Page 1 Mediant™ 2000 VoIP Media Gateway SIP Protocol User's Manual Version 6.4 November 2011 Document # LTRT-68814...
Page 3: Table Of Contents
SIP User's Manual Contents Table of Contents Overview ......................15 SIP Overview ......................16 Part I: Getting Started....................17 Assigning the VoIP LAN IP Address ..............19 Using CLI ....................... 19 Using the Web Interface ..................20 Using BootP/TFTP Server ..................21 Part II: Management Tools ..................23 Web-Based Management ..................
Page 4 Mediant 2000 CLI-Based Management ..................63 Configuring Telnet and SSH Settings ..............64 SNMP-Based Management ................65 Configuring SNMP Community Strings ..............65 Configuring SNMP Trap Destinations ..............66 Configuring SNMP Trusted Managers ..............68 Configuring SNMP V3 Users .................. 69 EMS-Based Management ..................
Page 5 SIP User's Manual Contents 10.6 DNS ........................113 10.6.1 Configuring the Internal DNS Table ...............113 10.6.2 Configuring the Internal SRV Table ...............115 10.7 NAT (Network Address Translation) Support ............116 10.7.1 STUN ........................116 10.7.2 First Incoming Packet Mechanism .................117 10.7.3 No-Op Packets ......................117 10.8 Configuring NFS Settings ..................
Page 6 Mediant 2000 12.7 Configuring Media Realms ................... 159 12.8 Configuring Media Security .................. 161 13 Services ......................163 13.1 Routing Based on LDAP Active Directory Queries ..........163 13.1.1 LDAP Overview .....................163 13.1.2 Configuring LDAP Settings ..................164 13.1.3 AD-Based Tel-to-IP Routing in Microsoft OCS 2007 Environment .......164 13.2 Least Cost Routing ....................
Page 7 SIP User's Manual Contents 18.2.2 Configuring Trunk Group Settings .................238 18.3 Manipulation ......................241 18.3.1 Configuring General Settings ................241 18.3.2 Configuring Number Manipulation Tables .............241 18.3.3 Configuring Redirect Number IP to Tel ..............246 18.3.4 Configuring Redirect Number Tel to IP ..............248 18.3.5 Mapping NPI/TON to SIP Phone-Context .............250 18.3.6 Numbering Plans and Type of Number ..............251 18.3.7 Configuring Release Cause Mapping ..............253...
Page 8 Mediant 2000 18.9.2.6 Step 6: Configure the Account Table ............ 307 18.9.2.7 Step 7: Configure IP Profiles for Voice Coders ........309 18.9.2.8 Step 8: Configure Inbound IP Routing ..........310 18.9.2.9 Step 9: Configure Outbound IP Routing ..........312 18.9.2.10 Step 10: Configure Destination Phone Number Manipulation ....
Page 9 SIP User's Manual Contents 22.3 Software Upgrade Wizard ..................366 22.4 Backing Up and Loading Configuration File ............369 23 Restoring Factory Defaults ................371 23.1 Restoring Defaults using CLI ................371 23.2 Restoring Defaults using an ini File ..............372 Part VI: Status, Performance Monitoring and Reporting ........373 24 System Status ....................
Page 10 Mediant 2000 A.1.2 Multiple Network Interfaces and VLAN Parameters ..........411 A.1.3 Static Routing Parameters ..................414 A.1.4 Quality of Service Parameters ................414 A.1.5 NAT and STUN Parameters ..................416 A.1.6 NFS Parameters ....................418 A.1.7 DNS Parameters ....................419 A.1.8 DHCP Parameters ....................420 A.1.9 NTP and Daylight Saving Time Parameters ............421...
Page 11 SIP User's Manual Contents A.11.6.2 TDM Bus and Clock Timing Parameters ..........532 A.11.6.3 CAS Parameters ................... 534 A.11.6.4 ISDN Parameters .................. 537 A.11.7 ISDN and CAS Interworking Parameters ..............544 A.11.8 Answer and Disconnect Supervision Parameters ..........561 A.11.9 Tone Parameters ....................564 A.11.9.1 Telephony Tone Parameters ..............
Page 12 Mediant 2000 C.2.33 Unsupported ......................630 C.2.34 Via ..........................630 C.2.35 Warning .........................631 C.2.36 Unknown Header ....................632 C.3 Structure Definitions ..................... 633 C.3.1 Event Structure ......................633 C.3.2 Host ........................633 C.3.3 MLPP ........................633 C.3.4 Privacy Struct ......................634 C.3.5 Reason Structure ....................634 C.3.6 SIPCapabilities ......................634 C.3.7...
Page 13: Weee Eu Directive
Notices Notice This document describes the AudioCodes Mediant 2000 Voice-over-IP (VoIP) media gateway, 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.
Page 14: Related Documentation
Related Documentation Manual Name SIP CPE Release Notes Product Reference Manual for SIP CPE Devices Mediant 2000 Hardware Installation Manual CPE Configuration Guide for IP Voice Mail Note: The scope of this document does not fully cover security aspects for deploying the device in your environment.
Page 15: Overview
Overview This manual provides you with the information for installing, configuring, and operating the Mediant 2000 SIP gateway (referred to throughout this manual as device). The device is a SIP-based Voice-over-IP (VoIP) media gateway. the device enables voice, fax, and data traffic to be sent over the same IP network.
Page 16: Sip Overview
Mediant 2000 The figure below illustrates a typical device applications VoIP network: SIP Overview Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol used on the gateway for creating, modifying, and terminating sessions with one or more participants. These sessions can include Internet telephone calls, media announcements, and conferences.
Page 17: Part I: Getting Started
Part I Getting Started Before you can begin configuring your device, you need to access it with the default LAN IP address and change this IP address to suit your networking scheme. Once modified, you can then access the device using the new LAN IP address. This section describes how to perform this initialization process.
Page 18 Reader’s Notes...
Page 19: Assigning The Voip Lan Ip Address
SIP User's Manual 2. Assigning the VoIP LAN IP Address Assigning the VoIP LAN IP Address This section describes how to change the default VoIP LAN IP address so that it corresponds to your networking scheme. The default VoIP LAN IP address is listed in the table below: Table 2-1: Default VoIP LAN IP Address IP Address Value...
Page 20: Using The Web Interface
Mediant 2000 At the prompt, typing the following command to change the network settings, and then press Enter: SCP IP <ip_address> <subnet_mask> <default_gateway> You must enter all three network parameters, each separated by a space, for example: SCP IP 10.13.77.7 255.255.0.0 10.13.0.1...
Page 21: Using Bootp/Tftp Server
SIP User's Manual 2. Assigning the VoIP LAN IP Address In the 'User Name' and 'Password' fields, enter the default login user name "Admin" (case-sensitive) and password "Admin" (case-sensitive), and then click OK; the device's Web interface is accessed. Open the Multiple Interface Table page (Configuration tab > VoIP menu > Network submenu >...
Page 22 Mediant 2000 Click the Add New Client icon. Figure 2-3: BootP Client Configuration Screen In the ‘Client MAC’ field, enter the device's MAC address. The MAC address is printed on the label located on the underside of the device. Ensure that the check box to the right of the field is selected in order to enable the client.
Page 23: Part Ii: Management Tools
Part II Management Tools This part provides an overview of the various management tools that can be used to configure the device and describes how to configure the management settings. The following management tools can be used to configure the device: ...
Page 24 Reader’s Notes...
Page 25: Web-Based Management
SIP User's Manual 3. Web-Based Management Web-Based Management The device's embedded Web server (hereafter referred to as the Web interface) provides FCAPS (fault management, configuration, accounting, performance, and security) functionality. The Web interface allows you to remotely configure the device for quick-and- easy deployment, including the loading of software (.cmp), configuration (.ini), and auxiliary files.
Page 26: Accessing The Web Interface
Mediant 2000 3.1.2 Accessing the Web Interface The procedure below describes how to access the Web interface. When initially accessing the Web interface, use Note: For assigning an IP address to the device, refer to the Installation Manual.  To access the Web interface: Open a standard Web browser (see ''Computer Requirements'' on page 25).
Page 27: Areas Of The Gui
SIP User's Manual 3. Web-Based Management Note: If access to the Web interface is denied ("Unauthorized") due to Microsoft Internet Explorer security settings, do the following: Delete all cookies in the Temporary Internet Files folder. If this does not resolve the problem, the security settings may need to be altered (continue with Step 2).
Page 28: Toolbar Description
Mediant 2000 3.1.4 Toolbar Description The toolbar provides frequently required command buttons, as described in the table below: Table 3-1: Description of Toolbar Buttons Icon Button Description Name Submit Applies parameter settings to the device (see ''Saving Configuration'' on page 350).
Page 29: Navigation Tree
SIP User's Manual 3. Web-Based Management 3.1.5 Navigation Tree The Navigation tree is located in the Navigation pane. It displays the menus pertaining to the selected menu tab on the Navigation bar and is used for accessing the configuration pages. The Navigation tree displays a tree-like structure of menus. You can drill-down to the required page item level to open its corresponding page in the Work pane.
Page 30: Displaying Navigation Tree In Basic And Full View
Mediant 2000 3.1.5.1 Displaying Navigation Tree in Basic and Full View You can view an expanded or reduced Navigation tree display regarding the number of listed menus and submenus. This is relevant when using the configuration tabs (Configuration, Maintenance, and Status & Diagnostics) on the Navigation bar.
Page 31 SIP User's Manual 3. Web-Based Management  To hide the Navigation pane: click the left-pointing arrow ; the pane is hidden and the button is replaced by the right-pointing arrow button.  To show the Navigation pane: click the right-pointing arrow ;...
Page 32: Working With Configuration
Mediant 2000 3.1.6 Working with Configuration Pages The configuration pages contain the parameters for configuring the device and are displayed in the Work pane, located to the right of the Navigation pane. 3.1.6.1 Accessing Pages The configuration pages are accessed by clicking the required page item in the Navigation tree.
Page 33 SIP User's Manual 3. Web-Based Management  Advanced Parameter List button with down-pointing arrow: click this button to display all parameters.  Basic Parameter List button with up-pointing arrow: click this button to show only common (basic) parameters. The figure below shows an example of a page displaying basic parameters only, and then showing advanced parameters as well, using the Advanced Parameter List button.
Page 34: Modifying And Saving Parameters
Mediant 2000 3.1.6.2.2 Showing / Hiding Parameter Groups Some pages provide groups of parameters, which can be hidden or shown. To toggle between hiding and showing a group, simply click the group title button that appears above each group. The button appears with a down-pointing or up-pointing arrow, indicating that it can be collapsed or expanded when clicked, respectively.
Page 35: Entering Phone Numbers
SIP User's Manual 3. Web-Based Management  To save configuration changes on a page to the device's volatile memory (RAM), do one of the following:  On the toolbar, click the Submit button.  At the bottom of the page, click the Submit button.
Page 36: Working With Tables
Mediant 2000 3.1.6.5 Working with Tables This section describes how to work with configuration tables, which are provided in basic or enhanced design (depending on the configuration page). 3.1.6.5.1 Basic Design Tables The basic design tables provide the following command buttons: ...
Page 37 SIP User's Manual 3. Web-Based Management Click Edit; the fields in the corresponding index row become available. Modify the values as required, and then click Apply; the new settings are applied.  To organize the index entries in ascending, consecutive order: ...
Page 38 Mediant 2000 3.1.6.5.2 Enhanced Design Tables The enhanced table structure includes the following buttons:  Add: adds a row entry to the table  Edit: edits the selected table row  Delete: deletes a selected table row  View/Unview: shows or hides all configuration settings of selected table rows ...
Page 39 SIP User's Manual 3. Web-Based Management  To view the configuration settings of an entry: Select the table row that you want to view, and then click the View/Unview button; a Details pane appears below the table, displaying the configuration settings of the selected row, as shown below: Figure 3-15: Displayed Details Pane To hide the Details pane, click the View/Unview button again.
Page 40: Searching For Configuration Parameters
Mediant 2000 3.1.7 Searching for Configuration Parameters The Web interface provides a search engine that allows you to search any ini file parameter that is configurable in the Web interface (i.e., has a corresponding Web parameter). You can search for a specific parameter (e.g., "EnableIPSec") or a substring of that parameter (e.g., "sec").
Page 41: Working With Scenarios
SIP User's Manual 3. Web-Based Management 3.1.8 Working with Scenarios The Web interface allows you to create your own "menu" with up to 20 pages selected from the menus in the Navigation tree (i.e., pertaining to the Configuration, Maintenance, and Status &...
Page 42 Mediant 2000 Click the Next button located at the bottom of the page; the Step is added to the Scenario and appears in the Scenario Step list: Figure 3-20: Creating a Scenario Repeat steps 5 through 8 to add additional Steps (i.e., pages).
Page 43: Accessing A Scenario
SIP User's Manual 3. Web-Based Management 3.1.8.2 Accessing a Scenario Once you have created the Scenario, you can access it at anytime by following the procedure below:  To access the Scenario: On the Navigation bar, select the Scenario tab; a message box appears, requesting you to confirm the loading of the Scenario.
Page 44: Editing A Scenario
Mediant 2000  In an opened Scenario Step (i.e., page appears in the Work pane), use the following navigation buttons: • Next: opens the next Step listed in the Scenario. • Previous: opens the previous Step listed in the Scenario.
Page 45: Saving A Scenario To A Pc
SIP User's Manual 3. Web-Based Management • Edit the Scenario Name: In the 'Scenario Name' field, edit the Scenario name. In the displayed page, click Next. • Remove a Step: In the Navigation tree, select the required Step; the corresponding page opens in the Work pane.
Page 46: Loading A Scenario To The Device
Mediant 2000 Click Save, and then in the 'Save As' window navigate to the folder to where you want to save the Scenario file. When the file is successfully downloaded to your PC, the 'Download Complete' window appears. Click Close to close the 'Download Complete' window.
Page 47: Quitting Scenario Mode
SIP User's Manual 3. Web-Based Management Click the Delete Scenario File button; a message box appears requesting confirmation for deletion. Figure 3-25: Message Box for Confirming Scenario Deletion Click OK; the Scenario is deleted and the Scenario mode closes. Note: You can also delete a Scenario using the following alternative methods: •...
Page 48: Creating A Login Welcome Message
Mediant 2000 3.1.9 Creating a Login Welcome Message You can create a Welcome message box (alert message) that appears after each successful login to the Web interface. The WelcomeMessage ini file parameter table allows you to create the Welcome message. Up to 20 lines of character strings can be defined for the message.
Page 49: Getting Help
SIP User's Manual 3. Web-Based Management 3.1.10 Getting Help The Web interface provides you with context-sensitive Online Help. The Online Help provides brief descriptions of parameters pertaining to the currently opened page.  To view the Help topic of a currently opened page: On the toolbar, click the Help button;...
Page 50: Logging Off The Web Interface
Mediant 2000 3.1.11 Logging Off the Web Interface You can log off the Web interface and re-access it with a different user account. For more information on Web User Accounts, see ''Configuring Web User Accounts'' on page 55.  To log off the Web interface: On the toolbar, click the Log Off button;...
Page 51: Using The Home Page
SIP User's Manual 3. Web-Based Management Using the Home Page By default, the Home page is displayed when you access the device's Web interface. The Home page provides you with a graphical display of the device's front panel, displaying color-coded status icons for monitoring the functioning of the device. The Home page also displays general device information (in the 'General Information' pane) such as the device's IP address and firmware version.
Page 52 Mediant 2000 The table below describes the areas of the Home page: Table 3-3: Description of the Areas of the Home Page Item # Description Displays the highest severity of an active alarm raised (if any) by the device: ...
Page 53: Assigning A Port Name
SIP User's Manual 3. Web-Based Management 3.2.1 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. ...
Page 54: Switching Between Modules
Mediant 2000 3.2.3 Switching Between Modules The device can house up to two modules, as discussed in previous sections. Since each module is a standalone gateway, the Home page displays only one of the modules to which you are connected. However, you can easily switch to the second module, by having the Web browser connect to the IP address of the other module.
Page 55: Configuring Web User Accounts
SIP User's Manual 3. Web-Based Management Configuring Web User Accounts To prevent unauthorized access to the Web interface, two Web user accounts are available (primary and secondary) with assigned user name, password, and access level. When you login to the Web interface, you are requested to provide the user name and password of one of these Web user accounts.
Page 56 Mediant 2000  To change the Web user accounts attributes: Open the Web User Accounts page (Configuration tab > System menu > Web User Accounts). Figure 3-34: WEB User Accounts Page (for Users with 'Security Administrator' Privileges) Note: If you are logged into the Web interface as the Security Administrator, both Web user accounts are displayed on the Web User Accounts page (as shown above).
Page 57 SIP User's Manual 3. Web-Based Management To change the user name of an account, perform the following: In the field 'User Name', enter the new user name (maximum of 19 case-sensitive characters). Click Change User Name; if you are currently logged into the Web interface with this account, the 'Enter Network Password' dialog box appears, requesting you to enter the new user name.
Page 58: Configuring Web Security Settings
Mediant 2000 Notes: • For security, it's recommended that you change the default user name and password. • A Web user with access level 'Security Administrator' can change all attributes of all the Web user accounts. Web users with an access level other than 'Security Administrator' can only change their own password and user name.
Page 59: Web Login Authentication Using Smart Cards
SIP User's Manual 3. Web-Based Management Web Login Authentication using Smart Cards You can enable Web login authentication using certificates from a third-party, common access card (CAC) with user identification. When a user attempts to access the device through the Web browser (HTTPS), the device retrieves the Web user’s login username (and other information, if required) from the CAC.
Page 60: Configuring Web And Telnet Access List
Mediant 2000 Configuring Web and Telnet Access List The Web & Telnet Access List page is used to define IP addresses (up to ten) that are permitted to access the device's Web, Telnet, and SSH interfaces. Access from an undefined IP address is denied. If no IP addresses are defined, this security feature is inactive and the device can be accessed from any IP address.
Page 61 SIP User's Manual 3. Web-Based Management To delete authorized IP addresses, select the Delete Row check boxes corresponding 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.
Page 62: Configuring Radius Settings
Mediant 2000 Configuring RADIUS Settings The RADIUS Settings page is used for configuring the Remote Authentication Dial In User Service (RADIUS) accounting parameters. For a description of these parameters, see ''Configuration Parameters Reference'' on page 409.  To configure RADIUS: Open the RADIUS Settings page (Configuration tab >...
Page 63: Mediant
SIP User's Manual 4. CLI-Based Management CLI-Based Management This section provides an overview of the CLI-based management and configuration relating to CLI management. The CLI can be accessed by using the RS-232 serial port or by using SSH or Telnet through the Ethernet interface.
Page 64: Configuring Telnet And Ssh Settings
Mediant 2000 Configuring Telnet and SSH Settings The Telnet/SSH Settings page is used to define Telnet and Secure Shell (SSH). For a description of these parameters, see ''Web and Telnet Parameters'' on page 422.  To define Telnet and SSH: Open the Telnet/SSH Settings page (Configuration tab >...
Page 65: Snmp-Based Management
SIP User's Manual 5. SNMP-Based Management SNMP-Based Management The device provides an embedded SNMP Agent to operate with a third-party SNMP Manager (e.g., element management system or EMS) for operation, administration, maintenance, and provisioning (OAMP) of the device. The SNMP Agent supports standard Management Information Base (MIBs) and proprietary MIBs, enabling a deeper probe into the interworking of the device.
Page 66: Configuring Snmp Trap Destinations
Mediant 2000 To delete a community string, select the Delete check box corresponding to the community string that you want to delete, and then click Submit. Table 5-1: 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).
Page 67 SIP User's Manual 5. SNMP-Based Management Parameter Description to receive SNMP traps.  [0] (Check box cleared) = Disabled (default)  [1] (Check box selected) = Enabled IP Address IP address of the remote host used as an SNMP Manager. [SNMPManagerTableIP_x] The device sends SNMP traps to these IP addresses.
Page 68: Configuring Snmp Trusted Managers
Mediant 2000 Configuring SNMP Trusted Managers The SNMP Trusted Managers page allows you to configure up to five SNMP Trusted Managers, based on IP addresses. By default, the SNMP agent accepts SNMP Get and Set requests from any IP address, as long as the correct community string is used in the request.
Page 69: Configuring Snmp V3 Users
SIP User's Manual 5. SNMP-Based Management Configuring SNMP V3 Users The SNMP v3 Users page allows you to configure authentication and privacy for up to 10 SNMP v3 users.  To configure the SNMP v3 users: Open the SNMP v3 Users page (Maintenance tab > System menu > Management submenu >...
Page 70 Mediant 2000 Parameter Description Privacy Key Privacy key. Keys can be entered in the form of a text password or [SNMPUsers_PrivKey] long hex string. Keys are always persisted as long hex strings and keys are localized. Group The group with which the SNMP v3 user is associated.
Page 71: Ems-Based Management
SIP User's Manual 6. 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.
Page 72 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 73: Ini File-Based Management
SIP User's Manual 7. INI File-Based Management INI File-Based Management The ini file is a text-based file (created using, for example, Notepad) that can contain any number of parameters settings. The ini file can be loaded to the device using the following methods: ...
Page 74: Configuring Ini File Table Parameters
Mediant 2000 7.1.2 Configuring ini File Table Parameters The ini file table parameters allow you to configure tables which can include multiple parameters (columns) and row entries (indices). When loading an ini file to the device, it's recommended to include only tables that belong to applications that are to be configured (dynamic tables of other applications are empty, but static tables are not).
Page 75: General Ini File Formatting Rules
SIP User's Manual 7. INI File-Based Management  Data lines must match the Format line, i.e., it must contain exactly the same number of Indices and Data fields and must be in exactly the same order.  A row in a table is identified by its table name and Index field. Each such row may appear only once in the ini file.
Page 76: Modifying An Ini File
Mediant 2000 Modifying an ini File You can modify an ini file currently used by the device. Modifying an ini file instead of loading an entirely new ini file preserves the device's current configuration.  To modify an ini file: Save the current ini file from the device to your PC, using the Web interface (see ''Backing Up and Loading Configuration File'' on page 369).
Page 77: Part Iii: General System Settings
Part III General System Settings This part provides general system configurations.
Page 78 Reader’s Notes...
Page 79: Configuring Certificates
SIP User's Manual 8. Configuring Certificates Configuring Certificates The Certificates page is used for configuring secure communication using HTTPS and SIP TLS. This page allows you to do the following:  Replace the device's certificate - see ''Replacing Device Certificate'' on page ...
Page 80 Mediant 2000 Open the Certificates page (Configuration tab > System menu > Certificates). Figure 8-1: Certificates Page SIP User's Manual Document #: LTRT-68814...
Page 81 SIP User's Manual 8. Configuring Certificates Under the Certificate Signing Request group, do the following: In the 'Subject Name [CN]' field, enter the DNS name. Fill in the rest of the request fields according to your security provider's instructions. Click Create CSR; a textual certificate signing request is displayed. Copy the text and send it to your security provider.
Page 82: Loading A Private Key
Mediant 2000 Loading a Private Key The device is shipped with a self-generated random private key, which cannot be extracted from the device. However, some security administrators require that the private key be generated externally at a secure facility and then loaded to the device through configuration.
Page 83: Mutual Tls Authentication
SIP User's Manual 8. Configuring Certificates 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.
Page 84: Self-Signed Certificates
Mediant 2000 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.
Page 85: Date And Time
SIP User's Manual 9. Date and Time Date and Time The date and time of the device can be configured manually or it can be obtained automatically from a Simple Network Time Protocol (SNTP) server. Configuring Manual Date and Time The date and time of the device can be configured manually.
Page 86: Configuring Automatic Date And Time Through Sntp Server
Mediant 2000 Configuring Automatic Date and Time through SNTP Server The Simple Network Time Protocol (SNTP) client functionality generates requests and reacts to the resulting responses using the NTP version 3 protocol definitions (according to RFC 1305). Through these requests and responses, the NTP client synchronizes the system time to a time source within the network, thereby eliminating any potential issues should the local system clock 'drift' during operation.
Page 87 SIP User's Manual 9. Date and Time The procedure below describes how to configure SNTP using the Web interface.  To configure SNTP using the Web interface: Open the Application Settings page (Configuration tab > System menu > Application Settings). Configure the NTP parameters: •...
Page 88 Mediant 2000 Configure daylight saving, if required: • 'Day Light Saving Time' (DayLightSavingTimeEnable) - enables daylight saving time • 'Start Time' (DayLightSavingTimeStart) and 'End Time' (DayLightSavingTimeEnd) - defines the period for which daylight saving time is relevant. • 'Offset' (DayLightSavingTimeOffset) - defines the offset in minutes to add to the time for daylight saving.
Page 89: Part Iv: Voip Settings
Part IV VoIP Settings This part describes the VoIP configurations.
Page 90 Reader’s Notes...
Page 91: Network
SIP User's Manual 10. Network Network This section describes the network-related configuration. 10.1 Ethernet Interface Configuration The device's Ethernet connection can be configured (using the ini file parameter EthernetPhyConfiguration) for one of the following modes:  Manual mode: • 10Base-T Half-Duplex or 10Base-T Full-Duplex •...
Page 92: Ethernet Interface Redundancy
Mediant 2000 10.2 Ethernet Interface Redundancy The device supports an Ethernet redundancy scheme. At the beginning of the start-up procedure, the device tests whether the ‘primary’ Ethernet interface is connected, by checking the existence of the Ethernet link carrier. If it's connected, the start-up procedure commences as usual.
Page 93 SIP User's Manual 10. Network Notes: • For more information and examples of network interfaces configuration, see ''Network Configuration'' on page 97. • When adding more than one interface, ensure that you enable VLANs using the 'VLAN Mode' (VlANMode) parameter. •...
Page 94 Mediant 2000  To configure IP network interfaces: Open the IP Settings page (Configuration tab > VoIP menu > Network submenu > IP Settings). Figure 10-1: IP Settings Page Note: The IP Settings page appears only on initial configuration (i.e., IP interfaces have never been configured) or after the device is restored to default settings.
Page 95 SIP User's Manual 10. Network Click Done to validate the interface. If the interface is not valid (e.g., if it overlaps with another interface in the table or if it does not adhere to the other rules as summarized in ''Multiple Interface Table Configuration Summary and Guidelines'' on page 102), a warning message is displayed.
Page 96 Mediant 2000 Parameter Description The prefix length is a Classless Inter-Domain Routing (CIDR) style presentation of a dotted-decimal subnet notation. The CIDR-style presentation is the latest method for interpretation of IP addresses. Specifically, instead of using eight-bit address blocks, it uses the...
Page 97: Network Configuration
SIP User's Manual 10. Network 10.3.1 Network Configuration The device allows you to configure multiple IP addresses with associated VLANs, using the Multiple Interface table. Complementing this table is the Routing table, which allows you to define static routing rules for non-local hosts/subnets. This section describes the various network configuration options offered by the device.
Page 98 Mediant 2000 10.3.1.1.1 Overview of Multiple Interface Table The Multiple Interfaces scheme allows you to define different IP addresses and VLANs in a table format, as shown below: Table 10-2: Multiple Interface Table Index Prefix Default VLAN Application Interface IP Address...
Page 99 SIP User's Manual 10. Network words, 192.168.0.0/16 is synonymous with 192.168.0.0 and a subnet 255.255.0.0 (Refer to http://en.wikipedia.org/wiki/Classless_Inter-Domain_Routing for more information). This CIDR notation lists the number of '1' bits in the subnet mask. So, a subnet mask of 255.0.0.0 (when broken down to its binary format) is represented by a prefix length of 8 (11111111 00000000 00000000 00000000), and a subnet mask of 255.255.255.252 is represented by a prefix length of 30 (11111111 11111111 11111111 11111100).
Page 100 Mediant 2000 10.3.1.1.2.3 VLAN ID Column This column defines the VLAN ID for each interface. This column must hold a unique value for each interface of the same address family. 10.3.1.1.2.4 Interface Name Column This column allows the configuration of a short string (up to 16 characters) to name this interface.
Page 101 SIP User's Manual 10. Network 10.3.1.1.3.4 Quality of Service Parameters The device allows you to specify values for Layer-2 and Layer-3 priorities, by assigning values to the following service classes:  Network Service class – network control traffic (ICMP, ARP) ...
Page 102 Mediant 2000 Application Traffic / Network Types Class-of-Service (Priority) Varies according to NTP settings Depends on traffic type: (EnableNTPasOAM):  Control: Premium control  OAMP  Management: Bronze  Control NFSServers_VlanType in the Gold NFSServers table 10.3.1.1.3.5 Assigning NTP Services to Application Types NTP applications can be associated with different application types (OAMP or Control) in different setups.
Page 103 SIP User's Manual 10. Network  Each network interface may be defined with a default gateway. This default gateway address must be in the same subnet as the associated interface. Additional routing rules may be specified in the Routing table (''Configuring the IP Routing Table'' on page 108).
Page 104: Setting Up Voip Networking
Mediant 2000 10.3.1.1.5 Troubleshooting the Multiple Interface Table If any of the Multiple Interface table guidelines are violated, the device falls back to a "safe mode" configuration, consisting of a single IPv4 interface without VLANs. For more information on validation failures, consult the Syslog messages.
Page 105 SIP User's Manual 10. Network BronzeServiceClassDiffServ = 10 ; Application Type for NTP applications: EnableNTPasOAM = 1 ; Multiple Interface Table Configuration: [InterfaceTable] FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes, InterfaceTable_InterfaceMode, InterfaceTable_IPAddress, InterfaceTable_PrefixLength, InterfaceTable_Gateway, InterfaceTable_VlanID, InterfaceTable_InterfaceName, InterfaceTable_PrimaryDNSServerIPAddress, InterfaceTable_SecondaryDNSServerIPAddress; InterfaceTable 0 = 6, 10, 192.168.85.14, 16, 192.168.0.1, 1, myAll, , ;...
Page 106 Mediant 2000 VLANS are not required and the 'Native' VLAN ID is irrelevant. Class of Service parameters may have default values. The required routing table features two routes: Table 10-8: Routing Table - Example 1 Destination Prefix Length Gateway Interface Metric 201.201.0.0...
Page 107 SIP User's Manual 10. Network Table 10-10: Routing Table - Example 2 Destination Prefix Length Gateway Interface Metric 176.85.49.0 192.168.0.1 All other parameters are set to their respective default values. The NTP application remains with its default application types. The corresponding ini file configuration is shown below: ;...
Page 108: Configuring The Ip Routing Table
Mediant 2000 One routing rule is required to allow remote management from a host in 176.85.49.0/24: Table 10-12: Routing Table - Example 3 Destination Subnet Destination Gateway Interface Metric Mask/Prefix Length 176.85.49.0 192.168.0.10 All other parameters are set to their respective default values. The NTP application remains with its default application types.
Page 109 SIP User's Manual 10. Network  To configure static IP routing: Open the IP Routing Table page (Configuration tab > VoIP menu > Network submenu > IP Routing Table). Figure 10-4: 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.
Page 110: Routing Table Columns
Mediant 2000 Parameter Description Gateway IP Address The IP address of the router (next hop) to which the packets [StaticRouteTable_Gateway] are sent if their destination matches the rules in the adjacent columns. Note: The Gateway address must be in the same subnet as the IP address of the interface over which you configure this static routing rule.
Page 111: Gateway Column
SIP User's Manual 10. Network 10.4.1.3 Gateway Column The Gateway column defines the IP address of the next hop used for traffic destined to the subnet/host as defined in the destination/mask columns. This gateway address must be on the same subnet as the IP address of the interface configured in the Interface column. 10.4.1.4 Interface Column This column defines the interface index (in the Multiple Interface table) from which the gateway address is reached.
Page 112: Routing Table Configuration Summary And Guidelines
Mediant 2000 10.4.2 Routing Table Configuration Summary and Guidelines The Routing table configurations must adhere to the following rules:  Up to 30 different static routing rules may be defined.  The Prefix Length replaces the dotted-decimal subnet mask presentation. This column must have a value of 0-31 for IPv4 interfaces.
Page 113: Dns
SIP User's Manual 10. Network  To configure QoS: Open the QoS Settings page (Configuration tab > VoIP menu > Network submenu > QoS Settings). Configure the QoS parameters as required. Click Submit to apply your changes. Save the changes to flash memory (see ''Saving Configuration'' on page 350). 10.6 You can use the device's embedded domain name server (DNS) or an external, third-party DNS to translate domain names into IP addresses.
Page 114 Mediant 2000  To configure the internal DNS table: Open the Internal DNS Table page (Configuration tab > VoIP menu > Network submenu > DNS submenu > Internal DNS Table). Figure 10-6: Internal DNS Table Page In the 'Domain Name' field, enter the host name to be translated. You can enter a string of up to 31 characters.
Page 115: Configuring The Internal Srv Table
SIP User's Manual 10. Network 10.6.2 Configuring the Internal SRV Table The Internal SRV Table page resolves host names to DNS A-Records. Three different A- Records can be assigned to each host name. Each A-Record contains the host name, priority, weight, and port. Notes: •...
Page 116: Nat (Network Address Translation) Support
Mediant 2000 10.7 NAT (Network Address Translation) Support Network Address Translation (NAT) is a mechanism that maps a set of internal IP addresses used within a private network to global IP addresses, providing transparent routing to end hosts. The primary advantages of NAT include (1) Reduction in the number of global IP addresses required in a private network (global IP addresses are only used to connect to the Internet);...
Page 117: First Incoming Packet Mechanism
SIP User's Manual 10. Network At the beginning of each call and if STUN is required (i.e., not an internal NAT call), the media ports of the call are mapped. The call is delayed until the STUN Binding Response (that includes a global IP:port) for each media (RTP, RTCP and T.38) is received. To enable STUN, perform the following: ...
Page 118: Configuring Nfs Settings
Mediant 2000 performed using the ini file parameter NoOpInterval. For a description of the RTP No-Op ini file parameters, see ''Networking Parameters'' on page 411.  RTP No-Op: The RTP No-Op support complies with IETF Internet-Draft draft-wing- avt-rtp-noop-03 ("A No-Op Payload Format for RTP"). This IETF document defines a No-Op payload format for RTP.
Page 119 SIP User's Manual 10. Network Configure the NFS parameters according to the table below. Click the Submit button; the remote NFS file system is immediately applied, which can be verified by the appearance of the 'NFS mount was successful' message in the Syslog server.
Page 120: Robust Receipt Of Media Streams
Mediant 2000 10.9 Robust Receipt of Media Streams This mechanism filters out unwanted RTP streams that are sent to the same port number on the device. These multiple RTP streams can result from traces of previous calls, call control errors, and deliberate attacks. When more than one RTP stream reaches the device on the same port number, the device accepts only one of the RTP streams and rejects the rest of the streams.
Page 121: Security
SIP User's Manual 11. Security Security This section describes the VoIP security-related configuration. 11.1 Configuring Firewall Settings The device provides an internal firewall, allowing you (the security administrator) to define network traffic filtering rules. You can add up to 50 ordered firewall rules. The access list provides the following firewall rules: ...
Page 122 Mediant 2000 Click one of the following buttons: • Apply: saves the new rule (without activating it). • Duplicate Rule: adds a new rule by copying a selected rule. • Activate: saves the new rule and activates it. • Delete: deletes the selected rule.
Page 123 SIP User's Manual 11. Security Parameter Description Source Port Defines the source UDP/TCP ports (on the remote host) [AccessList_Source_Port] from where packets are sent to the device. The valid range is 0 to 65535. Note: When set to 0, this field is ignored and any source port matches the rule.
Page 124 Mediant 2000 Parameter Description Packet Size Defines the maximum allowed packet size. [AccessList_Packet_Size] The valid range is 0 to 65535. Note: When filtering fragmented IP packets, this field relates to the overall (re-assembled) packet size, and not to the size of each fragment.
Page 125: Configuring General Security Settings
SIP User's Manual 11. Security 11.2 Configuring General Security Settings The General Security Settings page is used to configure various security features. For a description of the parameters appearing on this page, refer ''Configuration Parameters Reference'' on page 409.  To configure the general security parameters: Open the General Security Settings page (Configuration tab >...
Page 126: Configuring Ip Security Proposal Table
Mediant 2000 11.3 Configuring IP Security Proposal Table The IP Security Proposals Table page is used to configure Internet Key Exchange (IKE) with up to four proposal settings. Each proposal defines an encryption algorithm, an authentication algorithm, and a Diffie-Hellman group identifier. The same set of proposals applies to both Main mode and Quick mode.
Page 127: Configuring Ip Security Associations Table
SIP User's Manual 11. Security Parameter Name Description Diffie Hellman Group Determines the length of the key created by the [IPsecProposalTable_DHGroup] DH protocol for up to four proposals. For the ini file parameter, X depicts the proposal number (0 to 3). ...
Page 128 Mediant 2000  To configure the IPSec Association table: Open the ‘IP Security Associations Table page (Configuration tab > VoIP menu > Security submenu > IPSec Association Table). (Due to the length of the table, the figure below shows sections of this table.) Figure 11-3: IP Security Associations Table Page Add an Index or select the Index rule you want to edit.
Page 129 SIP User's Manual 11. Security Parameter Name Description Notes:  This parameter is applicable only if the Authentication Method parameter is set to pre-shared key.  The pre-shared key forms the basis of IPSec security and therefore, it should be handled with care (the same as sensitive passwords).
Page 130 Mediant 2000 Parameter Name Description Dead Peer Detection Mode Configures dead peer detection (DPD), according to RFC [IPsecSATable_DPDmode] 3706.  [0] DPD Disabled (default)  [1] DPD Periodic = DPD is enabled with message exchanges at regular intervals  [2] DPD on demand = DPD is enabled with on-demand checks - message exchanges as needed (i.e., before...
Page 131: Media
SIP User's Manual 12. Media Media This section describes the media-related configuration. 12.1 Configuring Voice Settings The Voice Settings page configures various voice parameters such as voice volume, silence suppression, and DTMF transport type. For a detailed description of these parameters, see ''Configuration Parameters Reference'' on page 409.
Page 132: Voice Gain (Volume) Control
Mediant 2000 12.1.1 Voice Gain (Volume) Control The device allows you to configure the level of the received (input gain) Tel-to-IP signal and the level of the transmitted (output gain) IP-to-Tel signal. The gain can be set between -32 and 31 decibels (dB).
Page 133: Echo Cancellation
SIP User's Manual 12. Media 12.1.3 Echo Cancellation The device supports adaptive linear (line) echo cancellation according to G.168-2002. Echo cancellation is a mechanism that removes echo from the voice channel. Echoes are reflections of the transmitted signal. In this line echo, echoes are generated when two-wire telephone circuits (carrying both transmitted and received signals on the same wire pair) are converted to a four-wire circuit.
Page 134: Fax And Modem Capabilities
Mediant 2000 12.2 Fax and Modem Capabilities This section describes the device's fax and modem capabilities, and includes the following main subsections:  Fax and modem operating modes (see ''Fax/Modem Operating Modes'' on page 135)  Fax and modem transport modes (see ''Fax/Modem Transport Modes'' on page 135) ...
Page 135: Fax/Modem Operating Modes
SIP User's Manual 12. Media 12.2.1 Fax/Modem Operating Modes The device supports two modes of operation:  Fax/modem negotiation that is not performed during the establishment of the call.  Voice-band data (VBD) mode for V.152 implementation (see ''V.152 Support'' on page 141): fax/modem capabilities are negotiated between the device and the remote endpoint at the establishment of the call.
Page 136: Fax / Modem Transport Mode
Mediant 2000 FaxRelayRedundancyDepth FaxRelayEnhancedRedundancyDepth parameters. Although this is a proprietary redundancy scheme, it should not create problems when working with other T.38 decoders. 12.2.2.1.1 Switching to T.38 Mode using SIP Re-INVITE In the Switching to T.38 Mode using SIP Re-INVITE mode, upon detection of a fax signal the terminating device negotiates T.38 capabilities using a Re-INVITE message.
Page 137: Fax Fallback
SIP User's Manual 12. Media  Dynamic Jitter Buffer Optimization Factor = 13 After a few seconds upon detection of fax V.21 preamble or super G3 fax signals, the device sends a second Re-INVITE enabling the echo canceller (the echo canceller is disabled only on modem transmission).
Page 138: Fax / Modem Nse Mode
Mediant 2000  FaxTransportMode = 2  V21ModemTransportType = 2  V22ModemTransportType = 2  V23ModemTransportType = 2  V32ModemTransportType = 2  V34ModemTransportType = 2  BellModemTransportType = 2  Additional configuration parameters: • FaxModemBypassCoderType • FaxBypassPayloadType • ModemBypassPayloadType •...
Page 139: Fax / Modem Transparent With Events Mode
SIP User's Manual 12. Media To configure NSE mode, perform the following configurations:  IsFaxUsed = 0  FaxTransportMode = 2  NSEMode = 1  NSEPayloadType = 100  V21ModemTransportType = 2  V22ModemTransportType = 2  V23ModemTransportType = 2 ...
Page 140: Rfc 2833 Ans Report Upon Fax/Modem Detection
Mediant 2000  Additional configuration parameters: • CodersGroup • DJBufOptFactor • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission. Instead, use the modes Bypass (see ''Fax/Modem Bypass Mode'' on page 137) or Transparent with Events (see ''Fax / Modem Transparent with Events Mode'' on page 139) for modem.
Page 141: Relay Mode For T.30 And V.34 Faxes
SIP User's Manual 12. Media Configure the following parameters to use bypass mode for V.34 faxes and T.38 for T.30 faxes:  FaxTransportMode = 1 (Relay)  V34ModemTransportType = 2 (Modem bypass)  V32ModemTransportType = 2  V23ModemTransportType = 2 ...
Page 142: Fax Transmission Behind Nat
Mediant 2000 Instead of using VBD transport mode, the V.152 implementation can use alternative relay fax transport methods (e.g., fax relay over IP using T.38). The preferred V.152 transport method is indicated by the SDP ‘pmft’ attribute. Omission of this attribute in the SDP content means that VBD mode is the preferred transport mechanism for voice-band data.
Page 143: Configuring Rtp/Rtcp Settings
SIP User's Manual 12. Media 12.3 Configuring RTP/RTCP Settings The RTP/RTCP Settings page configures the Real-Time Transport Protocol (RTP) and Real-Time Transport (RTP) Control Protocol (RTCP) parameters. For a detailed description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page 409.
Page 144: Dynamic Jitter Buffer Operation
Mediant 2000 12.3.1 Dynamic Jitter Buffer Operation Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames.
Page 145: Comfort Noise Generation
SIP User's Manual 12. Media 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 > VoIP menu > Media submenu >...
Page 146: Configuring Rfc 2833 Payload
Mediant 2000  Using NOTIFY messages according to IETF Internet-Draft draft-mahy-sipping- signaled-digits-01: DTMF digits are carried to the remote side using NOTIFY messages. To enable this mode, define the following: • RxDTMFOption = 0 • TxDTMFOption = 2 Note that in this mode, DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0).
Page 147 SIP User's Manual 12. Media  To configure RFC 2833 payload using the Web interface: Open the RTP/RTCP Settings page (Configuration tab > VoIP menu > Media submenu > RTP/RTCP Settings). Figure 12-4: RTP/RTCP Settings Page Configure the following parameters: •...
Page 148: Rtp Multiplexing (Throughpacket)
Mediant 2000 12.3.4 RTP Multiplexing (ThroughPacket) The device supports a proprietary method to aggregate RTP streams from several channels. This reduces the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP headers and reduces the packet/data transmission rate. This option reduces the load on network routers and can typically save 50% (e.g., for G.723) on IP bandwidth.
Page 149 SIP User's Manual 12. Media The procedure below describes how to configure RTP multiplexing using the Web interface.  To configure RTP multiplexing parameters: Open the RTP/RTCP Settings page (Configuration tab > VoIP menu > Media submenu > RTP/RTCP Settings). Figure 12-5: RTP/RTCP Settings Page Configure the following parameters: •...
Page 150: Configuring Rtp Base Udp Port
Mediant 2000 12.3.5 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.
Page 151 SIP User's Manual 12. Media To enable RTCP XR reporting, the VQMonEnable ini file parameter must be set to 1. In addition, the device must be installed with the appropriate Software Upgrade Key. For a detailed description of the RTCP XR ini file parameters, refer to the device's User's Manual. The procedure below describes how to configure RTCP XR using the Web interface.
Page 152: Configuring Ip Media Settings
Mediant 2000 12.4 Configuring IP Media Settings The IPMedia Settings page allows you to configure the IP media parameters. For a detailed description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 409.  To configure the IP media parameters: Open the IPMedia Settings page (Configuration tab >...
Page 153: Answer Machine Detector (Amd)
SIP User's Manual 12. Media 12.4.1 Answer Machine Detector (AMD) The device provides answering machine detection (AMD) capabilities that can detect for example, if a human voice or an answering machine is answering the call. AMD is useful for automatic dialing applications. The device supports up to four AMD parameter suites, where each parameter suite defines the AMD sensitivity levels of detection.
Page 154 Mediant 2000 Performance AMD Detection Sensitivity Success Rate for Live Calls Success Rate for Answering Machine 88.94% 94.31% 90.42% 91.64% 90.66% 91.30% 7 (Best for Live 94.72% 76.14% Calls) Table 12-2: Approximate AMD Detection High Sensitivity (Based on North American English)
Page 155 SIP User's Manual 12. Media Note: 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.
Page 156 Mediant 2000 INFO sip:sipp@172.22.2.9:5060 SIP/2.0 Via: SIP/2.0/UDP 172.22.168.249;branch=z9hG4bKac482466515 Max-Forwards: 70 From: sut <sip:3000@172.22.168.249:5060>;tag=1c419779142 To: sipp <sip:sipp@172.22.2.9:5060>;tag=1 Call-ID: 1-29753@172.22.2.9 CSeq: 1 INFO Contact: <sip:56700@172.22.168.249> Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB SCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway/v.6.40A.040.004 Content-Type: application/x-detect Content-Length: 34 Type= PTT SubType= SPEECH-START Upon detection of the end of voice (i.e., end of the greeting message of the answering machine), the device sends the Application server the following: INFO sip:sipp@172.22.2.9:5060 SIP/2.0...
Page 157: Configuring Automatic Gain Control (Agc)
SIP User's Manual 12. Media 12.4.2 Configuring Automatic Gain Control (AGC) Automatic Gain Control (AGC) adjusts the energy of the output signal to a required level (volume). This feature compensates for near-far gain differences. AGC estimates the energy of the incoming signal (from the IP or PSTN, determined by the parameter AGCRedirection), calculates the essential gain, and then performs amplification.
Page 158: Configuring Dsp Templates
Mediant 2000  To configure general media parameters: Open the General Media Settings page (Configuration tab > VoIP menu > Media submenu > General Media Settings). Configure the parameters as required. Click Submit to apply your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 350.
Page 159: Configuring Media Realms
SIP User's Manual 12. Media 12.7 Configuring Media Realms The Media Realm Table page allows you to define a pool of up to 64 SIP media interfaces, termed Media Realms. Media Realms allow you to divide a Media-type interface (defined in the Multiple Interface table - see ''Configuring IP Interface Settings'' on page 92) into several realms, where each realm is specified by a UDP port range.
Page 160 Mediant 2000 Table 12-3: Media Realm Table Parameter Descriptions Parameter Description Index Defines the required table index number. [CpMediaRealm_Index] Media Realm Name Defines an arbitrary, identifiable name for the Media Realm. [CpMediaRealm_MediaRealmName] The valid value is a string of up to 40 characters.
Page 161: Configuring Media Security
SIP User's Manual 12. Media 12.8 Configuring Media Security The Media Security page allows you to configure media security. For a detailed description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 409.  To configure media security: Open the Media Security page (Configuration tab >...
Page 162 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 163: Services
SIP User's Manual 13. Services Services This section describes configuration for various supported services. 13.1 Routing Based on LDAP Active Directory Queries The device supports Lightweight Directory Access Protocol (LDAP), allowing the device to make call routing decisions based on information stored on a third-party LDAP server (or Microsoft’s Active Directory-based enterprise directory server).
Page 164: Configuring Ldap Settings
Mediant 2000 13.1.2 Configuring LDAP Settings The LDAP Settings page is used for configuring the Lightweight Directory Access Protocol (LDAP) parameters. For a description of these parameters, see ''Configuration Parameters Reference'' on page 409. For an overview of LDAP, see ''Routing Based on LDAP Active Directory Queries'' on page 163.
Page 165 SIP User's Manual 13. Services Based on queries sent to the AD, this feature allows you to route incoming Tel calls to one of the following IP domains:  PBX/IP-PBX (for users yet to migrate to the OCS 2007 platform) ...
Page 166 Mediant 2000 Below is an example for configuring AD-based routing rules in the Outbound IP Routing Table (see ''Configuring Outbound IP Routing Table'' on page 256): Figure 13-2: Active Directory-based Routing Rules in Outbound IP Routing Table  First rule: sends call to IP-PBX (10.33.45.65) if AD query replies with prefix "PBX:"...
Page 167: Least Cost Routing
SIP User's Manual 13. Services 13.2 Least Cost Routing This section provides a description of the device's least cost routing (LCR) feature and how to configure it. 13.2.1 Overview The LCR feature enables the device to choose the outbound IP destination routing rule based on lowest call cost.
Page 168 Mediant 2000 Below are a few examples of how you can implement LCR:  Example 1: This example uses two different Cost Groups for routing local calls and international calls: Two Cost Groups are configured as shown below: Table 13-2: Configured Cost Groups for Local and International Calls...
Page 169 SIP User's Manual 13. Services • 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) ...
Page 170: Configuring Lcr
Mediant 2000 13.2.2 Configuring LCR The following main steps need to be done to configure LCR: Enable the LCR feature and configure the average call duration and default call connection cost - see ''Enabling the LCR Feature'' on page 170.
Page 171 SIP User's Manual 13. Services Parameter Description LCR Call Length Defines the average call duration (in minutes) and [RoutingRuleGroups_LCRAverageCallLength] is used to calculate the variable portion of the call cost. This is useful, for example, when the average call duration spans over multiple time bands. The LCR is calculated as follows: cost = call connect cost + (minute cost * average call duration) The valid value range is 0-65533.
Page 172: Configuring Cost Groups
Mediant 2000 13.2.2.2 Configuring Cost Groups The procedure below describes how to configure Cost Groups. Cost Groups are defined with a fixed call connection cost and a call rate (charge per minute). Once configured, you can configure Time Bands for each Cost Group. Up to 10 Cost Groups can be configured.
Page 173: Configuring Time Bands For Cost Groups
SIP User's Manual 13. Services 13.2.2.3 Configuring Time Bands for Cost Groups The procedure below describes how to configure Time Bands for a Cost Group. The time band defines the day and time range for which the time band is applicable (e.g., from Saturday 05:00 to Sunday 24:00) as well as the fixed call connection charge and call rate per minute for this interval.
Page 174: Assigning Cost Groups To Routing Rules
Mediant 2000 Parameter Description End Time Defines the day and time of day until when this time [CostGroupTimebands_EndTime] band is applicable. For a description of the valid values, see the parameter above. Connection Cost Defines the call connection cost during this time band.
Page 175: Control Network
SIP User's Manual 14. Control Network Control Network This section describes configuration of the network at the SIP control level. 14.1 Configuring SRD Table The SRD Settings page allows you to configure up to 32 signaling routing domains (SRD). An SRD is configured with a unique name and assigned a Media Realm (defined in the Media Realm table - see ''Configuring Media Realms'' on page 159).
Page 176 Mediant 2000  To configure SRDs: Open the SRD Settings page (Configuration tab > VoIP menu > Control Network submenu > SRD Table). From the 'SRD Index' drop-down list, select an index for the SRD, and then configure it according to the table below.
Page 177: Configuring Sip Interface Table
SIP User's Manual 14. Control Network Parameter Description this value becomes invalid in the SRD table.  For configuring Media Realms, see ''Configuring Media Realms'' on page 159. 14.2 Configuring SIP Interface Table The SIP Interface Table page allows you to configure up to 32 SIP signaling interfaces, referred to as SIP Interfaces.
Page 178 Mediant 2000 Parameter Description identical (including case-sensitive) to that configured in the 'Interface Name' in the Multiple Interface table (see ''Configuring IP Interface Settings'' on page 92). The default is "Not Configured". Note: SIP Interfaces that are assigned to a specific SRD must be defined with the same network interface.
Page 179: Configuring Ip Groups
SIP User's Manual 14. Control Network 14.3 Configuring IP Groups The IP Group Table page allows you to create up to 32 logical IP entities called IP Groups. An IP Group is an entity with a set of definitions such as a Proxy Set ID (see ''Configuring Proxy Sets Table'' on page 184), which represents the IP address of the IP Group.
Page 180 Mediant 2000  To configure IP Groups: Open the IP Group Table page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). Configure the IP group parameters according to the table below. Click Submit to apply your changes.
Page 181 SIP User's Manual 14. Control Network Parameter Description forwards these responses directly to the SIP users. To route a call to a registered user, a rule must be configured in the Outbound IP Routing Table table (see Configuring Outbound IP Routing Table on page 256). The device searches the dynamic database (by using the request URI) for an entry that matches a registered AOR or Contact.
Page 182 Mediant 2000 Parameter Description The SRD (defined in Configuring SRD Table on page 175) [IPGroup_SRD] associated with the IP Group. The default is 0. Note: For this parameter to take effect, a device reset is required. Media Realm Associates a Media Realm with the IP Group. The entered...
Page 183 SIP User's Manual 14. Control Network Parameter Description Groups. SIP Re-Routing Mode Determines the routing mode after a call redirection (i.e., a 3xx SIP response is received) or transfer (i.e., a SIP REFER [IPGroup_SIPReRoutingMode] request is received).  [0] Standard = INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI, according to the Refer-To header in the REFER message or Contact header in the 3xx response (default).
Page 184: Configuring Proxy Sets Table
Mediant 2000 Parameter Description Serving IP Group ID If configured, INVITE messages initiated from the IP Group are [IPGroup_ServingIPGroup] sent to this Serving IP Group (range 1 to 9). In other words, the INVITEs are sent to the address defined for the Proxy Set associated with this Serving IP Group.
Page 185 SIP User's Manual 14. Control Network  To add Proxy servers: Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control Network submenu > Proxy Sets Table). Figure 14-1: Proxy Sets Table Page From the 'Proxy Set ID' drop-down list, select an ID for the desired group. Configure the Proxy parameters according to the following table.
Page 186 Mediant 2000 Parameter Description sent according to the following preferences:  To the Trunk Group's Serving IP Group ID, as defined in the Trunk Group Settings table.  According to the Outbound IP Routing Table if the parameter PreferRouteTable is set to 1.
Page 187 SIP User's Manual 14. Control Network Parameter Description  [2] Using Register = Enables Keep-Alive with Proxy using SIP REGISTER messages. If set to 'Using Options', the SIP OPTIONS message is sent every user-defined interval (configured by the parameter ProxyKeepAliveTime). If set to 'Using Register', the SIP REGISTER message is sent every user-defined interval (configured by the RegistrationTime parameter).
Page 188: Configuring Nat Translation Per Ip Interface
Mediant 2000 Parameter Description percentage of the requests according to its' assigned weight. A single FQDN should be configured as a Proxy IP address. The Random Weights Load Balancing is not used in the following scenarios:  The Proxy Set includes more than one Proxy IP address.
Page 189 SIP User's Manual 14. Control Network The device’s priority method for performing NAT is as follows: Uses an external STUN server (STUNServerPrimaryIP parameter) to assign a NAT address to all interfaces. Uses the StaticNATIP parameter to define one NAT IP address for all interfaces. Uses the NATTranslation parameter to define NAT per interface.
Page 190: Multiple Sip Signaling And Media Interfaces Using Srds
Mediant 2000 14.6 Multiple SIP Signaling and Media Interfaces using SRDs The device supports the configuration of multiple, logical SIP signaling interfaces and media (RTP) interfaces. Multiple SIP and media interfaces allows you to:  Separate SIP and media traffic between different applications (i.e., SAS, Gateway\IP- to-IP) ...
Page 191 SIP User's Manual 14. Control Network Figure 14-3: Configuring SRDs and Assignment Typically, an SRD is defined per group of SIP UAs (e.g., proxies, IP phones, application servers, gateways, and softswitches) that communicate with each other. This provides these entities with VoIP services that reside on the same Layer-3 network (must be able to communicate without traversing NAT devices and must not have overlapping IP addresses).
Page 192 Mediant 2000 The figure below illustrates a typical scenario for implementing multiple SIP signaling interfaces. In this example, different SIP signaling interfaces and RTP traffic interfaces are assigned to Network 1 (ITSP A) and Network 2 (ITSP B). Below provides an example for configuring multiple SIP signaling and RTP interfaces. In...
Page 193 SIP User's Manual 14. Control Network Note that only the steps specific to multiple SIP signaling/RTP configuration are described in detail in the procedure below.  To configure multiple SIP signaling and RTP interfaces: Configure Trunk Group ID #1 in the Trunk Group Table page (Configuration tab > VoIP menu >...
Page 194 Mediant 2000 Configure SRDs in the SRD table (Configuration tab > VoIP menu > Control Network submenu > SRD Table): • SRD1 associated with media realm "Realm1". • SRD2 associated with media realm "Realm2". Figure 14-8: Defining SRDs Configure the SIP Interfaces in the SIP Interface Table page (Configuration tab >...
Page 195 SIP User's Manual 14. Control Network Configure IP Groups in the IP Group Table page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). The figure below configures IP Group for ITSP A. Do the same for ITSP B but for Index 2 with SRD 1 and Media Realm to "Realm2".
Page 196 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 197: Enabling Applications
SIP User's Manual 15. Enabling Applications Enabling Applications The device supports the following main applications:  Stand-Alone Survivability (SAS) application  IP2IP application The procedure below describes how to enable these applications. Once an application is enabled, the Web GUI provides menus and parameter fields relevant to the application. Notes: •...
Page 198 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 199: Coders And Profiles
SIP User's Manual 16. Coders and Profiles Coders and Profiles This section describes configuration of the coders and SIP profiles parameters. 16.1 Configuring Coders The Coders page allows you to configure up to 10 voice coders for the device to use. Each coder can be configured with packetization time (ptime), rate, payload type, and silence suppression.
Page 200: Configuring Coder Groups
Mediant 2000  To configure the device's coders: Open the Coders page (Configuration tab > VoIP menu > Coders And Profiles submenu > Coders). Figure 16-1: Coders Page From the 'Coder Name' drop-down list, select the required coder. From the 'Packetization Time' drop-down list, select the packetization time (in msec) for the selected coder.
Page 201 SIP User's Manual 16. Coders and Profiles Notes: • Each voice coder can appear only once per Coder Group. • For a list of supported coders and for configuring coders using the ini file, refer to the ini file parameter table CodersGroup, described in ''Configuration Parameters Reference'' on page 409.
Page 202: Configuring Tel Profile
Mediant 2000 16.3 Configuring Tel Profile The Tel Profile Settings page allows you to define up to nine SIP profiles for Tel calls (termed Tel Profiles). Each Tel Profile contains a set of parameters for configuring various behaviors, for example, used coder, silence suppression support, and echo canceler. Once configured, Tel Profiles can then be assigned to specific trunks (channels).
Page 203 SIP User's Manual 16. Coders and Profiles From the 'Profile ID' drop-down list, select the Tel Profile index. In the 'Profile Name' field, enter an arbitrary name that enables you to easily identify the Tel Profile. From the 'Profile Preference' drop-down list, select the priority of the Tel Profile, where 1 is the lowest priority and 20 the highest.
Page 204: Configuring Ip Profiles
Mediant 2000 16.4 Configuring IP Profiles The IP Profile Settings page allows you to define up to nine SIP profiles for IP calls (termed IP Profile). Each IP Profile contains a set of parameters for configuring various behaviors, for example, used coder, echo canceller support, and jitter buffer. Once configured, different IP Profiles can be assigned to specific inbound and outbound calls.
Page 205 SIP User's Manual 16. Coders and Profiles  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. In the 'Profile Name' field, enter an arbitrary name that allows you to easily identify the IP Profile.
Page 206 Mediant 2000 From the 'Profile Preference' drop-down list, select the priority of the IP Profile, where '1' is the lowest priority and '20' is the highest. If both IP and Tel profiles apply to the same call, the coders and other common parameters (noted by an asterisk) of the preferred Profile are applied to that call.
Page 207: Sip Definitions
SIP User's Manual 17. SIP Definitions SIP Definitions This section describes configuration of SIP parameters. 17.1 Configuring SIP General Parameters The SIP General Parameters page is used to configure general SIP parameters. For a description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 409.
Page 208: Configuring Advanced Parameters
Mediant 2000 Click Submit to apply your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 350. 17.2 Configuring Advanced Parameters The Advanced Parameters page allows you to configure advanced SIP control parameters. For a description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 409.
Page 209: Configuring Account Table
SIP User's Manual 17. SIP Definitions 17.3 Configuring Account Table The Account Table page allows you to define up to 32 Accounts per Trunk Group (Served Trunk Group) or source IP Group (Served IP Group). This is used for registration and/or digest authentication (user name and password) to a destination IP address (Serving IP Group).
Page 210 Mediant 2000 Table 17-1: Account Table Parameters Description Parameter Description Served Trunk Group The Trunk Group ID for which you want to register and/or [Account_ServedTrunkGroup] authenticate to a destination IP Group (i.e., Serving IP Group). For Tel-to-IP calls, the Served Trunk Group is the source Trunk Group from where the call originated.
Page 211 SIP User's Manual 17. SIP Definitions Parameter Description This parameter can be up to 49 characters. Register Enables registration. [Account_Register]  [0] No = Don't register  [1] Yes = Enables registration When enabled, the device sends REGISTER requests to the Serving IP Group.
Page 212: Configuring Proxy And Registration Parameters
Mediant 2000 17.4 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 409.
Page 213 SIP User's Manual 17. SIP Definitions Click the Register or Un-Register buttons to save your changes register/unregister the device to a Proxy/Registrar. Instead of registering the entire device, you can register specific entities ( Trunk Groups, and Accounts), by using the Register button located on the page in which these entities are configured.
Page 214 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 215: Gw And Ip To Ip
SIP User's Manual 18. GW and IP to IP GW and IP to IP This section describes configuration for the GW/IP2IP applications. Note: The "GW" and "IP2IP" applications refer to the Gateway and IP-to-IP applications respectively. 18.1 Digital PSTN This section describes configuration of the public switched telephone network (PSTN) parameters.
Page 216: Configuring Cas State Machines
Mediant 2000 18.1.2 Configuring CAS State Machines The CAS State Machine page allows you to modify various timers and other basic parameters to define the initialization of the CAS state machine without changing the state machine itself (no compilation is required). The change doesn't affect the state machine itself, but rather the configuration.
Page 217 SIP User's Manual 18. GW and IP to IP Table 18-1: CAS State Machine Parameters Description Parameter Description Generate Digit On Time Generates digit on-time (in msec). [CasStateMachineGenerateDigitOnTim The value must be a positive value. The default value is - 1 (use value from CAS state machine).
Page 218: Configuring Trunk Settings
Mediant 2000 18.1.3 Configuring Trunk Settings The Trunk Settings page allows you to configure the device's trunks. This includes selecting the PSTN protocol and configuring related parameters. Some parameters can be configured when the trunk is in service, while others require you to take the trunk out of service (by clicking the Stop button).
Page 219 SIP User's Manual 18. GW and IP to IP On the top of the page, a bar with Trunk number icons displays the status of each trunk, according to the following color codes: • Grey: Disabled • Green: Active • Yellow: RAI alarm (also appears when you deactivate a Trunk by clicking the Deactivate button) •...
Page 220: Configuring Digital Gateway Parameters
Mediant 2000 To save the changes to flash memory, see ''Saving Configuration'' on page 350. To reset the device, see ''Resetting the Device'' on page 347. Notes: • If the ‘Protocol Type’ field displays 'NONE' (i.e., no protocol type is selected) and no other trunks have been configured, after selecting a PRI protocol type, you must reset the device.
Page 221 SIP User's Manual 18. GW and IP to IP  To configure the digital gateway parameters: Open the Digital Gateway Parameters page (Configuration tab > VoIP menu > GW and IP to IP submenu > Digital Gateway submenu > Digital Gateway Parameters). Figure 18-4: Digital Gateway Parameters Page Configure the parameters as required.
Page 222: Tunneling Applications
Mediant 2000 18.1.5 Tunneling Applications This section discusses the device's support for VoIP tunneling applications. 18.1.5.1 TDM Tunneling The device's TDM Tunneling feature allows you to tunnel groups of digital trunk spans or timeslots (B-channels) over the IP network. TDM Tunneling utilizes the device's internal...
Page 223 SIP User's Manual 18. GW and IP to IP 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').
Page 224 Mediant 2000 [ \CodersGroup0 ] [TelProfile] FORMAT TelProfile_Index = TelProfile_ProfileName, TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP; TelProfile 1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$; TelProfile 2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$; [\TelProfile] Originating Side: ;E1_TRANSPARENT_31...
Page 225: Qsig Tunneling
SIP User's Manual 18. GW and IP to IP TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP; TelProfile_1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$ TelProfile_2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$ [\TelProfile] 18.1.5.1.1 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.
Page 226 Mediant 2000  Call setup (terminating device): After the terminating device receives a SIP INVITE request with a 'Content-Type: application/QSIG', it sends the encapsulated QSIG Setup message to the Tel side and sends a 200 OK response (no 1xx response is sent) to IP.
Page 227: Advanced Pstn Configuration
SIP User's Manual 18. GW and IP to IP 18.1.6 Advanced PSTN Configuration This section describes various advanced PSTN configurations. 18.1.6.1 Release Reason Mapping This section describes the available mapping mechanisms of SIP responses to Q.850 Release Causes and vice versa. The existing mapping of ISDN Release Causes to SIP Responses is described in ''Fixed Mapping of ISDN Release Reason to SIP Response'' on page and ''Fixed Mapping of SIP Response to ISDN Release Reason'' on page 229.
Page 228 Mediant 2000 ISDN Release Description Description Reason Response No route to destination Not found Channel unacceptable Not acceptable Call awarded and being delivered in an Server internal error established channel Normal call clearing User busy Busy here No user responding...
Page 229 SIP User's Manual 18. GW and IP to IP ISDN Release Description Description Reason Response Identified channel does not exist 502* Bad gateway Suspended call exists, but this call 503* Service unavailable identity does not Call identity in use 503* Service unavailable No call suspended 503*...
Page 230 Mediant 2000 ISDN Release SIP Response Description Description Reason Forbidden Call rejected Not found Unallocated number Method not allowed Service/option unavailable Not acceptable Service/option not implemented Proxy authentication required Call rejected Request timeout Recovery on timer expiry Conflict Temporary failure...
Page 231: Isdn Overlap Dialing
SIP User's Manual 18. GW and IP to IP 18.1.6.2 ISDN Overlap Dialing Overlap dialing is a dialing scheme used by several ISDN variants to send and/or receive called number digits one after the other (or several at a time). This is in contrast to en-bloc dialing in which a complete number is sent in one message.
Page 232: Isdn Non-Facility Associated Signaling (Nfas)
Mediant 2000 18.1.6.2.2 Interworking ISDN Overlap Dialing with SIP According to RFC 3578 The device supports the interworking of ISDN overlap dialing to SIP and vice versa, according to RFC 3578.  Interworking ISDN overlap dialing to SIP (Tel to IP): The device sends collected digits each time it receives them (initially from the ISDN Setup message and then from subsequent Q.931 Information messages) to the IP side, using subsequent SIP...
Page 233 SIP User's Manual 18. GW and IP to IP For example, to assign the first four T1 trunks to NFAS group #1, in which trunk #0 is the primary trunk and trunk #1 is the backup trunk, use the following configuration: NFASGroupNumber_0 = 1 NFASGroupNumber_1 = 1 NFASGroupNumber_2 = 1...
Page 234 Mediant 2000 For example, if four T1 trunks on a device are configured as a single NFAS group with Primary and Backup T1 trunks that is used with a DMS-100 switch, the following parameters should be used: NFASGroupNumber_0 = 1...
Page 235: Redirect Number And Calling Name (Display)
SIP User's Manual 18. GW and IP to IP 18.1.6.4 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: Table 18-4: Calling Name (Display) NT/TE Interface...
Page 236: Trunk Group
Mediant 2000 18.2 Trunk Group This section describes the configuration of the device's channels, which entails assigning them numbers and Trunk Group IDs. 18.2.1 Configuring Trunk Group Table The Trunk Group Table page allows you to define up to 120 Trunk Groups. A Trunk Group is a logical group of physical trunks and channels, and is assigned an ID.
Page 237 SIP User's Manual 18. GW and IP to IP Table 18-6: Trunk Group Table Parameters Parameter Description From Trunk Starting physical Trunk number in the Trunk Group. The [TrunkGroup_FirstTrunkId] number of listed Trunks depends on the device's hardware configuration. To Trunk Ending physical Trunk number in the Trunk Group.
Page 238: Configuring Trunk Group Settings
Mediant 2000 Parameter Description channels within Trunk Groups, using the parameter TrunkGroupSettings. Tel Profile ID The Tel Profile ID assigned to the channels pertaining to the [TrunkGroup_ProfileId] Trunk Group. Note: For configuring Tel Profiles, refer to the parameter TelProfile. 18.2.2 Configuring Trunk Group Settings The Trunk Group Settings page allows you to configure the settings of up to 120 Trunk Groups.
Page 239 SIP User's Manual 18. GW and IP to IP REGISTER sip:SipGroupName SIP/2.0 Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac862428454 From: <sip:101@GatewayName>;tag=1c862422082 To: <sip:101@GatewayName> Call-ID: 9907977062512000232825@10.33.37.78 CSeq: 3 REGISTER Contact: <sip:101@10.33.37.78>;expires=3600 Expires: 3600 User-Agent: Sip-Gateway/v.6.00A.008.002 Content-Length: 0 Table 18-7: Trunk Group Settings Parameters Parameter Description Trunk Group ID The Trunk Group ID that you want to configure.
Page 240 Mediant 2000 Parameter Description IP Group, according to the settings in the Account table (see ''Configuring Account Table'' on page 209). Notes:  To enable Trunk Group registrations, configure the global parameter IsRegisterNeeded to 1. This is unnecessary for 'Per Account' registration mode.
Page 241: Manipulation
SIP User's Manual 18. GW and IP to IP 18.3 Manipulation This section describes the configuration of number / name manipulation rules and various SIP to non-SIP mapping. 18.3.1 Configuring General Settings The General Settings page allows you to configure general manipulation parameters. For a description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 409.
Page 242 Mediant 2000 The device manipulates the number in the following order: Strips digits from the left of the number. Strips digits from the right of the number. Retains the defined number of digits. Adds the defined prefix. Adds the defined suffix.
Page 243 SIP User's Manual 18. GW and IP to IP  To configure number manipulation rules: Open the required 'Number Manipulation page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations submenu > Dest Number IP->Tel, Dest Number Tel->IP, Source Number IP->Tel, or Source Number Tel->IP);...
Page 244 Mediant 2000 Parameter Description Source IP Group The IP Group from where the IP-to-IP call originated. Typically, this IP Group of an incoming INVITE is determined/classified using the ‘Inbound IP Routing Table'. If not used (i.e., any IP Group), simply leave the field empty.
Page 245 SIP User's Manual 18. GW and IP to IP Parameter Description Web/EMS: Number of The number of digits that you want to retain from the right of the phone Digits to Leave number. For example, if you enter '4' and the phone number is 00165751234, then the new number is 1234.
Page 246: Configuring Redirect Number Ip To Tel
Mediant 2000 18.3.3 Configuring Redirect Number IP to Tel The Redirect Number IP > Tel page allows you to configure IP-to-Tel redirect number manipulation rules. This feature allows you to manipulate the value of the received SIP Diversion, Resource-Priority, or History-Info headers, which is then added to the Redirecting Number Information Element (IE) in the ISDN Setup message that is sent to the Tel side.
Page 247 SIP User's Manual 18. GW and IP to IP Parameter Description Web: Stripped Digits From Number of digits to remove from the right of the telephone number Right prefix. For example, if you enter 3 and the phone number is 5551234, EMS: Remove From Right the new phone number is 5551.
Page 248: Configuring Redirect Number Tel To Ip
Mediant 2000 18.3.4 Configuring Redirect Number Tel to IP The Redirect Number Tel > IP page allows you to configure Tel-to-IP Redirect Number manipulation rules. This feature manipulates the prefix of the redirect number received from the PSTN for the outgoing SIP Diversion, Resource-Priority, or History-Info header that is sent to IP.
Page 249 SIP User's Manual 18. GW and IP to IP Table 18-10: Redirect Number Tel to IP Parameters Description Parameter Description Source Trunk Group The Trunk Group from where the Tel call is received. To denote any Trunk Group, leave this field empty. Notes: ...
Page 250: Mapping Npi/Ton To Sip Phone-Context
Mediant 2000 18.3.5 Mapping NPI/TON to SIP Phone-Context The Phone-Context Table page allows you to map Numbering Plan Indication (NPI) and Type of Number (TON) to the SIP Phone-Context parameter. When a call is received from the ISDN, the NPI and TON are compared against the table and the matching Phone- Context value is used in the outgoing SIP INVITE message.
Page 251: Numbering Plans And Type Of Number
SIP User's Manual 18. GW and IP to IP Table 18-11: Phone-Context Parameters Description Parameter Description Add Phone Context As Prefix Determines whether the received Phone-Context parameter is added [AddPhoneContextAsPrefix] as a prefix to the outgoing ISDN SETUP message with Called and Calling numbers.
Page 252 Mediant 2000 Description National [2] A public number in complete national E.164 format, e.g., 6135551234. Subscriber [4] A public number in complete E.164 format representing a local subscriber, e.g., 5551234. Private [9] Unknown [0] A private number, but with no further information about the numbering plan.
Page 253: Configuring Release Cause Mapping
SIP User's Manual 18. GW and IP to IP 18.3.7 Configuring Release Cause Mapping The Release Cause Mapping page consists of two groups that allow the device to map up to 12 different SIP Response Codes to ITU-T Q.850 Release Cause Codes and vice versa, thereby overriding the hard-coded mapping mechanism (described in ''Release Reason Mapping'' on page 227).
Page 254: Sip Calling Name Manipulations
Mediant 2000 18.3.8 SIP Calling Name Manipulations You can configure manipulation rules for manipulating the calling name (i.e., caller ID) in the SIP message. This can include modifying or removing the calling name. SIP calling name manipulation is applicable to Tel-to-IP and IP-to-Tel calls.
Page 255: Manipulating Number Prefix
SIP User's Manual 18. GW and IP to IP 18.3.10 Manipulating Number Prefix The device supports a notation for adding a prefix where part of the prefix is first extracted from a user-defined location in the original destination or source number. This notation is entered in the 'Prefix to Add' field in the Number Manipulation tables (see ''Manipulation'' on page 241): x[n,l]y...
Page 256: Routing
Mediant 2000 18.4 Routing This section describes the configuration of call routing rules. 18.4.1 Configuring General Routing Parameters The Routing General Parameters page allows you to configure general routing parameters. For a description of these parameters, see ''Configuration Parameters Reference'' on page 409.
Page 257 SIP User's Manual 18. GW and IP to IP  Destination: User-defined IP destination. If the call matches the characteristics, the device routes the call to this destination. If the number dialed does not match the characteristics, the call is not made. The destination can be any of the following: •...
Page 258 Mediant 2000 Notes: When using a proxy server, you do not need to configure this table unless you require one of the following: • Fallback(alternative) routing if communication is lost with the proxy server. • IP security, whereby the device routes only received calls whose source IP addresses are defined in this table.
Page 259 SIP User's Manual 18. GW and IP to IP • A defined Release Reason code is received (see ''Configuring Alternative Routing Reasons'' on page 267). Alternative routing is typically implemented when there is no response to an INVITE message (after INVITE re-transmissions). The device then issues an internal 408 'No Response' implicit Release Reason.
Page 260 Mediant 2000 The previous figure displays the following outbound IP routing rules: • Rule 1 and Rule 2: For both rules, the called phone number prefix is 10, the caller's phone number prefix is 100, and the call is assigned IP Profile ID 1.
Page 261 SIP User's Manual 18. GW and IP to IP Parameter Description Web: Src. Host Prefix Defines the prefix of the SIP Request-URI host name in the From header of EMS: Source Host the incoming SIP INVITE message. If this routing rule is not required, leave Prefix the field empty.
Page 262 Mediant 2000 Parameter Description ''Configuring the Internal DNS Table'' on page 113).  If the string 'ENUM' is specified for the destination IP address, an ENUM query containing the destination phone number is sent to the DNS server. The ENUM reply includes a SIP URI used as the Request-URI in the outgoing INVITE and for routing (if a proxy is not used).
Page 263 SIP User's Manual 18. GW and IP to IP Parameter Description Status Displays the Quality of Service of the destination IP address:  "n/a" = Alternative Routing feature is disabled  "OK" = IP route is available  "Ping Error" = No ping to IP destination; route is unavailable ...
Page 264: Configuring Inbound Ip Routing Table
Mediant 2000 Parameter Description server or a domain name using DNS. In such scenarios, the INVITE is sent to all the queried LDAP or resolved IP addresses respectively. You can also use LDAP routing rules with standard routing rules for Forking Groups.
Page 265 SIP User's Manual 18. GW and IP to IP  To configure inbound IP routing rules: Open the Inbound IP Routing Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing submenu > IP to Trunk Group Routing). Figure 18-16: Inbound IP Routing Table The previous figure displays the following configured routing rules: •...
Page 266 Mediant 2000 Parameter Description Source Host Prefix The From URI host name prefix of the incoming SIP INVITE message. If this routing rule is not required, leave the field empty. Notes:  The asterisk (*) wildcard can be used to depict any prefix.
Page 267: Configuring Alternative Routing Reasons
SIP User's Manual 18. GW and IP to IP 18.4.4 Configuring Alternative Routing Reasons The Reasons for Alternative Routing page allows you to define up to five Release Reason codes for IP-to-Tel and Tel-to-IP call failure reasons. If a call is released as a result of one of these reasons, the device searches for an alternative route for the call.
Page 268: Configuring Call Forward Upon Busy Trunk
Mediant 2000 In the 'IP to Tel Reasons' group, select up to five different call failure reasons that invoke an alternative IP-to-Tel routing. In the 'Tel to IP Reasons' group, select up to five different call failure reasons that invoke an alternative Tel-to-IP routing.
Page 269: Dtmf And Supplementary
SIP User's Manual 18. GW and IP to IP 18.5 DTMF and Supplementary This section describes configuration of the DTMF and supplementary parameters. 18.5.1 Configuring DTMF and Dialing The DTMF & Dialing page is used to configure parameters associated with dual-tone multi- frequency (DTMF) and dialing.
Page 270 Mediant 2000  Emergency 911 calls - see Emergency E911 Phone Number Services on page  Multilevel Precedence and Preemption (MLPP) - see ''Multilevel Precedence and Preemption'' on page The device SIP users are only required to enable the Hold and Transfer features. By default, the Call Forward (supporting 30x redirecting responses) and Call Waiting (receipt of 182 response) features are enabled.
Page 271: Call Hold And Retrieve
SIP User's Manual 18. GW and IP to IP 18.5.2.1 Call Hold and Retrieve Call Hold and Retrieve:  The party that initiates the hold is called the holding party; the other party is called the held party. The device can't initiate Call Hold, but it can respond to hold requests and as such, it's a held party.
Page 272: Call Forward
Mediant 2000 • While speaking to C - transfer from active. The Explicit Call Transfer (ECT, according to ETS-300-367, 368, 369) supplementary service is supported for PRI trunks. This service provides the served user who has two calls to ask the network to connect these two calls together and release its connection to both parties.
Page 273: Message Waiting Indication
SIP User's Manual 18. GW and IP to IP 18.5.2.4 Message Waiting Indication The device supports Message Waiting Indication (MWI) according to IETF RFC 3842, including SUBSCRIBE (to an MWI server). Note: For more information on IP voice mail configuration, refer to the IP Voice Mail CPE Configuration Guide.
Page 274: Emergency E911 Phone Number Services
Mediant 2000 Interrogation request). Some support both these requests. Therefore, the device can be configured to disable this feature, or enable it with one of the following support: • Responds to MWI Activate requests from the PBX by sending SIP NOTIFY MWI messages (i.e., does not send MWI Interrogation messages).
Page 275 SIP User's Manual 18. GW and IP to IP SIPDefaultCallPriority parameter) is used if the incoming SIP INVITE or PRI Setup message contains an invalid priority or Precedence Level value respectively. For each MLPP call priority level, the Multiple Differentiated Services Code Points (DSCP) can be set to a value from 0 to 63.
Page 276: Configuring Voice Mail Parameters
Mediant 2000 The device receives SIP requests with preemption reason cause=5 in the following cases: • The softswitch performs a network preemption of an active call - the following sequence of events occurs: The softswitch sends the device a SIP BYE request with this Reason cause code.
Page 277: Advice Of Charge Services For Euro Isdn
SIP User's Manual 18. GW and IP to IP  To configure the Voice Mail parameters: Open the Voice Mail Settings page (Configuration tab > VoIP menu > GW and IP to IP submenu > Advanced Applications submenu > Voice Mail Settings). Figure 18-21: Voice Mail Settings Page Configure the parameters as required.
Page 278: Dialing Plan Features
Mediant 2000 the charge unit during the call. When the call ends, the devicesends an AoC-E Facility message to the PBX indicating the total number of charged units. To configure AoC: Ensure that the PSTN protocol for the E1 trunk line is Euro ISDN and set to network side.
Page 279 SIP User's Manual 18. GW and IP to IP Table 18-16: Digit Map Pattern Notations Notation Description Range of numbers (not letters). [n-m] (single dot) Repeat digits until next notation (e.g., T). Any single digit. Dial timeout (configured by the TimeBetweenDigits parameter). Short timer (configured by the TimeBetweenDigits parameter;...
Page 280: External Dial Plan File
Mediant 2000 18.6.2 External Dial Plan File The device allows you to select a specific Dial Plan (index) defined in an external Dial Plan file. This file is loaded to the device as a .dat file (binary file), converted from an ini file using the DConvert utility.
Page 281: Modifying Isdn-To-Ip Calling Party Number
SIP User's Manual 18. GW and IP to IP Notes: • If you are using an external Dial Plan file for dialing plans (see ''External Dial Plan File'' on page 280), the device first attempts to locate a matching digit pattern in the Dial Plan file, and if not found, then attempts to locate a matching digit pattern in the Digit Map (configured by the DigitMapping parameter).
Page 282: Dial Plan Prefix Tags For Ip-To-Tel Routing
Mediant 2000 take first Dial Plan rule example above (i.e., "0567811181,0,04343434181"), the received Calling Number Party of 0567811181 is changed to 04343434181 and sent to the IP with a SIP INVITE as follows: Via: SIP/2.0/UDP 211.192.160.214:5060;branch=z9hG4bK3157667347 From: <sip:04343434181@kt.co.kr:5060>;tag=de0004b1 To: sip:01066557573@kt.co.kr:5060 Call-ID: 585e60ec@211.192.160.214...
Page 283 SIP User's Manual 18. GW and IP to IP Define the external Dial Plan file with two routing tags (as shown below): • "LOCL" - for local calls • "LONG" - for long distance calls [ PLAN1 ] 42520[3-5],0,LOCL 425207,0,LOCL 42529,0,LOCL 425200,0,LONG 425100,0,LONG...
Page 284: Configuring Alternative Routing (Based On Connectivity And Qos)
Mediant 2000 18.7 Configuring Alternative Routing (Based on Connectivity and QoS) The Alternative Routing feature enables reliable routing of Tel-to-IP calls when a Proxy isn’t used. The device periodically checks the availability of connectivity and suitable Quality of Service (QoS) before routing. If the expected quality cannot be achieved, an alternative IP route for the prefix (phone number) is selected.
Page 285: Pstn Fallback
SIP User's Manual 18. GW and IP to IP 18.7.3 PSTN Fallback The PSTN Fallback feature enables the device to redirect PSTN originated calls back to the legacy PSTN network if a destination IP route is unsuitable (disallowed) for voice traffic at a specific time.
Page 286 Mediant 2000 User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.6.40.010.006 Supported: 100rel,em Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE, NOTIFY,PRACK,REFER,INFO Content-Type: application/sdp Content-Length: 208 o=AudiocodesGW 18132 74003 IN IP4 10.8.201.108 s=Phone-Call c=IN IP4 10.8.201.108 t=0 0 m=audio 4000 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20  F2 TRYING (10.8.201.161 >> 10.8.201.108): SIP/2.0 100 Trying...
Page 287 F5 ACK (10.8.201.108 >> 10.8.201.10): ACK sip:2000@10.8.201.161;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161>;tag=1c7345 Call-ID: 534366556655skKw-6000--2000@10.8.201.108 User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.6.40.010.006 CSeq: 18153 ACK Supported: 100rel,em Content-Length: 0 Note: Phone ‘6000’ goes on-hook and device 10.8.201.108 sends a BYE to device 10.8.201.161.
Page 288: Sip Message Authentication Example
The REGISTER request is sent to a Registrar/Proxy server for registration: REGISTER sip:10.2.2.222 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200 From: <sip: 122@10.1.1.200>;tag=1c17940 To: <sip: 122@10.1.1.200> Call-ID: 634293194@10.1.1.200 User-Agent: Audiocodes-Sip-Gateway/Mediant 2000/v.6.40.010.006 CSeq: 1 REGISTER Contact: sip:122@10.1.1.200: Expires:3600 Upon receipt of this request, the Registrar/Proxy returns a 401 Unauthorized response: SIP/2.0 401 Unauthorized...
Page 289 At this time, a new REGISTER request is issued with the following response: REGISTER sip:10.2.2.222 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200 From: <sip: 122@10.1.1.200>;tag=1c23940 To: <sip: 122@10.1.1.200> Call-ID: 654982194@10.1.1.200 Server: Audiocodes-Sip-Gateway/Mediant 2000/v.6.40.010.006 CSeq: 1 REGISTER Contact: sip:122@10.1.1.200: Expires:3600 Authorization: Digest, username: 122, realm="audiocodes.com”, nonce="11432d6bce58ddf02e3b5e1c77c010d2",...
Page 290: Trunk-To-Trunk Routing Example
Mediant 2000 18.8.3 Trunk-to-Trunk Routing Example This example describes two devices, each interfacing with the PSTN through four E1 spans. Device A is configured to route all incoming Tel-to-IP calls to Device B. Device B generates calls to the PSTN on the same E1 trunk on which the call was originally received (in Device A).
Page 291 SIP User's Manual 18. GW and IP to IP Scenario: In this example, the Enterprise wishes to connect its TDM PBX to two different ITSPs, by implementing the device in its network environment. It's main objective is for the device to route Tel-to-IP calls to these ITSPs according to a dial plan. The device is to register (on behalf of the PBX) to each ITSP, which implements two servers for redundancy and load balancing.
Page 292 Mediant 2000 In the Proxy Sets Table page (see ''Configuring Proxy Sets Table'' on page 184), configure two Proxy Sets and for each, enable Proxy Keep-Alive (using SIP OPTIONS) and 'round robin' load-balancing method: • Proxy Set #1 includes two IP addresses of the first ITSP (ITSP 1) - 10.33.37.77 and 10.33.37.79 - and using UDP.
Page 293 SIP User's Manual 18. GW and IP to IP In the Trunk Group Table page, enable the Trunks connected between the Enterprise's PBX and the device (Trunk Group ID #1), and between the local PSTN and the device (Trunk Group ID #2). In the Trunk Group Settings page, configure 'Per Account' registration for Trunk Group ID #1 (without serving IP Group) Figure 18-28: Configuring Trunk Group #1 for Registration per Account in Trunk Group...
Page 294: Ip-To-Ip Routing Application
Mediant 2000 18.9 IP-to-IP Routing Application The device's supports IP-to-IP VoIP call routing (or SIP Trunking). The IP-to-IP call routing application enables enterprises to seamlessly connect their IP-based PBX (IP-PBX) to SIP trunks, typically provided by an Internet Telephony Service Provider (ITSP). By implementing the device, enterprises can then communicate with PSTN networks (local and overseas) through ITSP's, which interface directly with the PSTN.
Page 295: Theory Of Operation
SIP User's Manual 18. GW and IP to IP RTP flow - Hold\Retrieve are the only exceptions that traverse the two legs.  ReINVITE: 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. ...
Page 296: Proxy Sets
Mediant 2000 18.9.1.1 Proxy Sets A Proxy Set is a group of up to five Proxy servers (for Proxy load balancing and redundancy), defined by IP address or fully qualified domain name (FQDN). The Proxy Set is assigned to IP Groups (of type SERVER only), representing the address of the IP Group to where the device sends the INVITE message (destination of the call).
Page 297: Inbound And Outbound Ip Routing Rules
SIP User's Manual 18. GW and IP to IP The device also supports the IP-to-IP call routing Survivability mode feature (refer to the figure below) for USER IP Groups. The device records (in its database) REGISTER messages sent by the clients of the USER IP Group. If communication with the Serving IP Group (e.g., IP-PBX) fails, the USER IP Group enters into Survivability mode in which the device uses its database for routing calls between the clients of the USER IP Group.
Page 298: Accounts
Mediant 2000 18.9.1.4 Accounts Accounts are used by the device to register to a Serving IP Group (e.g., an ITSP) on behalf of a Served IP Group (e.g., IP-PBX). This is necessary for ITSP's that require registration to provide services. Accounts are also used for defining user name/password for digest authentication (with or without registration) if required by the ITSP.
Page 299 SIP User's Manual 18. GW and IP to IP  Using SIP trunks, the IP-PBX connects (via the device) to two different ITSP's: • ITSP-A: ♦ Implements Proxy servers with fully qualified domain names (FQDN): "Proxy1.ITSP-A" and "Proxy2.ITSP-B", using TLS. ♦...
Page 300 Mediant 2000 The figure below provides an illustration of this example scenario: Figure 18-36: SIP Trunking Setup Scenario Example The steps for configuring the device according to the scenario above can be summarized as follows:  Enable the IP-to-IP feature (see ''Step 1: Enable the IP-to-IP Capabilities'' on page 301).
Page 301: Step 1: Enable The Ip-To-Ip Capabilities
SIP User's Manual 18. GW and IP to IP  Configure inbound IP routing rules (see ''Step 8: Configure Inbound IP Routing'' on page 310).  Configure outbound IP routing rules (see ''Step 9: Configure Outbound IP Routing'' on page 312). ...
Page 302: Step 3: Define A Trunk Group For The Local Pstn
Mediant 2000 18.9.2.3 Step 3: Define a Trunk Group for the Local PSTN For incoming and outgoing local PSTN calls with the IP-PBX, you need to define the Trunk Group ID (#1) for the T1 ISDN trunk connecting between the device and the local PSTN.
Page 303 SIP User's Manual 18. GW and IP to IP 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 18-40: Proxy Set ID #1 for ITSP-A Configure Proxy Set ID #2 for ITSP-B: From the 'Proxy Set ID' drop-down list, select 2.
Page 304 Mediant 2000 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 18-41: 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.
Page 305: Step 5: Configure The Ip Groups
SIP User's Manual 18. GW and IP to IP 18.9.2.5 Step 5: Configure the IP Groups This step describes how to create the IP Groups for the following entities in the network:  ITSP-A SIP trunk  ITSP-B SIP trunk ...
Page 306: Mediant
Mediant 2000 Contact User = name that is sent in the SIP Request Contact header for this IP Group (e.g., ITSP-B). Figure 18-44: Defining IP Group 2 Define IP Group #3 for the IP-PBX: From the 'Type' drop-down list, select SERVER.
Page 307: Step 6: Configure The Account Table
SIP User's Manual 18. GW and IP to IP Define IP Group #4 for the remote IP-PBX users: From the 'Type' drop-down list, select USER. In the 'Description' field, type an arbitrary name for the IP Group (e.g., IP-PBX). 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).
Page 308 Mediant 2000  To configure the Account table: Open the Account Table page (Configuration tab > VoIP menu > SIP Definitions submenu > Account Table). Figure 18-47: Defining Accounts for Registration Configure Account ID #1 for IP-PBX authentication and registration with ITSP-A: •...
Page 309: Step 7: Configure Ip Profiles For Voice Coders
SIP User's Manual 18. GW and IP to IP 18.9.2.7 Step 7: Configure IP Profiles for Voice Coders Since different voice coders are used by the IP-PBX (G.711) and the ITSP's (G.723), you need to define two IP Profiles:  Profile ID #1 - configured with G.711 for the IP-PBX ...
Page 310: Step 8: Configure Inbound Ip Routing
Mediant 2000 Figure 18-50: 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. 18.9.2.8 Step 8: Configure Inbound IP Routing This step defines how to configure the device for routing inbound (i.e., received) IP-to-IP...
Page 311 SIP User's Manual 18. GW and IP to IP Index #1: routes calls with prefix 9 (i.e., local calls) dialed from IP-PBX users to the local PSTN: • 'Dest Phone Prefix': enter "9" for the dialing prefix for local calls. •...
Page 312: Step 9: Configure Outbound Ip Routing
Mediant 2000 18.9.2.9 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. If the calls are destined to the Japanese market, then they are routed to ITSP-B;...
Page 313 SIP User's Manual 18. GW and IP to IP Index #4: routes IP calls received from the IP-PBX to ITSP-A: • 'Source IP Group ID': select 3 to indicate received (inbound) calls identified as belonging to the IP Group configured for the IP-PBX. •...
Page 314: Step 10: Configure Destination Phone Number Manipulation
Mediant 2000 18.9.2.10 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. This manipulation leaves only the four digits of the user's destination number coming from the ITSP's.
Page 315: Stand-Alone Survivability (Sas) Application
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application Stand-Alone Survivability (SAS) Application This section describes the Sand-Alone Survivability application. 19.1 Overview The device's Stand-Alone Survivability (SAS) feature ensures telephony communication continuity (survivability) for enterprises using hosted IP services (such as IP Centrex) or IP- PBX in cases of failure of these entities.
Page 316: Sas Outbound Mode
Mediant 2000 19.1.1.1 SAS Outbound Mode This section describes the SAS outbound mode, which includes the following states:  Normal state (see ''Normal State'' on page 316)  Emergency state (see ''Emergency State'' on page 316) 19.1.1.1.1 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).
Page 317: Sas Redundant Mode
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application The figure below illustrates the operation of SAS outbound mode in emergency state: Figure 19-2: SAS Outbound Mode in Emergency State (Example) When emergency state is active, SAS continuously attempts to communicate with the external proxy, using keep-alive SIP OPTIONS.
Page 318 Mediant 2000 19.1.1.2.1 Normal State In normal state, the UAs register and operate directly with the external proxy. Figure 19-3: SAS Redundant Mode in Normal State (Example) 19.1.1.2.2 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.
Page 319: Sas Routing
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.1.1.2.3 Exiting Emergency and Returning to Normal State Once the connection with the primary proxy is re-established, the following occurs:  UAs: switch back to operate with the primary proxy.  SAS: ignores REGISTER requests from the UAs, forcing the UAs to switch back to the primary proxy.
Page 320 Mediant 2000 The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the external proxy: Figure 19-6: Flowchart of INVITE from Primary Proxy in SAS Normal State SIP User's Manual Document #: LTRT-68814...
Page 321: Sas Routing In Emergency State
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.1.2.2 SAS Routing in Emergency State The flowchart below shows the routing logic for SAS in emergency state: Figure 19-7: Flowchart for SAS Emergency State Version 6.4 November 2011...
Page 322: Sas Configuration
Mediant 2000 19.2 SAS Configuration SAS supports various configuration possibilities, depending on how the device is deployed in the network and the network architecture requirements. This section provides step-by- step procedures on configuring the SAS application, using the device's Web interface.
Page 323: Configuring Common Sas Parameters
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.2.1.2 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).
Page 324 Mediant 2000 Figure 19-9: Configuring Common Settings In the 'SAS Proxy Set' field, enter the Proxy Set used for SAS. The SAS Proxy Set must be defined only for the following SAS modes: • Outbound mode: In SAS normal state, SAS forwards REGISTER and INVITE messages received from the UAs to the proxy servers defined in this Proxy Set.
Page 325: Configuring Sas Outbound Mode
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 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 19-10: Defining UAs' Proxy Server Click Submit to apply your settings.
Page 326: Configuring Sas Redundant Mode
Mediant 2000 19.2.3 Configuring SAS Redundant Mode This section describes how to configure the SAS redundant mode. These settings are in addition to the ones described in ''Configuring Common SAS Parameters'' on page 323. Note: The VoIP CPEs (such as IP phones or residential gateways) need to be...
Page 327: Gateway With Sas Outbound Mode
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.2.4.1 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 &...
Page 328: Gateway With Sas Redundant Mode
Mediant 2000 Disable use of user=phone in SIP URL: Open the SIP General Parameters page (Configuration tab > VoIP menu > SIP Definitions submenu > General Parameters). From the 'Use user=phone in SIP URL' drop-down list, select No. This instructs the Gateway application not to use user=phone in the SIP URL and therefore, REGISTER and INVITE messages use SIP URI.
Page 329 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 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 323).
Page 330: Advanced Sas Configuration
Mediant 2000 19.2.5 Advanced SAS Configuration This section describes the configuration of advanced SAS features that can be optionally implemented in your SAS deployment:  Manipulating incoming SAS Request-URI user part of REGISTER message (see ''Manipulating URI user part of Incoming REGISTER'' on page 330) ...
Page 331: Manipulating Destination Number Of Incoming Invite
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application After manipulation, SAS registers the user in its database as follows:  AOR: 976653434@10.33.4.226  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) ...
Page 332 Mediant 2000 rules to change the INVITE's destination number so that it matches that of the registered user in the database. This is done using the IP to IP Inbound Manipulation table. For example, in SAS emergency state, assume an incoming INVITE has a destination number "7001234"...
Page 333: Sas Routing Based On Sas Routing Table
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.2.5.3 SAS Routing Based on SAS Routing Table SAS routing based on rules configured in the SAS Routing table is applicable for SAS in the following states:  SAS in normal state, if the SASSurvivabilityMode parameter is set to 4 ...
Page 334 Mediant 2000 Note: The following parameters are not applicable to SAS and should be ignored: Destination IP Group ID, and Alternative Route Options. Table 19-1: SAS IP2IP Routing Table Parameters Parameter Description Matching Characteristics Source Username Prefix The prefix of the user part of the incoming INVITE’s source [IP2IPRouting_SrcUsernamePrefix] URI (usually the From URI).
Page 335 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application Parameter Description overridden and these fields take precedence.  [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.
Page 336: Blocking Calls From Unregistered Sas Users
Mediant 2000 19.2.5.4 Blocking Calls from Unregistered SAS Users To prevent malicious calls (for example, Service Theft), it is recommended to configure the feature for blocking SIP INVITE messages received from SAS users that are not registered in the SAS database. This applies to SAS in normal and emergency states.
Page 337 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application  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).
Page 338: Adding Sip Record-Route Header To Sip Invite
Mediant 2000 19.2.5.6 Adding SIP Record-Route Header to SIP INVITE You can configure SAS to add the SIP Record-Route header to SIP requests (e.g. INVITE) received from enterprise UAs. SAS then sends the request with this header to the proxy.
Page 339: Replacing Contact Header For Sip Messages
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.2.5.7 Replacing Contact Header for SIP Messages You can configure SAS to change the SIP Contact header so that it points to the SAS host. Therefore, this ensures that in the message, the top-most SIP Via header and the Contact header point to the same host.
Page 340: Viewing Registered Sas Users
Mediant 2000 19.3 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 SAS/SBC Registered Users'' on page 388. The maximum number of users that can be registered in the database is 250.
Page 341 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application The figure below illustrates an example of a SAS Cascading call flow configured using the SAS Routing table. In this example, a call is routed from SAS Gateway (A) user to a user on SAS Gateway (B). Figure 19-23: SAS Cascading Using SAS Routing Table - Example ...
Page 342 Mediant 2000 The figure below illustrates an example of a SAS Cascading call flow when configured using the SAS Redundancy feature. In this example, a call is initiated from a SAS Gateway (A) user to a user that is not located on any SAS gateway. The call is subsequently routed to the PSTN.
Page 343: Transcoding Using Third-Party Call Control
SIP User's Manual 20. Transcoding using Third-Party Call Control Transcoding using Third-Party Call Control The device supports transcoding using a third-party call control Application server. This support is provided by the following:  Using RFC 4117 (see ''Using RFC 4117'' on page 343) Note: Transcoding can also be implemented using the IP-to-IP (IP2IP) application.
Page 344 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 345: Part V: Maintenance
Part V Maintenance This part describes the maintenance procedures.
Page 346 Reader’s Notes...
Page 347: Basic Maintenance
SIP User's Manual 21. Basic Maintenance Basic Maintenance The Maintenance Actions page allows you to perform the following:  Reset the device - see ''Resetting the Device'' on page  Lock and unlock the device - see ''Locking and Unlocking the Device'' on page ...
Page 348 Mediant 2000  To reset the device: Open the Maintenance Actions page (see ''Basic Maintenance'' on page 345). Under the 'Reset Configuration' group, from the 'Burn To FLASH' drop-down list, select one of the following options: • Yes: The device's current configuration is saved (burned) to the flash memory prior to reset (default).
Page 349: Locking And Unlocking The Device
SIP User's Manual 21. Basic Maintenance 21.2 Locking and Unlocking the Device The Lock and Unlock options allow you to lock the device so that it doesn't accept any new calls. This is useful when, for example, you are uploading new software files to the device and you don't want any traffic to interfere with the process.
Page 350: Saving Configuration
Mediant 2000 21.3 Saving Configuration The Maintenance Actions page allows you to save (burn) the current parameter configuration (including loaded auxiliary files) to the device's non-volatile memory (i.e., flash). The parameter modifications that you make throughout the Web interface's pages are temporarily saved (to the volatile memory - RAM) when you click the Submit button on these pages.
Page 351: Software Upgrade
SIP User's Manual 22. Software Upgrade Software Upgrade The Software Update menu allows you to upgrade the device's software, install Software Upgrade Key, and load/save configuration file. This menu includes the following page items:  Load Auxiliary Files (see ''Loading Auxiliary Files'' on page 351) ...
Page 352 Mediant 2000 The Auxiliary files can be loaded to the device using one of the following methods:  Web interface.  TFTP: This is done by specifying the name of the Auxiliary file in an ini file (see Auxiliary and Configuration Files Parameters) and then loading the ini file to the device.
Page 353 SIP User's Manual 22. Software Upgrade The procedure below describes how to load Auxiliary files using the Web interface.  To load auxiliary files to the device using the Web interface: Open the Load Auxiliary Files page (Maintenance tab > Software Update menu > Load Auxiliary Files).
Page 354: Call Progress Tones File
Mediant 2000 You can also load auxiliary files using an ini file that is loaded to the device with BootP. Each auxiliary file has a specific ini file parameter that specifies the name of the auxiliary file that you want to load to the device with the ini file. For a description of these ini file parameters, see Auxiliary and Configuration Files Parameters on page 602.
Page 355 SIP User's Manual 22. Software Upgrade The Call Progress Tones section of the ini file comprises the following segments:  [NUMBER OF CALL PROGRESS TONES]: Contains the following key: 'Number of Call Progress Tones' defining the number of Call Progress Tones that are defined in the file.
Page 356: Prerecorded Tones File
Mediant 2000 • Modulation Freq [Hz]: Frequency of the modulated signal for AM tones (valid range from 1 to 128 Hz). • Signal Level [-dBm]: Level of the tone for AM tones. • AM Factor [steps of 0.02]: Amplitude modulation factor (valid range from 1 to 50).
Page 357: Cas Files
SIP User's Manual 22. Software Upgrade The raw data files must be recorded with the following characteristics:  Coders: G.711 A-law or G.711 µ-law  Rate: 8 kHz  Resolution: 8-bit  Channels: mono Once created, the PRT file can then be loaded to the device using AudioCodes' BootP/TFTP utility or the Web interface (see ''Loading Auxiliary Files'' on page 351).
Page 358 Mediant 2000 The Dial Plan file is first created using a text-based editor (such as Notepad) and saved with the file extension .ini. This ini file is then converted to a binary file (.dat) using the DConvert utility (refer to the Product Reference Manual). Once converted, it can then be loaded to the device using the Web interface (see ''Loading Auxiliary Files'' on page 351).
Page 359: User Information File
SIP User's Manual 22. Software Upgrade 22.1.5 User Information File The User Information file is a text-based file that can be used for mapping PBX extensions connected to the device to "global" IP numbers. The User Information file can be loaded to the device by using one of the following methods: ...
Page 360: Amd Sensitivity File
Mediant 2000 An example of a User Information file is shown in the figure below: Figure 22-1: Example of a User Information File Note: The last line in the User Information file must end with a carriage return (i.e., by pressing the <Enter> key).
Page 361 SIP User's Manual 22. Software Upgrade  Parameter Suite 2 with 3 sensitivity levels. <AMDSENSITIVITY> <PARAMETERSUIT> <PARAMETERSUITID>0</PARAMETERSUITID>  <NUMBEROFLEVELS>8</NUMBEROFLEVELS> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>15729</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>58163</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>32742</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>19923</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>50790</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>30720</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>...
Page 362: Loading Software Upgrade Key
Mediant 2000  <AMDCOEFFICIENTA>7340</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>64717</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>3840</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> </PARAMETERSUIT> <PARAMETERSUIT> <PARAMETERSUITID>2</PARAMETERSUITID>  <NUMBEROFLEVELS>3</NUMBEROFLEVELS> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>15729</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>58163</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>32742</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>5243</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>9830</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>24320</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL> ...
Page 363 SIP User's Manual 22. Software Upgrade Note: The Software Upgrade Key is an encrypted key. Each TPM utilizes a unique key.  To load a Software Upgrade Key: Open the Software Upgrade Key Status page (Maintenance tab > Software Update menu >...
Page 364 Mediant 2000 Follow one of the following procedures, depending on whether you are loading a single or multiple key S/N lines: • Single key S/N line: Open the Software Upgrade Key text file (using, for example, Microsoftׂ ◌ Notepad). Select and copy the key string and paste it into the field 'Add a Software Upgrade Key'.
Page 365: Loading Via Bootp/Tftp
SIP User's Manual 22. Software Upgrade 22.2.1 Loading via BootP/TFTP The procedure below describes how to load a Software Upgrade Key to the device using AudioCodes' BootP/TFTP Server utility (for more information on the BootP utility, refer to the Product Reference Manual). ...
Page 366: Software Upgrade Wizard
Mediant 2000 22.3 Software Upgrade Wizard The Software Upgrade Wizard allows you to upgrade the device's firmware (compressed .cmp file) as well as load an ini file and/or auxiliary files (typically loaded using the Load Auxiliary File page described in ''Loading Auxiliary Files'' on page 351). However, it is mandatory when using the wizard to first load a .cmp file to the device.
Page 367 SIP User's Manual 22. Software Upgrade  To load files using the Software Upgrade Wizard: Stop all traffic on the device using the Graceful Lock feature (refer to the warning bulletin above). Open the Software Upgrade wizard, by performing one of the following: •...
Page 368 Mediant 2000 Click the Next button; the wizard page for loading an ini file appears. You can now perform one of the following: • Load a new ini file: Click Browse, navigate to the ini file, and then click Send File;...
Page 369: Backing Up And Loading Configuration File
SIP User's Manual 22. Software Upgrade 22.4 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 PC, using the 'Configuration File page. The saved ini file includes only parameters that were modified and parameters with other than default values.
Page 370 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 371: Restoring Factory Defaults
SIP User's Manual 23. Restoring Factory Defaults Restoring Factory Defaults You can restore the device's configuration to factory defaults using one of the following methods:  Using the CLI (see ''Restoring Defaults using CLI'' on page 371)  Loading an empty ini file (see ''Restoring Defaults using an ini File'' on page 372) 23.1 Restoring Defaults using CLI The device can be restored to factory defaults using CLI, as described in the procedure...
Page 372: Restoring Defaults Using An Ini File
Mediant 2000 23.2 Restoring Defaults using an ini File You can restore the device to factory default settings by loading an empty ini file to the device, using the Web interface's Configuration File page (see ''Backing Up and Loading Configuration File'' on page 369). The only settings that are not restored to default are the management (OAMP) LAN IP address and the Web interface's login user name and password.
Page 373: Part Vi: Status, Performance Monitoring And Reporting
Part VI Status, Performance Monitoring and Reporting This part describes the status and performance monitoring procedures.
Page 374 Reader’s Notes...
Page 375: System Status
SIP User's Manual 24. System Status System Status This section describes how to view system status.  Syslog messages - see Viewing Syslog Messages on page  Device information - see ''Viewing Device Information'' on page  Ethernet port information - see ''Viewing Ethernet Port Information'' on page 24.1 Viewing Device Information The Device Information page displays the device's specific hardware and software product...
Page 376: Viewing Ethernet Port Information
Mediant 2000 24.2 Viewing Ethernet Port Information The Ethernet Port Information page displays read-only information on the Ethernet port connections. This includes information such as activity status, duplex mode, and speed. Notes: • The Ethernet Port Information page can also be accessed from the Home page (see ''Using the Home Page'' on page 51).
Page 377: Carrier-Grade Alarms
SIP User's Manual 25. Carrier-Grade Alarms Carrier-Grade Alarms This section describes how to view the following types of alarms:  Active alarms - see ''Viewing Active Alarms'' on page  Alarm history - see ''Viewing Alarm History'' on page 25.1 Viewing Active Alarms The Active Alarms page displays a list of currently active alarms.
Page 378: Viewing Alarm History
Mediant 2000 25.2 Viewing Alarm History The Alarms History page displays a list of alarms that have been raised and traps that have been cleared.  To view the list of history alarms:  Open the Alarms History page (Status & Diagnostics tab > System Status menu >...
Page 379: Performance Monitoring
SIP User's Manual 26. Performance Monitoring Performance Monitoring This section describes how to view the following performance monitoring graphs:  Trunk Utilization - see ''Viewing Trunk Utilization'' on page  MOS per Media Realm - see ''Viewing MOS per Media Realm'' on page 26.1 Viewing Trunk Utilization The Trunk Utilization page provides an X-Y graph that displays the number of active...
Page 380 Mediant 2000 For more graph functionality, see the following table: Table 26-1: Additional Graph Functionality for Trunk Utilization Button Description Add button Displays additional trunks in the graph. Up to five trunks can be displayed simultaneously in the graph. To view another trunk, click this button and then from the new 'Trunk' drop-down list, select the required trunk.
Page 381: Viewing Mos Per Media Realm
SIP User's Manual 26. Performance Monitoring 26.2 Viewing MOS per Media Realm The MOS Per Media Realm page displays statistics on Media Realms (configured in ''Configuring Media Realms'' on page 159). This page provides two graphs:  Upper graph: displays the Mean Opinion Score (MOS) quality in RTCP data per selected Media Realm.
Page 382 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 383: Voip Status
SIP User's Manual 27. VoIP Status VoIP Status This section describes how to view the following VoIP status and statistics:  Trunks and channels - see Viewing Trunks & Channels Status on page  IP network interface - see ''Viewing Active IP Interfaces'' on page ...
Page 384 Mediant 2000  To view the next eight trunks:  Click the Go To Page icon. Figure 27-1: Example of a Selected Page Icon for Displaying Trunks 17-24 The Trunks and Channels Status page uses the following color-coding icons to indicate the...
Page 385: Viewing Active Ip Interfaces
SIP User's Manual 27. VoIP Status 27.2 Viewing Active IP Interfaces The IP Interface Status page displays the device's active IP interfaces, which are configured in the Multiple Interface Table page (see ''Configuring IP Interface Settings'' on page 92).  To view the Active IP Interfaces page: ...
Page 386: Viewing Call Counters
Mediant 2000 27.4 Viewing Call Counters The IP to Tel Calls Count page and Tel to IP Calls Count page provide you with statistical information on incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. The statistical information is updated according to the release reason that is received after a call is terminated (during the same time as the end-of-call Call Detail Record or CDR message is sent).
Page 387 SIP User's Manual 27. VoIP Status Counter Description GWAPP_NORMAL_CALL_CLEAR increments the 'Number of Failed Calls due to No Answer' counter. The rest of the release reasons increment the 'Number of Failed Calls due to Other Failures' counter. Percentage of The percentage of established calls from attempted calls. Successful Calls (ASR) Indicates the number of calls that failed as a result of a busy line.
Page 388: Viewing Sas/Sbc Registered Users
Mediant 2000 27.5 Viewing SAS/SBC Registered Users The SAS/SBC Registered Users page displays a list of registered SAS users recorded in the device's database.  To view registered users:  Open the SAS/SBC Registered Users page (Status & Diagnostics tab > VoIP Status menu >...
Page 389: Viewing Call Routing Status
SIP User's Manual 27. VoIP Status 27.6 Viewing Call Routing Status The Call Routing Status page provides you with information on the current routing method used by the device. This information includes the IP address and FQDN (if used) of the Proxy server with which the device currently operates.
Page 390: Viewing Ip Connectivity
Mediant 2000 27.7 Viewing IP Connectivity The IP Connectivity page displays online, 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 256). Notes: •...
Page 391 SIP User's Manual 27. VoIP Status Column Name Description  Init = Connectivity queries not started (e.g., IP address not resolved).  Disable = The connectivity option is disabled, i.e., parameter 'Alt Routing Tel to IP Mode' (AltRoutingTel2IPMode ini) is set to 'None' or 'QoS'. Quality Status Determines the QoS (according to packet loss and delay) of the IP address.
Page 392 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 393: Reporting Information To External Party
SIP User's Manual 28. Reporting Information to External Party Reporting Information to External Party 28.1 Generating Call Detail Records The Call Detail Record (CDR) contains vital statistic information on calls made from the device. CDRs are generated at the end and optionally, at the beginning of each call (defined by the CDRReportLevel parameter).
Page 394 Mediant 2000 Field Name Description TrmSd Initiator of call release (IP, Tel, or Unknown) TrmReason Termination reason (see ''Release Reasons in CDR'' on page 395) Fax transaction during call InPackets Number of incoming packets Number of outgoing packets OutPackets PackLoss...
Page 395: Release Reasons In Cdr
SIP User's Manual 28. Reporting Information to External Party Field Name Description LocalMosCQ Local MOS for conversation quality RemoteMosCQ Remote MOS for conversation quality SourcePort Source RTP port DestPort Destination RTP port 28.1.2 Release Reasons in CDR The possible reasons for call termination which is represented in the CDR field TrmReason are listed below: ...
Page 396 Mediant 2000  "PREEMPTION_CIRCUIT_RESERVED_FOR_REUSE"  "GWAPP_NORMAL_CALL_CLEAR"  "GWAPP_USER_BUSY"  "GWAPP_NO_USER_RESPONDING"  "GWAPP_NO_ANSWER_FROM_USER_ALERTED"  "MFCR2_ACCEPT_CALL"  "GWAPP_CALL_REJECTED"  "GWAPP_NUMBER_CHANGED"  "GWAPP_NON_SELECTED_USER_CLEARING"  "GWAPP_INVALID_NUMBER_FORMAT"  "GWAPP_FACILITY_REJECT"  "GWAPP_RESPONSE_TO_STATUS_ENQUIRY"  "GWAPP_NORMAL_UNSPECIFIED"  "GWAPP_CIRCUIT_CONGESTION"  "GWAPP_USER_CONGESTION"  "GWAPP_NO_CIRCUIT_AVAILABLE"  "GWAPP_NETWORK_OUT_OF_ORDER"  "GWAPP_NETWORK_TEMPORARY_FAILURE"  "GWAPP_NETWORK_CONGESTION"...
Page 397: Supported Radius Attributes
SIP User's Manual 28. Reporting Information to External Party  "GWAPP_NO_CALL_SUSPENDED"  "GWAPP_CALL_HAVING_CALL_ID_CLEARED"  "GWAPP_INCOMPATIBLE_DESTINATION"  "GWAPP_INVALID_TRANSIT_NETWORK_SELECTION"  "GWAPP_INVALID_MESSAGE_UNSPECIFIED"  "GWAPP_NOT_CUG_MEMBER"  "GWAPP_CUG_NON_EXISTENT"  "GWAPP_MANDATORY_IE_MISSING"  "GWAPP_MESSAGE_TYPE_NON_EXISTENT"  "GWAPP_MESSAGE_STATE_INCONSISTENCY"  "GWAPP_NON_EXISTENT_IE"  "GWAPP_INVALID_IE_CONTENT"  "GWAPP_MESSAGE_NOT_COMPATIBLE"  "GWAPP_RECOVERY_ON_TIMER_EXPIRY"  "GWAPP_PROTOCOL_ERROR_UNSPECIFIED"  "GWAPP_INTERWORKING_UNSPECIFIED"...
Page 398 Mediant 2000 Attribute Attribute Value Purpose Example Number Name Format The call’s originator: H323-Call- Answer, Start Acc Answering (IP) or String Origin Originate etc Stop Acc Originator (PSTN) H323-Call- Protocol type or family Start Acc String VoIP Type used on this leg of the call...
Page 399 SIP User's Manual 28. Reporting Information to External Party Attribute Attribute Value Purpose Example Number Name Format during the call Physical port type of Start Acc device on which the call is String Asynchronous Stop Acc active Response Attributes The reason for failing H323-Return- 0 Request authentication (0 = ok,...
Page 400: Event Notification Using X-Detect Header
Mediant 2000 28.2 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. Event detection and notifications is performed using the SIP X-Detect message header and only when establishing a SIP dialog.
Page 401 SIP User's Manual 28. Reporting Information to External Party Table 28-4: Special Information Tones (SITs) Reported by the device Special Description First Tone Second Tone Third Tone Information Frequency Frequency Frequency Tones (SITs) Duration Duration Duration Name (Hz) (ms) (Hz) (ms) (Hz) (ms)
Page 402: Querying Device Channel Resources Using Sip Options
Mediant 2000 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 Max-Forwards: 70 From: "anonymous" <sip:anonymous@anonymous.invalid>;tag=1c25298 To: <sip:101@10.33.2.53;user=phone> Call-ID: 11923@10.33.2.53 CSeq: 1 INVITE Contact: <sip:100@10.33.2.53> X- Detect: Request=CPT,FAX SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906 From: "anonymous"...
Page 403: Part Vii: Diagnostics And Troubleshooting
Part VII Diagnostics and Troubleshooting This part describes the diagnostics procedures.
Page 404 Reader’s Notes...
Page 405: Configuring Syslog Settings
SIP User's Manual 29. Configuring Syslog Settings Configuring Syslog Settings The Syslog Settings page allows you to configure the device's embedded Syslog client. For a detailed description on the Syslog parameters, see ''Syslog, CDR and Debug Parameters'' on page 430. For viewing Syslog messages in the Web interface, see Viewing Syslog Messages on page 407.
Page 406 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 407: Viewing Syslog Messages
SIP User's Manual 30. Viewing Syslog Messages Viewing Syslog Messages The Message Log page displays Syslog debug messages sent by the device. You can select the Syslog messages in this page, and then copy and paste them into a text editor such as Notepad.
Page 408 Mediant 2000  To stop the Message Log:  Close the 'Message Log page by accessing any another page in the Web interface. SIP User's Manual Document #: LTRT-68814...
Page 409: Part Viii: Appendices
Part VIII Appendices This part includes various appendices.
Page 410 Reader’s Notes...
Page 411: A Configuration Parameters Reference
SIP User's Manual A. Configuration Parameters Reference Configuration Parameters Reference The device's configuration parameters, default values, and their descriptions are documented in this section. Parameters and values enclosed in square brackets ([...]) represent the ini file parameters and their enumeration values; parameters not enclosed in square brackets represent their corresponding Web interface and/or EMS parameters.
Page 412: Mediant
Mediant 2000 Parameter Description InterfaceTable_PrimaryDNSServerIPAddress, InterfaceTable_SecondaryDNSServerIPAddress, InterfaceTable_UnderlyingInterface; [\InterfaceTable] For example: InterfaceTable 0 = 0, 0, 192.168.85.14, 16, 0.0.0.0, 1, Management; InterfaceTable 1 = 2, 0, 200.200.85.14, 24, 0.0.0.0, 200, Control; InterfaceTable 2 = 1, 0, 211.211.85.14, 24, 211.211.85.1, 211, Media;...
Page 413 SIP User's Manual A. Configuration Parameters Reference Parameter Description [LocalOAMSubnetMask] Interface table. The default subnet mask is 0.0.0.0. Note: For this parameter to take effect, a device reset is required. Web: Default Gateway Address Defines the Default Gateway of the OAMP interface when EMS: Local Def GW operating in a single interface scenario without a Multiple [LocalOAMDefaultGW]...
Page 414: Static Routing Parameters
Mediant 2000 A.1.3 Static Routing Parameters The static routing parameters are described in the table below. Table A-3: Static Routing Parameters Parameter Description Static IP Routing Table Web/EMS: IP Routing Defines up to 30 static IP routing rules for the device. These rules can be...
Page 415 SIP User's Manual A. Configuration Parameters Reference Table A-4: QoS Parameters Parameter Description Layer-2 Class Of Service (CoS) Parameters (VLAN Tag Priority Field) Web: Network Priority Defines the VLAN priority (IEEE 802.1p) for Network Class of EMS: Network Service Class Priority Service (CoS) content.
Page 416: Nat And Stun Parameters
Mediant 2000 Parameter Description Web: Gold QoS Defines the DiffServ value for the Gold CoS content EMS: Gold Service Class Diff Serv (Streaming applications). [GoldServiceClassDiffServ] The valid range is 0 to 63. The default value is 26. Web: Bronze QoS...
Page 417 SIP User's Manual A. Configuration Parameters Reference Parameter Description  Use either the STUNServerPrimaryIP or the STUNServerDomainName parameter, with priority to the first one. NAT Parameters Web/EMS: NAT Traversal Enables the NAT mechanism. [DisableNAT]  [0] Enable  [1] Disable (default) Note: The compare operation that is performed on the IP address is enabled by default and is configured by the parameter EnableIPAddrTranslation.
Page 418: Nfs Parameters
Mediant 2000 Parameter Description DisableNAT to 0 and the parameter EnableIpAddrTranslation to A.1.6 NFS Parameters The Network File Systems (NFS) configuration parameters are described in the table below. Table A-6: NFS Parameters Parameter Description Defines the start of the range of numbers used for local UDP ports used [NFSBasePort] by the NFS client.
Page 419: Dns Parameters
SIP User's Manual A. Configuration Parameters Reference A.1.7 DNS Parameters The Domain name System (DNS) parameters are described in the table below. Table A-7: DNS Parameters Parameter Description Internal DNS Table Web: Internal DNS Table This parameter table defines the internal DNS table for resolving host EMS: DNS Information names into IP addresses.
Page 420: Dhcp Parameters
Mediant 2000 Parameter Description  For configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page 74. A.1.8 DHCP Parameters The Dynamic Host Control Protocol (DHCP) parameters are described in the table below. Table A-8: DHCP Parameters...
Page 421: Ntp And Daylight Saving Time Parameters
SIP User's Manual A. Configuration Parameters Reference A.1.9 NTP and Daylight Saving Time Parameters The Network Time Protocol (NTP) and daylight saving time parameters are described in the table below. Table A-9: 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 86.
Page 422: Management Parameters
Mediant 2000 Management Parameters This subsection describes the device's Web and Telnet parameters. A.2.1 General Parameters The general management parameters are described in the table below. Table A-10: General Management Parameters Parameter Description Web: Web and Telnet Defines up to ten IP addresses that are permitted to access the device's Access List Table Web interface and Telnet interfaces.
Page 423: Web Parameters
SIP User's Manual A. Configuration Parameters Reference A.2.2 Web Parameters The Web parameters are described in the table below. Table A-11: Web Parameters Parameter Description Web: Deny Access On Fail Count Defines the maximum number of login attempts after which the [DenyAccessOnFailCount] requesting IP address is blocked.
Page 424 Mediant 2000 Parameter Description addition, the following pages can't be accessed: 'Web User Accounts', 'Certificates', 'Regional Settings', 'Maintenance Actions' and all file-loading pages ('Load Auxiliary Files', 'Software Upgrade Wizard', and 'Configuration File'). Notes:  For this parameter to take effect, a device reset is required.
Page 425: Telnet Parameters
SIP User's Manual A. Configuration Parameters Reference A.2.3 Telnet Parameters The Telnet parameters are described in the table below. Table A-12: Telnet Parameters Parameter Description Web: Embedded Telnet Server Enables the device's embedded Telnet server. Telnet is disabled by EMS: Server Enable default for security.
Page 426 Mediant 2000 Parameter Description EMS: Keep Alive Trap Port Defines the port to which keep-alive traps are sent. [KeepAliveTrapPort] The valid range is 0 - 65534. The default is port 162. [SendKeepAliveTrap] Enables keep-alive traps and sends them every 9/10 of the time as defined by the NATBindingDefaultTimeout parameter.
Page 427 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: Address Manager. The device sends SNMP traps to this IP address. [SNMPManagerTableIP_x] Enter the IP address in dotted-decimal notation, e.g., 108.10.1.255. Web: Trap Port Defines the port number of the remote SNMP Manager. The device EMS: Port sends SNMP traps to this port.
Page 428: Serial Parameters
Mediant 2000 Parameter Description  For configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page A.2.5 Serial Parameters The RS-232 serial parameters are described in the table below. Table A-14: Serial Parameters Parameter Description [DisableRS232] Enables the device's RS-232 (serial) port.
Page 429: Debugging And Diagnostics Parameters
SIP User's Manual A. Configuration Parameters Reference Debugging and Diagnostics Parameters This subsection describes the device's debugging and diagnostic parameters. A.3.1 General Parameters The general debugging and diagnostic parameters are described in the table below. Table A-15: General Debugging and Diagnostic Parameters Parameter Description EMS: Enable Diagnostics...
Page 430: Syslog, Cdr And Debug Parameters
Mediant 2000 A.3.2 Syslog, CDR and Debug Parameters The Syslog, CDR and debug parameters are described in the table below. Table A-16: Syslog, CDR and Debug Parameters Parameter Description Web: Enable Syslog Determines whether the device sends logs and error messages EMS: Syslog enable generated by the device to a Syslog server.
Page 431 SIP User's Manual A. Configuration Parameters Reference Parameter Description connection and at the end of each call.  [4] Start & End & Connect Call = CDR report is sent to Syslog at the start, at connection, and at the end of each call. Notes: ...
Page 432 Mediant 2000 Parameter Description Messages operations according to the below user-defined filters. [ActivityListToLog]  [pvc] Parameters Value Change = Changes made on-the-fly to parameters.  [afl] Auxiliary Files Loading = Loading of auxiliary files.  [dr] Device Reset = Reset of device via the 'Maintenance Actions page.
Page 433: Resource Allocation Indication Parameters
SIP User's Manual A. Configuration Parameters Reference A.3.3 Resource Allocation Indication Parameters The Resource Allocation Indication (RAI) parameters are described in the table below. Table A-17: RAI Parameters Parameter Description [EnableRAI] Enables RAI alarm generation if the device's busy endpoints exceed a user-defined threshold.
Page 434 Mediant 2000 Parameter Description during start-up. The device stops after all packets are sent there's sending BootP requests when still no reply, the device loads from either BootP reply is received or flash. number of retries is reached.  [1] = 4 DHCP packets ...
Page 435: Security Parameters
SIP User's Manual A. Configuration Parameters Reference Security Parameters This subsection describes the device's security parameters. A.4.1 General Parameters The general security parameters are described in the table below. Table A-19: General Security Parameters Parameter Description Web: Voice Menu Defines the password for accessing the device's voice menu, used for Password configuring and monitoring the device.
Page 436: Https Parameters
Mediant 2000 Parameter Description AccessList_Allow_Type; [\AccessList] For example: AccessList 10 = mgmt.customer.com, , , 32, 0, 80, tcp, 1, OAMP, 0, 0, 0, allow; AccessList 22 = 10.4.0.0, , , 16, 4000, 9000, any, 0, , 0, 0, 0, block;...
Page 437 SIP User's Manual A. Configuration Parameters Reference Parameter Description to ‘RC4:EXP’, enabling RC-128bit encryption.  The value ‘ALL’ can be configured only if the “Strong Encryption” Software Upgrade Key is enabled. Web: HTTP Authentication Mode Determines the authentication mode used for the Web interface. EMS: Web Authentication Mode ...
Page 438: Srtp Parameters
Mediant 2000 A.4.3 SRTP Parameters The Secure Real-Time Transport Protocol (SRTP) parameters are described in the table below. Table A-21: SRTP Parameters Parameter Description Web: Media Security Enables Secure Real-Time Transport Protocol (SRTP). EMS: Enable Media Security  [0] Disable = SRTP is disabled (default).
Page 439 SIP User's Manual A. Configuration Parameters Reference Parameter Description a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:TAaxNnQt8/qLQMnDuG4vxYfWl6K7eBK/ufk04pR4|2^ 31|1:1 a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:bnuYZnMxSfUiGitviWJZmzr7OF3AiRO0l5Vnh0kH|2^ The first crypto line includes the MKI parameter "1:1". In the 200 OK response, the device selects one of the crypto lines (i.e., '2' or '3').
Page 440: Tls Parameters
Mediant 2000 A.4.4 TLS Parameters The Transport Layer Security (TLS) parameters are described in the table below. Table A-22: TLS Parameters Parameter Description Web/EMS: TLS Version Determines the supported versions of SSL/TLS (Secure Socket Layer/Transport Layer Security. [TLSVersion]  [0] SSL 2.0-3.0 and TLS 1.0 = SSL 2.0, SSL 3.0, and TLS 1.0 are supported (default).
Page 441: Ssh Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description the parameter TLSRemoteSubjectName. If a match is found, the connection is established. Otherwise, the connection is terminated. Web: TLS Client Verify Server Determines whether the device, when acting as a client for TLS Certificate connections, verifies the Server certificate.
Page 442: Ipsec Parameters
Mediant 2000 Parameter Description [SSHServerPort] Range is any valid port number. The default port is 22. Web: SSH Admin Key Defines the RSA public key for strong authentication for logging in [SSHAdminKey] to the SSH interface (if enabled). The value should be a base64-encoded string. The value can be a maximum length of 511 characters.
Page 443 SIP User's Manual A. Configuration Parameters Reference Parameter Description configure the Internet Key Exchange (IKE) and IP Security (IPSec) protocols. You can define up to 20 IPSec peers. The format of this parameter is as follows: [ IPsecSATable ] FORMAT IPsecSATable_Index = IPsecSATable_RemoteEndpointAddressOrName, IPsecSATable_AuthenticationMethod, IPsecSATable_SharedKey, IPsecSATable_SourcePort, IPsecSATable_DestPort,...
Page 444: Ocsp Parameters
Mediant 2000 Parameter Description  To support more than one Encryption / Authentication / DH Group proposal, for each proposal specify the relevant parameters in the Format line.  The proposal list must be contiguous.  For a detailed description of this table and to configure the table using the Web interface, see ''Configuring IP Security Proposal Table'' on page 126.
Page 445 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [1] Enable = RADIUS application is enabled. Note: For this parameter to take effect, a device reset is required. Web: Accounting Server IP Defines the IP address of the RADIUS accounting server. Address [RADIUSAccServerIP] Web: Accounting Port...
Page 446: Sip Media Realm Parameters
Mediant 2000 Parameter Description timeout doesn't reset, instead it continues decreasing.  [1] Reset Timer Upon Access = upon each access to a Web page, the timeout always resets (reverts to the initial value configured by RadiusLocalCacheTimeout). Web: Local RADIUS Password...
Page 447: Control Network Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description assigned to both an IP Group and SRD, the IP Group’s Media Realm takes precedence.  For a detailed description of all the parameters included in this ini file table parameter and for configuring Media Realms using the Web interface, see ''Configuring Media Realms'' on page 159.
Page 448 Mediant 2000 Parameter Description  For configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page 74. Account Table Web: Account Table This parameter table configures the Account table for EMS: SIP Endpoints > Account registering and/or authenticating (digest) Trunk Groups or IP [Account] Groups (e.g., an IP-PBX) to a Serving IP Group (e.g., an...
Page 449 SIP User's Manual A. Configuration Parameters Reference Parameter Description (now active) Proxy until the next failure, after which it works with the next redundant Proxy (default).  [1] Homing = device always tries to work with the primary Proxy server (i.e., switches back to the primary Proxy whenever it's available).
Page 450 Mediant 2000 Parameter Description  When this parameter is set to [1] and the INVITE sent to the Proxy fails, the device re-routes the call according to the Standard mode [0].  When this parameter is set to [2] and the INVITE fails, the device re-routes the call according to the Standard mode [0].
Page 451 SIP User's Manual A. Configuration Parameters Reference Parameter Description host name (according to the received weights) to locate up to four Proxy IP addresses. Therefore, if the first SRV query returns two domain names and the A-record queries return two IP addresses each, no additional searches are performed.
Page 452 Mediant 2000 Parameter Description Web/EMS: Cnonce Defines the Cnonce string used by the SIP server and client to [Cnonce] provide mutual authentication. The value is free format, i.e., 'Cnonce = 0a4f113b'. The default is 'Default_Cnonce'. Web/EMS: Mutual Authentication Determines the device's mode of operation when...
Page 453 SIP User's Manual A. Configuration Parameters Reference Parameter Description page 184.  For configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page 74. Proxy Set Table Web: Proxy Set Table This parameter table configures the Proxy Set ID table. It is EMS: Proxy Set used in conjunction with the ProxyIP ini file table parameter, [ProxySet]...
Page 454 Mediant 2000 Parameter Description Web: Registrar IP Address Defines the IP address (or FQDN) and port number (optional) EMS: Registrar IP of the Registrar server. The IP address is in dotted-decimal [RegistrarIP] notation, e.g., 201.10.8.1:<5080>. Notes:  If not specified, the REGISTER request is sent to the primary Proxy server.
Page 455 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Registration Time Threshold Defines a threshold (in seconds) for re-registration timing. If EMS: Time Threshold this parameter is greater than 0, but lower than the computed [RegistrationTimeThreshold] re-registration timing (according to the parameter RegistrationTimeDivider), the re-registration timing is set to the following: timing set by the Registration server in the SIP Expires header minus the value of the parameter...
Page 456 Mediant 2000 Parameter Description performed separately for each B-channel.  [1] Per Gateway = Single registration and authentication for the entire device (default). This is typically used for and digital modules. Web: Set Out-Of-Service On Enables setting a , trunk, or the entire device (i.e., all Registration Failure endpoints) to out-of-service if registration fails.
Page 457 SIP User's Manual A. Configuration Parameters Reference Parameter Description network  nonce - set to an empty value  response - set to an empty value For example: 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.
Page 458: Network Application Parameters
Mediant 2000 Parameter Description sequences. [PingPongKeepAliveTime] Defines the periodic interval (in seconds) after which a “ping” (double-CRLF) keep-alive is sent to a proxy/registrar, using the CRLF Keep-Alive mechanism. The default range is 5 to 2,000,000. The default is 120. The device uses the range of 80-100% of this user-defined value as the actual interval.
Page 459 SIP User's Manual A. Configuration Parameters Reference Parameter Description SIPInterface 0 = Voice, 2, 5060, 5060, 5061, 1; SIPInterface 1 = Voice, 2, 5070, 5070, 5071, 2; SIPInterface 2 = Voice, 0, 5090, 5000, 5081, 2; Notes:  This table can include up to 32 indices (where 0 is the first index). ...
Page 460: General Sip Parameters
Mediant 2000 Parameter Description for all interfaces. Uses the NATTranslation parameter to define NAT per interface.  If NAT is not configured (by any of the above-mentioned methods), the device sends the packet according to its IP address defined in the Multiple Interface table.
Page 461 SIP User's Manual A. Configuration Parameters Reference Parameter Description IP calls using cause 805. Web: QoS statistics in SIP Enables the device to include call quality of service (QoS) statistics in Release Call SIP BYE and SIP 200 OK response to BYE, using the proprietary SIP [QoSStatistics] header X-RTP-Stat.
Page 462 Mediant 2000 Parameter Description Web/EMS: Enable Early Enables the device to send a 18x response with SDP instead of a Media 18x, allowing the media stream to be established prior to the [EnableEarlyMedia] answering of the call.  [0] Disable = Early Media is disabled (default).
Page 463 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: 183 Message Behavior Defines the ISDN message that is sent when the 183 Session EMS: SIP 183 Behaviour Progress message is received for IP-to-Tel calls. [SIP183Behaviour]  [0] Progress = The device sends a Progress message. (default). ...
Page 464 Mediant 2000 Parameter Description EMS: Options User Part Defines the user part value of the Request-URI for outgoing SIP [OPTIONSUserPart] OPTIONS requests. If no value is configured, the configuration parameter ‘Username’ value is used. A special value is ‘empty’, indicating that no user part in the Request- URI (host part only) is used.
Page 465 SIP User's Manual A. Configuration Parameters Reference Parameter Description IPProfile parameter).  For more information on fax transport methods, see ''Fax/Modem Transport Modes'' on page 135. [HandleG711asVBD] Enables the handling of G.711 as G.711 VBD coder.  [0] = Disable (default). The device negotiates G.711 as a regular audio coder and sends an answer only with G.729 coder.
Page 466 Mediant 2000 Parameter Description  The device supports up to 100 simultaneous TLS sessions. Web: SIP UDP Local Port Defines the local UDP port for SIP messages. EMS: Local SIP Port The valid range is 1 to 65534. The default value is 5060.
Page 467 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: Destination Port The valid range is 1 to 65534. The default port is 5060. [SIPDestinationPort] Note: SIP responses are sent to the port specified in the Via header. Web: Use user=phone in SIP Determines whether the 'user=phone' string is added to the SIP URI and SIP To header.
Page 468 Mediant 2000 Parameter Description 486 - Busy Here Call Forward Busy (CFB) 600 - Busy Everywhere  If history reason is a Q.850 reason, it is translated to the SIP reason (according to the SIP-ISDN tables) and then to ISDN Redirect reason according to the table above.
Page 469 SIP User's Manual A. Configuration Parameters Reference Parameter Description - The device interworks ISDN Setup with an Off Hook Indicator of “Voice” to SIP INVITE with “tgrp=hotline;trunk- context=dsn.mil” in the Contact header. - The device interworks ISDN Setup with an Off Hook indicator of “Data”...
Page 470 Mediant 2000 Parameter Description  [0] Disable (default)  [1] Enable When this parameter is enabled, if the Request-URI in the received SIP INVITE includes the 'dtg' parameter, the device routes the call to the Trunk Group according to its value. This parameter is used instead of the 'tgrp/trunk-context' parameters.
Page 471 SIP User's Manual A. Configuration Parameters Reference Parameter Description REGISTER responses contain the “gruu” parameter in each Contact header. This parameter contains a SIP or SIPS URI that represents a GRUU corresponding to the UA instance that registered the contact. The server provides the same GRUU for the same AOR and instance-id when sending REGISTER again after registration expiration.
Page 472 Mediant 2000 Parameter Description increment the origin value even in scenarios where they want to re- negotiate, the device can assume that the remote party operates according to RFC 3264, and in cases where the origin field is not incremented, the device does not re-negotiate SDP capabilities.
Page 473 SIP User's Manual A. Configuration Parameters Reference Parameter Description unavailable to the requesting SIP client. The device maintains a list of available proxies, by using the Keep-Alive mechanism. The device checks the availability of proxies by sending SIP OPTIONS every keep-alive timeout to all proxies.
Page 474 Mediant 2000 Parameter Description [ForkingHandlingMode] response with a different SIP to-tag than the previous 18x response. These responses are typically generated (initiated) by Proxy / Application servers that perform call forking, sending the device's originating INVITE (received from SIP clients) to several destinations, using the same CallID.
Page 475 SIP User's Manual A. Configuration Parameters Reference Parameter Description [ZeroSDPHandling] Determines the device's response to an incoming SDP that includes an IP address of 0.0.0.0 in the SDP's Connection Information field (i.e., "c=IN IP4 0.0.0.0").  [0] = Sets the IP address of the outgoing SDP's c= field to 0.0.0.0 (default).
Page 476 Mediant 2000 Parameter Description [EnableUserInfoUsage] Information, see ''Loading Auxiliary Files'' on page 351.)  [0] Disable (default).  [1] Enable [HandleReasonHeader] 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.
Page 477 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [1] PI =1, [8] PI =8: For IP-to-Tel calls, if the parameter EnableEarlyMedia is set to 1, the device sends 180 Ringing with SDP in response to an ISDN Alerting or it sends a 183 Session Progress message with SDP in response to only the first received ISDN Proceeding or Progress message after a call is placed to PBX/PSTN over the trunk.
Page 478 Mediant 2000 Parameter Description  For this parameter to take effect, a device reset is required.  For more information on transcoding, see Transcoding using Third-Party Call Control on page 342. Web/EMS: Default Release Defines the default Release Cause (sent to IP) for IP-to-Tel calls...
Page 479 SIP User's Manual A. Configuration Parameters Reference Parameter Description [TransparentCoderPresent Determines the format of the Transparent coder representation in the ation] SDP.  [0] = clearmode (default)  [1] = X-CCD [IgnoreRemoteSDPMKI] Determines whether the device ignores the Master Key Identifier (MKI) if present in the SDP received from the remote side.
Page 480 Mediant 2000 Parameter Description the CPT file.  In case of an MLPP call, the device uses the value of this parameter plus 1 as the index of the Ringback tone in the CPT file (e.g., if this value is set to 1, then the index is 2, i.e., 1 + 1).
Page 481 SIP User's Manual A. Configuration Parameters Reference Parameter Description Note: For mapping specific SIT tones, you can use the SITQ850CauseForNC, SITQ850CauseForIC, SITQ850CauseForVC, and SITQ850CauseForRO parameters. Web/EMS: SIT Q850 Cause Defines the Q.850 cause value specified in the SIP Reason header For NC that is included in a 4xx response when SIT-NC (No Circuit Found [SITQ850CauseForNC]...
Page 482 Mediant 2000 Parameter Description Meridian). These behaviors are performed upon one of the following scenarios:  Physically disconnected from the network (i.e., Ethernet cable is disconnected).  The Ethernet cable is connected, but the device can't communicate with any host. Note that LAN Watch-Dog must be activated (the parameter EnableLANWatchDog set to 1).
Page 483 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Number of RTX Before Defines the number of retransmitted INVITE/REGISTER messages Hot-Swap before the call is routed (hot swap) to another Proxy/Registrar. EMS: Proxy Hot Swap Rtx The valid range is 1 to 30. The default value is 3. [HotSwapRtx] Note: This parameter is also used for alternative routing using the Outbound IP Routing Table'.
Page 484: Coders And Profile Parameters
Mediant 2000 Parameter Description  [0] Use Current Condition = The condition entered in this row must be matched in order to perform the defined action (default).  [1] Use Previous Condition = The condition of the rule configured directly above this row must be used in order to perform the defined action.
Page 485 SIP User's Manual A. Configuration Parameters Reference Parameter Description [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = g711Alaw64k, 20, 0, 255, 0; CodersGroup0 1 = eg711Ulaw, 10, 0, 71, 0; CodersGroup0 2 = eg711Ulaw, 10, 0, 71, 0; [ \CodersGroup0 ] [ CodersGroup1 ] FORMAT CodersGroup1_Index = CodersGroup1_Name,...
Page 486 Mediant 2000 Parameter Description 20 (default) 4.75 [0], Dynamic (0- Disable [0] [Amr] 5.15 [1], 127) Enable [1] 5.90 [2], 6.70 [3], 7.40 [4], 7.95 [5], 10.2 [6], 12.2 [7] (default) QCELP 20 (default), 40, 60, Always Always 12 Disable [0]...
Page 487 SIP User's Manual A. Configuration Parameters Reference Parameter Description IP Profile Table Web: IP Profile This parameter table configures the IP Profile table. Each IP Profile ID includes Settings a set of parameters (which are typically configured separately using their EMS: Protocol individual "global"...
Page 488 Mediant 2000 Parameter Description IpProfile_IsFaxUsed Fax Signaling Method IsFaxUsed IpProfile_JitterBufMi Dynamic Jitter Buffer DJBufMinDelay nDelay Minimum Delay IpProfile_JitterBufO Dynamic Jitter Buffer DJBufOptFactor ptFactor Optimization Factor IpProfile_IPDiffServ RTP IP DiffServ PremiumServiceClas sMediaDiffServ IpProfile_SigIPDiffS Signaling DiffServ PremiumServiceClas sControlDiffServ IpProfile_SCE EnableSilenceCompr ession IpProfile_RTPRedun...
Page 489 SIP User's Manual A. Configuration Parameters Reference Parameter Description IpProfile_FirstTxDtm First Tx DTMF Option TxDTMFOption fOption IpProfile_SecondTx Second Tx DTMF TxDTMFOption DtmfOption Option IpProfile_RxDTMFO Declare RFC 2833 in RxDTMFOption ption IpProfile_EnableHol Enable Hold EnableHold Input Gain InputGain IpProfile_InputGain IpProfile_VoiceVolu Voice Volume VoiceVolume Add IE In SETUP AddIEinSetup...
Page 490 Mediant 2000 Parameter Description IpProfile_SBCDivers Diversion Mode ionMode IpProfile_SBCHistor History Info Mode yInfoMode  The parameter IpPreference determines the priority of the IP Profile (1 to 20, where 20 is the highest preference). If both IP and Tel Profiles apply to the same call, the coders and common parameters (i.e., parameters configurable...
Page 491 SIP User's Manual A. Configuration Parameters Reference Parameter Description TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP, TelProfile_TimeForReorderTone, TelProfile_EnableDIDWink, TelProfile_IsTwoStageDial, TelProfile_DisconnectOnBusyTone, TelProfile_EnableVoiceMailDelay, TelProfile_DialPlanIndex, TelProfile_Enable911PSAP, TelProfile_SwapTelToIpPhoneNumbers, TelProfile_EnableAGC, TelProfile_ECNlpMode; TelProfile_DigitalCutThrough; [\TelProfile] For example: TelProfile 1 = ITSP_audio, 1, 0, 0, 10, 10, 46, 40, -11, 0, 0, 0, 0, 0, 1, 0, 0, 700, 0, -1, 255, 0, 1, 1, 1, -1, 1, 0, 0, 0, 0;...
Page 492 Mediant 2000 Parameter Description TelProfile_EnableEC Echo Canceler EnableEchoCanceller TelProfile_MWIAnal MWI Analog Lamp MWIAnalogLamp TelProfile_MWIDispl MWI Display MWIDisplay TelProfile_FlashHoo Flash Hook Period FlashHookPeriod kPeriod TelProfile_EnableEa Enable Early Media EnableEarlyMedia rlyMedia TelProfile_ProgressI Progress Indicator to ProgressIndicator2IP ndicator2IP TelProfile_TimeForR Time For Reorder TimeForReorderTone...
Page 493: Channel Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description  For a description of using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 74. A.10 Channel Parameters This subsection describes the device's channel parameters. A.10.1 Voice Parameters The voice parameters are described in the table below.
Page 494 Mediant 2000 Parameter Description Time [AnswerDetectorSilenceTime] Web: Answer Detector Currently, not supported. Redirection [AnswerDetectorRedirection] Web: Answer Detector Sensitivity Defines the Answer Detector sensitivity. EMS: Sensitivity The range is 0 (most sensitive) to 2 (least sensitive). The default [AnswerDetectorSensitivity] is 0.
Page 495: Coder Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description EchoCancellerLength, as it automatically acquires its value from this parameter. EMS: Echo Canceller Hybrid Defines the four-wire to two-wire worst-case Hybrid loss, the ratio Loss between the signal level sent to the hybrid and the echo level returning from the hybrid.
Page 496 Mediant 2000 Parameter Description EMS: VBR Coder Header Determines the format of the RTP header for VBR coders. Format  [0] = Payload only (no header, TOC, or m-factor) - similar to [VBRCoderHeaderFormat] RFC 3558 Header Free format (default). ...
Page 497: Dtmf Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: DSP Version Template Determines the DSP template used by the device. Each DSP Number template supports specific coders, channel capacity, and features. EMS: Version Template Number The default is DSP template 0. [DSPVersionTemplateNumber] You can load different DSP templates to digital modules using the syntax DSPVersionTemplateNumber=xy...
Page 498: Rtp, Rtcp And T.38 Parameters
Mediant 2000 Parameter Description EMS: Rx DTMF Relay Hang Defines the Voice Silence time (in msec) after playing DTMF or MF Over Time (msec) digits to the Tel/PSTN side that arrive as Relay from the IP side. [RxDTMFHangOverTime] Valid range is 0 to 2,000 msec. The default is 1,000 msec.
Page 499 SIP User's Manual A. Configuration Parameters Reference Parameter Description Notes:  When enabled, you can configure the payload type, using the RFC2198PayloadType parameter.  The RTP redundancy dynamic payload type can be included in the SDP, by using the EnableRTPRedundancyNegotiation parameter. ...
Page 500 Mediant 2000 Parameter Description type is used for the received DTMF packets. If negotiation isn't used, this payload type is used for receive and for transmit. Web/EMS: RFC 2833 RX Payload Defines the Rx RFC 2833 DTMF relay dynamic payload Type type.
Page 501 SIP User's Manual A. Configuration Parameters Reference Parameter Description  The UDP ports are allocated randomly to channels.  You can define a UDP port range per Media Realm (see Configuring Media Realms on page 159).  If RTP Base UDP Port is not a factor of 10, the following message is generated: 'invalid local RTP port'.
Page 502 Mediant 2000 Parameter Description Disables RTCP traffic when there is no RTP traffic. This [RTCPActivationMode] feature is useful, for example, to stop RTCP traffic that is typically sent when calls are put on hold (by an INVITE with 'a=inactive' in the SDP).
Page 503: Gateway And Ip-To-Ip Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: RTCP XR Collection Server Defines the IP address of the Event State Compositor EMS: Esc IP (ESC). The device sends RTCP XR reports to this server, [RTCPXREscIP] using SIP PUBLISH messages. The address can be configured as a numerical IP address or as a domain name.
Page 504 Mediant 2000 Parameter Description Web: V.23 Modem Transport Determines the V.23 modem transport type. Type  [0] Disable = Disable (Transparent) EMS: V23 Transport  [1] Enable Relay = N/A [V23ModemTransportType  [2] Enable Bypass = (default)  [3] Events Only = Transparent with Events Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 204).
Page 505 SIP User's Manual A. Configuration Parameters Reference Parameter Description thus, in these cases it is possible to configure the device to start the 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 per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 204).
Page 506 Mediant 2000 Parameter Description transport type is set to bypass or Transparent-with-Events. Web/EMS: Fax Bypass Defines the fax bypass RTP dynamic payload type. Payload Type The valid range is 96 to 120. The default value is 102. [FaxBypassPayloadType] EMS: Modem Bypass Defines the modem bypass dynamic payload type.
Page 507 SIP User's Manual A. Configuration Parameters Reference Parameter Description with NSE Bypass. The bypass packet interval is selected according to the FaxModemBypassBasicRtpPacketInterval parameter. Notes:  This feature can be used only if the VxxModemTransportType parameter is set to 2 (Bypass). ...
Page 508 Mediant 2000 Parameter Description and the parameter IsFaxUsed to 1, 2, or 3.  The "FAX" prefix in routing and manipulation tables is case- sensitive. Web: Detect Fax on Answer Determines when the device initiates a T.38 session for fax Tone transmission.
Page 509: Dtmf And Hook-Flash Parameters
SIP User's Manual A. Configuration Parameters Reference A.11.2 DTMF and Hook-Flash Parameters The DTMF and hook-flash parameters are described in the table below. Table A-37: DTMF and Hook-Flash Parameters Parameter Description Hook-Flash Parameters Web/EMS: Hook-Flash Code Defines the digit pattern used by the PBX to indicate a Hook Flash [HookFlashCode] event.
Page 510 Mediant 2000 Parameter Description DTMF Parameters EMS: Use End of DTMF Determines when the detection of DTMF events is notified. [MGCPDTMFDetectionPoint]  [0] = DTMF event is reported at the end of a detected DTMF digit.  [1] = DTMF event is reported at the start of a detected DTMF digit (default).
Page 511 SIP User's Manual A. Configuration Parameters Reference Parameter Description type as configured by the parameter RFC2833PayloadType. Removes DTMF digits in transparent mode (as part of the voice stream).  When TxDTMFOption is set to 0, the RFC 2833 payload type is set according to the parameter RFC2833PayloadType for both transmit and receive.
Page 512: Digit Collection And Dial Plan Parameters
Mediant 2000 Parameter Description Additional examples: 1664wpp102, 66644ppp503, and 7774w100pp200. Notes:  For this parameter to take effect, a device reset is required.  This parameter can also be configured per Tel Profile, using the TelProfile parameter. Web: Special Digit Defines the representation for ‘special’...
Page 513 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: Digit Map Pat terns ISDN overlap dialing). If the digit string (i.e., dialed number) matches [DigitMapping] one of the patterns in the digit map, the device stops collecting digits and establishes a call with the collected number. The digit map pattern can contain up to 52 options (rules), each separated by a vertical bar (|).
Page 514: Voice Mail Parameters
Mediant 2000 A.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. Table A-39: Voice Mail Parameters Parameter Description...
Page 515 SIP User's Manual A. Configuration Parameters Reference Parameter Description [WaitForBusyTime] Defines the time (in msec) that the device waits to detect busy and/or reorder tones. This feature is used for semi-supervised PBX call transfers (i.e., the LineTransferMode parameter is set to 2). The valid value range is 0 to 20000 (i.e., 20 sec).
Page 516 Mediant 2000 Parameter Description  When the RS-232 connection is used for SMDI messages (Serial SMDI), it cannot be used for other applications, for example, to access the Command Line Interface (CLI). Web/EMS: SMDI Timeout Defines the time (in msec) that the device waits for an SMDI Call Status message before or after a Setup message is received.
Page 517 SIP User's Manual A. Configuration Parameters Reference Parameter Description [MWIQsigMsgCentreldID Defines the Message Centred ID party number used for QSIG MWI PartyNumber] messages. If not configured (default), the parameter is not included in MWI (activate and deactivate) QSIG messages. The value is a string. Digit Patterns The following digit pattern parameters apply only to voice mail applications that use the DTMF communication method.
Page 518 Mediant 2000 Parameter Description Web: Forward on Do Not Defines the digit pattern used by the PBX to indicate 'call forward on do Disturb Digit Pattern not disturb' when the original call is received from an external line (not (External) an internal extension).
Page 519: Supplementary Services Parameters
SIP User's Manual A. Configuration Parameters Reference A.11.5 Supplementary Services Parameters This subsection describes the device's supplementary telephony services parameters. Caller ID Parameters A.11.5.1 The caller ID parameters are described in the table below. Table A-40: Caller ID Parameters Parameter Description Web: Asserted Identity Mode Determines whether the SIP header P-Asserted-Identity or P-...
Page 520: Call Waiting Parameters
Mediant 2000 Parameter Description the P-Asserted-Identity header in 180/200 OK responses, by setting the parameter AssertedIDMode to 1.  This parameter is applicable to ISDN, CAS interfaces. Web: Caller ID Transport Type Determines the device's behavior for Caller ID detection.
Page 521: Call Hold Parameters
SIP User's Manual A. Configuration Parameters Reference Call Hold Parameters A.11.5.4 The call hold parameters are described in the table below. Table A-43: Call Hold Parameters Parameter Description Web/EMS: Enable Hold Enables interworking of the Hold/Retrieve supplementary service from [EnableHold] PRI to SIP.
Page 522: Call Transfer Parameters
Mediant 2000 Call Transfer Parameters A.11.5.5 The call transfer parameters are described in the table below. Table A-44: Call Transfer Parameters Parameter Description Web/EMS: Enable Transfer Enables the Call Transfer feature. [EnableTransfer]  [0] Disable = Disable the call transfer service.
Page 523: Emergency Call Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description tone is played to the channel. EMS: Blind Transfer Disconnect Defines the duration (in milliseconds) for which the device waits for Timeout a disconnection from the Tel side after the Blind Transfer Code [BlindTransferDisconnectTim (KeyBlindTransfer) has been identified.
Page 524: Call Cut-Through Parameters
Mediant 2000 Call Cut-Through Parameters A.11.5.7 The call cut-through parameters are described in the table below. Table A-46: Call Cut-Through Parameters Parameter Description Enables PSTN CAS channels/endpoints to receive incoming IP calls even [DigitalCutThrough] if the B-channels are in off-hook state.
Page 525 SIP User's Manual A. Configuration Parameters Reference Parameter Description channels to receive the call, the device terminates one of the channel calls and sends the E911 call to that channel. The preemption is done only on a channel pertaining to the same Trunk Group for which the E911 call was initially destined and if the channel select mode (configured by the ChannelSelectMode parameter) is set to other than “By Dest Number”...
Page 526 Mediant 2000 Parameter Description Web: MLPP Normalized Defines the MLPP normalized service domain string. If the device Service Domain receives an MLPP ISDN incoming call, it uses the parameter (if EMS: Normalized Service different from ‘FFFFFF’) as a Service domain in the SIP Resource- Domain Priority header in outgoing INVITE messages.
Page 527 SIP User's Manual A. Configuration Parameters Reference Parameter Description Multiple Differentiated Services Code Points (DSCP) per MLPP Call Priority Level (Precedence) Parameters The MLPP service allows placement of priority calls, where properly validated users can preempt (terminate) lower-priority phone calls with higher-priority calls. For each MLPP call priority level, the DSCP can be set to a value from 0 to 63.
Page 528: Tty/Tdd Parameters
Mediant 2000 TTY/TDD Parameters A.11.5.9 The TTY (telephone typewriter) or telecommunications device for the deaf (TDD) is an electronic device for text communication via a telephone line for those with impaired hearing. The TTY/TDD parameters are described in the table below.
Page 529 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [11] T1 4ESS ISDN = ISDN PRI protocol for the Lucent™/AT&T™ 4ESS switch.  [12] T1 5ESS 9 ISDN = ISDN PRI protocol for the Lucent™/AT&T™ 5ESS-9 switch.  [13] T1 5ESS 10 ISDN = ISDN PRI protocol for the Lucent™/AT&T™...
Page 530 Mediant 2000 Parameter Description [ISDNDMSTimerT310] Overrides the T310 timer for the DMS-100 ISDN variant. T310 defines the timeout between the receipt of a Proceeding message and the receipt of an Alerting/Connect message. The valid range is 10 to 30. The default value is 10 (seconds).
Page 531 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web/EMS: Clock Master Determines the Tx clock source of the E1/T1 line. [ClockMaster]  [0] Recovered = Generate the clock according to the Rx of the E1/T1 line (default).  [1] Generated = Generate the clock according to the internal TDM bus.
Page 532: Tdm Bus And Clock Timing Parameters
Mediant 2000 Parameter Description  [EnableTDMoverIP] [0] Disable = Disabled (default).  [1] Enable = TDM Tunneling is enabled. When TDM Tunneling is enabled, the originating device automatically initiates SIP calls from all enabled B-channels pertaining to E1/T1/J1 spans that are configured with the 'Transparent' protocol.
Page 533 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS/Web: TDM Bus Local Defines the physical Trunk ID from which the device recovers (receives) Reference its clock synchronization. [TDMBusLocalReference The range is 0 to the maximum number of Trunks. The default is 0. Note: This parameter is applicable only if the parameter TDMBusClockSource is set to 4 and the parameter TDMBusPSTNAutoClockEnable is set to 0.
Page 534: Cas Parameters
Mediant 2000 Parameter Description Notes:  For this parameter to take effect, a device reset is required.  This parameter is applicable only when the TDMBusPSTNAutoClockEnable parameter is set to 1. Web: Auto Clock Trunk Defines the trunk priority for auto-clock fallback (per trunk parameter).
Page 535 SIP User's Manual A. Configuration Parameters Reference Parameter Description CASTableIndex_3 = 1 Note: You can define CAS tables per B-channel using the parameter CASChannelIndex. Web: Dial Plan Defines the CAS Dial Plan name that is used on a specific trunk EMS: Dial Plan Name (where x denotes the trunk ID).
Page 536 Mediant 2000 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.
Page 537: Isdn Parameters
SIP User's Manual A. Configuration Parameters Reference ISDN Parameters A.11.6.4 The ISDN parameters are described in the table below. Table A-52: ISDN Parameters Parameter Description Web: ISDN Termination Side Determines the ISDN termination side. EMS: Termination Side  [0] User side = ISDN User Termination Equipment (TE) [TerminationSide] side (default) ...
Page 538 Mediant 2000 Parameter Description channel that is used for signaling.  [1] BACKUP = Backup Trunk - contains a backup D- channel that is used if the primary D-channel fails.  [2] NFAS = NFAS Trunk - contains only 24 B-channels, without a signaling D-channel.
Page 539 SIP User's Manual A. Configuration Parameters Reference Parameter Description message if an empty called number is received in an incoming Setup message. This option is applicable to the overlap dialing mode. The device also plays a dial tone (for TimeForDialTone) until the next called number digits are received.
Page 540 Mediant 2000 Parameter Description message that contains the unknown Facility IE is rejected (default). Note: This option is applicable only to ISDN variants where a complete ASN1 decoding is performed on Facility IE.  [128] SEND USER CONNECT ACK = The Connect ACK message is sent in response to received Q.931 Connect;...
Page 541 SIP User's Manual A. Configuration Parameters Reference Parameter Description Notes:  To configure the device to support several ISDNIBehavior features, enter a summation of the individual feature values. For example, to support both [512] and [2048] features, set the parameter ISDNIBehavior is set to 2560 (i.e., 512 + 2048).
Page 542 Mediant 2000 Parameter Description 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). This allows more flexible application control on the UUI. When this bit is not set (default behavior), CC implements the UUI-protocol as specified in the ETS 300-403 standards for Implicit Service 1.
Page 543 SIP User's Manual A. Configuration Parameters Reference Parameter Description character in the called_nb, and is not restricted to extended digits only (i.e., 0-9,*,#).  [16384] DLCI REVERSED OPTION = Behavior bit used in the IUA interface groups to indicate that the reversed format of the DLCI field must be used.
Page 544: Isdn And Cas Interworking Parameters
Mediant 2000 A.11.7 ISDN and CAS Interworking Parameters The ISDN and CAS interworking parameters are described in the table below. Table A-53: ISDN and CAS Interworking Parameters Parameter Description ISDN Parameters Web: Send Local Time To ISDN Enables the device to send the date and time in the ISDN...
Page 545 SIP User's Manual A. Configuration Parameters Reference Parameter Description number is sent in the INVITE Request-URI user part. The device receives ISDN called number that is sent in the 'Overlap' mode. The ISDN Setup message is sent to IP only after the number (including the Sending Complete IE) is fully received (via Setup and/or subsequent Info Q.931 messages).
Page 546 Mediant 2000 Parameter Description the ISDN (for Tel-to-IP calls). This is interworked from the P- Asserted-Identity header in SIP 200 OK. The default is [0] (i.e., unknown). [ConnectedNumberPlan] 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).
Page 547 SIP User's Manual A. Configuration Parameters Reference Parameter Description duplicate all messages).  To define the format of encapsulated QSIG messages, use the QSIGTunnelingMode parameter.  Tunneling according to ECMA-355 is applicable to all ISDN variants (in addition to the QSIG protocol). ...
Page 548 Mediant 2000 Parameter Description If the incoming ISDN Setup message includes 'subaddress' values for the Called Number and/or the Calling Number, these values are mapped to the outgoing SIP INVITE message's ‘isub’ parameter in accordance with RFC 4715. [IgnoreISDNSubaddress] Determines whether the device ignores the Subaddress from the incoming ISDN Called and Calling numbers when sending to IP.
Page 549 SIP User's Manual A. Configuration Parameters Reference Parameter Description remains empty. Web: Play Busy Tone to Tel Enables the device to play a busy or reorder tone to the PSTN [PlayBusyTone2ISDN] after a Tel-to-IP call is released.  [0] Don't Play = Immediately sends an ISDN Disconnect message (default).
Page 550 Mediant 2000 Parameter Description ISDN/CAS protocol type doesn't play the RBT; PI = 8 is sent in an ISDN Alert message (unless the parameter ProgressIndicator2ISDN_ID is configured differently). If a 180 response is received, but the 'early media' voice channel is not opened, the device with CAS protocol type plays an RBT to the PSTN.
Page 551 SIP User's Manual A. Configuration Parameters Reference Parameter Description stop the trunk and then restart it for the update to take effect.  To determine the method for setting Out-Of-Service state for all trunks (i.e., per device), use the DigitalOOSBehavior parameter.
Page 552 Mediant 2000 Parameter Description [\CauseMapSIP2ISDN] Where,  SipResponse = SIP Response  IsdnReleaseCause = Q.850 Release Cause For example: CauseMapSIP2ISDN 0 = 480,50; CauseMapSIP2ISDN 0 = 404,3; When a SIP response is received (from the IP side), the device searches this mapping table for a match. If the SIP response is found, the Q.850 Release Cause assigned to it is sent to the...
Page 553 SIP User's Manual A. Configuration Parameters Reference Parameter Description [RemoveCallingName] ISDN calls for all trunks.  [0] Disable = Does not remove Calling Name (default).  [1] Enable = Removes Calling Name. Web: Remove Calling Name Enables the device to remove the Calling Name per trunk EMS: Remove Calling Name For (where x denotes the trunk number) for SIP-to-ISDN calls.
Page 554 Mediant 2000 Parameter Description [TrunkPSTNAlertTimeout_ID] outgoing calls to PSTN. This timer is used between the time that an ISDN Setup message is sent to the Tel side (IP-to-Tel call establishment) and a Connect message is received. If Alerting is received, the timer is restarted.
Page 555 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: Enable CIC Determines whether the Carrier Identification Code (CIC) is [EnableCIC] relayed to ISDN.  [0] = Do not relay the Carrier Identification Code (CIC) to ISDN (default).  [1] = CIC is relayed to the ISDN in Transit Network Selection (TNS) IE.
Page 556 Mediant 2000 Parameter Description the parameter SendIEonTG.  You can configure different IE data for Trunk Groups by defining this parameter for different IP Profile IDs (using the IPProfile parameter) and then assigning the required IP Profile ID in the Inbound IP Routing Table' (PSTNPrefix).
Page 557 SIP User's Manual A. Configuration Parameters Reference Parameter Description information. Note: This capability is applicable only to the NI-2 ISDN variant. [EarlyAnswerTimeout] Defines the time (in seconds) that the device waits for an ISDN Connect message from the called party (Tel side) after sending a Setup message.
Page 558 Mediant 2000 Parameter Description SendISDNTransferOnConnect must be set to 1.  The parameter SendISDNTransferOnConnect can be used to define if the 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).
Page 559 SIP User's Manual A. Configuration Parameters Reference Parameter Description  Tel-to-IP (network side): When a Facility message initiating an out-of-band blind transfer is received from the PBX, the device sends a SIP REFER message to the IP side (if the EnableNetworkISDNTransfer parameter is set to 1).
Page 560 Mediant 2000 Parameter Description The valid range is 1 to 10. The default is 4. Web/EMS: Enable Network ISDN Determines whether the device allows interworking of network- Transfer side received ECT/TBCT Facility messages (NI2 TBCT - Two B-channel Transfer and ETSI ECT - Explicit Call Transfer) to [EnableNetworkISDNTransfer] SIP REFER.
Page 561: Answer And Disconnect Supervision Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Via: SIP/2.0/UDP 192.168.13.2:5060 From: <sip:06@192.168.13.2:5060> <sip:4505656002@192.168.13.40:5060>;tag=13287 8796-1040067870294 Call-ID: 0010-0016-D69A7DA8-1@192.168.13.2 CSeq:2 INFO Content-Type: application/broadsoft Content-Length: 17 event flashhook Note: This parameter is applicable only to T1 CAS protocols. A.11.8 Answer and Disconnect Supervision Parameters The answer and disconnect supervision parameters are described in the table below.
Page 562 Mediant 2000 Parameter Description  [1] Yes (default) Notes:  The timeout is configured by the BrokenConnectionEventTimeout parameter.  This feature is applicable only if the RTP session is used without Silence Compression. If Silence Compression is enabled, the device doesn't detect a broken RTP connection.
Page 563 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [1] Packets Count = According to packet count.  [2] Voice/Energy Detectors = N/A.  [3] All = N/A. Note: For this parameter to take effect, a device reset is required.
Page 564: Tone Parameters
Mediant 2000 Parameter Description  This parameter is applicable only to CAS protocols.  This parameter can also be configured per Tel Profile, using the TelProfile parameter. A.11.9 Tone Parameters This subsection describes the device's tone parameters. Telephony Tone Parameters A.11.9.1...
Page 565 SIP User's Manual A. Configuration Parameters Reference Parameter Description Note: To enable the Cut-Through feature, use the DigitalCutThrough (for CAS channels) parameter. Web: Play Ringback Tone to Tel Enables the play of the ringback tone (RBT) to the Tel side EMS: Play Ring Back Tone To Tel and determines the method for playing the RBT.
Page 566: Tone Detection Parameters
Mediant 2000 Parameter Description doesn't play a ringback tone to IP and doesn't send 183 or 180+SDP responses.  This parameter can also be configured per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 204). Web: Play Local RBT on ISDN...
Page 567 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: User Defined Tone Enable Enables the detection of User Defined Tones signaling, [UserDefinedToneDetectorEnable] applicable for Special Information Tone (SIT) detection.  [0] = Disable (default)  [1] = Enable EMS: SIT Enable Enables SIT detection according to the ITU-T recommendation [SITDetectorEnable] E.180/Q.35.
Page 568: Metering Tone Parameters
Mediant 2000 Metering Tone Parameters A.11.9.3 The metering tone parameters are described in the table below. Table A-57: Metering Tone Parameters Parameter Description Web: Generate Metering Determines the method used to configure the metering tones that are Tones generated to the Tel side.
Page 569: Trunk Groups And Routing Parameters
SIP User's Manual A. Configuration Parameters Reference A.11.10 Trunk Groups and Routing Parameters The routing parameters are described in the table below. Table A-58: Routing Parameters Parameter Description Trunk Group Table Web: Trunk Group Table This parameter table is used to define and activate the device's EMS: SIP Endpoints >...
Page 570 Mediant 2000 Parameter Description  [2] Result Not Used = send MWI Interrogation message, but don't use its result. Instead, wait for MWI Activate requests from the PBX.  [3] Use Result = send MWI Interrogation messages, use its results, and use the MWI Activate requests.
Page 571 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [8] Trunk & Channel Cyclic Ascending = The device implements the Trunk Cyclic Ascending and Cyclic Ascending methods to select the channel. This method selects the next physical trunk (pertaining to the Trunk Group) and then selects the B-channel of this trunk according to the cyclic ascending method (i.e., selects the channel after the last allocated channel).
Page 572 Mediant 2000 Parameter Description Name is received from the Tel side, the IP Display Name remains empty (default).  [1] Yes = If a Tel Display Name is received, the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name.
Page 573 SIP User's Manual A. Configuration Parameters Reference Parameter Description PREFIX_TransportType, PREFIX_SrcTrunkGroupID, PREFIX_DestSRD, PREFIX_CostGroup, PREFIX_ForkingGroup; [\PREFIX] For example: PREFIX 0 = *, domain.com, *, 0, 255, $$, -1, , 1, , -1, -1, -1,,; PREFIX 1 = 20, 10.33.37.77, *, 0, 255, $$, -1, , 2, , 0, -1,,; Notes: ...
Page 574 Mediant 2000 Parameter Description DNSQueryType.  For available notations for depicting a range of multiple numbers, see ''Dialing Plan Notation for Routing and Manipulation'' on page 607.  For a description on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 74.
Page 575 SIP User's Manual A. Configuration Parameters Reference Parameter Description [IP2TelTaggingDestDialPlanInde Determines the Dial Plan index in the external Dial Plan file (.dat) in which string labels ("tags") are defined for tagging incoming IP-to-Tel calls. The special “tag” is added as a prefix to the called party number, and then the Inbound IP Routing Table' uses this “tag”...
Page 576: Alternative Routing Parameters
Mediant 2000 A.11.11 Alternative Routing Parameters The alternative routing parameters are described in the table below. Table A-59: Alternative Routing Parameters Parameter Description Web/EMS: Redundant Routing Determines the type of redundant routing mechanism when a Mode call can’t be completed using the main route.
Page 577 SIP User's Manual A. Configuration Parameters Reference Parameter Description  EMS: Alternative Routing [0] ICMP Ping (default) = Internet Control Message Protocol Telephone to IP Connection (ICMP) ping messages. Method  [1] SIP OPTIONS = The remote destination is considered [AltRoutingTel2IPConnMethod] offline if the latest OPTIONS transaction timed out.
Page 578 Mediant 2000 Parameter Description INVITE retransmissions), the device issues an internal 408 'No Response' implicit release reason.  The device sends the call to an alternative IP route only after the call has failed and the device has subsequently attempted twice to establish the call unsuccessfully.
Page 579: Number Manipulation Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description using transport protocol TCP, if Trunk Group ID 2 is unavailable: 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. Notes: ...
Page 580 Mediant 2000 Parameter Description Web: Tel2IP Default Redirect Determines the default redirect reason for Tel-to-IP calls when no Reason redirect reason (or “unknown”) exists in the received Q931 ISDN [Tel2IPDefaultRedirectReason Setup message. The device includes this default redirect reason in the SIP History-Info header of the outgoing INVITE.
Page 581 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Set TEL-to-IP Redirect Defines the redirect reason for Tel-to-IP calls. If redirect (diversion) Reason information is received from the Tel, the redirect reason is set to [SetTel2IpRedirectReason] the value of this parameter before the device sends it on to the IP. ...
Page 582 Mediant 2000 Parameter Description Tel-to-IP destination phone number manipulation. Therefore, this allows you to have different numbers for the called (i.e., SIP To header) and redirect (i.e., SIP Diversion header) numbers. Notes:  If the incoming ISDN-to-IP call includes a Redirect Number, this number is overridden by the new called number if this parameter is set to [1] or [2].
Page 583 SIP User's Manual A. Configuration Parameters Reference Parameter Description  Port Number [1] Yes [ReplaceEmptyDstWithPortN Note: This parameter is applicable only to Tel-to-IP calls and if the umber] called number is missing.  [0] = Leave Source Number empty (default). [CopyDestOnEmptySource] ...
Page 584 Mediant 2000 Parameter Description  [0] = Disabled (default)  [1] = Swap calling and called numbers Note: This parameter can also be configured per Tel Profile, using the TelProfile parameter. Web/EMS: Add Prefix to Defines a string prefix that is added to the Redirect number Redirect Number received from the Tel side.
Page 585 SIP User's Manual A. Configuration Parameters Reference Parameter Description CallingNameMapTel2Ip_SrcIPGroupID, CallingNameMapTel2Ip_RemoveFromLeft, CallingNameMapTel2Ip_RemoveFromRight, CallingNameMapTel2Ip_LeaveFromRight, CallingNameMapTel2Ip_Prefix2Add, CallingNameMapTel2Ip_Suffix2Add; [ \CallingNameMapTel2Ip ] Destination Phone Number Manipulation for IP-to-Tel Calls Table Web: Destination Phone This parameter table manipulates the destination number of IP-to- Number Manipulation Table for Tel calls.
Page 586 Mediant 2000 Parameter Description rules for complex number manipulation requirements (that generally require many rules).  [0] = Disable (default)  [1] = Enable Destination Phone Number Manipulation for Tel-to-IP Calls Table Web: Destination Phone This parameter table manipulates the destination number of Tel-to- Number Manipulation Table for IP calls.
Page 587 SIP User's Manual A. Configuration Parameters Reference Parameter Description 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; Notes:  This table parameter can include up to 120 indices.  The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 588 Mediant 2000 Parameter Description SourceNumberMapTel2Ip_RemoveFromRight, SourceNumberMapTel2Ip_LeaveFromRight, SourceNumberMapTel2Ip_Prefix2Add, SourceNumberMapTel2Ip_Suffix2Add, SourceNumberMapTel2Ip_IsPresentationRestricted, NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_SrcIPGroupID; [\SourceNumberMapTel2Ip] For example: SourceNumberMapTel2Ip 0 = 22,03,$$,0,0,$$,2,$$,667,$$,0,$$,$$; SourceNumberMapTel2Ip 0 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; Notes:  This table parameter can include up to 120 indices.  The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 589 SIP User's Manual A. Configuration Parameters Reference Parameter Description  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 Redirect Number IP -to-Tel Table Web: Redirect Number IP ->...
Page 590 Mediant 2000 Parameter Description RedirectNumberMapTel2Ip_IsPresentationRestricted, RedirectNumberMapTel2Ip_SrcTrunkGroupID, RedirectNumberMapTel2Ip_SrcIPGroupID; [\RedirectNumberMapTel2Ip] For example: RedirectNumberMapTel2Ip 1 = *, 4, 255, 255, 0, 0, 255, , 972, 255, 1, 2; Notes:  This parameter table can include up to 20 indices (1-20).  The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 591: Ldap Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description [AddPhoneContextAsPrefix] and Calling numbers.  [0] Disable = Disable (default).  [1] Enable = Enable. A.11.13 LDAP Parameters The Lightweight Directory Access Protocol (LDAP) parameters are described in the table below. For more information on routing based on LDAP, refer to ''Routing Based on LDAP Active Directory Queries'' on page 163.
Page 592: Least Cost Routing Parameters
Mediant 2000 Parameter Description Web: MS LDAP PBX Number attribute Defines the name of the attribute that represents the user name PBX number in the Microsoft AD database. [MSLDAPPBXNumAttributeName] The valid value is a string of up to 49 characters. The default is "telephoneNumber".
Page 593: Standalone Survivability Parameters
SIP User's Manual A. Configuration Parameters Reference A.12 Standalone Survivability Parameters The Stand-alone Survivability (SAS) parameters are described in the table below. Table A-63: SAS Parameters Parameter Description Web: Enable SAS Enables the Stand-Alone Survivability (SAS) feature. EMS: Enable  [0] Disable Disabled (default) [EnableSAS] ...
Page 594 Mediant 2000 Parameter Description IP address in the Record-Route header.  [0] Disable (default)  [1] Enable The Record-Route header is inserted in a request by a SAS proxy to force future requests in the dialog session to be routed through the SAS agent.
Page 595 SIP User's Manual A. Configuration Parameters Reference Parameter Description  EMS: Survivability Mode [0] Standard = Incoming INVITE and REGISTER requests [SASSurvivabilityMode] are forwarded to the defined Proxy list of SASProxySet in Normal mode and handled by the SAS application in Emergency mode (default).
Page 596 Mediant 2000 Parameter Description emergency calls. This valid value is a character string. The default is an empty string "". Web: SAS Inbound Manipulation Enables destination number manipulation in incoming INVITE Mode messages when SAS is in Emergency the state. The...
Page 597: Ip Media Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: SAS IP-to-IP Routing Table [IP2IPRouting] This parameter table 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,...
Page 598 Mediant 2000 Parameter Description Notes:  This parameter can also be configured per Tel Profile, using the TelProfile parameter.  For a description of AGC, see Automatic Gain Control (AGC) on page 157. Web: AGC Slope Determines the AGC convergence rate: EMS: Gain Slope ...
Page 599 SIP User's Manual A. Configuration Parameters Reference Parameter Description The range is 0 to -31. The default is -20. Note: For this parameter to take effect, a device reset is required. EMS: Maximal Gain Defines the maximum gain (in dB) by the AGC when [AGCMaxGain] activated.
Page 600 Mediant 2000 Parameter Description calls (i.e., voice detection). The default is 3. Notes:  This parameter is applicable only if the AMDSensitivityParameterSuit parameter is set to 0.  To enable the AMD feature, set the EnableDSPIPMDetectors parameter to 1. ...
Page 601 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Answer Machine Detector Beep Defines the AMD beep detection timeout (i.e., the duration Detection Timeout that the beep detector functions from when detection is EMS: Beep Detection Timeout initiated). This is used for detecting beeps at the end of an [AMDBeepDetectionTimeout] answering machine message.
Page 602: Auxiliary And Configuration Files Parameters
Mediant 2000 A.14 Auxiliary and Configuration Files Parameters This subsection describes the device's auxiliary and configuration files parameters. A.14.1 Auxiliary and Configuration File Name Parameters The configuration files (i.e., auxiliary files) can be loaded to the device using the Web interface or a TFTP session.
Page 603: Automatic Update Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Dial Plan Defines the Dial Plan name (up to 11-character strings) that is used EMS: Dial Plan Name on a specific trunk (denoted by x). [CasTrunkDialPlanName_x] Web: Dial Plan File Defines the name (and path) of the Dial Plan file (defining dial EMS: Dial Plan File Name plans).
Page 604 Mediant 2000 Parameter Description [ResetNow] Invokes an immediate device reset. This option can be used to activate offline (i.e., not on-the-fly) parameters that are loaded using the parameter IniFileUrl.  [0] = The immediate restart mechanism is disabled (default). ...
Page 605 SIP User's Manual A. Configuration Parameters Reference Parameter Description [TLSRootFileUrl] Defines the name of the TLS trusted root certificate file and the URL from where it can be downloaded. Note: For this parameter to take effect, a device reset is required. Defines the name of the TLS certificate file and the URL from where [TLSCertFileUrl] it can be downloaded.
Page 606 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 607: B Dialing Plan Notation For Routing And Manipulation
SIP User's Manual B. Dialing Plan Notation for Routing and Manipulation Dialing Plan Notation for Routing and Manipulation The device supports flexible dialing plan notations for depicting the prefix and/or suffix source and/or destination numbers and SIP URI user names in the routing and manipulation tables.
Page 608 Mediant 2000 Notation Description but (3-12) is not. [n,m,...] or (n,m,...) Represents multiple numbers. For example, to depict a one-digit number starting with 2, 3, 4, 5, or 6:  Prefix: [2,3,4,5,6]#  Suffix: (2,3,4,5,6)  Prefix with Suffix: [2,3,4,5,6](8,7,6) - prefix is denoted in square brackets;...
Page 609: Csip Message Manipulation Syntax
SIP User's Manual C. SIP Message Manipulation Syntax SIP Message Manipulation Syntax This section provides a detailed description on the support and syntax for configuring SIP message manipulation rules. For configuring message manipulation rules, see the parameter MessageManipulations. Actions The actions that can be done on SIP message manipulation in the Message Manipulations table are listed in the table below.
Page 610: Accept-Language
Mediant 2000 C.2.2 Accept-Language An example of the header is shown below: Accept-Language: da, en-gb;q=0.8, en;q=0.7 The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Below is a header manipulation example:...
Page 611: Call-Id
SIP User's Manual C. SIP Message Manipulation Syntax C.2.4 Call-Id An example of the header is shown below: Call-ID: JNIYXOLCAIWTRHWOINNR@10.132.10.128 The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes String Read Only...
Page 612: Cseq
Mediant 2000 C.2.6 Cseq An example of the header is shown below: CSeq: 1 INVITE The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Integer Read Only...
Page 613: Event
SIP User's Manual C. SIP Message Manipulation Syntax Below are header manipulation examples: Example 1 Rule: Add a Diversion header to all INVITE messages: MessageManipulations 0 = 1, invite, , header.Diversion, 0," '<tel:+101>;reason=unknown; counter=1;screen=no; privacy=off'", 0; Diversion: <tel:+101>;reason=user- Result: busy;screen=no;privacy=off;counter=1 Example 2 Rule: Modify the Reason parameter in the header to 1, see ''Reason (Diversion)''...
Page 614: From
Mediant 2000 Example 3 Rule: Modify the Event package enum: MessageManipulations 2 = 1, invite, , header.event.EVENTKEY.EVENTPACKAGE, 2, "'2'", 0; Event: refer;id=5678 Result: C.2.9 From An example of the header is shown below: From: <sip:555@10.132.10.128;user=phone>;tag=YQLQHCAAYBWKKRVIMWEQ The header properties are shown in the table below:...
Page 615: Min-Se And Min-Expires
SIP User's Manual C. SIP Message Manipulation Syntax Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes HistoryInfo String Read/Write Below are header manipulation examples: Example 1 Rule: Add a new History-Info header to the message: MessageManipulations 0 = 1, any, , header.History- Info, 0, '<sip:UserA@audc.mydomain.com;index=3>', 0 History-Info:sip:UserA@ims.example.com;index=1...
Page 616: P-Asserted-Identity
Mediant 2000 Example 2 Rule: Modify a Min-Expires header with the min-expires value and add an additional 0: MessageManipulations 0 = 1, Invite, , header.Min- Expires.param, 2, "header.Min-Expires.time + '0'", 0; Min-Expires: 340;3400 Result: Example 3 Rule: Modify a Min-Expires header changing the time to 700: MessageManipulations 0 = 1, Invite, , header.Min-...
Page 617: P-Called-Party-Id
SIP User's Manual C. SIP Message Manipulation Syntax Keyword Sub Types Attributes Name String Read/Write Param Param Read/Write URL Structure (see ''URL'' Read/Write on page 635) Below are header manipulation examples: Example 1 Rule: Add a P-Associated-Uri header to all INVITE response messages: MessageManipulations 5 = 1, register.response, ,header.P-Associated-URI, 0, '<sip:admin@10.132.10.108>', 0;...
Page 618: P-Charging-Vector
Mediant 2000 Example 3 Rule: Add a display name to the P-Called-Party-Id header: MessageManipulations 3 = 1, any, , header.p-called- party-id.name, 2, 'Secretary', 0; P-Called-Party-ID: Secretary Result: <sip:2000@gw.itsp.com>;p1=red C.2.15 P-Charging-Vector An example of the header is shown below: P-Charging-Vector: icid-value=1234bc9876e; icid-generated- at=192.0.6.8;...
Page 619: Privacy
SIP User's Manual C. SIP Message Manipulation Syntax Below are header manipulation examples: Example 1 Rule: Add a P-Preferred-Identity header to all messages: MessageManipulations 1 = 1, any, , header.P-Preferred- Identity, 0, "'Cullen Jennings <sip:fluffy@abc.com>'", P-Preferred-Identity: "Cullen Jennings" Result: <sip:fluffy@abc.com> Example 2 Rule: Modify the display name in the P-Preferred-Identity header:...
Page 620: Reason
Mediant 2000 Keyword Sub Types Attributes Capabilities SIPCapabilities Struct Read/Write Below are header manipulation examples: Add a Proxy-Require header to the message: Example 1 Rule: MessageManipulations 1 = 1, any, , header.Proxy- Require, 0, "'sec-agree'", 0; Proxy-Require: sec-agree Result: Example 2 Rule: Modify the Proxy-Require header to itsp.com:...
Page 621: Referred-By
SIP User's Manual C. SIP Message Manipulation Syntax Example 3 Rule: Modify the cause number: MessageManipulations 0 = 1, any, ,header.reason.reason.reason, 0, '483', 0; Reason: SIP ;cause=483 ;text="483 Too Many Hops" Result: Note: The protocol (SIP or Q.850) is controlled by setting the cause number to be greater than 0.
Page 622: Remote-Party-Id
Mediant 2000 Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Below are header manipulation examples: Example 1 Rule: Add a basic header: MessageManipulations 0 = 1, any, ,header.Refer-to, 0, "'<sip:referto@referto.com>'", 0; Refer-To: <sip:referto@referto.com> Result: Example 2 Rule: Add a Refer-To header with URI headers: MessageManipulations 0 = 1, any, ,header.Refer-to, 0,...
Page 623: Request-Uri
SIP User's Manual C. SIP Message Manipulation Syntax Keyword Sub Types Attributes URL Structure (see ''URL'' on page 635) Read/Write Below are header manipulation examples: Example 1 Rule: Add a Remote-Party-Id header to the message: MessageManipulations 0 = 1, invite, ,header.REMOTE- PARTY-ID, 0, "'<sip:999@10.132.10.108>;party=calling'", 0;...
Page 624: Require
Mediant 2000 Keyword Sub Types Attributes URL Structure (see ''URL'' Read/Write on page 635) Below are header manipulation examples: Example 1 Rule: Test the Request-URI transport type. If 1 (TCP), then modify the URL portion of the From header: MessageManipulations 1 = 1, Invite.request, "header.REQUEST-URI.url.user == '101'",...
Page 625: Resource-Priority
SIP User's Manual C. SIP Message Manipulation Syntax Example 2 Rule: If a Require header exists, then delete it: MessageManipulations 2 = 1, Invite, "header.require exists" ,header.require, 1, "''", 0; Result: The Require header is deleted. Example 3 Rule: Set the early media options tag in the header: MessageManipulations 0 = 0, invite, , header.require.earlymedia, 0, “1”...
Page 626: Server Or User-Agent
Mediant 2000 Below are header manipulation examples: Example 1 Rule: Add a Retry-After header: MessageManipulations 2 = 1, Invite, ,header.Retry- After, 0, "'3600'", 0; Retry-After: 3600 Result: Example 2 Rule: Modify the Retry-Time in the header to 1800: MessageManipulations 3 = 1, Invite, ,header.Retry-...
Page 627: Session-Expires
SIP User's Manual C. SIP Message Manipulation Syntax Below are header manipulation examples: Example 1 Rule: Add two Service-Route headers: MessageManipulations 1 = 1, Invite, ,header.service- route, 0, "'<P2.HOME.EXAMPLE.COM;lr>'", 0; MessageManipulations 2 = 1, Invite, ,header.service- route, 0, "'<sip:HSP.HOME.EXAMPLE.COM;lr>'", 0; Service-Route:<P2.HOME.EXAMPLE.COM;lr>...
Page 628: Subject
Mediant 2000 Example 3 Rule: Add a param called longtimer to the header: MessageManipulations 1 = 1, any, , header.Session- Expires.param.longtimer, 0, "'5'", 0; Session-Expires: 480;longtimer=5 Result: Example 4 Rule: Set the refresher to 1 (UAC): MessageManipulations 3 = 1, any, , header.session- expires.refresher, 2, '1', 0;...
Page 629: C.2.32 To
SIP User's Manual C. SIP Message Manipulation Syntax Below is a header manipulation example: Example 1 Rule: Add a Supported header: MessageManipulations 1 = 1, Invite, ,header.supported, 0, "'early-session", 0; Supported: early-session Result: Example 2 Rule: Set path in the Supported headers options tag: MessageManipulations 0 = 0, invite, , header.supported.path, 0, “true”, 0;...
Page 630: Unsupported
Mediant 2000 Example 4 Rule: Add a proprietary parameter to all To headers: MessageManipulations 6 = 1, invite.request, , header.to.param.artist, 0, "'singer'", 0; To: "Bob Dylan" Result: <sip:101@10.20.30.60:65100>;artist=singer C.2.33 Unsupported An example of the header is shown below: Unsupported: 100rel...
Page 631: Warning
SIP User's Manual C. SIP Message Manipulation Syntax Keyword Sub Types Attributes Branch String Read Only Host Host Structure (see ''Host'' Read Only on page 633) MAddrIp gnTIPAddress Read Only Param Param Read/Write Port Integer Read Only TransportType Enum TransportType (see Read Only ''TransportType'' on page 644)
Page 632: Unknown Header
Mediant 2000 C.2.36 Unknown Header An Unknown header is a SIP header that is not included in this list of supported headers. An example of the header is shown below: MYEXP: scooby, doo, goo, foo The header properties are shown in the table below:...
Page 633: Structure Definitions
SIP User's Manual C. SIP Message Manipulation Syntax Structure Definitions C.3.1 Event Structure The Event structure is used in the Event header (see ''Event'' on page 613). Table C-2: Event Structure Keyword Sub Types Attributes EventPackage Enum Event Package (see Read/Write ''Event Package'' on page 638)
Page 634: Privacy Struct
Mediant 2000 C.3.4 Privacy Struct This structure is applicable to the Privacy header (see ''Privacy'' on page 619). Table C-5: Privacy Structure Keyword Sub Types NONE Boolean HEADER Boolean SESSION Boolean USER Boolean CRITICAL Boolean IDENTITY Boolean HISTORY Boolean C.3.5 Reason Structure This structure is applicable to the Reason header (see ''Reason'' on page 620).
Page 635: Url
SIP User's Manual C. SIP Message Manipulation Syntax Keyword Sub Types History Boolean Unknown Boolean GRUU Boolean ResourcePriority Boolean TargetDialog Boolean SdpAnat Boolean C.3.7 This structure is applicable to the following headers:  Contact (see ''Contact'' on page 611)  Diversion (see ''Diversion'' on page 612) ...
Page 636: Random Type
Mediant 2000 Random Type Manipulation rules can include random strings and integers. An example of a manipulation rule using random values is shown below: MessageManipulations 4 = 1, Invite.Request, , Header.john, 0, rand.string.56.A.Z, 0; In this example, a header called "john" is added to all INVITE messages received by the device and a random string of 56 characters containing characters A through Z is added to the header.
Page 637: Copying Information Between Messages Using Variables
SIP User's Manual C. SIP Message Manipulation Syntax Copying Information between Messages using Variables You can use variables in SIP message manipulation rules to copy specific information from one message to another. Information from one message is copied to a variable and then information from that variable is copied to any subsequent message.
Page 638: Enum Definitions
Mediant 2000 Enum Definitions C.7.1 AgentRole These ENUMs are applicable to the Server or User-Agent headers (see ''Server or User- Agent'' on page 626). Table C-9: Enum Agent Role AgentRole Value Client Server C.7.2 Event Package These ENUMs are applicable to the Server or User-Agent (see ''Server or User-Agent'' on page 626) and Event (see ''Event'' on page 613) headers.
Page 639: Mlpp Reason Type
SIP User's Manual C. SIP Message Manipulation Syntax C.7.3 MLPP Reason Type These ENUMs are applicable to the MLPP Structure (see ''MLPP'' on page 633). Table C-11: Enum MLPP Reason Type Type Value PreEmption Reason MLPP Reason C.7.4 Number Plan These ENUMs are applicable to the Remote-Party-Id header (see ''Remote-Party-Id'' on page 622).
Page 640: Privacy
Mediant 2000 C.7.6 Privacy These ENUMs are applicable to the Remote-Party-Id (see ''Remote-Party-Id'' on page 622) and Diversion (see ''Diversion'' on page 612) headers. Table C-14: Enum Privacy Privacy Role Value Full C.7.7 Reason (Diversion) These ENUMs are applicable to the Diversion header (see ''Diversion'' on page 612).
Page 641 SIP User's Manual C. SIP Message Manipulation Syntax Reason Value SUBSCRIBE PRACK UPDATE PUBLISH LAST_REQUEST TRYING_100 RINGING_180 CALL_FORWARD_181 QUEUED_182 SESSION_PROGRESS_183 OK_200 ACCEPTED_202 MULTIPLE_CHOICE_300 MOVED_PERMANENTLY_301 MOVED_TEMPORARILY_302 SEE_OTHER_303 USE_PROXY_305 ALTERNATIVE_SERVICE_380 BAD_REQUEST_400 UNAUTHORIZED_401 PAYMENT_REQUIRED_402 FORBIDDEN_403 NOT_FOUND_404 METHOD_NOT_ALLOWED_405 NOT_ACCEPTABLE_406 AUTHENTICATION_REQUIRED_407 REQUEST_TIMEOUT_408 CONFLICT_409 GONE_410 LENGTH_REQUIRED_411 CONDITIONAL_REQUEST_FAILED_412 REQUEST_TOO_LARGE_413 REQUEST_URI_TOO_LONG_414...
Page 642 Mediant 2000 Reason Value UNKNOWN_RESOURCE_PRIORITY_417 BAD_EXTENSION_420 EXTENSION_REQUIRED_421 SESSION_INTERVAL_TOO_SMALL_422 SESSION_INTERVAL_TOO_SMALL_423 ANONYMITY_DISALLOWED_433 UNAVAILABLE_480 TRANSACTION_NOT_EXIST_481 LOOP_DETECTED_482 TOO_MANY_HOPS_483 ADDRESS_INCOMPLETE_484 AMBIGUOUS_485 BUSY_486 REQUEST_TERMINATED_487 NOT_ACCEPTABLE_HERE_488 BAD_EVENT_489 REQUEST_PENDING_491 UNDECIPHERABLE_493 SECURITY_AGREEMENT_NEEDED_494 SERVER_INTERNAL_ERROR_500 NOT_IMPLEMENTED_501 BAD_GATEWAY_502 SERVICE_UNAVAILABLE_503 SERVER_TIME_OUT_504 VERSION_NOT_SUPPORTED_505 MESSAGE_TOO_LARGE_513 PRECONDITION_FAILURE_580 BUSY_EVERYWHERE_600 DECLINE_603 DOES_NOT_EXIST_ANYWHERE_604 NOT_ACCEPTABLE_606 SIP User's Manual Document #: LTRT-68814...
Page 643: Reason (Remote-Party-Id)
SIP User's Manual C. SIP Message Manipulation Syntax C.7.9 Reason (Remote-Party-Id) These ENUMs are applicable to the Remote-Party-Id header (see ''Remote-Party-Id'' on page 622). Table C-17: Enum Reason (RPI) Reason Value Busy Immediate No Answer C.7.10 Refresher These ENUMs are used in the Session-Expires header (see ''Session-Expires'' on page 627).
Page 644: Screenind
Mediant 2000 C.7.12 ScreenInd These ENUMs are applicable to the Remote-Party-Id header (see ''Remote-Party-Id'' on page 622). Table C-20: Enum ScreenInd Screen Value User Provided User Passed User Failed Network Provided C.7.13 TransportType These ENUMs are applicable to the URL Structure (see ''URL'' on page 635) and the Via header (see ''Via'' on page 630).
Page 645: Actions And Types
SIP User's Manual C. SIP Message Manipulation Syntax Actions and Types Element Command Command Value Type Remarks Type Type IPGroup Match "==" String Returns true if the parameter equals to the value. "!=" String Returns true if the parameter not equals to the value.
Page 646 Mediant 2000 Element Command Command Value Type Remarks Type Type "Add" String Adds a new header to the end of the list. *Header "Remove" Removes the whole list from the message. Header Match "==" String Returns true if a header equals to the value.
Page 647 SIP User's Manual C. SIP Message Manipulation Syntax Element Command Command Value Type Remarks Type Type value. "contains" String Returns true if the header's parameter contains the string. "exists" Returns true if the header's parameter exists. Action "Modify" String Sets the header's parameter to the value.
Page 648 Mediant 2000 Element Command Command Value Type Remarks Type Type "!=" String Returns true if the string element not equals to the value. "contains" String Returns true if the value is found in the string element. Action "Modify" String Sets the string element to the value.
Page 649: Syntax
SIP User's Manual C. SIP Message Manipulation Syntax Syntax Rules table: Man Set ID Message Condition Action Element Action Type Action Type Value Rule <message- <match- <message- <action-type> <value> type> condition> element> message-type: Description: rule is applied only if this is the message's type Syntax: method "."...
Page 650 Mediant 2000 • param.call.dst.user != 100 • header.john exists • header.john exists AND header.to.host !contains “john” • header.from.user == 100 OR header.from.user == 102 OR header.from.user == match-type Description: comparison to be made Syntax: ( “==” / “!=” / “>” / “<” / “>=” / “<=” / “contains” / “exists” / “!exists” / “!contains”...
Page 651 SIP User's Manual C. SIP Message Manipulation Syntax header-index Description: header's index in the list of headers Syntax: integer Examples: If five Via headers arrive: ♦ 0 (default) – refers to the first Via header in the message ♦ 1 – the second Via header ♦...
Page 652 Mediant 2000 param-dir-element Description: direction relating to the classification Syntax: ( "src" / "dst" ) Examples: ♦ src – relates to the source ♦ dst – relates to the destination call-param-entity Description: parameters that can be accessed on the call Syntax: ( "user"...
Page 653 SIP User's Manual C. SIP Message Manipulation Syntax value: Description: value for action and match Syntax: ( string / message-element / param ) * ( "+" ( string / message-element / param ) ) Examples: • "itsp.com" • Header.from.user • Param.ipg.src.user •...
Page 654 Mediant 2000 Reader's Notes SIP User's Manual Document #: LTRT-68814...
Page 655: D Dsp Templates
SIP User's Manual D. DSP Templates DSP Templates This section lists the DSP templates supported by the device. Each DSP template provides support for specific voice coders (as well as channel capacity and various features). You can use the following parameters to select the required DSP template: ...
Page 656 Mediant 2000 DSP Template Number of Channels Default Setting With 128 ms EC With SRTP With IPM Detectors With IPM Detectors & SRTP Voice Coder EVRC QCELP GSM EFR iLBC SIP User's Manual Document #: LTRT-68814...
Page 657: E Selected Technical Specifications
SIP User's Manual E. Selected Technical Specifications Selected Technical Specifications The technical specifications of the Mediant 2000 are listed in the table below: Note: All specifications in this document are subject to change without prior notice. Table E-1: Mediant 2000 Functional Specifications...
Page 658 Mediant 2000 Function Specification Answer Machine Detects whether voice or an answering machine is answering the call. Detector Note: When implementing Answer Machine Detector, channel capacity may be reduced. Call Progress Tone 32 tones: single tone, dual tones or AM tones, programmable frequency Detection and &...
Page 659 SIP User's Manual E. Selected Technical Specifications Function Specification LED Indicators LED Indications on Power, ACT/Fail, T1/E1 status, LAN status, Swap ready indication Front Panel Connectors & Switches Rear Panel Trunks 1 to 8 and 9 to 16 Two 50-pin female Telco connectors (DDK57AE-40500-21D) or 8 RJ- 48c connectors for trunks 1 to 8 only Ethernet 1 and 2 Two 10/100Base-TX, RJ-45 shielded connectors...
Page 660 Mediant 2000 Function Specification Enclosure Dimensions 445 x 44 x 300 mm (17.5 x 1.75 x 12 inches) Weight Approx. 4.8 kg fully populated (16 spans); 4.2 kg for 1 span Installation 1U 19-inch 2-slot chassis; rack-, shelf-, or desktop-mount options.
Page 661 SIP User's Manual E. Selected Technical Specifications Function Specification Type Approvals  IC CS03; FCC part 68 Telecommunication Standards  Chassis and Host telecom card comply with IC CS03; FCC part 68; CTR 4, CTR 12 & CTR 13; JATE; TS.016; TSO; Anatel, Mexico Telecom, Russia CCC, ASIF S016, ASIF S038 ...
Page 662 User's Manual Ver. 6.4 www.audiocodes.com...

AudioCodes Mediant 2000 User Manual

1 Overview

Part I: Getting Started

2 Assigning the Voip LAN IP Address

Part II: Management Tools

3 Web-Based Management

4 CLI-Based Management

5 SNMP-Based Management

6 EMS-Based Management

7 INI File-Based Management

Part III: General System Settings

8 Configuring Certificates

9 Date and Time

Part IV: Voip Settings

10 Network

11 Security

12 Media

13 Services

14 Control Network

15 Enabling Applications

16 Coders and Profiles

17 SIP Definitions

18 GW and IP to IP

19 Stand-Alone Survivability (SAS) Application

20 Transcoding Using Third-Party Call Control

Part V: Maintenance

21 Basic Maintenance

22 Software Upgrade

Quick Links

Troubleshooting

Related Manuals for AudioCodes Mediant 2000

Summary of Contents for AudioCodes Mediant 2000