Local configuration server with BootP/TFTP provides an easy and efficient method for automatic configuration, where configuration occurs at a staging warehouse, as described below: Install AudioCodes AcBootP/TFTP utility program on a computer located in a staging warehouse. Prepare a standard configuration ini file and place it in the TFTP directory.
Mediant 600 & Mediant 1000 When the devices' LEDs turn green indicating that the files were successfully loaded, disconnect the devices and ship to the customer. Notes: • Typically, IP addressing at the customer site is done by DHCP. •...
40.4.4 TFTP Configuration using DHCP Option 66 This method is suitable when the customer's network contains a provisioning TFTP server for all network equipment, without being able to distinguish between AudioCodes and non- AudioCodes devices. Upon startup, the device searches for Option 66 in the DHCP response from the DHCP server.
NFS v2/v3 is also supported. 40.4.7 Configuration using AudioCodes EMS AudioCodes EMS server functions as a core-network provisioning server. The device's SNMP Manager should be configured with the IP address of the EMS server, using one of the methods detailed in the previous sections. As soon as a registered device contacts the EMS server through SNMP, the EMS server handles all required configuration automatically, upgrading software as needed.
Open the Advanced Parameters page (Configuration tab > VoIP menu > SIP Definitions > Advanced Parameters). Under the Misc Parameters group, set the 'SIP Remote Reset' parameter to Enable. Click Submit. Note: This SIP Event header value is proprietary to AudioCodes. Version 6.6 Mediant 600 & Mediant 1000...
With a paper clip or any other similar pointed object, press and hold down the reset pinhole button, located on the CPU module for at least 12 seconds (but no more than 25 seconds). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 41.3 Restoring Defaults using an ini File You can restore the device to factory default settings by loading an empty ini file to the device. This is done using the Web interface's Configuration File page (see 'Backing Up and Loading Configuration File' on page 526).
To delete a loaded file: Click the Delete button corresponding to the file that you want to delete. Deleting a file takes effect only after device reset (see 'Resetting the Device' on page 491). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 42.2 Viewing Ethernet Port Information The Ethernet Port Information page displays read-only information on the Ethernet port connections. Note: The Ethernet Port Information page can also be accessed from the Home page (see 'Viewing the Home Page' on page 61).
Open the Alarms History page (Status & Diagnostics tab > System Status menu > Carrier-Grade Alarms > Alarms History). For each alarm, the following information is provided: Severity: severity level of the alarm: • Critical (red) • Major (range) • Minor (yellow) Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 • Cleared (green) Source: unit from which the alarm was raised Description: brief explanation of the alarm Date: date and time that the alarm was generated You can view the next 20 alarms (if exist), by clicking the Go to page button.
As you increase the resolution, more data is displayed on the graph. The minimum resolution is about 30 seconds; the maximum resolution is about an hour. To pause the graph, click the Pause button; click Play to resume. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 44.2 Viewing Trunk Utilization The Trunk Utilization page provides an X-Y graph that displays the number of active channels per trunk over time. The x-axis indicates the time; the y-axis indicates the number of active trunk channels.
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Instead of using the buttons, you can use the slide ruler. As you increase the resolution, more data is displayed on the graph. The minimum resolution is about 30 seconds; the maximum resolution is about an hour. Version 6.6 Mediant 600 & Mediant 1000...
Gray Disabled Green Active - OK Yellow RAI Alarm LOS / LOF Alarm Blue AIS Alarm Light Orange D-Channel Alarm Dark Orange NFAS Alarm Purple Lower Layer Down (DS3 physical layer is disabled) Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 The status of the channels is depicted by color-coded icons, as described in the table below: Description of Color-Coded Icons for Channel Status Icon Color Label Description Light blue Inactive Channel is configured, but currently has no calls...
For more information, see 'Performing Manual D-Channel Switchover in NFAS Group' on page 288. Note: This page is applicable only to T1 ISDN protocols supporting NFAS, and only if the NFAS group is configured with two D-channels. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To view the status of the D-channels and NFAS groups: Open the NFAS Group & D-Channel Status page (Status & Diagnostics tab > VoIP Status menu > NFAS Group & D-Channel Status). Figure 45-4: NFAS Group & D-Channel Status Page 45.4...
Open the Call Counters page that you want to view (Status & Diagnostics tab > VoIP Status menu > IP to Tel Calls Count or Tel to IP Calls Count); the figure below shows the IP to Tel Calls Count page. Figure 45-6: Calls Count Page Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 The fields in this page are described in the following table: Call Counters Description Counter Description Number of Attempted Indicates the number of attempted calls. It is composed of established Calls and failed calls. The number of established calls is represented by the 'Number of Established Calls' counter.
URI to another URI (Contact) where the user might be available. Contact SIP URI that can be used to contact that specific instance of the User Agent for subsequent requests. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 45.8 Viewing Registration Status The Registration Status page displays whether the device as a whole, its endpoints (FXS / FXO / BRI), and SIP Accounts are registered to a SIP Registrar/Proxy server. To view the registration status: ...
N/A = Proxy server isn't defined. State OK = Communication with the Proxy server is in order. Fail = No response from any of the defined Proxies. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 45.10 Viewing IP Connectivity The IP Connectivity page displays on-line, read-only network diagnostic connectivity information on all destination IP addresses configured in the Outbound IP Routing Table page (see 'Configuring Outbound IP Routing Table' on page 323).
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This parameter is reset if no QoS information is received for 2 minutes. DNS Status DNS status can be one of the following: DNS Disable DNS Resolved DNS Unresolved Version 6.6 Mediant 600 & Mediant 1000...
Remote Address (IP, Port & SSRC) Payload Type Session Description Payload Description Sample Rate Frame Duration Frame Octets Frames per Packets Packet Loss Concealment Silence Suppression State Jitter Buffer Jitter Buffer Adaptive Jitter Buffer Rate Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Group Metric Name Jitter Buffer Nominal Jitter Buffer Max Jitter Buffer Abs Max Packet Loss Network Packet Loss Rate Jitter Buffer Discard Rate Burst Gap Loss Burst Loss Density Burst Duration Gap Loss Density Gap Duration...
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'RTCP XR Collection Server Transport Type' (RTCPXRESCTransportType) - determines the transport layer for outgoing SIP dialogs initiated by the device to the RTCP XR Collection Server. Click Submit. Reset the device for the settings to take effect. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 46.2 Generating Call Detail Records The Call Detail Record (CDR) contains vital statistic information on calls made from the device. The device can be configured to generate and report CDRs for various stages of the call, including SIP messages and/or media. You can configure when CDRs for a call are generated, for example, only at the end of the call or only at the start and end of the call.
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Mediant 600 & Mediant 1000 Field Name Description Source phone number type Note: This field is applicable only to the Gateway application. Source phone number plan Note: This field is applicable only to the Gateway application. Source phone number SrcPhoneNum...
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N/A - Reasons not belonging to above categories RedirectNumBeforeM Redirect number before manipulation SrdId SRD ID name SIPInterfaceId SIP interface ID ProxySetId Proxy Set ID IpProfileId IP Profile ID name MediaRealmId Media Realm name Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Field Name Description SigTransportType SIP signaling transport type (UDP, TCP, or TLS) TxRTPIPDiffServ Media IP DiffServ TxSigIPDiffServ Signaling IP DiffServ LocalRFactor Local R-factor RemoteRFactor Remote R-factor LocalMosCQ Local MOS for conversation quality RemoteMosCQ Remote MOS for conversation quality...
15 Stop Acc or blank digits long nas-ip- IP address of the Numeric 192.168.14. Start Acc address requesting device Stop Acc service-type Type of service Numeric 1: login Start Acc requested Stop Acc Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Vendor Attribute Attribute Specific Value Purpose Example Number Name Attribute Format (VSA) No. h323- SIP call identifier Up to 32 Start Acc incoming- octets Stop Acc conf-id h323-remote- IP address of the Numeric Stop Acc...
Mediant 600 & Mediant 1000 3fd61009 0e2f3cc5 (4923 30) h323-disconnect-cause = 22 (0x16) (4923 27) h323-call-type = VOIP (4923 26) h323-call-origin = Originate (4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5 46.4 Event Notification using X-Detect Header The device supports the sending of notifications to a remote party notifying the occurrence (or detection) of certain events on the media stream.
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Type = [AMD | CPT | FAX | PTT…] • Subtype = xxx (according to the defined subtypes of each type) Below is an example of SIP messages using the X-Detect header: INVITE sip:101@10.33.2.53;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906 Version 6.6 Mediant 600 & Mediant 1000...
A skip in the number sequence of messages indicates a loss of message packets. For example, in the below Syslog message generation, messages 238 through 300 were not received. In other words, three Syslog messages were lost Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 47.1.2 Identifying AudioCodes Syslog Messages using Facility Levels The device’s Syslog messages can easily be identified and distinguished from Syslog messages from other equipment, by setting its Facility level. The Facility levels of the device's Syslog messages are numerically coded with decimal values. Facility level may use any of the "local use"...
For viewing Syslog messages in the Web interface, see 'Viewing Syslog Messages' on page 588. • For a detailed description on the Syslog parameters, see 'Syslog, CDR and Debug Parameters' on page 630. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 To configure Syslog : Open the Syslog Settings page (Configuration tab > System menu > Syslog Settings). Figure 47-1: Syslog Settings Page Enable the Syslog feature by setting the 'Enable Syslog' to Enable. Define the Syslog server using the 'Syslog Server IP Address' and 'Syslog Server Port' parameters.
Click the Add button; the Add Record dialog box appears: Figure 47-3: Logging Filters Table - Add Record Dialog Box Configure the logging filter, as required. See the table below for a description of the parameters. Click Submit to save your changes. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Notes: • To configure the Syslog debug level, use the 'Debug Level' parameter (see 'Configuring Syslog' on page 583). • The Logging Filters table can also be configured using the table ini file parameter, LoggingFilters.
Wireshark condition "(ip.src == 1.1.1.1 or ip.src == 2.2.2.2) and ip.dst == 3.3.3.3" can be configured using the following two table row entries: ip.src == 1.1.1.1 and ip.dst == 3.3.3.3 ip.src == 2.2.2.2 and ip.dst == 3.3.3.3 Version 6.6 Mediant 600 & Mediant 1000...
You can select the Syslog messages in this page, and copy and paste them into a text editor such as Notepad. This text file (txt) can then be sent to AudioCodes Technical Support for diagnosis and troubleshooting. To activate the Web interface's Message Log: Enable Syslog (see 'Configuring Syslog' on page 583).
47. Syslog and Debug Recordings 47.6 Collecting Debug Recording Messages To collect debug recording packets, the open source program Wireshark is used. AudioCodes proprietary plug-in files for Wireshark, which are shipped in your software kit, are also required. Notes: •...
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Mediant 600 & Mediant 1000 For ISDN trace messages, the additional header, "NetBricks Trace" is added below the "AUDIOCODES DEBUG RECORDING" header, as shown in the example below: AUDIOCODES DEBUG RECORDING NetBricks Trace System time: 3559 Direction: Message received from internal server queue (73) From (Entity origination ID): DL_D (DL LAPD Q.921) (100)
Startup Test (automatic): This hardware test has minor impact in real-time. While this test is executed, the regular operation of the device is disabled. If an error is detected, an error message is sent to the Syslog. Version 6.6 Mediant 600 & Mediant 1000...
Line polarity (0 = Positive; 1 = Negative) • Message Waiting Indication (0 = Off; 1 = On) Note: Use the Analog Line testing mechanism only for monitoring and never when there are calls in progress. Version 6.6 Mediant 600 & Mediant 1000...
By default, you can configure up to five test calls. This maximum can be increased by installing the relevant Software License Key. For more information, contact your AudioCodes sales representative. • The Test Call Endpoint table can also be configured using the table ini file parameter Test_Call (see 'SIP Test Call Parameters' on page 629).
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Mediant 600 & Mediant 1000 Configure the test endpoint parameters as desired. See the table below for a description of these parameters. Click Submit to apply your settings. Test Call Table Parameters Parameter Description General Tab Endpoint URI Defines the endpoint's URI. This can be defined as a user or user@host.
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Note: This parameter is applicable only if 'Call Party' is set to Caller. Calls per Second Defines the number of calls per second. [Test_Call_CallsPer Note: This parameter is applicable only if 'Call Party' is set to Caller. Second] Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Test Mode Defines the test session mode. [Test_Call_TestMo [0] Once = (Default) The test runs until the lowest value between the following is reached: Maximum channels is reached for the test session, configured by 'Maximum Channels for Session'.
Call Attempts: The number of calls that were attempted. Total Established Calls: The total number of calls that were successfully established. Total Failed Attempts: The total number of calls that failed to be established. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Remote Disconnections Count: Number of calls that were disconnected by the remote side. Average CPS: The average calls per second. Test Status: Displays the status (brief description) as displayed in the 'Test Status' field (see 'Starting, Stopping and Restarting Test Calls' on page 598).
• For a full description of this parameter, see 'SIP Test Call Parameters' on page 629. • This call test is done on all SIP interfaces. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 50.4 Test Call Configuration Examples Below are a few examples of test call configurations. Single Test Call Scenario: This example describes the configuration of a simple test call scenario that includes a single test call between a simulated test endpoint on the device and a remote endpoint.
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The test call is done between two AudioCodes devices - Device A and Device B - with simulated test endpoints. This eliminates the need for phone users, who would otherwise need to answer and end calls many times for batch testing.
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Mediant 600 & Mediant 1000 Registration Test Call Scenario: This example describes the configuration for testing the registration and authentication (i.e., username and password) process of a simulated test endpoint on the device with an external proxy/registrar server. This is useful, for example, for verifying that endpoints located in the LAN can register with an external proxy and subsequently, communicate with one another.
03(abc): for any number that starts with 03 and ends with abc. 03(5xx): for any number that starts with 03 and ends with 5xx. 03(400,401,405): for any number that starts with 03 and ends with 400 or 401 or 405. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Notation Description Notes: The value n must be less than the value m. Only numerical ranges are supported (not alphabetical letters). For suffix ranges, the starting (n) and ending (m) numbers in the range must have the same number of digits.
[0] = Disable [1] = (Default) Enable For more information on Ethernet interface redundancy, see Ethernet Interface Redundancy on page 109. Note: For this parameter to take effect, a device reset is required. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.1.2 Multiple VoIP Network Interfaces and VLAN Parameters The IP network interfaces and VLAN parameters are described in the table below. IP Network Interfaces and VLAN Parameters Parameter Description Multiple Interface Table Web: Multiple Interface This table parameter configures the Multiple Interface table.
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: IP Routing Defines up to 30 static IP routing rules for the device. These rules can Table be associated with IP interfaces defined in the Multiple Interface table [StaticRouteTable] (InterfaceTable parameter). The routing decision for sending the outgoing IP packet is based on the source subnet/VLAN.
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Web: Bronze QoS Defines the DiffServ value for the Bronze CoS content (OAMP EMS: Bronze Service applications). Class Diff Serv The valid range is 0 to 63. The default is 10. [BronzeServiceClassDiff Serv] Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.1.5 NAT and STUN Parameters The Network Address Translation (NAT) and Simple Traversal of UDP through NAT (STUN) parameters are described in the table below. NAT and STUN Parameters Parameter Description STUN Parameters Web: Enable STUN Enables Simple Traversal of UDP through NATs (STUN).
Defines the start of the range of numbers used for local UDP ports used [NFSBasePort] by the NFS client. The maximum number of local ports is maximum channels plus maximum NFS servers. The valid range is 0 to 65535. The default is 47000. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description NFS Table Web: NFS Table This table parameter defines up to 16 NFS file systems so that the EMS: NFS Settings device can access a remote server's shared files and directories for [NFSServers] loading cmp, ini, and auxiliary files (using the Automatic Update mechanism).
When set to 0, the DHCP lease renewal is disabled. Otherwise, the renewal time is divided by this factor. Some DHCP-enabled routers perform better when set to 4. Note: For this parameter to take effect, a device reset is required. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.1.9 NTP and Daylight Saving Time Parameters The Network Time Protocol (NTP) and daylight saving time parameters are described in the table below. NTP and Daylight Saving Time Parameters Parameter Description NTP Parameters Note: For more information on Network Time Protocol (NTP), see 'Simple Network Time Protocol Support' on page 105.
When this duration expires, the password of the Web user [WebUserPassChangeIn must be changed. terval] The valid value is 0 to 100000, where 0 means that the password is always valid. The default is 1140. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Note: This parameter is applicable only when using the Web Users table, where the default value of the 'Password Age' parameter in the Web Users table inherits this parameter's value. Web: User inactivity timer...
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The file name must have the .dat extension and can be up to 47 characters. For loading a Scenario using the Web interface, see Loading a Scenario to the Device on page 57. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [WelcomeMessage] Enables and defines a Welcome message that appears on the Web Login page for logging in to the Web interface. The format of this parameter is as follows: [WelcomeMessage ] FORMAT WelcomeMessage_Index = WelcomeMessage_Text...
Defines the user-assigned asset tracking identifier object for the [ChassisPhysicalAssetID] device's chassis as specified by an EMS, and provides non- volatile storage of this information. The valid range is a string of up to 255 characters. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [ifAlias] Defines the textual name of the interface. The value is equal to the ifAlias SNMP MIB object. The valid range is a string of up to 64 characters. EMS: Keep Alive Trap Port Defines the port to which keep-alive traps are sent.
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Security can be enhanced by using Trusted Managers, which is an IP address from which the SNMP agent accepts and processes SNMP requests. If no values are assigned to these parameters any manager Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description can access the device. Trusted managers can work with all community strings. SNMP V3 Users Table Web/EMS: SNMP V3 Users This parameter table defines SNMP v3 users. The format of this [SNMPUsers]...
Enable LAN watchdog is relevant only if the Ethernet connection is full duplex. [LifeLineType] Defines the condition(s) upon which the Lifeline analog (FXS) feature is activated. The Lifeline feature can be activated upon a power outage, Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description physical disconnection of the LAN cable, or network failure (i.e., loss of IP connectivity). Upon any of these conditions, the Lifeline feature provides PSTN connectivity and thus call continuity for the FXS phone users.
Test_Call_CalledURI, Test_Call_RouteBy, Test_Call_IPGroupID, Test_Call_DestAddress, Test_Call_DestTransportType, Test_Call_SRD, Test_Call_ApplicationType, Test_Call_AutoRegister, Test_Call_UserName, Test_Call_Password, Test_Call_CallParty, Test_Call_MaxChannels, Test_Call_CallDuration, Test_Call_CallsPerSecond, Test_Call_TestMode, Test_Call_TestDuration, Test_Call_Play, Test_Call_ScheduleInterval; [\Test_Call] Note: For a description of this table, see 'Configuring Test Calls' on page 595. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.3.3 Syslog, CDR and Debug Parameters The Syslog, CDR and debug parameters are described in the table below. Syslog, CDR and Debug Parameters Parameter Description Web: Enable Syslog Determines whether the device sends logs and error messages (e.g., EMS: Syslog enable CDRs) generated by the device to a Syslog server.
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[23] = local use 7 (local7) Web: Activity Types to Defines the Activity Log mechanism of the device, which sends log Report via Activity Log messages to a Syslog server for reporting certain types of Web Messages Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [ActivityListToLog] operations according to the below user-defined filters. [pvc] Parameters Value Change = Changes made on-the-fly to parameters. Note that the ini file parameter, EnableParametersMonitoring can also be used to set this option, using values [0] (disable) or [1] (enable).
The range is 0 to 100%. The default is 90%. [RAILoopTime] Defines the time interval (in seconds) that the device periodically checks call resource availability. The valid range is 1 to 200. The default is 10. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.3.5 BootP Parameters The BootP parameters are described in the table below. The BootP parameters are special 'hidden' parameters. Once defined and saved in the device's flash memory, they are used even if they don't appear in the ini file.
[0] Disable (default). [1] Enable = disables TFTP and allows secure protocols such as HTTPS to fetch the device configuration. Note: For this parameter to take effect, a device reset is required. Firewall Table Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Internal This table parameter defines the device's access list (firewall), which Firewall Parameters defines network traffic filtering rules. [AccessList] The format of this parameter is as follows: [AccessList] FORMAT AccessList_Index = AccessList_Source_IP,...
Note: For this parameter to take effect, a device reset is required. Web/EMS: Media Determines the device's mode of operation when SRTP is used (i.e., Security Behavior when the parameter EnableMediaSecurity is set to 1). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [MediaSecurityBehavio [0] Preferable = (Default) The device initiates encrypted calls. However, if negotiation of the cipher suite fails, an unencrypted call is established. Incoming calls that don't include encryption information are accepted.
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[1] Disable EncryptionDisable Tx [RTCPEncryptionDisab leTx] Enables synchronization of the SRTP state between the device and a [ResetSRTPStateUpon Rekey] server when a new SRTP key is generated upon a SIP session expire. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description This feature ensures that the roll-over counter (ROC), one of the parameters used in the SRTP encryption/decryption process of the SRTP packets, is synchronized on both sides for transmit and receive packets. ...
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(‘*’) to replace parts of the domain name. The valid range is a string of up to 49 characters. Note: This parameter is applicable only if the parameter PeerHostNameVerificationMode is set to 1 or 2. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Web: Client Cipher Defines the cipher-suite string for TLS clients. String The valid value is up to 255 strings. The default is "ALL:!ADH". [TLSClientCipherString For example: TLSClientCipherString = 'EXP' This parameter complements the HTTPSCipherString parameter (which affects TLS servers).
The format of this parameter is as follows: [ IPsecSATable ] FORMAT IPsecSATable_Index = IPsecSATable_RemoteEndpointAddressOrName, IPsecSATable_AuthenticationMethod, IPsecSATable_SharedKey, IPsecSATable_SourcePort, IPsecSATable_DestPort, IPsecSATable_Protocol, IPsecSATable_Phase1SaLifetimeInSec, IPsecSATable_Phase2SaLifetimeInSec, IPsecSATable_Phase2SaLifetimeInKB, IPsecSATable_DPDmode, IPsecSATable_IPsecMode, IPsecSATable_RemoteTunnelAddress, IPsecSATable_RemoteSubnetIPAddress, IPsecSATable_RemoteSubnetPrefixLength, IPsecSATable_InterfaceName; [ \IPsecSATable ] For example: Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description IPsecSATable 1 = 0, 10.3.2.73, 0, 123456789, 0, 0, 0, 0, 28800, 3600, ; In the above example, a single IPSec/IKE peer (10.3.2.73) is configured. Pre-shared key authentication is selected, with the pre-shared key set to 123456789.
[ IDSRule ] FORMAT IDSRule_Index = IDSRule_Policy, IDSRule_RuleID, IDSRule_Reason, IDSRule_ThresholdScope, IDSRule_ThresholdWindow, IDSRule_MinorAlarmThreshold, IDSRule_MajorAlarmThreshold, IDSRule_CriticalAlarmThreshold; [ \IDSRule ] For a detailed description of this table, see 'Configuring IDS Policies' on page 149. IDS Match Table Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Web: IDS Match Table Defines target rules per IDS Policy. [IDSMatch] The format of the ini file parameter is: [ IDSMatch ] FORMAT IDSMatch_Index = IDSMatch_SIPInterface, IDSMatch_ProxySet, IDSMatch_Subnet, IDSMatch_Policy; [ \IDSMatch ] For a detailed description of this table, see 'Assigning IDS Policies' on page 152.
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Defines the time (in seconds) the locally stored user name and Password Cache Timeout password (verified by the RADIUS server) are valid. When this time [RadiusLocalCacheTime expires, the user name and password become invalid and a must be re- Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description out] verified with the RADIUS server. The valid range is 1 to 0xFFFFFF. The default is 300 (5 minutes). [-1] = Never expires. [0] = Each request requires RADIUS authentication. Web: RADIUS VSA...
Page 649
For a detailed description of this table, see 'Configuring Bandwidth Management per Media Realm' on page 188. Quality of Experience Parameters Web: Server IP Defines the IP address of AudioCodes Session Experience Manager CLI: server-ip (SEM) server to where the quality experience reports are sent. [QOEServerIP] Note: For this parameter to take effect, a device reset is required.
Mediant 600 & Mediant 1000 Parameter Description Web: Use Mos LQ Enables the reporting of the MOS-LQ (listening quality). If disabled, the [QOEUseMosLQ] MOS-CQ (conversational quality) is reported. MOS-LQ measures the quality of audio for listening purposes only. MOS-LQ does not take into account bi-directional effects such as delay and echo.
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Proxy Registration Parameters Web: Use Default Proxy Enables the use of a SIP proxy server. EMS: Proxy Used [0] No = (Default) Proxy isn't used and instead, the internal routing [IsProxyUsed] table is used. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [1] Yes = Proxy server is used. Define the IP address of the proxy server in the Proxy Sets table (see 'Configuring Proxy Sets Table' on page 219). Note: If you are not using a proxy server, you must define outbound IP call routing rules in the Outbound IP Routing Table (described in Configuring Outbound IP Routing Table on page 323).
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SIP Contact and Record-Route headers. [0] A-Record (default) [1] SRV [2] NAPTR If set to A-Record [0], no NAPTR or SRV queries are performed. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description If set to SRV [1] and the Proxy/Registrar IP address parameter, Contact/Record-Route headers, or IP address defined in the Routing tables contain a domain name, an SRV query is performed. The device uses the first host name received from the SRV query. The device then performs a DNS A-record query for the host name to locate an IP address.
Page 655
Proxy IP Table Web: Proxy IP Table This table parameter configures the Proxy Set table with Proxy Set IDs, EMS: Proxy IP each with up to five Proxy server IP addresses (or fully qualified domain Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [ProxyIP] name/FQDN). Each Proxy Set can be defined with a transport type (UDP, TCP, or TLS). The format of this parameter is as follows: [ProxyIP] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; [\ProxyIP] For example: ProxyIp 0 = 10.33.37.77, -1, 0;...
Page 657
Retry Time is re-sent if registration fails with a 4xx response or if there is no response from the Proxy/Registrar server. [RegistrationRetryTime] The default is 30 seconds. The range is 10 to 3600. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Registration Time Defines a threshold (in seconds) for re-registration timing. If this Threshold parameter is greater than 0, but lower than the computed re-registration EMS: Time Threshold timing (according to the parameter RegistrationTimeDivider), the re-...
Page 659
- set to the SIP URI of the domain name of the home network nonce - set to an empty value response - set to an empty value For example: Version 6.6 Mediant 600 & Mediant 1000...
Page 660
Mediant 600 & Mediant 1000 Parameter Description Authorization: Digest username=alice_private@home1.net, realm=”home1.net”, nonce=””, response=”e56131d19580cd833064787ecc” Note: This registration header is according to the IMS 3GPP TS24.229 and PKT-SP-24.220 specifications. Web: Add initial Route Enables the inclusion of the SIP Route header in initial registration or Header re-registration (REGISTER) requests sent by the device.
TCP Keep Alive Interval Defines the interval (in sec) between consecutive keep-alive probes, Time regardless of what the connection has exchanged in the meantime. [TCPKeepAliveInterval] The valid value is 10 to 65,000. The default is 10. Version 6.6 Mediant 600 & Mediant 1000...
(if Automatic Update has been enabled on the device) 'check-sync;reboot=true': triggers a device reset Note: The Event header value is proprietary to AudioCodes. Web/EMS: Max SIP Defines the maximum size (in Kbytes) for each SIP message that can Message Length [KB] be sent over the network.
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PR=<voice packets received> OR=<voice octets received> PL=<receive packet loss> JI=<jitter in ms> LA=<latency in ms> Below is an example of the X-RTP-Stat header in a SIP BYE message: Version 6.6 Mediant 600 & Mediant 1000...
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Progress message is received for IP-to-Tel calls. Analog: Defines the EMS: SIP 183 Behaviour response of the device upon receipt of a SIP 183 response. [SIP183Behaviour] [0] Progress = (Default) Digital: The device sends a Progress Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description message. Analog: A 183 response (without SDP) does not cause the device to play a ringback tone. [1] Alert = Digital: The device sends an Alerting message (upon receipt of a 183 response) instead of an ISDN Progress message.
Page 667
3), a 'gpmd' attribute is added to the SDP in the following format: For A-law: 'a=gpmd:8 vbd=yes;ecan=on' For µ-law: 'a=gpmd:0 vbd=yes;ecan=on' When this parameter is set to 1, 2, or 3, the parameter Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description FaxTransportMode is ignored. When this parameter is set to 0, T.38 might still be used without the control protocol's involvement. To completely disable T.38, set FaxTransportMode to a value other than 1.
Page 669
Enables setting of all TCP/TLS connections as persistent and therefore, Connection Persistent not released. Mode [0] = (Default) Disable. All TCP connections (except those that are [ReliableConnectionPer set to a proxy IP) are released if not used by any SIP Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description sistentMode] dialog\transaction. [1] = Enable - TCP connections to all destinations are persistent and not released unless the device reaches 70% of its maximum TCP resources. While trying to send a SIP message connection, reuse policy determines whether live connections to the specific destination are re- used.
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[2] Send and Receive = The functionality of outgoing SIP messages is identical to the functionality described for option [1]. In addition, for incoming SIP INVITEs, if the Request-URI includes a 'tgrp' parameter, the device routes the call according to that value (if Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description possible). The Contact header in the outgoing SIP INVITE (Tel-to-IP call) contains "tgrp=<source trunk group ID>;trunk-context=<gateway IP address>”. The <source trunk group ID> is the Trunk Group ID where incoming calls from Tel is received. For IP-Tel calls, the SIP 200 OK device's response contains “tgrp=<destination trunk group...
Page 673
For example, in call transfer where user A is talking to user B, and user A wants to transfer the call to user C. User A sends a REFER to user C: REFER sip:C@domain.com SIP/2.0 From: sip:A@domain.com;tag=99asd To: sip:C@domain.com Refer-To: (URI that identifies B's UA) Version 6.6 Mediant 600 & Mediant 1000...
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EMS: User Agent Display value>/software version' is used, for example: Info User-Agent: myproduct/v.6.40.010.006 [UserAgentDisplayInfo] If not configured, the default string, <AudioCodes product- name>/software version' is used, for example: User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.6.40.010.006 The maximum string length is 50 characters. Note: The software version number and preceding forward slash (/) cannot be modified.
Page 675
Party B is put on-hold while Party A blind transfers Party B to Party Party C answers and Party B is made un-hold. However, as Party C supports only G.729 coder, re-negotiation of the supported coder is required. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Multiple Determines whether the 'mptime' attribute is included in the outgoing Packetization Time SDP. Format [0] None = (Default) Disabled. EMS: Multi Ptime Format [1] PacketCable = Includes the 'mptime' attribute in the outgoing [MultiPtimeFormat] SDP - PacketCable-defined format.
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PlayRBTone2TEL parameter and disregards the subsequent forking 18x responses. [1] Sequential handling = If 18x with SDP is received, the device opens a voice stream according to the received SDP. The device re- Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description opens the stream according to subsequently received 18x responses with SDP, or plays a ringback tone if 180 response without SDP is received. If the first received response is 180 without SDP, the device responds according to the PlayRBTone2TEL parameter and processes the subsequent 18x forking responses.
Page 679
Determines whether the device uses the value of the incoming SIP Reason header for Release Reason mapping. [0] = Disregard Reason header in incoming SIP messages. [1] = (Default) Use the Reason header value for Release Reason Version 6.6 Mediant 600 & Mediant 1000...
Page 680
Mediant 600 & Mediant 1000 Parameter Description mapping. [EnableSilenceSuppInS Determines the device's behavior upon receipt of SIP Re-INVITE messages that include the SDP's 'silencesupp:off' attribute. [0] = (Default) Disregard the 'silecesupp' attribute. [1] = Handle incoming Re-INVITE messages that include the 'silencesupp:off' attribute in the SDP as a request to switch to the Voice-Band-Data (VBD) mode.
Page 681
[IP2IPTranscodingMode Note: This parameter is no longer valid and must not be used. Defines the voice transcoding mode (media negotiation) between two user agents for the IP-to-IP application. This parameter must always be Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description set to 1 when using the IP-to-IP application. [0] Only if Required = Do not force transcoding. Many of the media settings (such as gain control) are not implemented on the voice stream.
Page 683
If the device is the receiver and the remote SIP UA does not send a “CN” in the SDP, then no CNG occurs. If the remote side sends a “CN”, the device attempts to be compatible with the remote side and Version 6.6 Mediant 600 & Mediant 1000...
Page 684
Mediant 600 & Mediant 1000 Parameter Description even if the codec’s SCE is disabled, CNG occurs. If the ComfortNoiseNegotiation parameter is disabled, then the device does not send “CN” in the SDP. However, if the codec’s SCE is enabled, then CNG occurs.
Page 685
To activate the RTP Only feature without using ISDN / CAS signaling, you must do the following: Configure E1/T1 Transparent protocol type (set the ProtocoType parameter to 5 or 6). Enable the TDM-over-IP feature (set the EnableTDMoverIP Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description parameter to 1). To configure the RTP Only mode per trunk, use the RTPOnlyModeForTrunk_x parameter. If per trunk configuration (using the RTPOnlyModeForTrunk_x parameter) is set to a value other than the default, the RTPOnlyMode parameter value is ignored.
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Digital: To configure the method for setting digital trunks to Out-Of- Service, use the DigitalOOSBehavior parameter. Web/EMS: Graceful Busy Defines the timeout interval (in seconds) for Out-of-Service graceful Out Timeout [sec] shutdown mode for busy trunks (per trunk) if communication fails with a Version 6.6 Mediant 600 & Mediant 1000...
Page 688
Mediant 600 & Mediant 1000 Parameter Description [GracefulBusyOutTimeo Proxy server (or Proxy Set). In such a scenario, the device rejects new calls from the PSTN (Serving Trunk Group), but maintains currently active calls for this user-defined timeout. Once this timeout elapses, the device terminates currently active calls and takes the trunk out of service (sending the PSTN busy-out signal).
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SIP Message Manipulations Table Web: Message This table parameter defines manipulation rules for SIP header Manipulations messages. EMS: Message The format of this parameter is as follows: Manipulations [ MessageManipulations] [MessageManipulations] FORMAT MessageManipulations_Index = MessageManipulations_ManSetID, MessageManipulations_MessageType, Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description MessageManipulations_Condition, MessageManipulations_ActionSubject, MessageManipulations_ActionType, MessageManipulations_ActionValue, MessageManipulations_RowRole; [\MessageManipulations] For example, the below configuration changes the user part of the SIP From header to 200: MessageManipulations 1 = 0, Invite.Request, , Header.From.Url.User, 2, 200, 0; Note: For a detailed description of this table, see 'Configuring SIP Message Manipulation' on page 231.
Page 693
MWI Analog Lamp MWIAnalogLamp TelProfile_MWIDisplay MWI Display MWIDisplay TelProfile_FlashHookP Flash Hook Period FlashHookPeriod eriod TelProfile_EnableEarly Enable Early Media EnableEarlyMedia Media TelProfile_ProgressIndi Progress Indicator to IP ProgressIndicator2IP cator2IP TelProfile_TimeForReor Time For Reorder Tone TimeForReorderTone derTone Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description TelProfile_EnableDIDW Enable DID Wink EnableDIDWink TelProfile_IsTwoStage Dialing Mode IsTwoStageDial Dial TelProfile_DisconnectO Disconnect Call on DisconnectOnBusyTon nBusyTone Detection of Busy Tone TelProfile_EnableVoice Enable Voice Mail MailDelay Delay TelProfile_DialPlanInde Dial Plan Index DialPlanIndex TelProfile_Enable911P Enable 911 PSAP...
Page 695
Enables echo cancellation (i.e., echo from voice calls is removed). EMS: Echo Canceller Enable [0] Disable [EnableEchoCanceller] [1] Enable (default) Note: This parameter can also be configured in an IP Profile and/or a Tel Profile. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Max Echo Canceller Defines the maximum Echo Canceler Length (in msec), which is Length the maximum echo path delay (tail length) for which the echo [MaxEchoCancellerLength] canceller is designed to operate: ...
The default is DSP Template 0. Number You can load different DSP templates to analog and digital [DSPVersionTemplateNumbe modules using the syntax DSPVersionTemplateNumber=xy where: x = 0 or 1 for DSP templates of analog modules Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description y = 0 to 5 for DSP templates of digital and MPM modules Notes: For this parameter to take effect, a device reset is required. For a list of supported DSP templates, see DSP Templates on page 839.
[0] 0 = (Default) Disable. [1] 1 = Enable - previous voice payload packet is added to current packet. Notes: When enabled, you can configure the payload type, using the Version 6.6 Mediant 600 & Mediant 1000...
Page 700
Mediant 600 & Mediant 1000 Parameter Description RFC2198PayloadType parameter. The RTP redundancy dynamic payload type can be included in the SDP, by using the EnableRTPRedundancyNegotiation parameter. This parameter can also be configured in an IP Profile. Web: Enable RTP Redundancy...
Page 701
EMS: Remote Base UDP Port value, ThroughPacket™ (RTP multiplexing) is enabled. The device [RemoteBaseUDPPort] uses this parameter (and BaseUDPPort) to identify and distribute the payloads from the received multiplexed IP packet to the Version 6.6 Mediant 600 & Mediant 1000...
Page 702
Mediant 600 & Mediant 1000 Parameter Description relevant channels. The valid range is the range of possible UDP ports: 6,000 to 64,000. The default is 0 (i.e., RTP multiplexing is disabled). Notes: The value of this parameter on the local device must equal the value of BaseUDPPort on the remote device.
Page 703
Defines the IP address of the Event State Compositor (ESC). The Server device sends RTCP XR reports to this server, using SIP PUBLISH EMS: Esc IP messages. The address can be configured as a numerical IP Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description [RTCPXREscIP] address or as a domain name. Web: RTCP XR Report Mode Determines whether RTCP XR reports are sent to the Event State EMS: Report Mode Compositor (ESC) and defines the interval at which they are sent.
Page 705
T.38 fax session when the CNG tone is detected by the originating side. However, this mode is not recommended. Note: This parameter can also be configured in an IP Profile. Version 6.6 Mediant 600 & Mediant 1000...
Page 706
Mediant 600 & Mediant 1000 Parameter Description Web: Fax Relay Enhanced Defines the number of times that control packets are retransmitted Redundancy Depth when using the T.38 standard. EMS: Enhanced Relay The valid range is 0 to 4. The default is 2.
Page 707
0 for G.711 µ-Law). The parameters defining payload type for the 'old' Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass. The bypass packet interval is selected according to the FaxModemBypassBasicRtpPacketInterval parameter. Notes: Version 6.6 Mediant 600 & Mediant 1000...
Page 708
Mediant 600 & Mediant 1000 Parameter Description This feature can be used only if the VxxModemTransportType parameter is set to 2 (Bypass). If NSE mode is enabled, the SDP contains the following line: 'a=rtpmap:100 X-NSE/8000'. To use this feature: ...
[0] Not Supported = (Default) Hook-Flash indication is not sent. [1] INFO = Sends proprietary INFO message (Broadsoft) with Hook-Flash indication. The device sends the INFO message as follows: Content-Type: application/broadsoft; version=1.0 Content-Length: 17 event flashhook Version 6.6 Mediant 600 & Mediant 1000...
Page 710
Mediant 600 & Mediant 1000 Parameter Description [4] RFC 2833 = This option is currently not supported. [5] INFO (Lucent) = Sends proprietary SIP INFO message with Hook-Flash indication. The device sends the INFO message as follows: Content-Type: application/hook-flash...
Page 711
DTMFTransportType is automatically set to 0 (DTMF digits are erased from the RTP stream). When RFC 2833 (4) is selected, the device: Negotiates RFC 2833 payload type using local and remote Version 6.6 Mediant 600 & Mediant 1000...
Page 712
Mediant 600 & Mediant 1000 Parameter Description SDPs. Sends DTMF packets using RFC 2833 payload type according to the payload type in the received SDP. Expects to receive RFC 2833 packets with the same payload type as configured by the parameter RFC2833PayloadType.
Page 713
DTMF signaling (using SIP INFO/NOTIFY). EMS: Use Digit For [0] Special = (Default) Uses the strings ‘*’ and ‘#’. Special DTMF [1] Numeric = Uses the numerical values 10 and 11. [UseDigitForSpecialDTM Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.11.3 Digit Collection and Dial Plan Parameters The digit collection and dial plan parameters are described in the table below. Digit Collection and Dial Plan Parameters Parameter Description Web/EMS: Dial Plan Index Defines the Dial Plan index to use in the external Dial Plan file. The [DialPlanIndex] Dial Plan file is loaded to the device as a .dat file (converted using...
Page 715
[0] Disable = Use '*' or '#' to terminate number collection (refer [IsSpecialDigits] to the parameter UseDigitForSpecialDTMF). (Default.) [1] Enable = Allows '*' and '#' for telephone numbers dialed by a Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description user or for the endpoint telephone number. Note: These symbols can always be used as the first digit of a dialed number even if you disable this parameter. 52.11.4 Voice Mail Parameters The voice mail parameters are described in the table below. For more information on the Voice Mail application, refer to the CPE Configuration Guide for Voice Mail.
Page 717
Analog (FXO): After receiving a SIP REFER message from the IP side, the device sends a hook-flash to the PBX, and then dials the digits (that are received in the Refer-To header). The Version 6.6 Mediant 600 & Mediant 1000...
Page 718
Mediant 600 & Mediant 1000 Parameter Description device waits for connection of the transferred call and then completes the call transfer by releasing the line. If speech is not detected, the transfer is cancelled, the device sends a SIP NOTIFY message with a failure reason in the NOTIFY body...
Page 719
Defines the digit pattern used by the PBX to indicate 'call forward on Answer Digit Pattern no answer' when the original call is received from an internal (Internal) extension. EMS: Digit Pattern The valid range is a 120-character string. Forward On No Answer [DigitPatternForwardOn NoAnswer] Version 6.6 Mediant 600 & Mediant 1000...
Page 720
Mediant 600 & Mediant 1000 Parameter Description Web: Forward on Do Not Defines the digit pattern used by the PBX to indicate 'call forward on Disturb Digit Pattern do not disturb' when the original call is received from an internal (Internal) extension.
The information configured in this Caller ID table is sent in the SIP INVITE message's From header. [CallerDisplayInfo] The format of this parameter is as follows: [CallerDisplayInfo] FORMAT CallerDisplayInfo_Index = CallerDisplayInfo_DisplayString, Version 6.6 Mediant 600 & Mediant 1000...
Page 722
Mediant 600 & Mediant 1000 Parameter Description CallerDisplayInfo_IsCidRestricted, CallerDisplayInfo_Module, CallerDisplayInfo_Port; [\CallerDisplayInfo] Where, Module = Module number, where 1 denotes the module in Slot 1. Port = Port number, where 1 denotes Port 1 of a module. For example: CallerDisplayInfo 0 = Susan C.,0,1,1;...
Page 723
Determines when Caller ID is generated. Mode [0] = (Default) Caller ID is generated between the first two rings. [AnalogCallerIDTimingM [1] = The device attempts to find an optimized timing to generate Version 6.6 Mediant 600 & Mediant 1000...
Page 724
Mediant 600 & Mediant 1000 Parameter Description ode] the Caller ID according to the selected Caller ID type. Notes: This parameter is applicable only to FXS interfaces. If this parameter is set to 1 and used with distinctive ringing, the Caller ID signal doesn't change the distinctive ringing timing.
Call Waiting Parameters The call waiting parameters are described in the table below. Call Waiting Parameters Parameter Description Web/EMS: Enable Call Enables the Call Waiting feature. Waiting [0] Disable [EnableCallWaiting] [1] Enable (Default) Version 6.6 Mediant 600 & Mediant 1000...
Page 726
Mediant 600 & Mediant 1000 Parameter Description For digital interfaces: If enabled and the device initiates a Tel-to-IP call to a destination that is busy, it plays a call waiting ringback tone to the caller. The tone is played only if the destination returns a 182 "Queued"...
Page 727
The SIP INFO message is according to Broadsoft's application server definition. Below is an example of such an INFO message: INFO sip:06@192.168.13.2:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.13.40:5060;branch=z9hG4bK040066422630 From: <sip:4505656002@192.168.13.40:5060>;tag=1455352915 To: <sip:06@192.168.13.2:5060> Call-ID:0010-0008@192.168.13.2 CSeq:342168303 INFO Content-Length:28 Content-Type:application/broadsoft play tone CallWaitingTone1 Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.11.5.3 Call Forwarding Parameters The call forwarding parameters are described in the table below. Call Forwarding Parameters Parameter Description Web: Enable Call Enables the Call Forwarding feature. Forward [0] Disable [EnableForward] [1] Enable (Default) For FXS interfaces, the Call Forward table (FwdInfo parameter) must be defined to use the Call Forward service.
[0] No (default) Subscription [1] Yes [EnableMWISubscripti Notes: To configure the MWI server address, use the MWIServerIP parameter. To configure whether the device subscribes per endpoint or per the Version 6.6 Mediant 600 & Mediant 1000...
Page 730
Mediant 600 & Mediant 1000 Parameter Description entire device, use the parameter SubscriptionMode. Web: MWI Server IP Defines the MWI server's IP address. If provided, the device subscribes to Address this IP address. The MWI server address can be configured as a EMS: MWI Server IP numerical IP address or as a domain name.
This parameter is applicable only to FXS interfaces. This Reminder Ring feature can be disabled using the DisableReminderRing parameter. Disables the reminder ring, which notifies the FXS user of a call on hold Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description [DisableReminderRing] or a waiting call when the phone is returned to on-hook position. [0] = (Default) The reminder ring feature is active. In other words, if a call is on hold or there is a call waiting and the phone is changed from offhook to onhook, the phone rings (for a duration defined by the CHRRTimeout parameter) to "remind"...
Page 733
The valid value range is 0 to 1,000,000. The default is 0. Web: QSIG Path Enables QSIG transfer for IP-to-Tel and Tel-to-IP calls. Replacement Mode [0] IP2QSIGTransfer = (Default) Enables IP-to-QSIG transfer. [QSIGPathReplacement [1] QSIG2IPTransfer = Enables QSIG-to-IP transfer. Mode] Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Defines the maximum duration (timeout) to wait between call Setup and [ReplaceTel2IPCallingN Facility with Redirecting Number for replacing the calling number (for Tel- umTimeout] to-IP calls). The valid value range is 0 to 10,000 msec. The default is 0.
Page 735
Conference Mode is used. EMS: 3 Way Mode [0] AudioCodes Media Server = (Default) The Conference-initiating [3WayConferenceMode] INVITE (sent by the device) uses the ConferenceID concatenated with a unique identifier as the Request-URI. This same Request-URI is set as the Refer-To header value in the REFER messages that are sent to the two remote parties.
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Conference ID Defines the Conference Identification string. [ConferenceID] The valid value is a string of up to 16 characters. The default is "conf". For 3-way conferencing using an external media server: The device uses this identifier in the conference-initiating INVITE that is sent to the media server when the Enable3WayConference parameter is set to 1.
Page 737
E911 Tandem Switches and the PSAP, using analog loop-start lines. The FXO device can be installed instead of an E911 switch, connected directly to PSAP DID loop-start lines. [0] = Disable (default) [1] = Enable Version 6.6 Mediant 600 & Mediant 1000...
Page 738
Mediant 600 & Mediant 1000 Parameter Description Notes: This parameter is applicable only to FXO interfaces. This parameter can also be configured in a Tel Profile. Web/EMS: Emergency Defines a list of “emergency” numbers. Numbers For FXS: When one of these numbers is dialed, the outgoing INVITE [EmergencyNumbers] message includes the SIP Priority and Resource-Priority headers.
Page 739
‘FFFFFF’) as a Service domain in the SIP Resource-Priority Domain header in outgoing INVITE messages. If the parameter is configured to [MLPPNormalizedServic ‘FFFFFF’, the Resource-Priority header is set to the MLPP Service Version 6.6 Mediant 600 & Mediant 1000...
Page 740
Mediant 600 & Mediant 1000 Parameter Description eDomain] Domain obtained from the Precedence IE. The valid value is 6 hexadecimal digits. The default is ‘000000’. Note: This parameter is applicable only to the MLPP NI-2 ISDN variant with CallPriorityMode set to 1.
Page 741
The valid range is -1 to 63. The default is -1. [MLPPFlashOverOverRT Note: If set to -1, the DiffServ value is taken from the global parameter PDSCP] PremiumServiceClassMediaDiffServ or as defined in IP Profiles per call. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.11.5.9 Call Cut-Through Parameters The call cut-through parameters are described in the table below. Call Cut-Through Parameters Parameter Description Web: Enable Calls Cut Enables FXS endpoints to receive incoming IP calls while the port is in Through off-hook state.
The FXS interface generates a Wink signal upon detection of an off-hook state, instead of playing a dial tone. Example: (Wink) KP I(I) xxx-xxxx ST (Off Hook) Where: I = one or two information digits x = ANI Version 6.6 Mediant 600 & Mediant 1000...
Page 744
Mediant 600 & Mediant 1000 Parameter Description Note: The FXO interface generates such MF digits when the Enable911PSAP parameter is set to 1. [2] Double Wink = Double-wink signaling. The FXS interface generates the first Wink upon detection of an off-hook state in the line.
SIP INVITE to the softswitch with a user-defined code in the SIP To header, representing the reason for the call forward. For more information on BRI call forwarding, see 'BRI Call Forwarding' on page 356. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Call Forward Defines the prefix code for activating Call Forward Unconditional sent to Unconditional the softswitch. [SuppServCodeCFU] The valid value is a string. The default is an empty string. Note: The string must be enclosed in single apostrophe (e.g., ‘*72’).
[17] E1 AUSTEL ISDN = ISDN PRI protocol for the Australian Telecom. [18] E1 HKT ISDN = ISDN PRI (E1) protocol for the Hong Kong - HKT. [19] E1 KOR ISDN = ISDN PRI protocol for Korean Operator Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description (similar to ETSI). [20] T1 HKT ISDN = ISDN PRI (T1) protocol for the Hong Kong - HKT. [21] E1 QSIG = ECMA 143 QSIG over E1 [22] E1 TNZ = ISDN PRI protocol for Telecom New Zealand (similar to ETSI) ...
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Mediant 600 & Mediant 1000 Parameter Description CRC6 [E] T1 FRAMING F72 = T1 72-Frame multiframe (SLC96) [F] T1 FRAMING ESF CRC6 J2 = J1 Extended SuperFrame with CRC6 (Japan) Note: This parameter is not configurable for BRI interfaces; the device automatically uses the BRI framing method.
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SIP calls from all enabled B-channels pertaining to E1/T1/J1 spans that are configured with the 'Transparent' protocol. The called number of each call is the internal phone number of the B-channel from where the call Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description originates. The 'The Inbound IP Routing Table is used to define the destination IP address of the terminating device. The terminating device automatically answers these calls if its E1/T1 protocol is set to 'Transparent' (ProtocolType = 5).
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This parameter is applicable only when the TDMBusPSTNAutoClockEnable parameter is set to 1. Web: Auto Clock Trunk Priority Defines the trunk priority for auto-clock fallback (per trunk EMS: Auto Trunk Priority parameter). Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description [AutoClockTrunkPriority] 0 to 99 = priority, where 0 (default) is the highest. 100 = the SW never performs a fallback to that trunk (usually used to mark untrusted source of clock).
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To configure this parameter, the trunk must first be stopped. Only one of these formats can be implemented; not both. When this parameter is not configured, a single CAS table for Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description the entire trunk is used, configured by the parameter CASTableIndex. [CASTablesNum] Defines how many CAS protocol configurations files are loaded. The valid range is 1 to 8. Note: For this parameter to take effect, a device reset is required.
EMS: Group Number group number (NFAS member) per trunk. [NFASGroupNumber_x] [0] = (Default) Non-NFAS trunk. [1] to [12] = NFAS group number. Trunks that belong to the same NFAS group have the same Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description number. With NFAS, you can use a single D-channel to control multiple PRI interfaces. Notes: For this parameter to take effect, a device reset is required. This parameter is applicable only to T1 ISDN protocols.
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[0] = Disable (default). [1] NO STATUS ON UNKNOWN IE = Q.931 Status message isn't sent if Q.931 received message contains an unknown/unrecognized IE. By default, the Status message is sent. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Note: This value is applicable only to ISDN variants in which sending of Status message is optional. [2] NO STATUS ON INV OP IE = Q.931 Status message isn't sent if an optional IE with invalid content is received. By default, the Status message is sent.
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[128] USE E1 PRI = PRI interface type is forced to E1. [256] START WITH B CHAN OOS = B-channels start in the Out-Of-Service state (OOS). [512] CHAN ALLOC LOWEST = CC allocates B-channels Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description starting from the lowest available B-channel id. [1024] CHAN ALLOC HIGHEST = CC allocates B-channels starting from the highest available B-channel id. [16384] CC_TRANSPARENT_UUI bit: The UUI-protocol implementation of CC is disabled allowing the application to freely send UUI elements in any primitive, regardless of the UUI-protocol requirements (UUI Implicit Service 1).
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If this bit is not set (default): 0x9F = Networking Extensions. If this bit is set: 0x91 = ROSE. [3] = Use options [0] and [2] above. Note: For this parameter to take effect, a device reset is required. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.11.7 ISDN and CAS Interworking Parameters The ISDN and CAS interworking parameters are described in the table below. ISDN and CAS Interworking Parameters Parameter Description ISDN Parameters Web: Send Local Time Determines the device's handling of the date and time sent in the ISDN...
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Q.931 Information messages. [ConnectedNumberTyp Defines the Numbering Type of the ISDN Q.931 Connected Number IE that the device sends in the Connect message to the ISDN (for Tel-to-IP Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description calls). This is interworked from the P-Asserted-Identity header in SIP 200 OK. The default is [0] (i.e., unknown). [ConnectedNumberPla Defines the Numbering Plan of the ISDN Q.931 Connected Number IE that the device sends in the Connect message to the ISDN (for Tel-to-IP calls).
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[1] = BCD (Binary Coded Decimal) - allows up to 40 characters (digits and letters). Indicates that the 'isub' parameter value needs to be encoded using BCD when translated to an ISDN message. [2] = User Specified Version 6.6 Mediant 600 & Mediant 1000...
Page 768
Mediant 600 & Mediant 1000 Parameter Description For IP-to-Tel calls, if the incoming SIP INVITE message includes subaddress values in the 'isub' parameter for the Called Number (in the Request-URI) and/or the Calling Number (in the From header), these values are mapped to the outgoing ISDN Setup message.
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Calling Party Category SIP and PRI. [EnableCallingPartyCat [0] Disable = (Default) Don't relay the CPC between SIP and PRI. egory] [1] Enable = The CPC is relayed between SIP and PRI. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description If enabled, the CPC received in the Originating Line Information (OLI) IE of an incoming ISDN Setup message is relayed to the From/P-Asserted- Identity headers using the 'cpc' parameter in the outgoing INVITE message, and vice versa.
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Defines the Alert Timeout (ISDN T301 timer) in seconds for outgoing Timeout calls to PSTN, per trunk. This timer is used between the time that an [TrunkPSTNAlertTimeo ISDN Setup message is sent to the Tel side (IP-to-Tel call Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description ut_x] establishment) and a Connect message is received. If Alerting is received, the timer is restarted. The range is 1 to 600. The default is 180. Note: The x in the ini file parameter name denotes the trunk number, where 0 is Trunk 1.
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IP Profiles (using the IPProfile parameter) and then assigning the required IP Profile ID in the Inbound IP Routing Table (PSTNPrefix). This feature is similar to that of the EnableISDNTunnelingIP2Tel parameter. If both parameters are configured, the Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description EnableISDNTunnelingIP2Tel parameter takes precedence. Web: Trunk Groups to Defines Trunk Group IDs (up to 50 characters) from where the optional Send IE ISDN IE (defined by the parameter AddIEinSetup) is sent. For example: EMS: List Of Trunk '1,2,4,10,12,6'.
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TBCT/ECT transfer is performed after receipt of Alerting or Connect messages. For RLT, the transfer is always done after receipt of Connect (SendISDNTransferOnConnect is set to 1). This transfer can be performed between B-channels from Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description different trunks or Trunk Groups, by using the parameter EnableTransferAcrossTrunkGroups. The device initiates the ECT process after receiving a SIP REFER message only for trunks that are configured to User side. ...
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[0] Disable = Rejects ISDN transfer requests. [1] Enable = (Default) The device sends a SIP REFER message to the remote call party if ECT/TBCT Facility messages are received from the ISDN side (e.g., from a PBX). Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [DisableFallbackTransf Enables "hairpin" TDM transfer upon ISDN (ECT, RLT, or TBCT) call erToTDM] transfer failure. When this feature is enabled and an ISDN call transfer failure occurs, the device sends a SIP NOTIFY message with a SIP 603 Decline response.
Notes: This parameter is applicable to IP-to-Tel and Tel-to-IP calls. This parameter is applicable only to ISDN and CAS protocols. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Disconnect Determines whether the device disconnects a call when a dial tone is on Dial Tone detected from the PBX. [DisconnectOnDialTon [0] Disable = (Default) Call is not released. [1] Enable = Call is released if a dial tone is detected on the device's FXO port.
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[1] Enable = (Default) Call is released upon detection of busy or Detection End Tones reorder (fast busy) tone. [DisconnectOnBusyTo Notes: Digital interfaces: This parameter is applicable only to CAS protocols. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Analog interfaces: This parameter is applicable only to FXO interfaces. This parameter is also applicable to the IP-to-IP application. This parameter can also be configured in a Tel Profile. Polarity (Current) Reversal for Call Release (Analog Interfaces) Parameters [SetDefaultLinePolarity Defines the FXO line polarity, required for DID signaling.
When enabled, the device uses the incoming audio from this connector instead of playing the held tone defined in the Call Progress Tones (CPT) file. [0] = (Default) Disable [1] = Enable Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Note: EnableHold must be set to 1 to enable this feature. Enables playing a held tone to an IP-to-IP leg instead of placing it [PlayHeldToneForIP2IP] on hold. [0] = (Default) Disabled. The device interworks the re-INVITE with 'a=inactive' from one SIP leg to another SIP leg.
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The valid values are 0 to 30. The default is 0 (i.e., no reorder tone is played). Note: To enable the Cut-Through feature, use the DigitalCutThrough (for CAS channels) or CutThrough (for FXS channels) parameters. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Enable Comfort Determines whether the device plays a comfort tone (Tone Type Tone #18) to the FXS/FXO endpoint after a SIP INVITE is sent and [EnableComfortTone] before a SIP 18x response is received.
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RTP packets. At this stage, if the device receives additional 18x responses, it does not resume playing the local ringback tone. Note that for ISDN trunks, this option is applicable only if the LocalISDNRBSource parameter Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description is set to 1. Note: This parameter is applicable to the Gateway and IP-to-IP applications. Web: Play Ringback Tone to Determines the playing method of the ringback tone to the trunk Trunk side per trunk.
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ISDN's Two B Channel Transfer (TBCT), Release Line Trunk EMS: Play RBT On ISDN (RLT), or Explicit Call Transfer (ECT) call transfers to the Transfer originator when the second leg receives an ISDN Alerting or [PlayRBTOnISDNTransfer] Progress message. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description [0] Don't Play (default) [1] Play Notes: For Blind transfer, the local ringback tone is played to first call PSTN party when the second leg receives the ISDN Alerting or Progress message.
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Defines the deviation (in Hz) allowed for the detection of each CPT Deviation signal frequency. [CPTDetectorFrequencyDevi The valid range is 1 to 30. The default is 10. ation] Note: For this parameter to take effect, a device reset is required. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 52.11.9.3 Metering Tone Parameters The metering tone parameters are described in the table below. Metering Tone Parameters Parameter Description Web: Generate Determines the method used to configure the metering tones that are Metering Tones generated to the Tel side.
For example, if this parameter is enabled and the prefix string for the external line is defined as "9" (using the parameter Prefix2ExtLine) and the FXS user wants to make a call to destination "123", the device collects Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description and sends all the dialed digits, including the prefix string, as "9123" to the IP destination number. Note: This parameter is applicable only to FXS interfaces. Hook Flash Parameters Web: Flash Keys Determines the hook-flash key sequence for FXS interfaces.
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Dial the telephone number to which the phone automatically dials after a configurable delay (terminate the number with #); a confirmation tone is heard. Web: Hot-line Defines the keypad sequence to deactivate the delayed hotline option. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Deactivate After the sequence is pressed, a confirmation tone is heard. EMS: Hot Line Deactivation [KeyHotLineDeact] Keypad Feature - Transfer Parameters Note: See the description of the KeyBlindTransfer parameter for this feature. Keypad Feature - Call Waiting Parameters Web: Call Waiting Defines the keypad sequence to activate the Call Waiting option.
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Call Progress Tones) is performed directly with the PBX without the device's intervention. Note: This parameter can also be configured in a Tel Profile. Web/EMS: Waiting For Dial Tone Determines whether or not the device waits for a dial tone Version 6.6 Mediant 600 & Mediant 1000...
Page 798
Mediant 600 & Mediant 1000 Parameter Description [IsWaitForDialTone] before dialing the phone number for IP-to-Tel (FXO) calls. [0] No [1] Yes (default) When one-stage dialing and this parameter are enabled, the device dials the phone number (to the PSTN/PBX line) only after it detects a dial tone.
TrunkGroup 0 = 2, 0, 1, 30, 50000, 0, 0, 1; The configuration below assigns BRI channels 1 through 4 of Module 2 to Trunk Group ID 2 with phone numbers 208 to 211: Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description TrunkGroup 1 = 2, 0, 1, 4, 208, 0, 0 ,2; Notes: Trunk Group ID 1 is denoted as 0 in the table. This parameter can appear up to four times per module.
Page 801
Profiles server is used. EMS: Use Routing Table [0] Disable = (Default) Don't use internal routing table. For Host Names [1] Enable = Use the Outbound IP Routing Table. [AlwaysUseRouteTable] Notes: Version 6.6 Mediant 600 & Mediant 1000...
Page 802
Mediant 600 & Mediant 1000 Parameter Description This parameter appears only if the 'Use Default Proxy' parameter is enabled. The domain name is used instead of a Proxy name or IP address in the INVITE SIP URI. Web/EMS: Tel to IP...
Page 803
For more information on this feature, see Dial Plan Prefix Tags for IP-to-Tel Routing on page 510. Determines the method in which the Redirect Number is sent to the [EnableETSIDiversion] Tel side. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [0] = (Default) Q.931 Redirecting Number Information Element (IE). [1] = ETSI DivertingLegInformation2 in a Facility IE. Web: Add CIC Determines whether to add the Carrier Identification Code (CIC) as a [AddCicAsPrefix] prefix to the destination phone number for IP-to-Tel calls.
DNS host name is not resolved. Notes: QoS is quantified according to delay and packet loss calculated according to previous calls. QoS statistics are reset if no new data is received within two minutes. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description To receive quality information (displayed in the 'Quality Status' and 'Quality Info.' fields in 'Viewing IP Connectivity' on page 558) per destination, this parameter must be set to 2 or 3. Web: Alt Routing Tel to IP...
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Note: For a detailed description of this table, see 'Alternative Routing to Trunk upon Q.931 Call Release Cause Code' on page 340. Forward On Busy Trunk Destination Table Web/EMS: Forward On This table parameter configures the Forward On Busy Trunk Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description Busy Trunk Destination Destination table. This table allows you to define an alternative IP [ForwardOnBusyTrunkD destination if a trunk is busy for IP-to-Tel calls. est] The format of this parameter is as follows:...
Page 809
IP, the redirect reason is set to the [SetIp2TelRedirectReason value of this parameter before the device sends it on to the Tel. [-1] Not Configured (default) [0] Unkown [1] Busy Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [2] No Reply [3] Network Busy [4] Deflection [9] DTE out of Order [10] Forwarding DTE [13] Transfer [14] PickUp [15] Systematic/Unconditional Note: This parameter is applicable only to digital PSTN interfaces (ISDN).
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(i.e., called number) for Tel-to-IP calls. [AddTrunkGroupAsPrefix] [0] No = (Default) Don't add Trunk Group ID as prefix. [1] Yes = Add Trunk Group ID as prefix to called number. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Notes: This option can be used to define various routing rules. To use this feature, you must configure the Trunk Group IDs (see Configuring Trunk Group Table on page 291). Web: Add Trunk ID as...
Page 813
The valid range is an 8-character string. The default is an empty string. Web: Add Number Plan and Determines whether the TON/PLAN parameters are included in the Type to RPI Header Remote-Party-ID (RPID) header. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description EMS: Add Ton 2 RPI [0] No [AddTON2RPI] [1] Yes (default) If the Remote-Party-ID header is enabled (EnableRPIHeader = 1) and AddTON2RPI = 1, it's possible to configure the calling and called number type and number plan using the Number Manipulation tables for Tel-to-IP calls.
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This parameter table manipulates the source number for IP-to-Tel calls. Number Manipulation Table The format of this parameter is as follows: for IP > Tel Calls [SourceNumberMapIp2Tel] EMS: SIP Manipulations > FORMAT SourceNumberMapIp2Tel_Index = Source IP to Telcom SourceNumberMapIp2Tel_DestinationPrefix, Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [SourceNumberMapIP2Tel SourceNumberMapIp2Tel_SourcePrefix, SourceNumberMapIp2Tel_SourceAddress, SourceNumberMapIp2Tel_NumberType, SourceNumberMapIp2Tel_NumberPlan, SourceNumberMapIp2Tel_RemoveFromLeft, SourceNumberMapIp2Tel_RemoveFromRight, SourceNumberMapIp2Tel_LeaveFromRight, SourceNumberMapIp2Tel_Prefix2Add, SourceNumberMapIp2Tel_Suffix2Add, SourceNumberMapIp2Tel_IsPresentationRestricted; [\SourceNumberMapIp2Tel] For example: SourceNumberMapIp2Tel 0 = 22,03,$$,$$,$$,$$,2,667,$$,$$; SourceNumberMapIp2Tel 1 = 034,01,1.1.1.1,$$,0,2,$$,$$,972,$$,10; Note: For a detailed description of this table, see 'Configuring Source/Destination Number Manipulation' on page 299.
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NPI and TON to the SIP 'phone-context' parameter, Phone Context and vice versa. [PhoneContext] The format for this parameter is as follows: [PhoneContext] FORMAT PhoneContext_Index = PhoneContext_Npi, PhoneContext_Ton, PhoneContext_Context; [\PhoneContext] For example: PhoneContext 0 = 0,0,unknown.com PhoneContext 1 = 1,1,host.com Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description PhoneContext 2 = 9,1,na.e164.host.com Note: For a detailed description of this table, see 'Mapping NPI/TON to SIP Phone-Context' on page 313. Web/EMS: Add Phone Determines whether the received Phone-Context parameter is added as...
For example: LDAPBindDN = "CN=Search user,OU=Labs,DC=OCSR2,DC=local" Note: The DN is used to uniquely name an Active Directory object. Web: LDAP Search Dn Defines up to three search DNs for LDAP search queries. These are Version 6.6 Mediant 600 & Mediant 1000...
Page 820
Mediant 600 & Mediant 1000 Parameter Description EMS: Search DN the DN subtrees where the search is done. This parameter is [LDAPSearchDN] mandatory for the search. The format of this parameter is as follows: [LdapSearchDNs ] FORMAT LdapSearchDNs_Index = LdapSearchDNs_Base_Path;...
Enables the Stand-Alone Survivability (SAS) feature. EMS: Enable [0] Disable (default) [EnableSAS] [1] Enable When enabled, the device receives the registration requests from different SIP entities in the local network and then forwards them to the Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description defined proxy. If the connection to the proxy fails ('Emergency Mode'), the device serves as a proxy by allowing calls internal to the local network or outgoing to PSTN. Note: For this parameter to take effect, a device reset is required.
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Proxy list of SASProxySet in Normal [SASSurvivabilityMode] mode and handled by the SAS application in Emergency mode. [1] Always Emergency = The SAS application does not use Keep- Version 6.6 Mediant 600 & Mediant 1000...
Page 824
Mediant 600 & Mediant 1000 Parameter Description Alive messages towards the SASProxySet, instead it always operates in Emergency mode (as if no Proxy in the SASProxySet is available). [2] Ignore Register = Use regular SAS Normal/Emergency logic (same as option [0]), but when in Normal mode incoming REGISTER requests are ignored.
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This table parameter configures the IP-to-IP Routing table for SAS routing rules. The format of this parameter is as follows: [IP2IPRouting] FORMAT IP2IPRouting_Index = IP2IPRouting_SrcIPGroupID, IP2IPRouting_SrcUsernamePrefix, IP2IPRouting_SrcHost, IP2IPRouting_DestUsernamePrefix, IP2IPRouting_DestHost, IP2IPRouting_DestType, IP2IPRouting_DestIPGroupID, IP2IPRouting_DestSRDID, IP2IPRouting_DestAddress, IP2IPRouting_DestPort, IP2IPRouting_DestTransportType, IP2IPRouting_AltRouteOptions; [\IP2IPRouting] For example: Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description IP2IPRouting 1 = -1, *, *, *, *, 0, -1, -1, , 0, -1, 0; Note: For a detailed description of this table parameter, see 'SAS Routing Based on IP-to-IP Routing Table' on page 437.
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SIP INVITE message with a SIP URI that includes this string. The default is ‘ivr’. For example: INVITE sip:ivr@10.2.3.4 Subsequent INFO messages carry the requests and responses. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description Web: Transcoding ID Defines the Transcoding identification string (up to 16 characters) [TranscodingID] used for identifying an incoming Transcoding call. The default is ‘trans’. For more information on Transcoding, see NetAnn Interface on page 467.
Page 831
Note: This parameter can also be configured in an IP Profile. [AMDMaxGreetingTime] EMS: AMD Max Post Defines the maximum duration of silence from after the greeting time Silence Greeting Time is over (defined by AMDMaxGreetingTime) until the AMD decision. [AMDMaxPostGreetingS Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description ilenceTime] Note: This parameter can also be configured in an IP Profile. EMS: Time Out Defines the timeout (in msec) between receiving Connect messages [AMDTimeout] from the ISDN and sending AMD results. The valid range is 1 to 30,000. The default is 2,000 (i.e., 2 seconds).
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The range is 0 - 65,535. The default is 10. Note: For this parameter to take effect, a device reset is required. [RTSPEnabled] Activates the RTSP functionality. Version 6.6 Mediant 600 & Mediant 1000...
Mediant 600 & Mediant 1000 Parameter Description [0] = Disable (default) [1] = Activate Note: For this parameter to take effect, a device reset is required. [RTSPMaxPorts] Defines the number of channels that can be simultaneously active in RTSP sessions.
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0 is Trunk 1. Web: Dial Plan File Defines the name (and path) of the Dial Plan file. This file should be EMS: Dial Plan File Name created using AudioCodes DConvert utility (refer to DConvert Utility [DialPlanFileName] User's Guide). [UserInfoFileName] Defines the name (and path) of the file containing the User Information data.
Mediant 600 & Mediant 1000 52.17 Automatic Update Parameters The automatic update of software and configuration files parameters are described in the table below. Automatic Update of Software and Configuration Files Parameters Parameter Description General Automatic Update Parameters [AutoUpdateCmpFile] Enables the Automatic Update mechanism for the cmp file.
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Note: For this parameter to take effect, a device reset is required. [TLSCertFileUrl] Defines the name of the TLS certificate file and the URL from where it can be downloaded. Note: For this parameter to take effect, a device reset is required. Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Parameter Description [TLSPkeyFileUrl] Defines the URL for downloading a TLS private key file using the Automatic Update facility. [UserInfoFileURL] Defines the name of the User Information file and the path to the server (IP address or FQDN) on which it is located.
• The number of channels refers to the maximum channel capacity of the device. • For additional DSP templates, contact your AudioCodes representative. 53.1 Analog Interfaces The DSP templates for analog interfaces are shown in the table below. DSP Firmware Templates for Analog (FXS/FXO) Interfaces...
Mediant 600 & Mediant 1000 Notes: IPM Features refers to the configuration that includes at least one of the following: • Mounted MPM module in Slot #6 for conference applications. • IPM detectors (e.g., Answer Detector) are enabled. • The IP Media Channels featured is enabled.
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6 or 16 Assembly Slot no. Supplementary Capabilities Number of Channels Voice Coder G.711 A-law/Mm- law PCM G.727 G.726 ADPCM G.723.1 G.729 A, B GSM FR MS GSM iLBC EVRC QCELP GSM EFR G.722 Transparent Version 6.6 Mediant 600 & Mediant 1000...
• The compliance and regulatory information can be downloaded from AudioCodes Web site at http://www.audiocodes.com/library. 54.1 Mediant 600 The table below lists the main technical specifications of the Mediant 600. Mediant 600 Functional Specifications Function Specification Interfaces E1/T1/J1 1, 2 or fractional (15 DS0) span spans using RJ-48c connectors...
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Mediant 600 & Mediant 1000 Function Specification Signaling E1/T1 CAS E&M, Loop Start, Feature Group-D, E911CAMA, R2 MFC, numerous protocol and country variants ISDN PRI ETSI/EURO, ANSI NI2, DMS-100, 5ESS, VN3, VN4, VN6 QSIG (Basic Call and Supplementary Services) and other variants Control &...
Supports up to 600 registered users in its Registration database. SAS ensures call continuity between LAN SIP clients upon connectivity failure with IP Centrex services (e.g., WAN IP PBX). Max. Capacity Registered Users (IP-to-IP, SAS) Version 6.6 Mediant 600 & Mediant 1000...
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Mediant 600 & Mediant 1000 Function Specification OSN Server Platform - Embedded, Partner application platform for third-party services OSN1: Intel™ Celeron™ 600 Mhz OSN2: Intel Pentium M 1.4 GHz OSN1: One SODIMM slot 512M or 1G RAM Memory ...
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