Page 1 Mediant™ 3000 VoIP Media Gateway SIP Protocol User’s Manual Version 6.4 November 2011 Document # LTRT-89712...
Page 3: Table Of Contents
Assigning the VoIP LAN IP Address ..............27 Using CLI ....................... 27 Using the Web Interface ..................28 Using BootP/TFTP Server ..................29 Assigning an IP Address for Mediant 3000 HA ............31 2.4.1 Using the Web Interface ..................32 2.4.2 Using the EMS ......................
Page 4 10.3.1.1 Multiple Network Interfaces and VLANs ..........111 10.3.1.2 Setting Up VoIP Networking ..............119 10.3.1.3 Getting Started with Mediant 3000 in High Availability Mode ....123 10.4 Configuring the IP Routing Table ................. 130 10.4.1 Routing Table Columns ..................132...
Page 5 SIP User's Manual Contents 10.4.1.1 Destination Column ................132 10.4.1.2 Prefix Length Column ................132 10.4.1.3 Gateway Column ................... 132 10.4.1.4 Interface Column ................... 132 10.4.1.5 Metric Column ..................133 10.4.1.6 State Column ..................133 10.4.2 Routing Table Configuration Summary and Guidelines ........133 10.4.3 Troubleshooting the Routing Table ...............134 10.5 Configuring QoS Settings ..................
Page 6 Mediant 3000 12.3.3.1 Configuring DTMF Transport Types ............170 12.3.3.2 Configuring RFC 2833 Payload ............172 12.3.4 Configuring RTP Base UDP Port ................173 12.3.5 RTP Control Protocol Extended Reports (RTCP XR) ...........173 12.3.5.1 Configuring RTCP XR using Web Interface .......... 174 12.4 Configuring IP Media Settings ................
Page 7 SIP User's Manual Contents 18.1.1 TDM and Timing ....................245 18.1.1.1 Configuring TDM Bus Settings .............. 245 18.1.1.2 Configuring Digital PCM Settings ............246 18.1.1.3 Configuring System Timing ..............246 18.1.2 Configuring Transmission Settings ................250 18.1.3 Configuring CAS State Machines ................251 18.1.4 Configuring Trunk Settings ..................253 18.1.5 Configuring Digital Gateway Parameters ..............256 18.1.6 Tunneling Applications ...................257...
Page 8 Mediant 3000 18.7.2 Determining the Availability of Destination IP Addresses ........327 18.7.3 PSTN Fallback .......................327 18.8 SIP Call Routing Examples .................. 328 18.8.1 SIP Call Flow Example ..................328 18.8.2 SIP Message Authentication Example ..............330 18.8.3 Trunk-to-Trunk Routing Example ................333 18.8.4 SIP Trunking between Enterprise and ITSPs ............334 18.9 IP-to-IP Routing Application .................
Page 9 SIP User's Manual Contents 19.1.11.2 BroadSoft's Shared Phone Line Call Appearance for SBC Survivability 19.1.11.3 Call Survivability for Call Centers ............386 19.1.12 Call Forking ......................389 19.1.12.1 Initiating SIP Call Forking ..............389 19.1.12.2 SIP Forking Initiated by SIP Proxy Server ..........389 19.1.13 Alternative Routing on Detection of Failed SIP Response ........389 19.1.14 Active SBC Call Continuity during HA Blade Switchover ........390 19.2 SBC Configuration ....................
Page 10 Mediant 3000 20.1.2.1 SAS Routing in Normal State ..............447 20.1.2.2 SAS Routing in Emergency State ............449 20.2 SAS Configuration ....................449 20.2.1 General SAS Configuration ...................450 20.2.1.1 Enabling the SAS Application ............... 450 20.2.1.2 Configuring Common SAS Parameters ..........450 20.2.2 Configuring SAS Outbound Mode .................453...
Page 11 SIP User's Manual Contents Part VI: Status, Performance Monitoring and Reporting ........505 26 System Status ....................507 26.1 Viewing Device Information .................. 507 26.2 Viewing Ethernet Port Information ............... 508 26.3 Viewing Timing Module Information ..............509 26.4 Viewing Hardware Components Status ............... 510 27 Carrier-Grade Alarms ..................
Page 12 Mediant 3000 A.1.1 Ethernet Parameters ....................549 A.1.2 Multiple Network Interfaces and VLAN Parameters ..........550 A.1.3 Static Routing Parameters ..................553 A.1.4 Quality of Service Parameters ................553 A.1.5 NAT and STUN Parameters ..................555 A.1.6 NFS Parameters ....................557 A.1.7 DNS Parameters ....................558 A.1.8 DHCP Parameters ....................559...
Page 13 SIP User's Manual Contents A.12.5.9 TTY/TDD Parameters ................674 A.12.6 PSTN Parameters ....................675 A.12.6.1 General Parameters ................675 A.12.6.2 TDM Bus and Clock Timing Parameters ..........679 A.12.6.3 CAS Parameters ................... 683 A.12.6.4 ISDN Parameters .................. 686 A.12.6.5 DS3 Parameters ..................693 A.12.6.6 SDH/SONET Parameters ..............
Page 14 D DSP Templates ....................821 D.1 Mediant 3000 Full Chassis ................... 822 D.2 Mediant 3000 16 E1 / 21 T1 ................. 823 D.3 Mediant 3000 with Single T3 ................825 D.4 DSP Template Mix Feature for Mediant 3000 ............826 Selected Technical Specifications ..............
Page 15: Weee Eu Directive
SIP User's Manual Notices Notice This document describes the AudioCodes Mediant 3000 SIP media gateway, housed with TP- 8410 SIP blade(s) or TP-6310 SIP blade(s). Information contained in this document is believed to be accurate and reliable at the time of printing.
Page 16: Related Documentation
Note: Throughout this manual, unless otherwise specified, the following naming conventions are used: • The term device refers to the Mediant 3000 housing either TP-8410 or TP-6310 blades. • The term blade refers to the TP-8410 blade or TP-6310 blade.
Page 17: Overview
SIP User's Manual 1. Overview Overview The Mediant 3000 (hereafter referred to as device) is a SIP-based Voice-over-IP (VoIP) media gateway, offering an integrated voice media gateway functionality for voice, data, and fax streaming over IP networks. The device addresses mid-density applications deployed in IP networks, by delivering up to 2,016 simultaneous voice channels.
Page 18: Mediant 3000 High Availability System
Monitors system components to detect any hardware failures • Handles switchover procedures to overcome possible failures For more details on the HA system, see 'Mediant 3000 High Availability System' on page 18. Mediant 3000 High Availability System The High Availability architecture of the Mediant 3000 provides the following main functionality: ...
Page 19 HA system with TP-8410 and TP-6310 are shown respectively in the figures below: Figure 1-2: High Availability System 1+1 of Mediant 3000 with TP-8410 Figure 1-3: High Availability System 1+1 of Mediant 3000 with TP-6310 On the front panel, the active blade occupies Slot 1 while the redundant blade occupies Slot 3.
Page 20: Initialization Process
For the device to be configured for HA, the following installation stages must be performed:  Hardware configuration setup (refer to Mediant 3000 and IPmedia 3000 Installation Manual). The system can be set up initially with only one VoP blade and one SA/M3K blade.
Page 21: Blade Failure Detection
OAMP interface. This allows the device's blade to send reports to a Syslog server in case of major device failure.  The Mediant 3000 HA is loaded from flash (when loading from BootP/TFTP on first configuration setup, HA is disabled). 1.1.3...
Page 22: Functional Block Diagrams
Mediant 3000 Functional Block Diagrams The functional block diagrams of the Mediant 3000 with the TP-6310 blade and with the TP-8410 blade are shown in the figures below, respectively: Figure 1-4: Mediant 3000/TP-6310 Functional Block Diagram SIP User's Manual Document #: LTRT-89712...
Page 23 SIP User's Manual 1. Overview Figure 1-5: Mediant 3000/TP-8410 Functional Block Diagram Version 6.4 November 2011...
Page 24: Sip Overview
Mediant 3000 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 25: Part I: Getting Started
Part I Getting Started Before you can begin configuring your device, you need to access it with the default VoIP LAN IP address and change this IP address, if required, 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 26 Reader’s Notes...
Page 27: Assigning The Voip Lan Ip Address
This section describes how to change the default VoIP LAN IP address for Simplex configuration so that it corresponds to your networking scheme. To assign IP addresses to High Availability (HA) configuration, see Assigning an IP Address for Mediant 3000 HA on page 31.
Page 28: Using The Web Interface
Mediant 3000 At the prompt, type the following command to access the configuration folder, and then press Enter: conf At the prompt, type the following command to view the current network settings, and then press Enter: GCP IP At the prompt, typing the following command to change the network settings, and then press Enter: SCP IP <ip_address>...
Page 29: Using Bootp/Tftp Server
Disconnect the computer from the device or hub / switch (depending on the connection used in Step 2) and reconnect the device to your network. Using BootP/TFTP Server You can assign an IP address to the device, using the supplied AudioCodes BootP/TFTP Server utility. Notes: •...
Page 30 Mediant 3000 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 31: Assigning An Ip Address For Mediant 3000 Ha
2. Assigning the VoIP LAN IP Address Assigning an IP Address for Mediant 3000 HA This section describes how to assign IP addresses to the Mediant 3000 HA, using BootP and the Web interface (refer to 'Using the Web Interface' on page 32), or using AudioCodes EMS management tool (refer to 'Using the EMS' on page 34).
Page 32: Using The Web Interface
The procedure below describes how to initially assign a global (public) IP address to Mediant 3000 HA, using BootP and the Web interface.  To initially assign a global IP address to Mediant 3000 HA, using the Web interface: Assign the device's two blades with new private IP addresses that correspond to your network IP addressing scheme, using BootP: Ensure that both blades are connected to the network.
Page 33 SIP User's Manual 2. Assigning the VoIP LAN IP Address Define the global IP address: Access the Web interface with the IP address that you assigned to the blade in Slot 1. Open the Multiple Interface Table page (Configuration tab > VoIP menu > Network submenu >...
Page 34: Using The Ems
Mediant 3000 2.4.2 Using the EMS The procedure below describes how to initially assign a global IP address to Mediant 3000 HA using AudioCodes EMS management tool.  To initially assign a global IP address to Mediant 3000 HA, using the EMS: Power down the chassis.
Page 35 Mediant 3000 is now configured in the EMS Server with the private IP address of the blade in Slot 1. In the Mediant 3000 Status window, only the active blade in Slot 1 is displayed (i.e., Mediant 3000 is not yet configured for High Availability).
Page 36 Right-click the active blade, and then choose Configuration > Upload; the EMS server synchronizes with the blade's configuration settings. Verify that the Mediant 3000 is up and running (by performing a ping to its IP address). If it is up and running, the EMS’s 'Status' screen displays a graphic representation of it.
Page 37: Part Ii: Management Tools
Command Line Interface (CLI) - see 'CLI-Based Management' on page  Configuration INI file - see 'INI File-Based Management' on page  AudioCodes Element Management System - see 'EMS-Based Management' on page  Simple Network Management Protocol (SNMP) browser software - see 'SNMP-Based Management' on page Notes: •...
Page 38 Reader’s Notes...
Page 39: 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 40: Accessing The Web Interface
Mediant 3000 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 39).
Page 41: 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 42: Toolbar Description
Mediant 3000 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 478).
Page 43: 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 44: Displaying Navigation Tree In Basic And Full View
Mediant 3000 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 45: Showing / Hiding The Navigation Pane
SIP User's Manual 3. Web-Based Management 3.1.5.2 Showing / Hiding the Navigation Pane The Navigation pane can be hidden to provide more space for elements displayed in the Work pane. This is especially useful when the Work pane displays a table that's wider than the Work pane and to view all the columns, you need to use scroll bars.
Page 46: Working With Configuration
Mediant 3000 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 47 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 48: Modifying And Saving Parameters
Mediant 3000 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 49: Entering Phone Numbers
SIP User's Manual 3. Web-Based Management with the lightning symbol are not changeable on-the-fly and require a device reset (see 'Resetting the Device' on page 475) before taking effect. Notes: • Parameters saved to the volatile memory (by clicking Submit), revert to their previous settings after a hardware or software reset (or if the device is powered down).
Page 50: Working With Tables
Mediant 3000 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 51 SIP User's Manual 3. Web-Based Management  To edit an index table entry: In the 'Index' column, select the index corresponding to the table row that you want to edit. Click Edit; the fields in the corresponding index row become available. Modify the values as required, and then click Apply;...
Page 52 Mediant 3000 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 53 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 54: Searching For Configuration Parameters
Mediant 3000 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 55: 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 56 Mediant 3000 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 57: 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 58: Editing A Scenario
Mediant 3000 To navigate between Scenario Steps, you can perform one of the following:  In the Navigation tree, click the required Scenario Step.  In an opened Scenario Step (i.e., page appears in the Work pane), use the following navigation buttons: •...
Page 59: Saving A Scenario To A Pc
SIP User's Manual 3. Web-Based Management • Edit the Step Name: In the Navigation tree, select the required Step. In the 'Step Name' field, modify the Step name. In the page, click Next. • Edit the Scenario Name: In the 'Scenario Name' field, edit the Scenario name. In the displayed page, click Next.
Page 60: Loading A Scenario To The Device
Mediant 3000 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 61: 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 62: Creating A Login Welcome Message
Mediant 3000 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 63: 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 64: Logging Off The Web Interface
Mediant 3000 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 70.  To log off the Web interface: On the toolbar, click the Log Off button;...
Page 65: 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 66 Mediant 3000 To perform these operations, see 'Basic Maintenance' on page 473.  High Availability: status of the device's HA mode: • "Not Operational" - HA is not supported • "Stand Alone" - device is in Simplex mode • "Operational" - device is in HA mode •...
Page 67 SIP User's Manual 3. Web-Based Management Item # Description 7 & 8 (Applicable only to 6310 Series blades.) Dual Ethernet port status icons (Eth 1 and Eth 2):  (gray): No link  (green): Active Ethernet link  (yellow): Redundant link You can also view detailed Ethernet port information in the Ethernet Port Information page (see Viewing Ethernet Port Information on page 508), by clicking the icon.
Page 68: High Availability Status
VoIP blades (TP-6310 or TP-8410) and two SA/M3K Alarms, Status and Synchronization ("SAT") blades, as shown in the figure below: Figure 3-31: Home Page for Mediant 3000 HA Device The status of the HA mode is indicated in the General Information pane's 'High Availability' field: ...
Page 69 SIP User's Manual 3. Web-Based Management The Active blade is indicated by the following:  The darker-shaded blade in the graphical display of the device's chassis.  The color of the ACT icon: • (gray): Single blade (i.e., Simplex mode) •...
Page 70: Configuring Web User Accounts
Mediant 3000 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 71: Mediant
SIP User's Manual 3. Web-Based Management  To change the Web user accounts attributes: Open the Web User Accounts page (Configuration tab > System menu > Web User Accounts). Figure 3-32: 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 72 Mediant 3000 Click Change Password; 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 password. To prevent user access after a specific number of failed logins, do the following:...
Page 73: Configuring Web Security Settings
This feature is enabled using the EnableMgmtTwoFactorAuthentication parameter. Note: For specific integration requirements for implementing a third-party smart card for Web login authentication, contact your AudioCodes representative.  To login to the Web interface using CAC: Insert the Common Access Card into the card reader.
Page 74: Configuring Web And Telnet Access List
Mediant 3000 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 75: Configuring Radius Settings
SIP User's Manual 3. Web-Based Management 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 547. ...
Page 76 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 77: Cli-Based Management
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 78: Configuring Telnet And Ssh Settings
Mediant 3000 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 561.  To define Telnet and SSH: Open the Telnet/SSH Settings page (Configuration tab >...
Page 79: 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 80: Configuring Snmp Trap Destinations
Mediant 3000 Configure the SNMP community strings parameters according to the table below. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 478. To delete a community string, select the Delete check box corresponding to the community string that you want to delete, and then click Submit.
Page 81: Configuring Snmp Trusted Managers
SIP User's Manual 5. SNMP-Based Management Table 5-2: SNMP Trap Destinations Parameters Description Parameter Description SNMP Manager Determines the validity of the parameters (IP address and [SNMPManagerIsUsed_x] port number) of the corresponding SNMP Manager used to receive SNMP traps.  [0] (Check box cleared) = Disabled (default) ...
Page 82: Configuring Snmp V3 Users
Mediant 3000 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 83 SIP User's Manual 5. SNMP-Based Management Parameter Description Authentication Key Authentication key. Keys can be entered in the form of a text [SNMPUsers_AuthKey] password or long hex string. Keys are always persisted as long hex strings and keys are localized. Privacy Key Privacy key.
Page 84 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 85: 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 86 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 87: Ini File-Based Management
 Web interface (see 'Backing Up and Loading Configuration File' on page 502)  AudioCodes' BootP/TFTP utility (refer to the Product Reference Manual)  Any standard TFTP server When loaded to the device, the configuration settings of the ini file are saved to the device's non-volatile memory.
Page 88: Configuring Ini File Table Parameters
Mediant 3000 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 89: 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 90: Modifying An Ini File
The file may be loaded to the device using TFTP or HTTP. These protocols are not secure and are vulnerable to potential hackers. To overcome this security threat, the AudioCodes' TrunkPack Downloadable Conversion Utility (DConvert) utility allows you to binary-encode (encrypt) the ini file before loading it to the device (refer to the Product Reference Manual).
Page 91: Part Iii: General System Settings
Part III General System Settings This part provides general system configurations.
Page 92 Reader’s Notes...
Page 93: 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 94 Mediant 3000 Open the Certificates page (Configuration tab > System menu > Certificates). Figure 8-1: Certificates Page SIP User's Manual Document #: LTRT-89712...
Page 95 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 96: Loading A Private Key
Mediant 3000 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 97: 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 98: Self-Signed Certificates
Mediant 3000 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 99: 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. Manual Date and Time The date and time of the device can be configured manually. 9.1.1 Configuring Date and Time Manually using Web Interface The Regional Settings page allows you to define and view the device's internal date and...
Page 100: Automatic Date And Time Through Sntp Server
Mediant 3000 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 101: Configuring Sntp Using Web Interface
SIP User's Manual 9. Date and Time 9.2.1 Configuring SNTP using Web Interface 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).
Page 102 Mediant 3000 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 103: Part Iv: Voip Configuration
Part IV VoIP Configuration This part describes the VoIP configurations.
Page 104 Reader’s Notes...
Page 105: 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 Full-Duplex •...
Page 106: Configuring Ip Interface Settings
Mediant 3000 After start-up is complete and the operational software is running, the device continues to use the Ethernet port used for software upload. The device switches over from one Ethernet port to the other each time an Ethernet link carrier-loss is detected on the active Ethernet port, and if the Ethernet link of the other port is operational.
Page 107 SIP User's Manual 10. Network Notes: • For more information and examples of network interfaces configuration, see 'Network Configuration' on page 111. • When adding more than one interface, ensure that you enable VLANs using the 'VLAN Mode' (VlANMode) parameter. •...
Page 108 Mediant 3000 To view network interfaces that are currently active, click the IP Interface Status Table button. For a description of this display, see 'Viewing Active IP Interfaces' on page 521. Table 10-1: Multiple Interface Table Parameters Description Parameter Description...
Page 109 SIP User's Manual 10. Network Parameter Description flash again. This enables the device to operate with a temporary address for initial management and configuration while retaining the address to be used for deployment. Web/EMS: Prefix Length Defines the Classless Inter-Domain Routing (CIDR)-style [InterfaceTable_PrefixLength] representation of a dotted decimal subnet notation.
Page 110 [0] Disable (default).  [1] Enable. Notes:  This parameter is applicable only to Mediant 3000 with TP- 8410.  For this parameter to take effect, a device reset is required.  When the parameter is enabled, VLANs are not supported (i.e., VlANMode is set to 1).
Page 111: 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 112 Mediant 3000 Note: When Physical Network Separation of network traffic is enabled (using the parameter EnableNetworkPhysicalSeparation), user VLANs supported. 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...
Page 113 SIP User's Manual 10. Network 10.3.1.1.2.1 IP Address and Prefix Length Columns These columns allow the user to configure an IPv4/IPv6 IP address and its related subnet mask. The Prefix Length column holds the Classless Inter-Domain Routing (CIDR)-style representation of a dotted-decimal subnet notation. The CIDR-style representation uses a suffix indicating the number of bits which are set in the dotted-decimal format, in other 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).
Page 114 Mediant 3000 Table 10-4: Separate Routing Table Example Destination Prefix Length Gateway Interface Metric Status 17.17.0.0 192.168.0.1 Active 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. One IPv4 interface and one IPv6 interface may share the same VLAN ID, allowing hybrid networks on a single broadcast domain.
Page 115 SIP User's Manual 10. Network The Native' VLAN ID is configurable using the VlanNativeVlanId parameter (refer to the Setting up your System sub-section below). The default value of the 'Native' VLAN ID is 1. Note: If VlanNativeVlanId is not configured (i.e., its default value of 1 occurs), but one of the interfaces has a VLAN ID configured to 1, this interface is still related to the 'Native' VLAN.
Page 116 Mediant 3000 Application Traffic / Network Types Class-of-Service (Priority) ARP listener Determined by the initiator of the Network request SNMP Traps Management Bronze DNS client Varies according to DNS settings: Depends on traffic type:   OAMP Control: Premium Control ...
Page 117 SIP User's Manual 10. Network 10.3.1.1.5 Multiple Interface Table Configuration Summary and Guidelines Multiple Interface table configuration must adhere to the following rules:  Up to 32 different interfaces may be defined.  The indices used must be in the range between 0 and 31. ...
Page 118 Mediant 3000 traffic from the interface which VLAN ID equals to the 'Native' VLAN ID are tagged with VLAN ID 0 (priority tag).  Quality of Service parameters specify the priority field for the VLAN tag (IEEE 802.1p) and the DiffServ field for the IP headers. These specifications relate to service classes.
Page 119: Setting Up Voip Networking
SIP User's Manual 10. Network 10.3.1.2 Setting Up VoIP Networking 10.3.1.2.1 Using the ini File When configuring the network configuration using the ini File, use a textual presentation of the Interface and Routing Tables, as well as some other parameters. The following shows an example of a full network configuration, consisting of all the parameters described in this section: ;...
Page 120 Mediant 3000 Notes: • Lines that begin with a semicolon are considered a remark and are ignored. • When using the ini file, the Multiple Interface table must have the prefix and suffix to allow the INI File parser to correctly recognize and parse the table.
Page 121 SIP User's Manual 10. Network Example 2 - Three VoIP Interfaces, One for each Application Exclusively: the Multiple Interface table is configured with three interfaces, one exclusively for each application type: one interface for OAMP applications, one for Call Control applications, and one for RTP Media applications: Table 10-9: Multiple Interface Table - Example 2 Allowed...
Page 122 V6CntrlMedia Control Manual VLANs are required. The Native' VLAN ID is the same VLAN ID as the AudioCodes Management interface (index 0). 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...
Page 123: Getting Started With Mediant 3000 In High Availability Mode
Active module in Slot 1 and the Redundant module in Slot 3). Each blade in the Mediant 3000 system boots as standalone. The blade is also assigned its own private address (which may have been acquired via BootP/DHCP or configured manually) which is used for maintenance only (prior to entering HA mode).
Page 124 211.211.85.1 211.211.85. myMediaIF VLANs are required. The Native' VLAN ID is the same VLAN ID as the AudioCodes Management interface (Index 0). One routing rule is required, to allow remote management from a host in 176.85.49.0 / 24: Table 10-16: Routing Table...
Page 125 SIP User's Manual 10. Network The ini file matching this configuration can be written as follows: ; Interface Table Configuration: [InterfaceTable] FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes, InterfaceTable_IPv6InterfaceMode, InterfaceTable_IPAddress, InterfaceTable_PrefixLength, InterfaceTable_Gateway, InterfaceTable_VlanID, InterfaceTable_InterfaceName, InterfaceTable_PrimaryDNSServerIPAddress, InterfaceTable_SecondaryDNSServerIPAddress; InterfaceTable 0 = 0, 0, 192.168.85.14, 16, 192.168.0.1, 1, ManagementIF, , ;...
Page 126 Physical Network Separation can be configured in the Web (or by using the ini file parameter EnableNetworkPhysicalSeparation). Notes: • Physical Network Separation is supported only for Mediant 3000 housing TP-8410. • Physical Network Separation supports three interfaces (OAMP, Control, and Media) and all three must be configured when operating in Physical Network Separation mode.
Page 127 SIP User's Manual 10. Network For more information on cabling these interfaces, refer to the Mediant 3000 and IPmedia 3000 Installation Manual. The figure below illustrates the connectivity of the device when operating in network physical interfaces separation mode. Figure 10-2: Separate Physical Network Interfaces Version 6.4...
Page 128 Verify that the Syslog displays the following message: "Updating Flash to work in Network Separation Mode in the next Boot". If the Mediant 3000 is in High Availability (HA) mode, remove the blade that you configured above, and then repeat steps 1 through 5 for the second blade (using the identical ini file).
Page 129 Verify that the Syslog displays the following message: "Updating Flash to work in Non Network Separation Mode in the next Boot". If the Mediant 3000 is in High Availability (HA) mode, remove the blade that you configured above, insert the second blade, and then repeat steps 1 through 6 for the second blade (using the identical ini file).
Page 130: Configuring The Ip Routing Table
Mediant 3000 10.4 Configuring the IP Routing Table The IP Routing Table page allows you to define up to 30 static IP routing rules for the device. These rules can be associated with a network interface (defined in the Multiple Interface table) and therefore, the routing decision is based on the source subnet/VLAN.
Page 131 SIP User's Manual 10. Network Table 10-17: IP Routing Table Description Parameter Description Destination IP Address Specifies the IP address of the destination host/network. [StaticRouteTable_Destination] Prefix Length Specifies the subnet mask of the destination host/network. [StaticRouteTable_PrefixLength] The address of the host/network you want to reach is determined by an AND operation that is applied to the fields 'Destination IP Address' and 'Destination Mask'.
Page 132: Routing Table Columns
Mediant 3000 10.4.1 Routing Table Columns Each row of the Routing table defines a static routing rule. Traffic destined to the subnet specified in the routing rule is re-directed to the defined gateway, reachable through the specified interface. The IP Routing table consists of the following:...
Page 133: Metric Column
SIP User's Manual 10. Network Figure 10-5: Interface Column 10.4.1.5 Metric Column The Metric column must be set to 1 for each static routing rule. 10.4.1.6 State Column The State column displays the state of each static route. Possible values are "Active" and "Inactive".
Page 134: Troubleshooting The Routing Table
Mediant 3000 10.4.3 Troubleshooting the Routing Table When adding a new static routing rule, the added rule passes a validation test. If errors are found, the routing rule is rejected and is not added to the IP Routing table. Failed routing validations may result in limited connectivity (or no connectivity) to the destinations specified in the incorrect routing rule.
Page 135: 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 478). 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 136: Configuring The Internal Srv Table
Mediant 3000  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 137: Nat (Network Address Translation) Support
SIP User's Manual 10. Network  To configure the Internal SRV table: Open the Internal SRV Table page (Configuration tab > VoIP menu > Network submenu > DNS submenu > Internal SRV Table). Figure 10-7: Internal SRV 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 138: Stun
Mediant 3000 messages and therefore, can’t change local to global addresses. Two different streams traverse through NAT: signaling and media. A device (located behind a NAT) that initiates a signaling path has problems in receiving incoming signaling responses (they are blocked by the NAT server).
Page 139: First Incoming Packet Mechanism
You can control the payload type with which the No-Op packets are sent. This is performed using the RTPNoOpPayloadType ini parameter (see 'Networking Parameters' on page 549). AudioCodes’ default payload type is 120.  T.38 No-Op: T.38 No-Op packets are sent only while a T.38 session is activated. Sent packets are a duplication of the previously sent frame (including duplication of the sequence number).
Page 140: Configuring Nfs Settings
Mediant 3000 10.8 Configuring NFS Settings Network File System (NFS) enables the device to access a remote server's shared files and directories, and to handle them as if they're located locally. You can configure up to 16 different NFS file systems. As a file system, the NFS is independent of machine types, operating systems, and network architectures.
Page 141 SIP User's Manual 10. Network Notes: • To avoid terminating current calls, a row must not be deleted or modified while the device is currently accessing files on that remote NFS file system. • The combination of 'Host Or IP' and 'Root Path' must be unique for each row in the table.
Page 142: Robust Receipt Of Media Streams
Mediant 3000 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 143: 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 25 ordered firewall rules. The access list provides the following firewall rules: ...
Page 144 Mediant 3000 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 145 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 146 Mediant 3000 Parameter Description Packet Size 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 147: 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 547.  To configure the general security parameters: Open the General Security Settings page (Configuration tab >...
Page 148 Mediant 3000  To configure IP Security Proposals: Open the ‘IP Security Proposals Table page (Configuration tab > VoIP menu > Security submenu > IPSec Proposal Table). Figure 11-3: IP Security Proposals Table In the figure above, four proposals are defined.
Page 149: Configuring Ip Security Associations Table
SIP User's Manual 11. Security 11.4 Configuring IP Security Associations Table The IP Security Associations Table page allows you to configure up to 20 peers (hosts or networks) for IP security (IPSec)/IKE. Each of the entries in the IPSec Security Association table controls both Main Mode and Quick Mode configuration for a single peer Note: You can also configure the IP Security Associations table using the ini file...
Page 150 Mediant 3000 Parameter Name Description Authentication Method Selects the method used for peer authentication during IKE [IPsecSATable_AuthenticationMetho main mode.  [0] Pre-shared Key (default)  [1] RSA Signature = in X.509 certificate Note: For RSA-based authentication, both peers must be provisioned with certificates signed by a common CA.
Page 151 SIP User's Manual 11. Security Parameter Name Description IPSec SA Lifetime (Kbs) Determines the maximum volume of traffic (in kilobytes) for [IPsecSATable_Phase2SaLifetimeInK which the negotiated IPSec SA (Quick mode) is valid. After this specified volume is reached, the SA is re-negotiated. The default value is 0 (i.e., the value is ignored).
Page 152 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 153: 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 547.
Page 154: Silence Suppression (Compression)
Mediant 3000  To configure gain control using the Web interface: Open the Voice Settings page (Configuration tab > VoIP menu > Media submenu > Voice Settings). Figure 12-2: Voice Volume Parameters in Voice Settings Page Configure the following parameters: •...
Page 155 SIP User's Manual 12. Media non-linear echo. To support this feature, the Forced Transcoding feature must be enabled so that the device uses DSPs. The procedure below describes how to configure echo cancellation using the Web interface:  To configure echo cancellation using the Web interface: Configure line echo cancellation: Open the Voice Settings page (Configuration tab >...
Page 156: Fax And Modem Capabilities
Mediant 3000 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 157)  Fax and modem transport modes (see 'Fax/Modem Transport Modes' on page 157) ...
Page 157: Fax/Modem Operating Modes
SIP User's Manual 12. Media Configure the parameters as required. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 478. Note: Some SIP parameters override these fax and modem parameters (see the parameter IsFaxUsed, and V.152 parameters in Section 'V.152 Support' on page 167).
Page 158: Fax Relay Mode
Mediant 3000 12.2.2.1 T.38 Fax Relay Mode In Fax Relay mode, fax signals are transferred using the T.38 protocol. T.38 is an ITU standard for sending fax across IP networks in real-time mode. The device currently supports only the T.38 UDP syntax.
Page 159: Fax / Modem Transport Mode
SIP User's Manual 12. Media • FaxRelayECMEnable • FaxRelayMaxRate 12.2.2.2 G.711 Fax / Modem Transport Mode In this mode, when the terminating device detects fax or modem signals (CED or AnsAM), it sends a Re-INVITE message to the originating device requesting it to re-open the channel in G.711 VBD with the following adaptations: ...
Page 160: Fax / Modem Nse Mode
Tip: When the remote (non-AudioCodes’) gateway uses G711 coder for voice and doesn’t change the coder payload type for fax or modem transmission, it is recommended to use the Bypass mode with the following configuration: •...
Page 161: Fax / Modem Transparent With Events Mode
The voice channel is optimized for fax/modem transmission (same as for usual bypass mode). The parameters defining payload type for the proprietary AudioCodes’ Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass. When configured for NSE mode, the device includes in its SDP the following line:...
Page 162: Rfc 2833 Ans Report Upon Fax/Modem Detection
Mediant 3000 To configure fax / modem transparent mode, use the following parameters:  IsFaxUsed = 0  FaxTransportMode = 0  V21ModemTransportType = 0  V22ModemTransportType = 0  V23ModemTransportType = 0  V32ModemTransportType = 0  V34ModemTransportType = 0 ...
Page 163: Bypass Mechanism For V.34 Fax Transmission
SIP User's Manual 12. Media Notes: • For T.38 Version 3 support, ensure that the relevant DSP firmware template is installed on the device (see DSP Templates on page a list of supported DSP templates) • The CNG detector is disabled (CNGDetectorMode = 0) in all the subsequent examples.
Page 164: Fax Relay For Sg3 Fax Machines
Mediant 3000 12.2.3.3 V.34 Fax Relay for SG3 Fax Machines Super Group 3 (SG3) is a standard for fax machines that support speeds of up to 33.6 kbps through V.34 half duplex (HD) modulation. This section describes how to configure the device to support V.34 (SG3) fax relay based on the ITU Specification T.38 version 3.
Page 165 Contact: <sip:318@10.8.6.55:5060> Supported: em,100rel,timer,replaces,path,resource-priority,sdp- anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB SCRIBE,UPDATE Remote-Party-ID: <sip:318@10.8.211.250>;party=calling;privacy=off;screen=no;screen- ind=0;npi=1;ton=0 Remote-Party-ID: <sip:2001@10.8.211.250>;party=called;npi=1;ton=0 User-Agent: Audiocodes-Sip-Gateway-/v.6.00A.013.007 Content-Type: application/sdp Content-Length: 433 o=AudiocodesGW 1938931006 1938930708 IN IP4 10.8.6.55 s=Phone-Call c=IN IP4 10.8.6.55 t=0 0 m=audio 6010 RTP/AVP 18 97 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no...
Page 166: Modem Relay
V.150.1 Software Upgrade Key. • The V.150.1 feature has been tested with certain IP phones. For more details, please contact your AudioCodes sales representative.  To configure V.150.1 Modem relay: In the Coders table, select the v1501mr coder.
Page 167: Support
SIP User's Manual 12. Media 12.2.5 V.152 Support The device supports the ITU-T recommendation V.152 (Procedures for Supporting Voice- Band Data over IP Networks). Voice-band data (VBD) is the transport of modem, facsimile, and text telephony signals over a voice channel of a packet network with a codec appropriate for such signals.
Page 168: Configuring Rtp/Rtcp Settings
Mediant 3000 least one IP packet from the LAN to the WAN through the firewall. If the firewall blocks T.38 packets sent from the termination IP fax, the fax fails. To overcome this, the device sends No-Op (“no-signal”) packets to open a pinhole in the NAT for the answering fax machine.
Page 169: Dynamic Jitter Buffer Operation
SIP User's Manual 12. Media 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 170: Comfort Noise Generation
Mediant 3000 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 > RTP/RTCP Settings).
Page 171: Mediant
SIP User's Manual 12. Media  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 172: Configuring Rfc 2833 Payload
Mediant 3000 12.3.3.2 Configuring RFC 2833 Payload The procedure below describes how to configure the RFC 2833 payload using the Web interface:  To configure RFC 2833 payload using the Web interface: Open the RTP/RTCP Settings page (Configuration tab > VoIP menu > Media submenu >...
Page 173: Configuring Rtp Base Udp Port
The maximum (when all channels are required) UDP port range is calculated as follows:  Mediant 3000/TP-6310: BaseUDPport to (BaseUDPport + 4031*10) - for example, if the BaseUDPPort is set to 6,000, then the UDP port range is 6,000 to 46,310 ...
Page 174: Configuring Rtcp Xr Using Web Interface
Mediant 3000 XR measures VoIP call quality such as packet loss, delay, signal / noise / echo levels, estimated R-factor, and mean opinion score (MOS). RTCP XR measures these parameters using metrics (refer to the Product Reference Manual). RTCP XR messages containing key call-quality-related metrics are exchanged periodically (user-defined) between the device and the SIP UA.
Page 175: Configuring Ip Media Settings
SIP User's Manual 12. Media 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 547. ...
Page 176: Answer Machine Detector (Amd)
Mediant 3000 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 177 SIP User's Manual 12. Media Performance AMD Detection Sensitivity Success Rate for Live Calls Success Rate for Answering Machine 90.42% 91.64% 90.66% 91.30% 7 (Best for Live 94.72% 76.14% Calls) Table 12-3: Approximate AMD Detection High Sensitivity (Based on North American English) Performance AMD Detection Sensitivity...
Page 178 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 179 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 180: Automatic Gain Control (Agc)
Mediant 3000 12.4.2 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 181: Configuring General Media Settings
SIP User's Manual 12. Media 12.5 Configuring General Media Settings The General Media Settings page allows you to configure various media parameters. For a detailed description of the parameters appearing on this page, see 'Configuration Parameters Reference' on page 547. ...
Page 182: Configuring Media Realms
208) or SRDs (in the SRD table - see 'Configuring SRD Table' on page 203). For each Media Realm you can configure Quality of Experience parameters and their thresholds for reporting to the AudioCodes SEM server used for monitoring the quality of calls. For configuring this, see 'Configuring Quality of Experience Parameters per Media Realm' on page 186.
Page 183 SIP User's Manual 12. Media  To define a Media Realm: Open the Media Realm Table page (Configuration tab > VoIP menu > Media submenu > Media Realm Configuration). Click the Add button; the following appears: Figure 12-9: Add Record Dialog Box Configure the parameters as required.
Page 184 (not some with and some without).  The available UDP port range is calculated using the BaseUDPport parameter:  Mediant 3000/TP-6310: BaseUDPport to BaseUDPport + 4031*10. For example, if BaseUDPPort is 6000 (default), then the available port range is 6000-46310 ...
Page 185: 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 547.  To configure media security: Open the Media Security page (Configuration tab >...
Page 186: Configuring Quality Of Experience Parameters Per Media Realm
The QoE feature is available only if the device is installed with the relevant Software Upgrade Key. • To configure the address of the AudioCodes Session Experience Manager (SEM) server to where the device reports the QoE, see 'Configuring Server for Media Quality of Experience' on page 189.
Page 187 SIP User's Manual 12. Media Click the Add button; the Add Record dialog box appears: Figure 12-11: Add Record Dialog Box for QoE The figure above shows value thresholds for the MOS parameter, which are assigned using pre-configured values of the Low Sensitivity profile. In this example setting, if the MOS value changes by 0.1 (hysteresis) to 3.3 or 3.5, the device sends a report to the SEM indicating this change.
Page 188 Mediant 3000 Parameter Description Profile Defines the pre-configured threshold profile to use. [QOERules_Profile]  No Profile = No profile is used and you need to define the thresholds in the parameters described below.  Low Sensitivity = Automatically sets the thresholds to low sensitivity values.
Page 189: Configuring Server For Media Quality Of Experience
12. Media 12.10 Configuring Server for Media Quality of Experience The device can be configured to report voice (media) quality of experience to AudioCodes Session Experience Manager (SEM) server, a plug-in for AudioCodes EMS. The reports include real-time metrics of the quality of the actual call experience and processed by the SEM.
Page 190 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 191: 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 192: Configuring Ldap Settings
Mediant 3000 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 547. For an overview of LDAP, see 'Routing Based on LDAP Active Directory Queries' on page 191.
Page 193 SIP User's Manual 13. Services The device queries the AD using the destination number. The device's AD queries return up to three user phone number IP destinations, each pertaining to one of the IP domains listed above. The device routes the call according to the following priority: OCS SIP address: The call is routed to Mediation Server (which then routes the call to the OCS client).
Page 194 Mediant 3000 Below is an example for configuring AD-based routing rules in the Outbound IP Routing Table (see 'Configuring Outbound IP Routing Table' on page 298): 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 195: 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. Note: The LCR feature is applicable only to the GW/IP2IP application. 13.2.1 Overview The LCR feature enables the device to choose the outbound IP destination routing rule based on lowest call cost.
Page 196 Mediant 3000 If four matching routing rules are located in the routing table and each one is assigned a different Cost Group as listed in the table above, then the rule assigned Cost Group "D" is selected. Note that for one minute, Cost Groups "A" and "D" are identical, but due to the average call duration, Cost Group "D"...
Page 197 SIP User's Manual 13. Services The device calculates the optimal route in the following index order: 3, 1, 2, and then 4, due to the following logic: • Index 1 - Cost Group "A" has the lowest connection cost and minute cost •...
Page 198: Configuring Lcr
Mediant 3000 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 198.
Page 199: Configuring Cost Groups
SIP User's Manual 13. Services Parameter Description LCR Call Length Defines the average call duration (in minutes) and is used to [RoutingRuleGroups_LCRAverage calculate the variable portion of the call cost. This is useful, for CallLength] example, when the average call duration spans over multiple time bands.
Page 200: Configuring Time Bands For Cost Groups
Mediant 3000 Configure the parameters as required. For a description of the parameters, see the table below. Click Submit; the entry is added to the Cost Group table. Table 13-8: Cost Group Table Description Parameter Description Index Defines the table index entry.
Page 201 SIP User's Manual 13. Services  To configure Time Bands for a Cost Group: Open the Cost Group Table page (Configuration tab > VoIP menu > Services submenu > Least Cost Routing > Cost Group Table). Select a Cost Group for which you want to assign Time Bands, and then click the Time Band link located below the table;...
Page 202: Assigning Cost Groups To Routing Rules
Mediant 3000 13.2.2.4 Assigning Cost Groups to Routing Rules Once you have configured your Cost Groups, you need to assign them to routing rules in the Outbound IP Routing table. For more information, see 'Configuring Outbound IP Routing Table' on page 298.
Page 203: 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 182).
Page 204 Mediant 3000  To configure SRDs: Open the SRD Settings page (Configuration tab > VoIP menu > Control Network submenu > SRD Table). Figure 14-1: SRD Settings Page From the 'SRD Index' drop-down list, select an index for the SRD, and then configure it according to the table below.
Page 205 SIP User's Manual 14. Control Network Parameter Description Internal SRD Media Anchoring Determines whether the device performs media anchoring or not on [SRD_IntraSRDMediaAnchori media for the SRD.  [0] Anchor Media (default) = RTP traverses the device and each leg uses a different coder or coder parameters. ...
Page 206: Configuring Sip Interface Table
Mediant 3000 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. A SIP Interface consists of a combination of ports (UDP, TCP, and TLS), associated with a specific IP address (IPv4 / IPv6) , and for a specific application (i.e., SAS, Gateway\IP2IP, and SBC).
Page 207 SIP User's Manual 14. Control Network Parameter Description Interfaces must be defined with the same network interface (e.g., "SIP1"). Application Type Defines the application type associated with the SIP Interface. [SIPInterface_ApplicationType]  [0] GW/IP2IP (default) = IP-to-IP routing application and regular gateway functionality ...
Page 208: Configuring Ip Groups
Mediant 3000 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 215), which represents the IP address of the IP Group.
Page 209 SIP User's Manual 14. Control Network  To configure IP Groups: Open the IP Group Table page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). Figure 14-2: IP Group Table Configure the IP group parameters according to the table below. Click Submit to apply your changes.
Page 210 Mediant 3000 Parameter Description USER-type IP Group. Each SIP request sent by a user of this IP Group is proxied to the Serving IP Group. For registrations, the device updates its internal database with the AOR and contacts of the users.
Page 211 SIP User's Manual 14. Control Network Parameter Description SIP Group Name The SIP Request-URI host name used in INVITE and REGISTER [IPGroup_SIPGroupName] messages sent to the IP Group, or the host name in the From header of INVITE messages received from the IP Group. If not specified, the value of the global parameter, ProxyName (see 'Configuring Proxy and Registration Parameters' on page 244) is used instead.
Page 212 Mediant 3000 Parameter Description Routing Mode Defines the routing mode for outgoing SIP INVITE messages. [IPGroup_RoutingMode]  [-1] Not Configured = The routing is according to the selected Serving IP Group. If no Serving IP Group is selected, the device routes the call according to the Outbound IP Routing Table' (see Configuring Outbound IP Routing Table on page 298).
Page 213 SIP User's Manual 14. Control Network Parameter Description Enable Survivability Determines whether Survivability mode is enabled for USER-type IP [IPGroup_EnableSurvivabilit Groups.  [0] Disable (default).  [1] Enable if Necessary = Survivability mode is enabled. The device records in its database the registration messages sent by the clients belonging to the USER-type IP Group.
Page 214 Mediant 3000 Parameter Description  This parameter is applicable only to SERVER-type IP Groups.  This classification is not relevant in cases where multiple IP Groups use the same Proxy Set. Max Number Of Registered Maximum number of users belonging to this IP Group that can register Users with the device.
Page 215: Configuring Proxy Sets Table
SIP User's Manual 14. Control Network 14.4 Configuring Proxy Sets Table The Proxy Sets Table page allows you to define Proxy Sets. A Proxy Set is a group of Proxy servers defined by IP address or fully qualified domain name (FQDN). You can define up to 32 Proxy Sets, each with a unique ID number and up to five Proxy server addresses.
Page 216 Mediant 3000  To add Proxy servers: Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control Network submenu > Proxy Sets Table). Figure 14-3: Proxy Sets Table Page From the 'Proxy Set ID' drop-down list, select an ID for the desired group.
Page 217 SIP User's Manual 14. Control Network Parameter Description  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 218 Mediant 3000 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...
Page 219 SIP User's Manual 14. Control Network Parameter Description sends the requests in such a fashion that each Proxy receives a 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: ...
Page 220: Configuring Nat Translation Per Ip Interface
Mediant 3000 14.5 Configuring NAT Translation per IP Interface The NAT Translation table defines NAT rules for translating source IP addresses per VoIP interface (SIP control and RTP media traffic) into NAT IP addresses (global). This allows, for example, the separation of VoIP traffic between different ISTP’s, and topology hiding of internal IP addresses to the “public”...
Page 221: Multiple Sip Signaling And Media Interfaces Using Srds
SIP User's Manual 14. Control Network 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, and SBC) ...
Page 222 Mediant 3000  Define it as a destination SRD for IP-to-IP routing rules (see 'Configuring IP-to-IP Routing Table' on page 402). Routing from one SRD to another is possible, where each routing destination (IP Group or destination address) indicates the SRD to which it belongs.
Page 223 SIP User's Manual 14. Control Network The figure below illustrates two SRD's - one for Network-1 and one for Network-2. Each application (i.e., SAS, Gateway\IP2IP, and SBC) pertains to the same SRD, but each has its own SIP interface. The figure below illustrates the SBC call flow between an enterprises LAN (IP PBX) and an ITSP (Network-2) implementing different interfaces (IP addresses and ports) for RTP packets and SIP signaling.
Page 224 Mediant 3000 Below provides an example for configuring multiple SIP signaling and RTP interfaces. In this example, the device serves as the interface between the enterprise's PBX (connected using an E1/T1 trunk) and two ITSP's, as shown in the figure below:...
Page 225 SIP User's Manual 14. Control Network Configure the trunk in the Trunk Settings page (Configuration tab > VoIP menu > GW and IP to IP submenu > Trunk Group > Trunk Group Settings). Configure the IP interfaces in the Multiple Interface table (Configuration tab > VoIP menu >...
Page 226 Mediant 3000 Configure Proxy Sets in the Proxy Sets Table page (Configuration tab > VoIP menu > Control Network submenu > Proxy Sets Table). The figure below configures ITSP A. Do the same for ITSP B but for Proxy Set 2 with IP address 212.179.95.100 and SRD 2.
Page 227 SIP User's Manual 14. Control Network Configure IP-to-Trunk Group routing in the Inbound IP Routing Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing submenu > IP to Trunk Group Routing): Figure 14-14: Defining IP-to-Trunk Group Routing Configure Trunk Group-to-IP routing in the Outbound IP Routing Table page (Configuration tab >...
Page 228 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 229: Enabling Applications
Software Upgrade Key supporting the application (see 'Loading Software Upgrade Key' on page 493). • The SAS application is not applicable to Mediant 3000 HA. • For configuring the SAS application, see 'Stand-Alone Survivability (SAS) Application' on page 443.
Page 230 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 231: 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 232: Configuring Coder Groups
Mediant 3000 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. The packetization time determines how many coder payloads are combined into a single RTP packet.
Page 233: Configuring Tel Profile
SIP User's Manual 16. Coders and Profiles  To configure Coder Groups: Open the Coder Group Settings page (Configuration tab > VoIP menu > Coders And Profiles submenu > Coders Group Settings). Figure 16-2: Coder Group Settings Page From the 'Coder Group ID' drop-down list, select a Coder Group ID. From the 'Coder Name' drop-down list, select the first coder for the Coder Group.
Page 234 Mediant 3000 Note: You can also configure Tel Profiles using the ini file table parameter TelProfile (see 'Configuration Parameters Reference' on page 547).  To configure Tel Profiles: Open the Tel Profile Settings page (Configuration tab > VoIP menu > Coders And Profiles submenu >...
Page 235 SIP User's Manual 16. Coders and Profiles From the 'Profile Preference' drop-down list, select the priority of the Tel Profile, where 1 is the lowest priority and 20 the highest. If both IP and Tel profiles apply to the same call, the coders and other common parameters (noted by an asterisk in the description of the parameter TelProfile) of the preferred Profile are applied to that call.
Page 236: Configuring Ip Profiles
Mediant 3000 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 237 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). Figure 16-4: IP Profile Settings Page 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 238 Mediant 3000 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 239: 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 547.
Page 240: Configuring Advanced Parameters
Mediant 3000 Configure the parameters as required. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 478. 17.2 Configuring Advanced Parameters The Advanced Parameters page allows you to configure advanced SIP control parameters.
Page 241: 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 242 Mediant 3000 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 243 SIP User's Manual 17. SIP Definitions Parameter Description 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. In addition, to activate registration, you also need to set the parameter 'Registration Mode' to 'Per Account' in the Trunk Group Settings table for the specific Trunk Group.
Page 244: Configuring Proxy And Registration Parameters
Mediant 3000 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 547.
Page 245: 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 246: Configuring Digital Pcm Settings
Mediant 3000 18.1.1.2 Configuring Digital PCM Settings The Digital PCM Settings page allows you to configure the PCM companding law in input/output TDM bus, and PCM and ABCD patterns. For a description of these parameters, see 'PSTN Parameters' on page 675.
Page 247 SIP User's Manual 18. GW and IP to IP 18.1.1.3.1 Clock Settings In a traditional TDM service network such as PSTN, both ends of the TDM connection must be synchronized. If synchronization is not achieved, voice frames are either dropped (to prevent a buffer overflow condition) or inserted (to prevent an underflow condition).
Page 248 Mediant 3000 When the device has BITS capability, configure the following parameter in addition to the above parameters:  TMMode =2 - defines the BITS mode (set this to 2 for line interface clock) Notes: • When operating in line recovery mode and the device is installed with the BITS timing hardware, when the source link/trunk clock reference fails, both active and redundant blades change to hold-over mode.
Page 249 SIP User's Manual 18. GW and IP to IP  TMReferenceValidationTime: In order for a BITS reference to be returned to service, the device must validate the clock. This parameter defines the validation time: TMReferenceValidationTime = [1 - 15] min (default 1 min) - time were reference must have an alarm cleared before the device declares it as a valid reference ...
Page 250: Configuring Transmission Settings
T3/DS3 or SONET/SDH). For a description of the parameters related to transmission type, see 'PSTN Parameters' on page 675. Notes: • Transmission settings are applicable only to Mediant 3000 with TP-6310. • When applying your settings, traffic disturbances may be experienced. ...
Page 251: Configuring Cas State Machines
SIP User's Manual 18. GW and IP to IP Click Submit to apply your changes. Reset the device and save your settings to the flash memory (see 'Resetting the Device' on page 475). For SDH/SONET, to view the 'KLM Numbering Mapping Table page, click Go to the KLM Mapping Table button.
Page 252 Mediant 3000 Notes: • Don't modify the default values unless you fully understand the implications of the changes and know the default values. Every change affects the configuration of the state machine parameters and the call process related to the trunk you are using with this state machine.
Page 253: Configuring Trunk Settings
SIP User's Manual 18. GW and IP to IP Parameter Description Collet ANI In some cases, when the state machine handles the ANI [CasStateMachineCollectANI] collection (not related to MFCR2), you can control the state machine to collect ANI or discard ANI. ...
Page 254 Mediant 3000  To configure the trunks: Open the Trunk Settings page (Configuration tab > VoIP menu > PSTN submenu > Trunk Settings). Figure 18-7: Trunk Settings Page (Partial Display) On the top of the page, a bar with Trunk number icons displays the status of each trunk, according to the following color codes: •...
Page 255 SIP User's Manual 18. GW and IP to IP Note: If the Trunk scroll bar displays all available trunks, the scroll bar buttons are unavailable. After you have selected a trunk, the following is displayed: • The read-only 'Trunk ID' field displays the selected trunk number. •...
Page 256: Configuring Digital Gateway Parameters
Mediant 3000 18.1.5 Configuring Digital Gateway Parameters The Digital Gateway Parameters page allows you to configure miscellaneous digital parameters. For a description of these parameters, see 'Configuration Parameters Reference' on page 547.  To configure the digital gateway parameters: Open the Digital Gateway Parameters page (Configuration tab > VoIP menu > GW and IP to IP submenu >...
Page 257: Tunneling Applications
SIP User's Manual 18. GW and IP to IP 18.1.6 Tunneling Applications This section discusses the device's support for VoIP tunneling applications. 18.1.6.1 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.
Page 258 Mediant 3000 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'). Note: For TDM over IP, the parameter CallerIDTransportType must be set to '0' (disabled), i.e., transparent.
Page 259 SIP User's Manual 18. GW and IP to IP [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 260: Qsig Tunneling
Mediant 3000 TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP; TelProfile_1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$ TelProfile_2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$ [\TelProfile] 18.1.6.1.1 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 261: Advanced Pstn Configuration
SIP User's Manual 18. GW and IP to IP Setup message to the Tel side and sends a 200 OK response (no 1xx response is sent) to IP. The 200 OK response includes an encapsulated QSIG Call Proceeding message (without waiting for a Call Proceeding message from the Tel side). If tunneling is disabled and the incoming INVITE includes a QSIG body, a 415 response is sent.
Page 262 Mediant 3000 18.1.7.1.1 Reason Header The device supports the Reason header according to RFC 3326. The Reason header conveys information describing the disconnection cause of a call:  Sending Reason header: If a call is disconnected from the Tel side (ISDN), the Reason header is set to the received Q.850 cause in the appropriate message...
Page 263 SIP User's Manual 18. GW and IP to IP ISDN Release Description Description Reason Response No circuit available Service unavailable Network out of order Service unavailable Temporary failure Service unavailable Switching equipment congestion Service unavailable Access information discarded 502* Bad gateway Requested channel not available 503* Service unavailable...
Page 264 Mediant 3000 ISDN Release Description Description Reason Response Message not compatible with call state or message type non-existent or not 409* Conflict implemented Information element non-existent or not 480* Not found implemented Invalid information elements contents 501* Not implemented Message not compatible with call state...
Page 265: Isdn Overlap Dialing
SIP User's Manual 18. GW and IP to IP ISDN Release SIP Response Description Description Reason exist 482* Loop detected Interworking Too many hops Interworking Address incomplete Invalid number format Ambiguous Unallocated number Busy here User busy Not acceptable here Normal, unspecified Server internal error Temporary failure...
Page 266 Mediant 3000 By default (see the ISDNINCallsBehavior parameter), the device plays a dial tone to the ISDN user side when it receives an empty called number from the ISDN. In this scenario, the device includes the Progress Indicator in the SetupAck ISDN message that it sends to the ISDN side.
Page 267: Isdn Non-Facility Associated Signaling (Nfas)
SIP User's Manual 18. GW and IP to IP  DigitMapping  MuteDTMFInOverlap For configuring ISDN overlap dialing using the Web interface, see 'Configuring Trunk Settings' on page 253. 18.1.7.3 ISDN Non-Facility Associated Signaling (NFAS) In regular T1 ISDN trunks, a single 64 kbps channel carries signaling for the other 23 B- channels of that particular T1 trunk.
Page 268 Mediant 3000 To define an explicit Interface ID for a T1 trunk (that is different from the default), use the following parameters:  ISDNIBehavior_x = 512 (x = 0 to the maximum number of trunks identifying the device's physical trunk) ...
Page 269 SIP User's Manual 18. GW and IP to IP If there is no NFAS Backup trunk, the following configuration should be used: ISDNNFASInterfaceID_0 = 0 ISDNNFASInterfaceID_1 = 2 ISDNNFASInterfaceID_2 = 3 ISDNNFASInterfaceID_3 = 4 ISDNIBehavior = 512 ;This parameter should be added because of ;ISDNNFASInterfaceID coniguration above NFASGroupNumber_0 = 1 NFASGroupNumber_1 = 1...
Page 270: Sdh / Sonet Configuration
Mediant 3000 18.1.7.4 SDH / SONET Configuration Note: This section is only relevant for PSTN STM-1 / OC-3 ports on the TP-6310 blade. The device supports both STM-1 and OC-3 optical fiber transmission modes. Optical fiber transmission is configured using the parameter PSTNTransmissionType (set to 1) and the fiber mode is defined using the parameter SdhFbrGrp_SdhSonetMode (set to 1 for STM-1, or 2 for OC-3).
Page 271 Once the scheme has been selected, the corresponding table is automatically determined for 63 or 84 tributaries (trunks). Note: This section is only applicable to Mediant 3000 with the TP-6310 blade(s). 18.1.7.4.1.1 E1 Trunk Enumeration (SDH Mappings) The following table is used for converting internal STM-1 (KLM) numbering to sequential trunk numbering for API references.
Page 272 Mediant 3000 Trunk ETSI GR-253 Timeslots TUG-3 TUG-2 TU-12 TUG-3 TUG-2 TU-12 TUG-3 TUG-2 TU-12 SIP User's Manual Document #: LTRT-89712...
Page 273 SIP User's Manual 18. GW and IP to IP Trunk ETSI GR-253 Timeslots TUG-3 TUG-2 TU-12 TUG-3 TUG-2 TU-12 TUG-3 TUG-2 TU-12 18.1.7.4.1.2 T1 Trunk Enumeration (Sonet Mappings) The following table is used for converting internal OC-3 numbering to sequential trunk numbering for API references.
Page 274 Mediant 3000 Trunk ETSI GR-253 Timeslots STS-1 VT1.5 STS-1 VT1.5 STS-1 VT1.5 SIP User's Manual Document #: LTRT-89712...
Page 275 SIP User's Manual 18. GW and IP to IP Trunk ETSI GR-253 Timeslots STS-1 VT1.5 STS-1 VT1.5 STS-1 VT1.5 Version 6.4 November 2011...
Page 276: Redirect Number And Calling Name (Display)
Mediant 3000 18.1.7.5 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 18-6: Calling Name (Display)
Page 277: Trunk Group
SIP User's Manual 18. GW and IP to IP 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 278 Mediant 3000 Table 18-8: 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. The number...
Page 279: Configuring Trunk Group Settings
SIP User's Manual 18. GW and IP to IP Parameter Description 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 280 Mediant 3000 An example is shown below of a REGISTER message for registering endpoint "101" using registration Per Endpoint mode. The "SipGroupName" in the Request-URI is defined in the IP Group table (see 'Configuring IP Groups' on page 208). REGISTER sip:SipGroupName SIP/2.0 Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac862428454...
Page 281 SIP User's Manual 18. GW and IP to IP Parameter Description according to the settings in the Account table (see 'Configuring Account Table' on page 241). Notes:  To enable Trunk Group registrations, configure the global parameter IsRegisterNeeded to 1. This is unnecessary for 'Per Account' registration mode.
Page 282: Manipulation
Mediant 3000 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 547.
Page 283 SIP User's Manual 18. GW and IP to IP 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.
Page 284 Mediant 3000  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); the relevant Manipulation table page is displayed (e.g., 'Source Phone Number Manipulation Table...
Page 285 SIP User's Manual 18. GW and IP to IP 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 286 Mediant 3000 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 287: Configuring Redirect Number Ip To Tel
SIP User's Manual 18. GW and IP to IP 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 288 Mediant 3000 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 289: Configuring Redirect Number Tel To Ip
SIP User's Manual 18. GW and IP to IP 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 290 Mediant 3000 Table 18-12: 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:  The value -1 indicates that this field is ignored in the rule.
Page 291: Mapping Npi/Ton To Sip Phone-Context
SIP User's Manual 18. GW and IP to IP 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 292 Mediant 3000 Table 18-13: 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.  [0] Disable (default) ...
Page 293: Numbering Plans And Type Of Number
SIP User's Manual 18. GW and IP to IP 18.3.6 Numbering Plans and Type of Number The IP-to-Tel destination or source number manipulation tables allow you to classify numbers by their Numbering Plan Indication (NPI) and Type of Number (TON). The device supports all NPI/TON classifications used in the standard.
Page 294: Configuring Release Cause Mapping
Mediant 3000 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 261).
Page 295: Sip Calling Name Manipulations
SIP User's Manual 18. GW and IP to IP 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 296: Manipulating Number Prefix
Mediant 3000 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 282): x[n,l]y...
Page 297: Routing
SIP User's Manual 18. GW and IP to IP 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 547.
Page 298: Configuring Outbound Ip Routing Table
Mediant 3000 18.4.2 Configuring Outbound IP Routing Table The Outbound IP Routing Table page allows you to configure up to 180 Tel-to-IP/outbound IP call routing rules. The device uses these rules to route calls from the Tel or IP to user- defined IP destinations.
Page 299 SIP User's Manual 18. GW and IP to IP Since each call must have a destination IP Group (even in cases where the destination type is not to an IP Group), in cases when the IP Group is not specified, the SRD's default IP Group is used (the first defined IP Group that belongs to the SRD).
Page 300 Mediant 3000 In addition to basic outbound IP routing, supports the following features:  Least cost routing (LCR): If the LCR feature is enabled, the device searches the routing table for matching routing rules and then selects the one with the lowest call cost.
Page 301 SIP User's Manual 18. GW and IP to IP  To configure outbound IP routing rules: Open the Outbound IP Routing Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing submenu > Tel to IP Routing). Figure 18-20: Tel to IP Routing Page The figure above displays the following outbound IP routing rules: •...
Page 302 Mediant 3000 From the 'Routing Index' drop-down list, select the range of entries that you want to add. Configure the routing rules according to the table below. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 478.
Page 303 SIP User's Manual 18. GW and IP to IP Parameter Description Web/EMS: Source Defines the prefix and/or suffix of the calling (source) telephone number. For Phone Prefix example, [100-199](100,101,105) depicts a number that starts with 100 to 199 and ends with 100, 101 or 105. For a description of notations that you can use to represent single and multiple numbers (ranges), see 'Dialing Plan Notation for Routing and Manipulation Tables' on page 775.
Page 304 Mediant 3000 Parameter Description Web: Dest IP Group Defines the IP Group to where you want to route the call. The SIP INVITE message is sent to the IP address defined for the Proxy Set ID associated EMS: Destination IP with the IP Group.
Page 305 SIP User's Manual 18. GW and IP to IP Parameter Description Forking Group Defines a forking group ID for the routing rule. This enables forking of incoming Tel calls to two or more IP destinations. The device sends simultaneous INVITE messages and handles multiple SIP dialogs until one of the calls is answered.
Page 306: Configuring Inbound Ip Routing Table
Mediant 3000 18.4.3 Configuring Inbound IP Routing Table The Inbound IP Routing Table page allows you to configure up to 120 inbound call routing rules:  For IP-to-IP routing: identifying IP-to-IP calls and assigning them to IP Groups (referred to as Source IP Groups). These IP-to-IP calls, now pertaining to an IP Group, can later be routed to an outbound destination IP Group (see Configuring Outbound IP Routing Table on page 298).
Page 307 SIP User's Manual 18. GW and IP to IP The previous figure displays the following configured routing rules: • Rule 1: If the incoming IP call destination phone prefix is between 10 and 19, the call is assigned settings configured for IP Profile ID 2 and routed to Trunk Group ID 1.
Page 308: Configuring Alternative Routing Reasons
Mediant 3000 Parameter Description The prefix can include up to 49 digits. Source IP Address The source IP address of the incoming IP call (obtained from the Contact header in the INVITE message) that can be used for routing decisions.
Page 309 SIP User's Manual 18. GW and IP to IP can be configured, for example, when there is no response to an INVITE message (after INVITE re-transmissions), the device issues an internal 408 'No Response' implicit release reason. Notes: • To enable alternative routing using the IP-to-Tel routing table, set the parameter RedundantRoutingMode to 1 (default).
Page 310: Configuring Call Forward Upon Busy Trunk
Mediant 3000 18.4.5 Configuring Call Forward upon Busy Trunk The Forward on Busy Trunk Destination page allows you to configure forwarding (call redirection) of IP-to-Tel calls to a different (alternative) IP destination, using SIP 3xx responses upon the following scenario: ...
Page 311: 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 312 Mediant 3000  Message waiting indication (MWI)- see 'Message Waiting Indication' on page  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.
Page 313: 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 314: Call Forward
Mediant 3000 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 315: 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 316: Emergency E911 Phone Number Services
Mediant 3000 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 317 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 318: Denial Of Collect Calls
Mediant 3000 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 319: Configuring Voice Mail Parameters
SIP User's Manual 18. GW and IP to IP 18.5.3 Configuring Voice Mail Parameters The Voice Mail Settings page allows you to configure the voice mail parameters. For a description of these parameters, see 'Configuration Parameters Reference' on page 547. Notes: •...
Page 320: Advice Of Charge Services For Euro Isdn
Mediant 3000 18.5.4 Advice of Charge Services for Euro ISDN Advice of charge (AOC) is a pre-billing function that tasks the rating engine with calculating the cost of using a service and relaying that information back to the customer thus, allowing users to obtain charging information for all calls during the call (AOC-D) or at the end of the call (AOC-E), or both.
Page 321: Dialing Plan Features
SIP User's Manual 18. GW and IP to IP 18.6 Dialing Plan Features This section discusses various dialing plan features supported by the device:  Digit mapping (see 'Digit Mapping' on page 321)  External Dial Plan file containing dial plans (see 'External Dial Plan File' on page 322) ...
Page 322: External Dial Plan File
Mediant 3000 In the example, the rule "00[1-7]xxx" denotes dialed numbers that begin with 00, and then any digit from 1 through 7, followed by three digits (of any number). Once the device receives these digits, it does not wait for additional digits, but starts sending the collected digits (dialed number) immediately.
Page 323 SIP User's Manual 18. GW and IP to IP An example of a Dial Plan file with indices (in ini-file format before conversion to binary .dat) is shown below: [ PLAN1 ] ; Area codes 02, 03, - phone numbers include 7 digits. 02,7 03,7 ;...
Page 324: Modifying Isdn-To-Ip Calling Party Number
Mediant 3000 18.6.2.1 Modifying ISDN-to-IP Calling Party Number The device can use the Dial Plan file to change the Calling Party Number value (source number) of the incoming ISDN call when sending to IP. For this feature, the Dial Plan file supports the following syntax: <ISDN Calling Party Number>,0,<new calling number>...
Page 325: Dial Plan Prefix Tags For Ip-To-Tel Routing
SIP User's Manual 18. GW and IP to IP 18.6.3 Dial Plan Prefix Tags for IP-to-Tel Routing The device supports the use of string labels (or "tags") in the external Dial Plan file 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 326: Configuring Alternative Routing (Based On Connectivity And Qos)
Mediant 3000 Assign the different tag prefixes to different Trunk Groups in the Inbound IP Routing Table' (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing submenu > IP to Trunk Group Routing): • The Dest. Phone Prefix' field is set to the value "LOCL" and this rule is assigned to a local Trunk Group (e.g.
Page 327: Alternative Routing Mechanism
SIP User's Manual 18. GW and IP to IP 18.7.1 Alternative Routing Mechanism When the device routes a Tel-to-IP call, the destination number is compared to the list of prefixes defined in the Outbound IP Routing Table (described in 'Configuring the Outbound IP Routing Table' on page 298).
Page 328: Sip Call Routing Examples
F1 INVITE (10.8.201.108 >> 10.8.201.161): INVITE sip:2000@10.8.201.161;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161> Call-ID: 534366556655skKw-6000--2000@10.8.201.108 CSeq: 18153 INVITE Contact: <sip:8000@10.8.201.108;user=phone> User-Agent: Audiocodes-Sip-Gateway/Mediant 3000/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...
Page 329  F2 TRYING (10.8.201.161 >> 10.8.201.108): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161> Call-ID: 534366556655skKw-6000--2000@10.8.201.108 Server: Audiocodes-Sip-Gateway/Mediant 3000/v.6.40.010.006 CSeq: 18153 INVITE Content-Length: 0  F3 RINGING 180 (10.8.201.161 >> 10.8.201.108): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161>;tag=1c7345...
Page 330: Sip Message Authentication Example
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 3000/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 331 Since the algorithm is MD5: • The username is equal to the endpoint phone number 122. • The realm return by the proxy is audiocodes.com. • The password from the ini file is AudioCodes. • The equation to be evaluated is (according to RFC this part is called A1) ‘122:audiocodes.com:AudioCodes’.
Page 332 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 3000/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 333: Trunk-To-Trunk Routing Example
SIP User's Manual 18. GW and IP to IP 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 334: Sip Trunking Between Enterprise And Itsps
Mediant 3000 18.8.4 SIP Trunking between Enterprise and ITSPs By implementing the device's enhanced and flexible routing capabilities, you can design complex routing schemes. This section provides an example of an elaborate routing scheme for SIP trunking between an Enterprise's PBX and two Internet Telephony Service Providers (ITSP), using the device.
Page 335 SIP User's Manual 18. GW and IP to IP  To configure call routing between an Enterprise and two ITSPs: Enable the device to register to a Proxy/Registrar server using the parameter IsRegisterNeeded. In the Proxy Sets Table page (see 'Configuring Proxy Sets Table' on page 215), configure two Proxy Sets and for each, enable Proxy Keep-Alive (using SIP OPTIONS) and 'round robin' load-balancing method: •...
Page 336 Mediant 3000 In the IP Group Table page (see 'Configuring IP Groups' on page 208), configure the two IP Groups #1 and #2. Assign Proxy Sets #1 and #2 to IP Groups #1 and #2 respectively. Figure 18-31: Configuring IP Groups #1 and #2 in the IP Group Table Page...
Page 337 SIP User's Manual 18. GW and IP to IP In the Inbound IP Routing Table page, configure IP-to-Tel routing for calls from ITSPs to Trunk Group ID #1 (see 1 below) and from the device to the local PSTN (see 2 below).
Page 338: Ip-To-Ip Routing Application
Mediant 3000 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 339: Theory Of Operation
SIP User's Manual 18. GW and IP to IP  UPDATE: terminated at each leg independently and may cause only changes in the RTP flow - Hold\Retrieve are the only exceptions that traverse the two legs.  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 340: Proxy Sets
Mediant 3000 Number manipulation can be performed at both legs (inbound and outbound). The following subsections discuss the main terms associated with the IP-to-IP call routing application. 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).
Page 341 SIP User's Manual 18. GW and IP to IP Figure 18-37: IP-to-IP Routing/Registration/Authentication of Remote IP-PBX Users (Example) 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.
Page 342: Inbound And Outbound Ip Routing Rules
Mediant 3000 18.9.1.3 Inbound and Outbound IP Routing Rules The device's IP-to-IP call routing is performed using the following two routing rule stages: Inbound IP Routing Mapping Rule: Identifies the received call as an IP-to-IP call based on various characteristics such as the call's source IP address, and assigns it to an IP Group.
Page 343 SIP User's Manual 18. GW and IP to IP This example assumes the following:  The device has the public IP address 212.25.125.136 and is connected to the enterprise's firewall/NAT demilitarized zone (DMZ) network, providing the interface between the IP-PBX, and two ITSP's and the local PSTN. ...
Page 344 Mediant 3000 The figure below provides an illustration of this example scenario: Figure 18-40: 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 345).
Page 345: 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 354).  Configure outbound IP routing rules (see 'Step 9: Configure Outbound IP Routing' on page 355). ...
Page 346: Step 3: Define A Trunk Group For The Local Pstn
Mediant 3000 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 347 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 18-44: 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 348: Step 5: Configure The Ip Groups
Mediant 3000 Configure Proxy Set ID #3 for the IP-PBX: From the 'Proxy Set ID' drop-down list, select 3. In the 'Proxy Address' column, enter the IP address of the IP-PBX (e.g., 10.15.4.211). From the 'Transport Type' drop-down list corresponding to the IP address entered above, select UDP".
Page 349 SIP User's Manual 18. GW and IP to IP Contact User = name that is sent in the SIP Request's Contact header for this IP Group (e.g., ITSP-A). Figure 18-47: Defining IP Group 1 Define IP Group #2 for ITSP-B: From the 'Type' drop-down list, select SERVER.
Page 350 Mediant 3000 Define IP Group #3 for the IP-PBX: From the 'Type' drop-down list, select SERVER. In the 'Description' field, type an arbitrary name for the IP Group (e.g., IP-PBX). From the 'Proxy Set ID' drop-down lists, select 3 (represents the IP address, configured in , for communicating with this IP Group).
Page 351: Step 6: Configure The Account Table
SIP User's Manual 18. GW and IP to IP From the 'Serving IP Group ID' drop-down list, select 3 (i.e. the IP Group for the IP-PBX). Figure 18-50: Defining IP Group 4 Note: No Serving IP Groups are defined for ITSP-A and ITSP-B. Instead, the Outbound IP Routing table (see 'Step 9: Configure Outbound IP Routing' on page 355) is used to configure outbound call routing for calls originating from these ITSP IP Groups.
Page 352: Step 7: Configure Ip Profiles For Voice Coders
Mediant 3000 Configure Account ID #1 for IP-PBX authentication and registration with ITSP-A: • In the 'Served IP Group' field, enter "3" to indicate that authentication is performed on behalf of IP Group #3 (i.e., the IP-PBX). • In the 'Serving IP Group' field, enter "1" to indicate that registration/authentication is with IP Group #1 (i.e., ITSP-A).
Page 353 SIP User's Manual 18. GW and IP to IP Configure Coder Group ID #2 for the ITSP's (as shown in the figure below): From the 'Coder Group ID' drop-down list, select 2. From the 'Coder Name' drop-down list, select G.723.1. Click Submit.
Page 354: Step 8: Configure Inbound Ip Routing
Mediant 3000 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 calls. The table in which this is configured uses the IP Groups that you defined in 'Step 5: Configure the IP Groups' on page 348.
Page 355: Step 9: Configure Outbound Ip Routing
SIP User's Manual 18. GW and IP to IP • 'Source IP Group ID': enter "2" to assign these calls to IP Group pertaining to ITSP-B. Index #5: identifies all IP calls received from IP-PBX remote users: • 'Source Host Prefix': enter "PBXuser". This is the host name that appears in the From header of the Request URI received from remote IP-PBX users.
Page 356 Mediant 3000 Index #2: routes IP calls received from ITSP-B to the IP-PBX: • 'Source IP Group ID': select 2 to indicate received (inbound) calls identified as belonging to the IP Group configured for ITSP-B. • 'Dest Phone Prefix' and 'Source Phone Prefix': enter the asterisk (*) symbol to indicate all destinations and callers respectively.
Page 357: Step 10: Configure Destination Phone Number Manipulation
SIP User's Manual 18. GW and IP to IP 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.
Page 358 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 359: Session Border Controller
SIP User's Manual 19. Session Border Controller Session Border Controller 19.1 SBC Overview This section provides a detailed description of the device's SBC application. This application performs transparent transcoding of voice coders for IP-to-IP calls without affecting the SIP signaling messages. The SBC application supports up to 1,008 concurrent SBC sessions and provides the following main features: ...
Page 360: Sip Network Definitions
Mediant 3000  SIP normalization: The device supports SIP normalization, whereby the SBC application can overcome interoperability problems between SIP user agents. This is achieved by the following: • Manipulation of SIP URI user and host parts • Connection to ITSP SIP trunks on behalf of an IP-PBX - the device can register and utilize user and password to authenticate for the IP-PBX ...
Page 361 SIP User's Manual 19. Session Border Controller The flowchart below illustrates this process: Figure 19-1: Routing Process Version 6.4 November 2011...
Page 362: Determining Source And Destination Url
Mediant 3000 19.1.2.1 Determining Source and Destination URL The SIP protocol has more than one URL in a dialog establishing request that might represent the source and destination URL. When handling an incoming request, the device determines which SIP headers are used for source and destination URLs. Once these URLs are determined, the input user and host are taken from them.
Page 363: Sbc Ip-To-Ip Routing
SIP User's Manual 19. Session Border Controller The flowchart below illustrates the classification process: Figure 19-2: Classification Process (Identifying IP Group or Rejecting Call) 19.1.2.3 SBC IP-to-IP Routing The device's SBC application employs a comprehensive and flexible routing scheme:  Routing rules according to Layer-3/4 and SIP characteristics ...
Page 364: Ip-To-Ip Inbound And Outbound Manipulation
Mediant 3000 • Destination IP Group (address defined by Proxy Set associated with the IP Group) with the ability of load balancing and redundancy • ENUM query  Alternative Routing  Routing between two different Layer-3 networks  Transport protocol translator (UDP to TCP to TLS) ...
Page 365 SIP User's Manual 19. Session Border Controller Manipulated destination user and host are performed on the following SIP headers: Request-URI, To, and Remote-Party-ID (if exists). Manipulated source user and host are performed on the following SIP headers: From, P-Asserted (if exists), P-Preferred (if exists), and Remote-Party-ID (if exists).
Page 366: Sip Header Manipulation
Mediant 3000 Below is an example of a call flow and consequent SIP URI manipulations: The SIP message manipulations in the example above (contributing to typical topology hiding) are as follows: SIP Manipulation From Inbound Source SIP URI User Name...
Page 367: User Registration And Internal Database
SIP User's Manual 19. Session Border Controller  Configurable identity hiding (information related to identity of subscribers for example, P-Asserted-Identity, Referred-By, Identity and Identity-Info).  Apply conditions per rule - the condition can be on parts of the message or call’s parameters.
Page 368: Internal Database
Mediant 3000 of type USER. If classification fails or the source IP Group is not of type USER, the registration is rejected. Routing: The REGISTER routing is performed using the IP2IP Routing table: • The destination type can be an IP Group, specific IP address, Request-URI, or ENUM query (can also use DNS queries).
Page 369: Routing Using Internal Database
SIP User's Manual 19. Session Border Controller The device's database can include up to 2000 registered SBC users (for a fully populated device). The database has the following limitations:  Maximum of five contacts per AOR  The same contact cannot belong to more than one AOR ...
Page 370: Registration Restriction Control
Mediant 3000 19.1.3.5 Registration Restriction Control The device provides flexibility in controlling user's registration:  Limiting Number of Registrations per Source SRD and/or IP Group: You can limit the number of users that can register with the device. This limitation can be applied per source IP Group and/or SRD.
Page 371 SIP User's Manual 19. Session Border Controller The media capabilities exchanged in an offer/answer transactions include the following:  Media types (Audio, Secure Audio, Video, Fax, Text...)  IP addresses and ports of the media flow  Media flow mode (send receive, receive only, send only, inactive) ...
Page 372: Media Anchoring Without Transcoding (Transparent)
Mediant 3000 19.1.4.1 Media Anchoring without Transcoding (Transparent) To direct the RTP to flow through the device (for NAT traversal, firewall and security), all IP address fields in the SDP are modified:  Origin: IP address, session and version id ...
Page 373 SIP User's Manual 19. Session Border Controller assigned to each IP Group. Therefore, each offer destined to specific IP Groups include this coder. In the scenario depicted in the figure below, the IP phone on the Network-1 side initiates a call to the IP phone on the Network-2.
Page 374: No Media Anchoring
Mediant 3000  INVITE without SDP, offer in 180, and answer in PRACK  PRACK and UPDATE transactions can also be used for initiating subsequent offer\answer transactions before the INVITE 200 OK response.  In a SIP dialog life time, media characteristics after originally determined by the first offer\answer transaction can be changed by using subsequent offer\answer transactions.
Page 375: Interworking Dtmf Methods
SIP User's Manual 19. Session Border Controller Notes: • No Media Anchoring can be used when the SBC does not do NAT traversal (for media) where all the users are in the same domain. • No Media Anchoring calls cannot operate simultaneously with the following SBC features: - Force transcoding - Extension Coders...
Page 376: Transcoding Modes
Mediant 3000 • [3]: INFO, Nortel • [4]: INFO, Korea The chosen DTMF method determines (for each leg) which DTMF method is used for sending DTMF’s. If the device interworks between different DTMF methods and one of the methods is In-band\RFC 2833, detection and generation of DTMF methods requires DSP allocation.
Page 377 SIP User's Manual 19. Session Border Controller In addition to restricting the use of coders, the device can prioritize the coders listed in the SDP offer. This feature is referred to as Coder Preference. This is done on both SBC legs: ...
Page 378: Srtp-Rtp Transcoding
Mediant 3000 policies on SDP (in this order) returns an empty coders list, the second leg rejects the call (SIP 488, or ACK and BYE). Below is an example, assuming that Allowed Coders list (ordered) includes G711A-law (PCMA), G729, and G711U-law (PCMU), and Extension Coder is G729.
Page 379: Multiple Rtp Media Streams Per Call Session
SIP User's Manual 19. Session Border Controller If one of the above transcoding prerequisites is not met:  Any value other than “As is” is discarded.  If the incoming offer is SRTP, force transcoding, coder transcoding, and DTMF extensions are not applied. Transcoding between RTP and SRTP requires two DSP allocation.
Page 380: Sip Dialog Admission Control
Mediant 3000 19.1.6 SIP Dialog Admission Control The device allows you to limit the number of concurrent calls (SIP dialogs). These call limits can be applied per SRD and/or IP Group, and per user (identified by its registered contact). This feature can be useful for implementing Service Level Agreements (SLA) policies.
Page 381: Handling Sip 3Xx Redirect Responses
If the user is successfully identified, the SIP message request is processed. Note: This feature is applicable only to Mediant 3000 in Simplex mode. 19.1.9 Handling SIP 3xx Redirect Responses By default, the device's handling of SIP 3xx responses is to send the Contact header unchanged.
Page 382: Interworking Sip Diversion And History-Info Headers
Mediant 3000 The process of this feature is described using an example: The device receives the Redirect server's SIP 3xx response (e.g., Contact: <sip:User@IPPBX:5060;transport=tcp;param=a>;q=0.5). The device replaces the Contact header value with the special prefix and database key value as user part, and with the device's URL as host part (e.g., Contact: <sip:Prefix_Key_User@SBC:5070;transport=udp>;q=0.5).
Page 383: Call Survivability
SIP User's Manual 19. Session Border Controller Parameter Value SIP Header Present in Received SIP Message HistoryInfoMode = Add Diversion History-Info Headers are synced and sent. converted to converted to DiversionMode = Add History-Info. Diversion. HistoryInfoMode = Diversion removed. History-Info Both removed.
Page 384: Broadsoft's Shared Phone Line Call Appearance For Sbc Survivability
Mediant 3000 Below is an example of an XML body received from the BroadWorks server: <?xml version="1.0" encoding="utf-8"?> <BroadsoftDocument version="1.0" content="subscriberData"> <phoneNumbers> <phoneNumber>2403645317</phoneNumber> <phoneNumber>4482541321</phoneNumber> </phoneNumbers> <aliases> <alias>sip:bob@broadsoft.com</alias> <alias>sip:rhughes@broadsoft.com</alias> </aliases> <extensions> <extension>5317</extension> <extension>1321</extension> </extensions> </BroadSoftDocument> 19.1.11.2 BroadSoft's Shared Phone Line Call Appearance for SBC Survivability The device can provide redundancy for BroadSoft's Shared Call Appearance feature.
Page 385 SIP User's Manual 19. Session Border Controller To configure this capability, you need to configure a shared-line, inbound manipulation rule for registration requests to change the destination number of the secondary extension numbers (e.g. 601 and 602) to the primary extension (e.g., 600). In addition, call forking must also be enabled.
Page 386: Call Survivability For Call Centers
Mediant 3000 19.1.11.3 Call Survivability for Call Centers The device supports call survivability for call centers. When a communication failure (e.g., in the network) occurs with the remote voice application server responsible for handling the call center application (such as IVR), the device routes the incoming calls received from the customer (i.e., from the TDM gateway) to the call center agents.
Page 387 SIP User's Manual 19. Session Border Controller Figure 19-10: Call Survivability for Call Center  To configure call survivability for a call center application: Configure IP Groups in the IP Group table (see 'Configuring IP Groups' on page 208) for the following entities: •...
Page 388 Mediant 3000 In the SBC IP2IP Routing table (see 'Configuring SBC IP-to-IP Routing' on page 402), configure the following IP-to-IP routing rules: • For normal operation: ♦ Routing from TDM Gateway to Application server. ♦ Routing from Application server to call center agents.
Page 389: Call Forking
SIP User's Manual 19. Session Border Controller 19.1.12 Call Forking 19.1.12.1 Initiating SIP Call Forking The SBC device enables call forking, whereby an incoming call is forked to multiple SBC users (destinations). In such a scenario, upon an incoming call, all the extensions of a user ring simultaneously and the first extension to pick up the call receives the call and all other extensions stop ringing.
Page 390: Active Sbc Call Continuity During Ha Blade Switchover
Mediant 3000 19.1.14 Active SBC Call Continuity during HA Blade Switchover The device maintains active SBC call sessions during a blade switchover (when in High Availability/HA mode). Upon blade failure, the standby blade takes over from the previously active blade and current SBC calls are preserved (and not disconnected).
Page 391: Configuring Admission Control
SIP User's Manual 19. Session Border Controller Configure the parameters as required. Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 478. 19.2.2 Configuring Admission Control The Admission Control page allows you to define up to 100 rules for limiting the number of concurrent calls (SIP dialogs).
Page 392 Mediant 3000 Table 19-2: Admission Control Parameters Parameter Description Limit Type Limitation rule defined per IP group or SRD.  [0] IP Group (default)  [1] SRD IP Group ID IP Group to which you want to apply the SIP dialog limit. To apply the rule to all IP Groups, set this parameter to -1 (default).
Page 393: Configuring Allowed Coder Groups
SIP User's Manual 19. Session Border Controller Parameter Description MaxBurst The maximum number of tokens (SIP dialogs) that the bucket can hold, where 0 is unlimited (default). The device only accepts a SIP dialog if a token exists in the bucket. Once the SIP dialog is accepted, a token is removed from the bucket.
Page 394: Configuring Sip Message Policy Rules
Mediant 3000  To configure Allowed Coder Groups: Open the Allowed Coders Group page (Configuration tab > VoIP menu > SBC submenu > Allowed Coders Group). Figure 19-13: Allowed Coders Group Page From the 'Allowed Coders Group ID' drop-down list, select an ID for the Allowed Coder Group.
Page 395 SIP User's Manual 19. Session Border Controller  Option to send 400 "Bad Request" response if message request is rejected  Blacklist and whitelist for defined SIP methods (e.g., INVITE)  Blacklist and whitelist for defined SIP bodies  To configure SIP message policy rules: Open the Message Policy Table page (Configuration tab >...
Page 396 Mediant 3000 Parameter Description Max Body Length Defines the maximum SIP message body length. This is [MessagePolicy_MaxBodyLength] the value of the Content-Length header. The valid value is up to 512 characters. Max Num Headers Defines the maximum number of headers.
Page 397: Routing Sbc
SIP User's Manual 19. Session Border Controller 19.2.5 Routing SBC This section describes the configuration of the routing entities for the SBC application. These include the following:  Classification rules - see 'Configuring the Classification Table' on page  Condition rules - see 'Configuring Condition Rules' on page ...
Page 398 Mediant 3000 The flowchart below illustrates the classification process: Figure 19-15: Classification Process (Identifying IP Group or Rejecting Call) Notes: • Incoming REGISTER messages are saved in the device’s registration database and sent to a destination only if they are associated with a source IP Group that is of USER type.
Page 399 SIP User's Manual 19. Session Border Controller  To configure classification rules: Open the Classification Table page (Configuration tab > VoIP menu > SBC submenu > Routing SBC submenu > Classification Table). Click the Add button; the following appears: Figure 19-16: Classification Table Page The figure above shows an example classification rule that identifies an incoming SIP dialog to IP Group ID #4, if its source IP address is 10.8.6.15, source port is 5060, SIP transport is TLS, and matches the Message Condition rule 1.
Page 400 Mediant 3000 Parameter Description Source IP Address Defines the source IP address (in dotted-decimal notation) of the [Classification_SrcAddress] incoming SIP dialog. Notes:  If this parameter is not configured or is configured as an ‘*’ (asterisk), then any source IP address is accepted.
Page 401: Configuring Condition Rules
SIP User's Manual 19. Session Border Controller Parameter Description Action Type Defines a whitelist or blacklist for incoming SIP dialog requests that [Classification_ActionType] match the characteristics of the classification rule.  [0] Deny = Blocks incoming SIP dialogs that match the characteristics of the Classification rule (blacklist).
Page 402: Configuring Sbc Ip-To-Ip Routing
Mediant 3000 19.2.5.3 Configuring SBC IP-to-IP Routing The IP2IP Routing Table page configures up to 120 SBC IP-to-IP routing rules. This table provides enhanced IP-to-IP call routing capabilities for routing received SIP dialog messages (e.g., INVITE) to a destination IP address. The SIP message is routed according to a routing rule whose configured input characteristics (e.g., Source IP Group) match the...
Page 403 SIP User's Manual 19. Session Border Controller  To configure SBC IP-to-IP routing rules: Open the IP2IP Routing Table page (Configuration tab > VoIP menu > SBC submenu > Routing SBC submenu > IP to IP Routing Table). Click the Add button; the Add Record dialog box appears: Figure 19-18: SBC IP2IP Routing Table - Add Record Dialog Box Add an entry and then configure it according to the table below.
Page 404 Mediant 3000 Parameter Description Source Username Prefix Defines the prefix of the user part of the incoming SIP [IP2IPRouting_SrcUsernamePrefix] dialog's source URI (usually the From URI). The default is "*". Note: The prefix can be a single digit or a range of digits.
Page 405 SIP User's Manual 19. Session Border Controller Parameter Description Operation Routing Rule (when match occurs in characteristics) Destination Type Determines the destination type to which the outgoing SIP [IP2IPRouting_DestType] dialog is sent.  [0] IP Group (default) = The SIP dialog is sent to the IP Group’s Proxy Set (SERVER-type IP Group) or registered contact from the database (if USER-type IP Group).
Page 406 Mediant 3000 Parameter Description addresses.  If the selected destination IP Group ID is type USER, the request is routed according to the IP Group specific database (i.e., only to registered users of the selected database).  If the selected destination IP Group ID is ANY USER ([- 2]), the request is routed according to the general database (i.e., any matching registered user).
Page 407: Configuring Alternative Routing Reasons
SIP User's Manual 19. Session Border Controller Parameter Description Alternative Route Options Determines whether this routing rule is the main routing rule [IP2IPRouting_AltRouteOptions] or an alternative routing rule (to the rule defined directly above it in the table).  [0] Route Row (default) = Main routing rule - the device first attempts to route the call to this route if the incoming SIP dialog's input characteristics matches this rule.
Page 408: Manipulations Sbc
Mediant 3000 Notes: • Alternative routing occurs even if this table is not configured upon scenarios where no response, ICMP, or a SIP 408 response is received. • SIP requests pertaining to an SRD or IP Group that reach the call limit...
Page 409 SIP User's Manual 19. Session Border Controller Notes: • For more information on the syntax for configuring SIP message manipulation rules in the Message Manipulation table, see 'SIP Message Manipulation Description' on page 777. • The values entered in the table are not case-sensitive. •...
Page 410 Mediant 3000 Add an entry and then configure it according to the table below. Click the Apply button to save your changes. To save the changes to flash memory, see 'Saving Configuration' on page 478. Table 19-7: Message Manipulations Parameters...
Page 411: Configuring Ip-To-Ip Inbound Manipulations
SIP User's Manual 19. Session Border Controller Parameter Description Action Value Defines a value (string) that you want to use in the manipulation. [ActionValue] The syntax is as follows: string/<message-element>/<call-param> "+" string/<message- element>/<call-param> For example:  'itsp.com'  header.from.url.user  param.call.dst.user ...
Page 412 Mediant 3000  To configure IP-to-IP inbound manipulation rules: Open the IP to IP Inbound Manipulation page (Configuration tab > VoIP menu > SBC submenu > Manipulations SBC submenu > IP to IP Inbound). Figure 19-21: IP to IP Inbound Manipulation Page The figure above shows a manipulation configuration example that removes the destination URI user name prefix "976"...
Page 413 SIP User's Manual 19. Session Border Controller Parameter Description Source Host Defines the source SIP URI host name - full name (usually in the From [SrcHost] header). For any host name, enter the asterisk "*" symbol (default). Destination Username Defines the prefix of the destination SIP URI user name (usually in the Prefix Request-URI).
Page 414: Configuring Ip-To-Ip Outbound Manipulations
Mediant 3000 19.2.6.3 Configuring IP-to-IP Outbound Manipulations The IP to IP Outbound Manipulation page allows you to configure up to 100 manipulation rules for manipulating SIP URI user part (source and destination) of outbound SIP dialog requests. Manipulation rules in the table are located according to the source IP Group, and source and destination host and user prefixes and can be applied to a user-defined SIP request type (e.g., INVITE, OPTIONS, SUBSCRIBE, and /or REGISTER).
Page 415 SIP User's Manual 19. Session Border Controller Parameter Description Note: Additional manipulation can only be performed on a different SIP URI (either source or destination) to the rule configured in the row above (defined by the parameter ManipulatedURI). Source IP Group ID Defines the IP Group from where the INVITE is received.
Page 416 Mediant 3000 Parameter Description Prefix to Add Defines the number or string that you want added to the front of the [Prefix2Add] user name. For example, if you enter 'user' and the user name is "john", the new user name is "userjohn".
Page 417: Sbc Configuration Examples
SIP User's Manual 19. Session Border Controller 19.3 SBC Configuration Examples This section provides SBC configuration examples. Note: The examples described in this section are for reference only. Modifications to device configuration should be made to suit your networking environment. 19.3.1 Basic SBC Scenario This example assumes the following: ...
Page 418: Step 1: Enable The Sbc Application
Mediant 3000 19.3.1.1 Step 1: Enable the SBC Application The procedure below describes how to enable the SBC application. Once enabled, the SBC-specific parameters/pages become available in the Web interface.  To enable SBC: Open the Applications Enabling page (Configuration tab > VoIP menu >...
Page 419: Step 3: Define Ip Addresses For Media/Control
SIP User's Manual 19. Session Border Controller Click Submit. Save the settings to flash memory ("burn") and reset the device (see 'Saving Configuration' on page 478). 19.3.1.3 Step 3: Define IP Addresses for Media/Control The procedure below describes how to configure two IP interfaces, one for each network. ...
Page 420 Mediant 3000  To configure SIP and RTP interfaces for each network leg: For each network leg, configure a Media Realm for the RTP traffic in the Media Realm Table page (Configuration tab > VoIP menu > Media submenu >Media Realm Configuration): •...
Page 421 SIP User's Manual 19. Session Border Controller For each network leg, configure a SIP signaling interface in the SIP Interface Table page (Configuration tab > VoIP menu > Control Network submenu > SIP Interface Table): • Add a SIP interface for Network-1 leg: 'Network Interface' = "NW1"...
Page 422: Step 5: Define Proxy Sets Per Network
Mediant 3000 19.3.1.5 Step 5: Define Proxy Sets per Network The procedure below describes how to configure Proxy Sets for each network leg. The Proxy Set defines the IP addresses of the IP-PBX's located in each network.  To configure Proxy Sets: Open the Proxy Sets Table page (Configuration tab >...
Page 423: Step 6: Define Ip Groups Per Network
SIP User's Manual 19. Session Border Controller Click Submit. Figure 19-31: Defining Proxy Set for IP PBX in Network-2 Save the settings to flash memory ("burn") and reset the device (see 'Saving Configuration' on page 478). 19.3.1.6 Step 6: Define IP Groups per Network An IP Group is a convenient way to represent a SIP User Agent (client or server) entity, which in our example are the IP-PBX's in each network.
Page 424 Mediant 3000 Click Submit. Figure 19-32: Defining IP Group for Network-1 IP-PBX Add an IP Group for Network-2 IP-PBX with the following values: 'Index' = 2 'Type' = SERVER 'Description' = "IP-PBX NW2" 'Proxy Set ID' = "2" - defined previously which depicts the IP address of the IP 'SRD' = "1"...
Page 425: Step 7: Define Ip-To-Ip Routing Rules
SIP User's Manual 19. Session Border Controller Click Submit. Figure 19-33: Defining IP Group for Network-2 IP-PBX Save the settings to flash memory ("burn") and reset the device (see 'Saving Configuration' on page 478). 19.3.1.7 Step 7: Define IP-to-IP Routing Rules The procedure below describes how to configure IP-to-IP routing rules for routing SIP signaling and calls between IP Group 1 (Network-1) and IP Group 2 (Network-2).
Page 426: Sbc-To-Pstn Routing
Mediant 3000 'Destination SRD = "0" Click Apply Figure 19-34: IP-to-IP Routing Rules between IP Groups 19.3.2 SBC-to-PSTN Routing This example describes how to setup the device for SBC-to-PSTN routing. This example is based on the general scenario described in 'Basic SBC Scenario' on page 417, but in addition the device is now connected to the PSTN network through an E1/T1 trunk interface.
Page 427: Step 1: Define Sip Interface For Pstn
SIP User's Manual 19. Session Border Controller 19.3.2.1 Step 1: Define SIP Interface for PSTN The procedure below describes how to configure a SIP signaling interface for the PSTN. This SIP interface is configured for the "GW/IP2IP" application using port 5070 on the Network-1 SRD.
Page 428: Step 3: Define Ip Group For Pstn
Mediant 3000 From the 'SRD Index' drop-down list, select 0. This associates the Proxy Set with the Network-1 SRD. It allows the device to classify calls by Proxy Set for this SRD. Figure 19-37: Defining Device as Proxy Set Click Submit.
Page 429: Step 4: Define Ip-To-Ip Routing Rules For Pstn
SIP User's Manual 19. Session Border Controller From the 'Classify By Proxy Set' drop-down list, select Disable. Figure 19-38: Defining IP Group for PSTN Click Submit. Save the settings to flash memory ("burn") and reset the device (see 'Saving Configuration' on page 478). 19.3.2.4 Step 4: Define IP-to-IP Routing Rules for PSTN The procedure below describes how to configure IP-to-IP routing rules.
Page 430 Mediant 3000 In the 'Add' field, enter "2", and then click Add; the new entry is added as Index #2 after Index #1. The previous Index #2 is now shifted down to Index #3. From the 'Source IP Group ID' drop-down list, select 1. This is the IP Group to which the Network-1 IP-PBX belongs.
Page 431: Step 5: Define Trunk Group For Pstn
SIP User's Manual 19. Session Border Controller 19.3.2.5 Step 5: Define Trunk Group for PSTN The procedure below describes how to configure the E1/T1 Trunk that is connected between the device (TRUNK module) and the PSTN. This is done by defining a Trunk Group.
Page 432: Step 6: Define Ip-To-Tel Routing Rules
Mediant 3000 19.3.2.6 Step 6: Define IP-to-Tel Routing Rules The procedure below describes how to configure IP-to-Tel routing rules. In the example, you need to configure the a rule that routes IP calls with destination prefix 9 to the E1/T1 trunk/PSTN network (i.e., Trunk Group 1).
Page 433: Step 1: Define Coder Groups
SIP User's Manual 19. Session Border Controller 19.3.3.1 Step 1: Define Coder Groups The procedure below describes how to configure a Coder Group (with G.711) for the Network-1 IP-PBX and a Coder Group (with G.729) for the Network-2 IP-PBX. These Coder Groups are later assigned to the IP Profiles of Network-1 IP-PBX and Network-2 IP- PBX.
Page 434 Mediant 3000 Click Submit. Figure 19-44: Defining IP Profile for Network-1 Add IP Profile #2 for Network-2: From the 'Profile ID' drop-down list, select 2. In the 'Profile Name', enter a brief description (e.g., "Network-2"). From the 'Extension Coders Group ID', select 2. This is the Coder Group that you defined previously.
Page 435: Step 4: Assign Ip Profiles To Ip Groups
SIP User's Manual 19. Session Border Controller 19.3.3.4 Step 4: Assign IP Profiles to IP Groups The procedure below describes how to assign the previously defined IP Profiles to the IP Groups for Network-1 and Network-2. This stage assumes that you have defined IP Groups for Network-1 (e.g., #1) and Network-2 (e.g., #2).
Page 436: Sbc Rtp-Srtp Transcoding
Mediant 3000 19.3.4 SBC RTP-SRTP Transcoding This section describes how to configure an RTP-SRTP transcoding scenario. This example is based on the previous examples and only describes the configuration of the transcoding feature itself. It assumes that the other elements such as SRD's and IP Groups have already been configured.
Page 437 SIP User's Manual 19. Session Border Controller Click Submit. Figure 19-49: RTP-to-SRTP Transcoding for Network-2 Version 6.4 November 2011...
Page 438: Sbc Sip Uri Manipulation
Mediant 3000 19.3.5 SBC SIP URI Manipulation This example enhances that described in 'Basic SBC Scenario' on page to include SIP URI user and host parts manipulation. In this example, the following manipulations are performed:  SIP URI host part: For the SIP INVITE sent from Network-1 IP PBX (e.g., IP Group ID #1) to Network-2 IP PBX (i.e., IP Group ID #2), the URI host name is replaced with...
Page 439: Step 2: Define Sip Uri User Manipulation
SIP User's Manual 19. Session Border Controller 19.3.5.2 Step 2: Define SIP URI User Manipulation The procedure below describes how to configure SIP URI user manipulation. In the example, the SIP INVITE containing the destination URI user name prefix "976" sent from Network-1 IP Group, must be manipulated so that the prefix is removed (i.e., "976") in the outgoing INVITE to Network-2 IP PBX.
Page 440: Sip Header Manipulation
Mediant 3000 19.3.6 SIP Header Manipulation This section provides an example on how to configure a SIP message manipulation rule that adds a P-Asserted-Identity header with the user part from the From header, to all received (inbound) INVITE messages. For example, if the device receives an INVITE from user "1000", it adds a P-Asserted-Identity header to the sent INVITE with the value...
Page 441: Step 2: Assign Message Manipulation Rule To Ip Group
SIP User's Manual 19. Session Border Controller The manipulation rule is shown in the figure below: Figure 19-52: SIP Header Manipulation Example Click Apply. 19.3.6.2 Step 2: Assign Message Manipulation Rule to IP Group The procedure below describes how to assign the configured SIP message manipulation rule (rule #1) to the IP Group belonging to the LAN users.
Page 442 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 443: Stand-Alone Survivability (Sas) Application
The SAS application is available only if the device is installed with the SAS Software Upgrade Key. • The SAS feature is applicable only to Mediant 3000 Simplex (not HA). • Throughput this section, the term user agent (UA) refers to the enterprise's LAN phone user (i.e., SIP telephony entities such as IP...
Page 444: Sas Outbound Mode
Mediant 3000 20.1.1.1 SAS Outbound Mode This section describes the SAS outbound mode, which includes the following states:  Normal state (see 'Normal State' on page 444)  Emergency state (see 'Emergency State' on page 444) 20.1.1.1.1 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 445: Sas Redundant Mode
SIP User's Manual 20. Stand-Alone Survivability (SAS) Application The figure below illustrates the operation of SAS outbound mode in emergency state: Figure 20-2: 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 446 Mediant 3000 20.1.1.2.1 Normal State In normal state, the UAs register and operate directly with the external proxy. Figure 20-3: SAS Redundant Mode in Normal State (Example) 20.1.1.2.2 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 447: Sas Routing
SIP User's Manual 20. Stand-Alone Survivability (SAS) Application 20.1.1.2.3 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 448: Mediant
Mediant 3000 The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the external proxy: Figure 20-6: Flowchart of INVITE from Primary Proxy in SAS Normal State SIP User's Manual Document #: LTRT-89712...
Page 449: Sas Routing In Emergency State
SIP User's Manual 20. Stand-Alone Survivability (SAS) Application 20.1.2.2 SAS Routing in Emergency State The flowchart below shows the routing logic for SAS in emergency state: Figure 20-7: Flowchart for SAS Emergency State 20.2 SAS Configuration SAS supports various configuration possibilities, depending on how the device is deployed in the network and the network architecture requirements.
Page 450: General Sas Configuration
Note: The SAS application is available only if the device is installed with the SAS Software Upgrade Key. If your device is not installed with the SAS feature, contact your AudioCodes representative.  To enable the SAS application: Open the Applications Enabling page (Configuration tab > VoIP menu >...
Page 451 SIP User's Manual 20. Stand-Alone Survivability (SAS) Application Note: This SAS port must be different than the device's local gateway port (i.e., that defined for the 'SIP UDP/TCP/TLS Local Port' parameter in the SIP General Parameters page - Configuration tab > VoIP menu > SIP Definitions > General Parameters).
Page 452 Mediant 3000 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 453: Configuring Sas Outbound Mode
SIP User's Manual 20. Stand-Alone Survivability (SAS) Application 20.2.2 Configuring SAS Outbound Mode This section describes how to configure the SAS outbound mode. These settings are in addition to the ones described in 'Configuring Common SAS Parameters' on page 450. Note: The VoIP CPEs (such as IP phones or residential gateways) need to be defined so that their proxy and registrar destination addresses and ports are...
Page 454: Configuring Gateway Application With Sas
Mediant 3000 20.2.4 Configuring Gateway Application with SAS If you want to run both the Gateway and SAS applications on the device, the configuration described in this section is required. The configuration steps depend on whether the Gateway application is operating with SAS in outbound mode or SAS in redundant mode.
Page 455 SIP User's Manual 20. Stand-Alone Survivability (SAS) Application In the first 'Proxy Address' field, enter the IP address and port of the device (in the format x.x.x.x:port). This is the port as defined in the 'SAS Local UDP/TCP/TLS Port' field (see 'Configuring Common SAS Parameters' on page 450).
Page 456: Gateway With Sas Redundant Mode
Mediant 3000 20.2.4.2 Gateway with SAS Redundant Mode The procedure below describes how to configure the Gateway application with SAS redundant mode.  To configure Gateway application with SAS redundant mode: Define the proxy servers for the Gateway application: Open the Proxy & Registration page (Configuration tab > VoIP menu > SIP Definitions submenu >...
Page 457 SIP User's Manual 20. Stand-Alone Survivability (SAS) Application Disable the use of user=phone in the 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 SIP URL and therefore, REGISTER and INVITE messages use SIP URI.
Page 458: Advanced Sas Configuration
Mediant 3000 20.2.5 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 458) ...
Page 459: Manipulating Destination Number Of Incoming Invite
SIP User's Manual 20. 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 460 Mediant 3000 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 461: Sas Routing Based On Sas Routing Table
SIP User's Manual 20. Stand-Alone Survivability (SAS) Application 20.2.5.3 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 462 Mediant 3000 Note: The following parameters are not applicable to SAS and should be ignored: Destination IP Group ID, Destination SRD ID, and Alternative Route Options. Table 20-1: SAS IP2IP Routing Table Parameters Parameter Description Matching Characteristics Source Username Prefix The prefix of the user part of the incoming INVITE’s source...
Page 463 SIP User's Manual 20. 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 464: Blocking Calls From Unregistered Sas Users
Mediant 3000 20.2.5.4 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 465 SIP User's Manual 20. 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 466: Adding Sip Record-Route Header To Sip Invite
Mediant 3000 20.2.5.6 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 467: Replacing Contact Header For Sip Messages
SIP User's Manual 20. Stand-Alone Survivability (SAS) Application 20.2.5.7 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 468: Viewing Registered Sas Users
Mediant 3000 20.3 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 524. The maximum number of users that can be registered in the database is 2000.
Page 469 SIP User's Manual 20. 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 20-22: SAS Cascading Using SAS Routing Table - Example ...
Page 470 Mediant 3000 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 471: Transcoding Using Third-Party Call Control
In the example below, an Application Server sends a special INVITE that consists of two media lines to perform transcoding between G.711 and G.729: m=audio 20000 RTP/AVP 0 c=IN IP4 A.example.com m=audio 40000 RTP/AVP 18 c=IN IP4 B.example.com Note: This feature is supported only by Mediant 3000 Simplex (not HA). Version 6.4 November 2011...
Page 472: Using Rfc 4240 - Netann 2-Party Conferencing
Mediant 3000 21.2 Using RFC 4240 - NetAnn 2-Party Conferencing Transcoding bridges (or translates) between two remote network locations, each of which uses a different coder and/or a different DTMF and fax transport types. The device supports IP-to-IP transcoding. It creates a transcoding call that is similar to a dial-in, two- party conference call.
Page 473: Part V: Maintenance
Part V Maintenance This part describes the maintenance procedures.
Page 474 Reader’s Notes...
Page 475: Basic Maintenance
SIP User's Manual 22. 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 476 Mediant 3000  To reset the device: Open the Maintenance Actions page (see 'Basic Maintenance' on page 473). 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 477: Locking And Unlocking The Device
SIP User's Manual 22. Basic Maintenance 22.2 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 478: Saving Configuration
Mediant 3000 22.3 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 479: High Availability Maintenance
SIP User's Manual 23. High Availability Maintenance High Availability Maintenance The High Availability Maintenance page allows you to perform a switch-over between the Active and Redundant blades. In addition, this page allows you to reset the Redundant blade. Notes: • When performing a blade switchover or a Redundant blade reset, the HA mode becomes temporarily unavailable.
Page 480 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 481: Software Upgrade
SIP User's Manual 24. 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 481) ...
Page 482 Mediant 3000 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 483 SIP User's Manual 24. 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 484: Call Progress Tones File
Mediant 3000 Repeat steps 2 through 3 for each file you want to load. Save the loaded auxiliary files to flash memory, see 'Saving Configuration' on page and reset the device (if you have loaded a Call Progress Tones file), see 'Resetting the Device' on page 475.
Page 485 SIP User's Manual 24. Software Upgrade tone's definition lines to the first tone definition in the ini file. The device reports dial tone detection if either of the two tones is detected. The Call Progress Tones section of the ini file comprises the following segments: ...
Page 486: Prerecorded Tones File
Mediant 3000 • Fourth Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the fourth cadence on-off cycle. Can be omitted if there isn't a fourth cadence. • Carrier Freq [Hz]: Frequency of the carrier signal for AM tones.
Page 487: Cas Files
 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 481). The prerecorded tones are played repeatedly. This allows you to record only part of the tone and then play the tone for the full duration.
Page 488 Mediant 3000  Prefix tags (for IP-to-Tel routing): Provides enhanced routing rules based on Dial Plan prefix tags. For more information, see Dial Plan Prefix Tags for IP-to-Tel Routing on page 325. The Dial Plan file is first created using a text-based editor (such as Notepad) and saved with the file extension .ini.
Page 489: User Information File
SIP User's Manual 24. Software Upgrade 24.1.5 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 or for creating an SBC users database. The User Information file can be loaded to the device by using one of the following methods: ...
Page 490: User Information File For Sbc Users Database
Mediant 3000 An example of a User Information file is shown in the figure below: Figure 24-1: 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 491: Amd Sensitivity File
SIP User's Manual 24. Software Upgrade where:  LocalUser identifies the user and is used as the URI user part for the AOR in the database  UserName is the user's authentication username.  Password is the user's authentication password. ...
Page 492 Mediant 3000 </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>19923</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>50790</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>30720</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>10486</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>57344</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>25600</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>8389</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>62259</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>23040</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>  <AMDCOEFFICIENTA>10486</AMDCOEFFICIENTA> <AMDCOEFFICIENTB>50790</AMDCOEFFICIENTB> <AMDCOEFFICIENTC>28160</AMDCOEFFICIENTC> </AMDSENSITIVITYLEVEL> <AMDSENSITIVITYLEVEL>...
Page 493: Loading Software Upgrade Key
 Web interface  BootP/TFTP configuration utility (see Loading via BootP/TFTP on page 495)  AudioCodes’ EMS (refer to EMS User’s Manual or EMS Product Description) Warning: Do not modify the contents of the Software Upgrade Key file. Notes: •...
Page 494 Mediant 3000  To load a Software Upgrade Key: Open the Software Upgrade Key Status page (Maintenance tab > Software Update menu > Software Upgrade Key). Backup your current Software Upgrade Key as a precaution so that you can re-load this backup key to restore the device's original capabilities if the new key doesn’t...
Page 495: Loading Via Bootp/Tftp
Verify that the content of the file has not been altered. 24.2.1 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 496: Software Upgrade Wizard
(i.e., maintain existing configuration) running on the device. For Mediant 3000 HA, the wizard also allows you to perform Hitless Upgrade (non-traffic affecting upgrade), whereby the upgrade process begins only after all current calls have been terminated.
Page 497 System Reset Upgrade: both blades immediately reset with the newly loaded .cmp file. Note: For Mediant 3000 HA, if you choose Hitless Upgrade, you can upload only a .cmp file (auxiliary files and ini files cannot be uploaded as well). Version 6.4...
Page 498 Mediant 3000 If you want to load only a .cmp file, then click the Reset button to reset the device with the newly loaded .cmp file, utilizing the existing configuration (ini) and auxiliary files. To load additional files, skip to Step 7.
Page 499 SIP User's Manual 24. Software Upgrade After the device resets, the End of Process wizard page appears displaying the new .cmp and auxiliary files loaded to the device. Figure 24-4: End Process Wizard Page Click End Process to close the wizard; the Web Login dialog box appears. Enter your login user name and password, and then click OK;...
Page 500: Hitless Software Upgrade
Mediant 3000 24.3.1 Hitless Software Upgrade The Mediant 3000 HA system allows you to upgrade the software (SW) version (i.e., cmp file) running on the device without disrupting current calls. This non-affecting traffic upgrade feature is referred to as Hitless Software Upgrade.
Page 501 SIP User's Manual 24. Software Upgrade Both blades now operate with the new SW version and a switchback is issued to return the system to its original state. The previously Active blade now becomes active, and the previously Redundant blade resets once more to return to redundant state.
Page 502: Backing Up And Loading Configuration File
Mediant 3000 24.4 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 503: Restoring Factory Defaults
Restoring Defaults using CLI The device can be restored to factory defaults using CLI, as described in the procedure below. Note: Restoring factory defaults using CLI is supported only by Mediant 3000 Simplex.  To restore factory defaults using CLI:...
Page 504: Restoring Defaults Using An Ini File
Mediant 3000 25.2 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 502). 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 505: Part Vi: Status, Performance Monitoring And Reporting
Part VI Status, Performance Monitoring and Reporting This part describes the status and performance monitoring procedures.
Page 506 Reader’s Notes...
Page 507: System Status
This information can help you expedite troubleshooting. Capture the page and e-mail it to AudioCodes Technical Support personnel to ensure quick diagnosis and effective corrective action. This page also displays any loaded files used by the device (stored in the RAM) and allows you to remove them.
Page 508: Viewing Ethernet Port Information
Open the Ethernet Port Information page (Status & Diagnostics tab > System Status menu > Ethernet Port Information). Figure 26-2: Ethernet Port Information Page When Physical Network Separation is enabled on Mediant 3000 with TP-8410 (using the parameter EnableNetworkPhysicalSeparation - see Networking Parameters on page 549), the Ethernet Port Information page...
Page 509: Viewing Timing Module Information
SIP User's Manual 26. System Status The different Ethernet ports are located on the Mediant 3000 chassis as follows:  Control: Port 1A - bottom PEM module; Port 1B - top PEM module.  OAMP: Port 2A - bottom PEM module; Port 2B - top PEM module.
Page 510: Viewing Hardware Components Status
Mediant 3000 26.4 Viewing Hardware Components Status The Components Status page provides read-only, real-time status of the device's chassis components such as slot occupants, fans, and power supply units. Note: You can also access this page from the Home page (see 'Using the Home Page' on page 65).
Page 511: Carrier-Grade Alarms
SIP User's Manual 27. 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 27.1 Viewing Active Alarms The Active Alarms page displays a list of currently active alarms.
Page 512: Viewing Alarm History
Mediant 3000 27.2 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 513: Performance Monitoring
SIP User's Manual 28. 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 ...
Page 514 Mediant 3000 From the 'Trunk' drop-down list, select the trunk for which you want to view active channels. For more graph functionality, see the following table: Table 28-1: 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.
Page 515: Viewing Mos Per Media Realm
SIP User's Manual 28. Performance Monitoring 28.2 Viewing MOS per Media Realm The MOS Per Media Realm page displays statistics on Media Realms (configured in 'Configuring Media Realms' on page 182). This page provides two graphs:  Upper graph: displays the Mean Opinion Score (MOS) quality in RTCP data per selected Media Realm.
Page 516: Viewing Quality Of Experience
Mediant 3000 28.3 Viewing Quality of Experience The Quality Of Experience page provides statistical information on calls per SRD or IP Group. The statistics can be further filtered to display incoming and/or outgoing call direction, and type of SIP dialog (INVITE, SUBSCRIBE, or all).
Page 517: Viewing Average Call Duration
SIP User's Manual 28. Performance Monitoring From the 'Type' drop-down list, select the SIP message type: • Invite - INVITE • Subscribe - SUBSCRIBE • Other - all SIP messages To refresh the charts, click Refresh. To reset the counters, click Reset Counters. 28.4 Viewing Average Call Duration The Average Call Duration page displays information about a specific SRD or IP Group.
Page 518 Mediant 3000 From the 'SRD/IpGroup' drop-down list, select whether you want to view information for an SRD or IP Group. From the 'Index' drop-down list, select the SRD or IP Group index. Use the Zoom In button to increase the displayed time resolution or the Zoom Out button to decrease it.
Page 519: Voip Status
SIP User's Manual 29. 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 520 Mediant 3000  To view the next eight trunks:  Click the Go To Page icon. Figure 29-2: 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 521: Viewing Active Ip Interfaces
SIP User's Manual 29. VoIP Status  To view detailed channel information of a trunk's channel: Click a required channel pertaining to a trunk for which you want to view information; the 'Basic Channel Information page appears, displaying basic information about the channel: Figure 29-3: Basic Channel Information Page To view additional channel information, click the buttons (SIP, Basic, RTP/RTCP, and...
Page 522: Viewing Performance Statistics
Mediant 3000 29.3 Viewing Performance Statistics The Basic Statistics page provides read-only, device performance statistics. This page is refreshed every 60 seconds. The duration that the currently displayed statistics has been collected is displayed above the statistics table.  To view performance statistics: ...
Page 523 SIP User's Manual 29. VoIP Status The fields in this page are described in the following table: Table 29-2: Call Counters Description Counter Description Number of Attempted Indicates the number of attempted calls. It is composed of established Calls and failed calls. The number of established calls is represented by the 'Number of Established Calls' counter.
Page 524: Viewing Sas/Sbc Registered Users
Mediant 3000 Counter Description Number of Failed Calls Indicates the number of calls that failed due to unavailable resources or due to No Resources a device lock. The counter is incremented as a result of one of the following release reasons: ...
Page 525: Viewing Call Routing Status
SIP User's Manual 29. VoIP Status 29.6 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 526: Viewing Ip Connectivity
Mediant 3000 29.7 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 298). Notes: •...
Page 527 SIP User's Manual 29. 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 528 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 529: Reporting Information To External Party
SIP User's Manual 30. Reporting Information to External Party Reporting Information to External Party 30.1 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 530 Mediant 3000 Field Name Description TrmSd Initiator of call release (IP, Tel, or Unknown) TrmReason Termination reason (see 'Release Reasons in CDR' on page 531) Fax transaction during call InPackets Number of incoming packets Number of outgoing packets OutPackets PackLoss...
Page 531: Release Reasons In Cdr
SIP User's Manual 30. Reporting Information to External Party Field Name Description RemoteRFactor Remote R-factor LocalMosCQ Local MOS for conversation quality RemoteMosCQ Remote MOS for conversation quality SourcePort Source RTP port DestPort Destination RTP port 30.1.2 Release Reasons in CDR The possible reasons for call termination which is represented in the CDR field TrmReason are listed below: ...
Page 532 Mediant 3000  "GWAPP_CALL_AWARDED_AND "  "GWAPP_PREEMPTION"  "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" ...
Page 533: Cdr Fields For Sbc
SIP User's Manual 30. Reporting Information to External Party  "GWAPP_IDENTIFIED_CHANNEL_NOT_EXIST"  "GWAPP_SUSPENDED_CALL_BUT_CALL_ID_NOT_EXIST"  "GWAPP_CALL_ID_IN_USE"  "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"...
Page 534: Cdr Fields For Sbc Media
Mediant 3000 CDR Field Name Description TransportType Transport type (UDP, TCP, or TLS) SrcURI Source URI SrcURIBeforeMap Source URI before manipulation DstURI Destination URI Destination URI before manipulation DstURIBeforeMap Durat Call duration TrmSd Termination side (local or remote) Termination reason...
Page 535: Supported Radius Attributes
SIP User's Manual 30. Reporting Information to External Party CDR Field Name Description PacketInterval Coder packet interval LocalRtpIp Local RTP IP address LocalRtpPort Local RTP port RemoteRtpIp Remote RTP IP address RemoteRtpPort Remote RTP port InPackets Number of received packets Number of sent packets OutPackets LocalPackLoss...
Page 536 Mediant 3000 Attribute Attribute Value Purpose Example Number Name Format H323- IP address of the remote Remote- Numeric Stop Acc gateway Address H323-Conf- Up to 32 Start Acc H.323/SIP call identifier octets Stop Acc H323-Setup- Setup time in NTP format...
Page 537 SIP User's Manual 30. Reporting Information to External Party Attribute Attribute Value Purpose Example Number Name Format A unique accounting Start Acc identifier - match start & String 34832 Stop Acc stop For how many seconds the Numeric Stop Acc user received the service Number of packets Numeric...
Page 538: Event Notification Using X-Detect Header
Mediant 3000 30.2 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 539 SIP User's Manual 30. Reporting Information to External Party Table 30-6: 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 540: Querying Device Channel Resources Using Sip Options
Mediant 3000 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" <sip:anonymous@anonymous.invalid>;tag=1c25298 To: <sip:101@10.33.2.53;user=phone>;tag=1c19282 Call-ID: 11923@10.33.2.53 CSeq: 1 INVITE Contact: <sip:101@10.33.2.53>...
Page 541: Part Vii: Diagnostics
Part VII Diagnostics This part describes the diagnostics procedures.
Page 542 Reader’s Notes...
Page 543: Configuring Syslog Settings
SIP User's Manual 31. 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 569. For viewing Syslog messages in the Web interface, see Viewing Syslog Messages on page 545.
Page 544 Mediant 3000 Reader's Notes SIP User's Manual Document #: LTRT-89712...
Page 545: 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. This text file (txt) can then be sent to AudioCodes Technical Support for diagnosis and troubleshooting.
Page 546 Mediant 3000  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-89712...
Page 547: Part Viii: Appendices
Part VIII Appendices Thie part includes various appendices.
Page 548 Reader’s Notes...
Page 549: A Configuration Parameters Reference
SIP User's Manual 32. Viewing Syslog Messages 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 550: Multiple Network Interfaces And Vlan Parameters
Mediant 3000 A.1.2 Multiple Network Interfaces and VLAN Parameters The IP network interfaces and VLAN parameters are described in the table below. Table A-2: Network Interfaces and VLAN Parameters Parameter Description Multiple Interface Table Web: Multiple Interface Table This parameter table configures the Multiple Interface table EMS: IP Interface Settings for configuring logical IP addresses.
Page 551  For this parameter to take effect, a device reset is required.  This parameter is applicable only to Mediant 3000 with TP-8410.  When the parameter is enabled, user VLANs are not supported (i.e., VlANMode is set to 1).
Page 552 Mediant 3000 Parameter Description  If this parameter is enabled, separate physical networks for Media, OAMP, and Control network types (using the parameter EnableNetworkPhysicalSeparation) is disabled.  VLANs are available only when booting the device from flash. When booting using BootP/DHCP protocols, VLANs are disabled to allow easier maintenance access.
Page 553: Static Routing Parameters
SIP User's Manual 32. Viewing Syslog Messages 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 Table associated with IP interfaces defined in the Multiple Interface table [StaticRouteTable]...
Page 554 Mediant 3000 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 EMS: Network Service Class Priority Class of Service (CoS) content. [VLANNetworkServiceClassPriority] The valid range is 0 to 7.
Page 555: Nat And Stun Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Gold QoS Defines the DiffServ value for the Gold CoS content EMS: Gold Service Class Diff Serv (Streaming applications). The valid range is 0 to 63. The default value is 26. [GoldServiceClassDiffServ] Web: Bronze QoS Defines the DiffServ value for the Bronze CoS...
Page 556 Mediant 3000 Parameter Description Notes:  For this parameter to take effect, a device reset is required.  Use either the STUNServerPrimaryIP or the STUNServerDomainName parameter, with priority to the first one. NAT Parameters Web/EMS: NAT Traversal Enables the NAT mechanism.
Page 557: Nfs Parameters
SIP User's Manual 32. Viewing Syslog Messages 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 558: Dns Parameters
Mediant 3000 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 559: Dhcp Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Table Parameters' on page 88. A.1.8 DHCP Parameters The Dynamic Host Control Protocol (DHCP) parameters are described in the table below. Table A-8: DHCP Parameters Parameter Description Web: Enable DHCP Enables Dynamic Host Control Protocol (DHCP) functionality. EMS: DHCP Enable ...
Page 560: Ntp And Daylight Saving Time Parameters
Mediant 3000 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 100.
Page 561: Management Parameters
SIP User's Manual 32. Viewing Syslog Messages 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 562: Web Parameters
Mediant 3000 A.2.2 Web Parameters The Web parameters are described in the table below. Table A-11: Web Parameters Parameter Description Web: Deny Acces On Fail Count Defines the maximum number of login attempts after which the requesting IP address is blocked.
Page 563 SIP User's Manual 32. Viewing Syslog Messages Parameter Description 'Configuration File'). Notes:  For this parameter to take effect, a device reset is required.  To return to read/write after you have applied read- only using this parameter (set to 1), you need to reboot your device with an ini file that doesn't include this parameter, using the BootP/TFTP Server utility (refer to the Product Reference Manual).
Page 564: Telnet Parameters
Mediant 3000 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 565 SIP User's Manual 32. Viewing Syslog Messages Parameter Description [SendKeepAliveTrap] Enables keep-alive traps and sends them every 9/10 of the time as defined by the NATBindingDefaultTimeout parameter.  [0] = Disable  [1] = Enable Note: For this parameter to take effect, a device reset is required. Defines the base product system OID.
Page 566 Mediant 3000 Parameter Description [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. [SNMPManagerTrapPort_x The valid SNMP trap port range is 100 to 4000. The default port is 162.
Page 567: Serial Parameters
SIP User's Manual 32. Viewing Syslog Messages 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.  [0] = Enabled (default)  [1] = Disabled The RS-232 serial port can be used to change the networking parameters and view error/notification messages.
Page 568: Debugging And Diagnostics Parameters
Mediant 3000 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 569: Syslog, Cdr And Debug Parameters
SIP User's Manual 32. Viewing Syslog Messages 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 570 Mediant 3000 Parameter Description  [3] Connect & End Call = CDR report is sent to Syslog at 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.
Page 571 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Activity Types to Defines the Activity Log mechanism of the device, which sends log Report via Activity Log messages (to a Syslog server) for reporting certain types of Web Messages operations according to the below user-defined filters.
Page 572: Heartbeat Packet Parameters
Mediant 3000 A.3.3 Heartbeat Packet Parameters The Heartbeat packet parameters are described in the table below. The device sends a heartbeat packet to ensure that the far-end is passing traffic. Table A-17: Heartbeat Packet Parameters Parameter Description EMS: APS IP Address...
Page 573: Bootp Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description [RAILoopTime] Defines the time interval (in seconds) that the device periodically checks call resource availability. The valid range is 1 to 200. The default is 10. [EnableAutoRAITransmitBER] Enables the device to send RAI when the bit error rate (BER) is above 0.001.
Page 574: Security Parameters
Mediant 3000 Parameter Description [BootPDelay] Defines the interval between the device's startup and the first BootP/DHCP request that is issued by the device.  [1] = 1 second (default).  [2] = 3 second.  [3] = 6 second. ...
Page 575 SIP User's Manual 32. Viewing Syslog Messages Parameter Description [EnableSecureStartup] Enables the Secure Startup mode. In this mode, downloading the ini file to the device is restricted to a URL provided in initial configuration (see the parameter IniFileURL) or using DHCP. ...
Page 576: Https Parameters
Mediant 3000 A.4.2 HTTPS Parameters The Secure Hypertext Transport Protocol (HTTPS) parameters are described in the table below. Table A-21: HTTPS Parameters Parameter Description Web: Secured Web Connection Determines the protocol used to access the Web interface. (HTTPS)  [0] HTTP and HTTPS (default).
Page 577: Srtp Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description  For this parameter to take effect, a device reset is required.  For a description on implementing client certificates, see 'Client Certificates' on page 97. [HTTPSRootFileName] Defines the name of the HTTPS trusted root certificate file to be loaded using TFTP.
Page 578 Mediant 3000 Parameter Description negotiation of the cipher suite fails, the call is terminated. Incoming calls that don't include encryption information are rejected.  [2] Disable = The IP Profile for which this parameter is set does not support encrypted calls (i.e., SRTP).
Page 579 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web/EMS: SRTP offered Defines the offered crypto suites (cipher encryption algorithms) for Suites SRTP. [SRTPofferedSuites]  [0] = All available crypto suites (default)  [1] CIPHER SUITES AES CM 128 HMAC SHA1 80 = device uses AES-CM encryption with a 128-bit key and HMAC-SHA1 message authentication with a 80-bit tag.
Page 580: Tls Parameters
Mediant 3000 A.4.4 TLS Parameters The Transport Layer Security (TLS) parameters are described in the table below. Table A-23: TLS Parameters Parameter Description Web/EMS: TLS Version Determines the supported versions of SSL/TLS (Secure Socket [TLSVersion] Layer/Transport Layer Security.  [0] SSL 2.0-3.0 and TLS 1.0 = SSL 2.0, SSL 3.0, and TLS 1.0 are supported (default).
Page 581 SIP User's Manual 32. Viewing Syslog Messages Parameter Description terminated. Web: TLS Client Verify Server Determines whether the device, when acting as a client for TLS Certificate connections, verifies the Server certificate. The certificate is EMS: Verify Server Certificate verified with the Root CA information. [VerifyServerCertificate] ...
Page 582: Ssh Parameters
Mediant 3000 A.4.5 SSH Parameters Secure Shell (SSH) parameters are described in the table below. Table A-24: SSH Parameters Parameter Description Web/EMS: Enable SSH Server Enables the device's embedded SSH server. [SSHServerEnable]  [0] Disable (default)  [1] Enable Web/EMS: Server Port Defines the port number for the embedded SSH server.
Page 583: Ipsec Parameters
SIP User's Manual 32. Viewing Syslog Messages A.4.6 IPSec Parameters The Internet Protocol security (IPSec) parameters are described in the table below. Table A-25: IPSec Parameters Parameter Description IPSec Parameters Web: Enable IP Security Enables IPSec on the device. EMS: IPSec Enable ...
Page 584: Ocsp Parameters
Mediant 3000 Parameter Description Web: IP Security Proposal Table EMS: IPSec Proposal Table [IPSecProposalTable] This parameter table defines up to four IKE proposal settings, where each proposal defines an encryption algorithm, an authentication algorithm, and a Diffie-Hellman group identifier. [ IPsecProposalTable ]...
Page 585: Radius Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Server Port Defines the OCSP server's TCP port number. EMS: OCSP Server Port The default port number is 2560. [OCSPServerPort] Web: Default Response Determines the default OCSP behavior when the server cannot be When Server Unreachable contacted.
Page 586 Mediant 3000 Parameter Description Defines the time interval (measured in seconds) that the device [RadiusTO] waits for a response before a RADIUS retransmission is issued. The valid range is 1 to 30. The default value is 10. Web: RADIUS Authentication Defines the IP address of the RADIUS authentication server.
Page 587: Sip Media Realm Parameters
SIP User's Manual 32. Viewing Syslog Messages SIP Media Realm Parameters The Media Realm parameters are described in the table below. Table A-28: Media Realm Parameters Parameter Description Media Realm Table Web: Media Realm Table This parameter table defines the Media Realm table. The Media Realm table allows you to divide a Media-type interface (defined in the Multiple EMS: Protocol Definition >...
Page 588: Quality Of Experience Reporting
Mediant 3000 Quality of Experience Reporting The Quality of Experience parameters are described in the table below. Table A-29: Quality of Experience Parameters Parameter Description Defines the IP address of the Session Experience Manager (SEM) [QOEServerIP] server. Note: For this parameter to take effect, a device reset is required.
Page 589: Control Network Parameters
SIP User's Manual 32. Viewing Syslog Messages Control Network Parameters A.8.1 IP Group, Proxy, Registration and Authentication Parameters The proxy server, registration and authentication SIP parameters are described in the table below. Table A-30: Proxy, Registration and Authentication SIP Parameters Parameter Description IP Group Table...
Page 590 Mediant 3000 Parameter Description 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 Groups (e.g., an IP-PBX) to a Serving IP Group (e.g., an [Account] Internet Telephony Service Provider - ITSP).
Page 591 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [1] Homing = device always tries to work with the primary Proxy server (i.e., switches back to the primary Proxy whenever it's available). Note: To use this Proxy Redundancy mechanism, you need to enable the keep-alive with Proxy option, by setting the parameter EnableProxyKeepAlive to 1 or 2.
Page 592 Mediant 3000 Parameter Description 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]. If DNS resolution fails, the device attempts to route the call to the Proxy.
Page 593 SIP User's Manual 32. Viewing Syslog Messages Parameter Description returns two domain names and the A-record queries return two IP addresses each, no additional searches are performed. If set to NAPTR [2], an NAPTR query is performed. If it is successful, an SRV query is sent according to the information received in the NAPTR response.
Page 594 Mediant 3000 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 595 SIP User's Manual 32. Viewing Syslog Messages Parameter Description 215.  For configuring ini file table parameters, see 'Configuring ini File Table Parameters' on page 88. 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 596 Mediant 3000 Parameter Description [RegistrarIP] notation, e.g., 201.10.8.1:<5080>. Notes:  If not specified, the REGISTER request is sent to the primary Proxy server.  When a port number is specified, DNS NAPTR/SRV queries aren't performed, even if the parameter DNSQueryType is set to 1 or 2.
Page 597 SIP User's Manual 32. Viewing Syslog Messages 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 598 Mediant 3000 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 599 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  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 600: Network Application Parameters
Mediant 3000 Parameter Description [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 601 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Notes:  This table can include up to 32 indices (where 0 is the first index).  Each SIP Interface must have a unique signaling port (i.e., no two SIP Interfaces can share the same port - no port overlapping). ...
Page 602: General Sip Parameters
Mediant 3000 Parameter Description  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. General SIP Parameters The general SIP parameters are described in the table below.
Page 603 SIP User's Manual 32. Viewing Syslog Messages Parameter Description 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 604 Mediant 3000 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 605 SIP User's Manual 32. Viewing Syslog Messages 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.  [0] Progress = The device sends a Progress message. (default). [SIP183Behaviour] ...
Page 606 Mediant 3000 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 607 SIP User's Manual 32. Viewing Syslog Messages Parameter Description IPProfile parameter).  For more information on fax transport methods, see 'Fax/Modem Transport Modes' on page 157. [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 608 Mediant 3000 Parameter Description  The device supports up to 500 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 609 SIP User's Manual 32. Viewing Syslog Messages Parameter Description msec. Web: SIP Destination Port Defines the SIP destination port for sending initial SIP requests. 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.
Page 610 Mediant 3000 Parameter Description 480 - Temporarily Unavailable 487 - Request Terminated 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 611 SIP User's Manual 32. Viewing Syslog Messages Parameter Description INVITE sip:1234567;tgrp=hotline-ccdata;trunk- context=dsn.mil@example.com  For ISDN-to-IP calls: - 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 612 Mediant 3000 Parameter Description 'tgrp' parameter), use the parameter UseBroadsoftDTG. [UseBroadsoftDTG] Determines whether the device uses the 'dtg' parameter for routing IP-to-Tel calls to a specific Trunk Group.  [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.
Page 613 When configured, the string Info <UserAgentDisplayInfo value>/software version' is used, for example: [UserAgentDisplayInfo] User-Agent: myproduct/v.6.40.010.006 If not configured, the default string, <AudioCodes product- name>/software version' is used, for example: User-Agent: Audiocodes-Sip-Gateway-Mediant 3000/v.6.40.010.006 The maximum string length is 50 characters.
Page 614 Mediant 3000 Parameter Description expires, SDP negotiation is irrelevant and thus, the origin field is not changed. Even though some SIP devices don’t follow this behavior and don’t increment the origin value even in scenarios where they want to re-...
Page 615 SIP User's Manual 32. Viewing Syslog Messages Parameter Description (Service Unavailable) responses to indicate an unavailable service. The Retry-After header is used with the 503 (Service Unavailable) response to indicate how long the service is expected to be unavailable to the requesting SIP client. The device maintains a list of available proxies, by using the Keep-Alive mechanism.
Page 616 Mediant 3000 Parameter Description from the P-Called-Party-ID header. Web/EMS: Forking Handling Determines how the device handles the receipt of multiple SIP 18x Mode forking responses for Tel-to-IP calls. The forking 18x response is the [ForkingHandlingMode] response with a different SIP to-tag than the previous 18x response.
Page 617 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web/EMS: Gateway Name Defines a name for the device (e.g., device123.com'). [SIPGatewayName] Notes:  Ensure that the name defined is the one with which the Proxy is configured to identify the device. ...
Page 618 Mediant 3000 Parameter Description The default value is null. EMS: Use URL In Refer To Defines the source for the SIP URI set in the Refer-To header of Header outgoing REFER messages. [UseAORInReferToHeader]  [0] = Use SIP URI from Contact header of the initial call (default).
Page 619 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  '5' is the Trunk number  '8' is the B-channel  'IP=192.168.13.1' is the device's IP address  [-1] Not Configured = for ISDN spans, the progress indicator (PI) Web/EMS: Progress Indicator that is received in ISDN Proceeding, Progress, and Alerting to IP messages is used as described in the options below.
Page 620 Mediant 3000 Parameter Description stream. The device passes packets RTP to RTP packets without any processing.  [1] Force = Force transcoding on the outgoing IP leg. The device interworks the media by implementing DSP transcoding. (default) Web: Enable RFC 4117 Enables transcoding of calls according to RFC 4117.
Page 621 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Manipulation tables to leave only the last 3 digits (for example) for sending to a PBX. EMS: Use SIP URI For Defines the URI format in the SIP Diversion header. Diversion Header ...
Page 622 Mediant 3000 Parameter Description occur.  If the device is the receiver and the remote SIP UA does not send a “CN” in the SDP, then no CNG occurs. If the remote side sends a “CN”, the device attempts to be compatible with the remote side and even if the codec’s SCE is disabled, CNG occurs.
Page 623 SIP User's Manual 32. Viewing Syslog Messages Parameter Description ProtocoType parameter to 5 or 6).  Enable the TDM-over-IP feature (set the EnableTDMoverIP parameter to 1).  To configure the RTP Only mode per trunk, use the RTPOnlyModeForTrunk_ID parameter.  If per trunk configuration (using the RTPOnlyModeForTrunk_ID parameter) is set to a value other than the default, the RTPOnlyMode parameter value is ignored.
Page 624 Mediant 3000 Parameter Description Web/EMS: SIT Q850 Cause Defines the Q.850 cause value specified in the SIP Reason header For VC that is included in a 4xx response when SIT-VC (Vacant Circuit - non- [SITQ850CauseForVC] registered number Special Information Tone) is detected from the PSTN for IP-to-Tel calls.
Page 625 SIP User's Manual 32. Viewing Syslog Messages Parameter Description all the physical trunks pertaining to that Trunk Group are set to the Busy-Out condition. Each trunk uses the proper Out-Of-Service method according to the selected ISDN/CAS variant.  You can use the parameter DigitalOOSBehavior to select the method for setting digital trunks to Out-Of-Service.
Page 626 Mediant 3000 Parameter Description [\MessageManipulations] Where:  ManSetID = Defines a Manipulation Set ID for the rule. You can define the same Manipulation Set ID for multiple rules and thereby, create a group of rules that you can assign to an IP entity. The...
Page 627: Coders And Profile Parameters
SIP User's Manual 32. Viewing Syslog Messages A.10 Coders and Profile Parameters The profile parameters are described in the table below. Table A-33: Profile Parameters Parameter Description Coders Table / Coder Groups Table Web: Coders This parameter table defines the device's coders. Up to five groups of coders Table/Coder Group can be defined, where each group can consist of up to 10 coders.
Page 628 Mediant 3000 Parameter Description G.711A- 10, 20 (default), 30, Always Dynamic (0- law_VBD 40, 50, 60, 80, 100, 127) [g711AlawV G.711U- 10, 20 (default), 30, Always Dynamic (0- law_VBD 40, 50, 60, 80, 100, 127) [g711UlawV G.722 20 (default), 40, 60,...
Page 629 SIP User's Manual 32. Viewing Syslog Messages Parameter Description EVRC-B 20 (default), 40, 60, Variable Dynamic (0- (4GV) 80, 100, 120 127) [EvrcB] (default), 1/8 [1], 1/4 [2], 1/2 [3], Full [4] iLBC 20 (default), 40, 60, Dynamic (0- Disable [0] [iLBC] 80, 100, 120 (default)
Page 630 Mediant 3000 Parameter Description  For information on V.152 (and implementation of T.38 and VBD coders), see 'V.152 Support' on page 167.  For a description of using ini file table parameters, see 'Configuring ini File Table Parameters' on page 88.
Page 631 SIP User's Manual 32. Viewing Syslog Messages Parameter Description IpProfile_IpPreferen Profile Preference IpProfile_CodersGro Coder Group CodersGroup upID IpProfile_IsFaxUsed Fax Signaling Method IsFaxUsed IpProfile_JitterBufMi Dynamic Jitter Buffer DJBufMinDelay nDelay Minimum Delay IpProfile_JitterBufO Dynamic Jitter Buffer DJBufOptFactor Optimization Factor ptFactor IpProfile_IPDiffServ RTP IP DiffServ PremiumServiceClas sMediaDiffServ Signaling DiffServ...
Page 632 Mediant 3000 Parameter Description IpProfile_CallLimit Number of Calls Limit IpProfile_Disconnec Disconnect on Broken DisconnectOnBroken tOnBrokenConnecti Connection Connection 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...
Page 633 SIP User's Manual 32. Viewing Syslog Messages Parameter Description IpProfile_AMDMaxP AMD Max Post AMDMaxPostGreetin ostSilenceGreetingT Silence Greeting gSilenceTime Time 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).
Page 634 Mediant 3000 Parameter Description 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_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 635 SIP User's Manual 32. Viewing Syslog Messages Parameter Description TelProfile_EnableCu Enable Current EnableCurrentDiscon rrentDisconnect Disconnect nect TelProfile_EnableDi Enable Digit Delivery EnableDigitDelivery gitDelivery TelProfile_EnableEC Echo Canceler EnableEchoCanceller TelProfile_MWIAnal MWI Analog Lamp MWIAnalogLamp TelProfile_MWIDispl MWI Display MWIDisplay TelProfile_FlashHoo Flash Hook Period FlashHookPeriod kPeriod Enable Early Media EnableEarlyMedia...
Page 636: Channel Parameters
Mediant 3000 Parameter Description to the same call, the coders and common parameters (i.e., parameters configurable in both IP and Tel Profiles) of the preferred profile are applied to that call. If the Tel and IP Profiles are identical, the Tel Profile parameters take precedence.
Page 637 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Delay Answer Detector and the time that the detector actually starts to [AnswerDetectorActivityDel operate. The valid range is 0 to 1023. The default is 0. Web: Answer Detector Silence Currently, not supported. Time [AnswerDetectorSilenceTim Web: Answer Detector...
Page 638 Mediant 3000 Parameter Description  [1] Acoustic Echo suppressor - netw = Echo canceller for IP side. Web: Attenuation Intensity Defines the acoustic echo suppressor signals identified as echo [AcousticEchoSuppAttenuat attenuation intensity. ionIntensity] The valid range is 0 to 3. The default is 0.
Page 639: Coder Parameters
SIP User's Manual 32. Viewing Syslog Messages A.11.2 Coder Parameters The coder parameters are described in the table below. Table A-35: Coder Parameters Parameter Description Web: G729EV local MBS Defines the maximal bit rate (in kilobits per second - Kbps) that can EMS: G729 EV Local MBS be used by the G.729EV coder for a specific channel.
Page 640 Mediant 3000 Parameter Description  [5] 20 KBPS  [6] 22 KBPS  [7] 24 KBPS  [8] 26 KBPS  [9] 28 KBPS  [10] 30 KBPS  [11] 32 KBPS  [15] Undefined Web/EMS: Microsoft RTA Enables Forward Error Correction (FEC) for the Microsoft real-time Forward Error Correction Mode audio narrowband coder (MS RTA NB).
Page 641 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Note: Bandwidth Efficient mode is not supported; the mode is always Octet-aligned. Web: DSP Template Mix Table EMS: VoP Media Provisioning > General Settings [DSPTemplates] This parameter table allows the device to use a combination of two DSP templates and determines the percentage of DSP resources allocated per DSP template.
Page 642: Dtmf Parameters
Mediant 3000 A.11.3 DTMF Parameters The dual-tone multi-frequency (DTMF) parameters are described in the table below. Table A-36: DTMF Parameters Parameter Description Web/EMS: DTMF Transport Determines the DTMF transport type. Type  [0] DTMF Mute = Erases digits from voice stream and doesn't [DTMFTransportType] relay to remote.
Page 643: Rtp, Rtcp And T.38 Parameters
SIP User's Manual 32. Viewing Syslog Messages A.11.4 RTP, RTCP and T.38 Parameters The RTP, RTCP and T.38 parameters are described in the table below. Table A-37: RTP/RTCP and T.38 Parameters Parameter Description Web: Dynamic Jitter Buffer Minimum Defines the minimum delay (in msec) for the Dynamic Jitter Delay Buffer.
Page 644 Mediant 3000 Parameter Description using the RFC2198PayloadType parameter.  The RTP redundancy dynamic payload type can be included in the SDP, by using the EnableRTPRedundancyNegotiation parameter.  This parameter can also be configured per IP Profile, using the IPProfile parameter.
Page 645 The range of possible UDP ports is 6,000 to 64,000. The default base UDP port is 6000. Once this parameter is configured, the UDP port range (lower to upper boundary) is calculated as follows:  Mediant 3000/TP-6310: BaseUDPport to (BaseUDPport + 4031*10)  Mediant 3000/TP-8410: BaseUDPport to (BaseUDPport + 4031*10) Notes: ...
Page 646 Mediant 3000 Parameter Description required.  The UDP ports are allocated randomly to channels.  You can define a UDP port range per Media Realm (see Configuring Media Realms on page 182).  If RTP Base UDP Port is not a factor of 10, the following message is generated: 'invalid local RTP port'.
Page 647 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [1] Enable = Enables Note: For this parameter to take effect, a device reset is required. Web: Minimum Gap Size Defines the voice quality monitoring - minimum gap size EMS: GMin (number of frames).
Page 648: Gateway And Ip-To-Ip Parameters
Mediant 3000 A.12 Gateway and IP-to-IP Parameters A.12.1 Fax and Modem Parameters The fax and modem parameters are described in the table below. Table A-38: Fax and Modem Parameters Parameter Description Web: Fax Transport Mode Determines the fax transport mode used by the device.
Page 649 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  Type [0] Disable = Disable (Transparent) EMS: V32 Transport  [1] Enable Relay = N/A [V32ModemTransportType]  [2] Enable Bypass = (default)  [3] Events Only = Transparent with Events Notes: ...
Page 650 Mediant 3000 Parameter Description mode is not recommended. Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see 'Configuring IP Profiles' on page 236). Web: T38 Version Determines the T.38 fax relay version. [SIPT38Version] ...
Page 651 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Fax Relay ECM Enable Enables Error Correction Mode (ECM) mode during fax relay. EMS: Relay ECM Enable  [0] Disable. [FaxRelayECMEnable]  [1] Enable (default). Web: Fax/Modem Bypass Coder Determines the coder used by the device when performing Type fax/modem bypass.
Page 652 Mediant 3000 Parameter Description EMS: Enable Inband Network Enables in-band network detection related to fax/modem. Detection  [0] = Disable (default) [EnableFaxModemInbandNetw  [1] = Enable. When this parameter is enabled on Bypass and orkDetection] transparent with events mode (VxxTransportType is set to 2 or...
Page 653 SIP User's Manual 32. Viewing Syslog Messages Parameter Description and T.38 coders, the UDP port published in SDP for RTP and for T38 must be different. Therefore, set the T38UseRTPPort parameter to 0. Web/EMS: T.38 Max Datagram Defines the maximum size of a T.38 datagram that the device can Size receive.
Page 654 Mediant 3000 Parameter Description  For this parameter to take effect, a device reset is required.  This parameter is applicable only if the IsFaxUsed parameter is set to 1 (T.38 Relay). [T38FaxSessionImmediateStart Enables fax transmission of T.38 “no-signal” packets to the terminating fax machine.
Page 655: Dtmf And Hook-Flash Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description [V1501SPRTTransportChannel 2MaxWindowSize] Web: SPRT Transport Ch.3 Max Defines the maximum payload size for V.150.1 SPRT Transport Payload Size Channel 3. [V1501SPRTTransportChannel The range is 140 to 256. The default is 140. 3MaxPayloadSize] A.12.2 DTMF and Hook-Flash Parameters The DTMF and hook-flash parameters are described in the table below.
Page 656 Mediant 3000 Parameter Description Where 16 is the DTMF code for hook flash.  [7] INFO (HUAWAEI) = Sends a SIP INFO message with Hook- Flash indication. The device sends the INFO message as follows: Content-Length: 17 Content-Type: application/sscc event=flashhook Note: The device can interwork DTMF HookFlashCode to SIP INFO messages with Hook Flash indication.
Page 657 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  When out-of-band DTMF transfer is used ([1], [2], [3], or [5]), the parameter DTMFTransportType is automatically set to 0 (DTMF digits are erased from the RTP stream).  When RFC 2833 (4) is selected, the device: Negotiates RFC 2833 payload type using local and remote SDPs.
Page 658: Digit Collection And Dial Plan Parameters
Mediant 3000 Parameter Description If the called number in IP-to-Tel call includes the characters 'w' or 'p', the device places a call with the first part of the called number (before 'w' or 'p') and plays DTMF digits after the call is answered. If the character 'w' is used, the device waits for detection of a dial tone before it starts playing DTMF digits.
Page 659 SIP User's Manual 32. Viewing Syslog Messages Parameter Description [Tel2IPSourceNumberMappi Defines the Dial Plan index in the external Dial Plan file for the Tel- ngDialPlanIndex] to-IP Source Number Mapping feature. The valid value range is 0 to 7, defining the Dail Plan index [Plan x] in the Dial Plan file.
Page 660: Voice Mail Parameters
Mediant 3000 Parameter Description that are received from the PSTN or IP during overlap dialing. [TimeBetweenDigits] When this inter-digit timeout expires, the device uses the collected digits to dial the called destination number. The valid range is 1 to 10. The default value is 4.
Page 661 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Others >> 302 If the device receives a Request-URI that includes a 'target' and 'cause' parameter, the 'target' is mapped to the Redirect phone number and the 'cause' is mapped to the Redirect number reason. [WaitForBusyTime] Defines the time (in msec) that the device waits to detect busy and/or reorder tones.
Page 662 Mediant 3000 Parameter Description Web: MWI On Digit Pattern Defines the digit code used by the device to notify the PBX of messages EMS: MWI On Code waiting for a specific extension. This code is added as prefix to the [MWIOnCode] dialed number.
Page 663 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Forward on No Defines the digit pattern used by the PBX to indicate 'call forward on no Answer Digit Pattern answer' when the original call is received from an internal extension. (Internal) The valid range is a 120-character string.
Page 664 Mediant 3000 Parameter Description Web: Internal Call Digit Defines the digit pattern used by the PBX to indicate an internal call. Pattern The valid range is a 120-character string. EMS: Digit Pattern Internal Call [DigitPatternInternalCall] Web: External Call Digit Defines the digit pattern used by the PBX to indicate an external call.
Page 665: Supplementary Services Parameters
SIP User's Manual 32. Viewing Syslog Messages A.12.5 Supplementary Services Parameters This subsection describes the device's supplementary telephony services parameters. Caller ID Parameters A.12.5.1 The caller ID parameters are described in the table below. Table A-42: Caller ID Parameters Parameter Description Web: Asserted Identity Mode Determines whether the SIP header P-Asserted-Identity or P-...
Page 666: Call Waiting Parameters
Mediant 3000 Parameter Description 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. EMS: Transport Type  [0] Disable = The caller ID signal is not detected - DTMF digits [CallerIDTransportType] remain in the voice stream.
Page 667: Call Hold Parameters
SIP User's Manual 32. Viewing Syslog Messages Call Hold Parameters A.12.5.4 The call hold parameters are described in the table below. Table A-45: Call Hold Parameters Parameter Description Web/EMS: Enable Hold Enables interworking of the Hold/Retrieve supplementary service from [EnableHold] PRI to SIP.
Page 668: Call Transfer Parameters
Mediant 3000 Call Transfer Parameters A.12.5.5 The call transfer parameters are described in the table below. Table A-46: Call Transfer Parameters Parameter Description Web/EMS: Enable Transfer Enables the Call Transfer feature. [EnableTransfer]  [0] Disable = Disable the call transfer service.
Page 669 SIP User's Manual 32. Viewing Syslog Messages Parameter Description After the destination phone number is collected, it is sent to the transferee in a SIP REFER request in a Refer-To header. The call is then terminated and a confirmation tone is played to the channel.
Page 670: Emergency Call Parameters
Mediant 3000 Emergency Call Parameters A.12.5.6 The emergency call parameters are described in the table below. Table A-47: Emergency Call Parameters Parameter Description Web/EMS: Emergency Defines a list of “emergency” numbers. Numbers For CAS, and ISDN: These emergency numbers are used for the...
Page 671: Mlpp Parameters
SIP User's Manual 32. Viewing Syslog Messages MLPP Parameters A.12.5.8 The Multilevel Precedence and Preemption (MLPP) parameters are described in the table below. Table A-49: MLPP Parameters Parameter Description Web/EMS: Call Priority Mode Enables priority calls handling. [CallPriorityMode]  [0] Disable = Disable (default). ...
Page 672 Mediant 3000 Parameter Description  [9] 9 = FLASH-OVERRIDE-OVERRIDE If the incoming SIP INVITE request doesn't contain a valid priority value in the SIP Resource-Priority header, the default value is used in the Precedence IE (after translation to the relevant ISDN Precedence value) of the outgoing PRI Setup message.
Page 673 SIP User's Manual 32. Viewing Syslog Messages Parameter Description If MLPPDefaultServiceDomain is set to 'FFFFFF', the device interworks the non-MLPP ISDN call to non-MLPP SIP call, and the outgoing INVITE does not contain the Resource-Priority header. The valid value is a 6 hexadecimal digits. The default is "000000". Note: This parameter is applicable only to the MLPP NI-2 ISDN variant with CallPriorityMode set to 1.
Page 674: Tty/Tdd Parameters
Mediant 3000 Parameter Description [MLPPFlashRTPDSCP] Note: If set to -1, the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call (using the parameter IPProfile). Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash-Override precedence call Flash Override level.
Page 675: Pstn Parameters
SIP User's Manual 32. Viewing Syslog Messages A.12.6 PSTN Parameters This subsection describes the device's PSTN parameters. General Parameters A.12.6.1 The general PSTN parameters are described in the table below. Table A-51: General PSTN Parameters Parameter Description Web/EMS: Protocol Type Defines the PSTN protocol for all the Trunks.
Page 676 Mediant 3000 Parameter Description  [20] T1 HKT ISDN = ISDN PRI (T1) protocol for the Hong Kong - HKT.  [21] E1 QSIG = ECMA 143 QSIG over E1  [22] E1 TNZ = ISDN PRI protocol for Telecom New Zealand (similar to ETSI) ...
Page 677 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web/EMS: Trace Level Defines the trace level: [TraceLevel]  [0] No Trace (default)  [1] Full ISDN Trace  [2] Layer 3 ISDN Trace  [3] Only ISDN Q.931 Messages Trace ...
Page 678 Mediant 3000 Parameter Description [LineCode_x] Same as the description for parameter LineCode, but for a specific trunk ID (where 0 depicts the first trunk). [AdminState] Defines the administrative state for all trunks.  [0] = Lock the trunk; stops trunk traffic to configure the trunk protocol type.
Page 679: Tdm Bus And Clock Timing Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Note: For this parameter to take effect, a device reset is required. Web: Enable TDM Tunneling Enables TDM tunneling. EMS: TDM Over IP  [0] Disable = Disabled (default).  [1] Enable = TDM Tunneling is enabled. [EnableTDMoverIP] When TDM Tunneling is enabled, the originating device automatically initiates SIP calls from all enabled B-channels...
Page 680 Mediant 3000 Parameter Description Web/EMS: Idle ABCD Pattern Defines the ABCD (CAS) Pattern that is applied to the CAS [IdleABCDPattern] signaling bus when the channel is idle. The valid range is 0x0 to 0xF. The default is -1 (i.e., default pattern is 0000).
Page 681 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [TDMBusPSTNAutoClockEnabl [1] Enable = Recovers the clock from any connected synchronized slave E1/T1 line. If this trunk loses its synchronization, the device attempts to recover the clock from the next trunk. Note that initially, the device attempts to recover the clock from the trunk defined by the parameter TDMBusLocalReference.
Page 682 Synchronization or Line Synchronization modes (refer to the parameter TMMode).  This parameter is applicable to Mediant 3000 systems housing TP-8410 blades. For Mediant 3000 systems housing TP-6310 blades, configure this parameter to [0] to use a 12 ppm reference.
Page 683: Cas Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: E1 External Reference Defines the transmission power (in ohm) between the timing Transmit Line Build Out module on the SAT blade and the E1 external reference clock. EMS: E1 Line Build Out ...
Page 684 Mediant 3000 Parameter Description Web: CAS Table per Trunk Defines the CAS protocol per trunk (where x denotes the trunk EMS: Trunk CAS Table Index ID) from a list of CAS protocols defined by the parameter [CASTableIndex_x] CASFileName_x. For example, the below configuration specifies Trunks 0 and 1 to use the E&M Winkstart CAS (E_M_WinkTable.dat) protocol,...
Page 685 SIP User's Manual 32. Viewing Syslog Messages Parameter Description index>, (e.g., "1-10:1,11-31:3"). Every B-channel (including 16 for E1) must belong to a channel group. Below is an example for configuring an E1 CAS trunk (Trunk 5) with several CAS variants: ProtocolType_5 = 8 CASFILENAME_2='E1_R2D' CASFILENAME_7= E_M_ImmediateTable_A-Bit.txt'...
Page 686: Isdn Parameters
Mediant 3000 Parameter Description  [-1] Default = Default value. Web: Digit Signaling System Defines which Signaling System to use in both directions [CASStateMachineDigitSignaling (detection\generation). System]  [0] DTMF = Uses DTMF signaling.  [1] MF = Uses MF signaling (default).
Page 687 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  For this parameter to take effect, a device reset is required.  This parameter is applicable only to T1 ISDN protocols.  For more information on NFAS, see 'ISDN Non-Facility Associated Signaling (NFAS)' on page 267.
Page 688 Mediant 3000 Parameter Description  [2048] CHAN ID IN FIRST RS = The device sends Channel ID in the first response to an incoming Q.931 Call Setup message. Otherwise, the Channel ID is sent only if the device requires changing the proposed Channel ID (default).
Page 689 SIP User's Manual 32. Viewing Syslog Messages Parameter Description which sending of Status message is optional.  [2] NO STATUS ON INV OP IE = Q.931 Status message isn't sent if an optional IE with invalid content is received. By default, the Status message is sent. Note: This option is applicable only to ISDN variants in which sending of Status message is optional.
Page 690 Mediant 3000 Parameter Description  [1073741824] QSI ENCODE INTEGER = If this bit is set, INTEGER ASN.1 type is used in operator coding (compliant to new ECMA standards); otherwise, OBJECT IDENTIFIER ASN.1 type is used. Note: This option is applicable only to QSIG.
Page 691 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [256] START WITH B CHAN OOS = B-channels start in the Out-Of-Service state (OOS).  [512] CHAN ALLOC LOWEST = CC allocates B- channels starting from the lowest available B-channel id. ...
Page 692 Mediant 3000 Parameter Description Note: This option is applicable only to the E10 variant.  [1024] = Numbering plan/type for T1 IP-to-Tel calling numbers are defined according to the manipulation tables or according to the RPID header (default). Otherwise, the plan/type for T1 calls are set according to the length of the calling number.
Page 693: Ds3 Parameters
SIP User's Manual 32. Viewing Syslog Messages DS3 Parameters A.12.6.5 The DS3 parameters are described in the table below. Note: DS3 interface is applicable only to Mediant 3000 with TP-6310. Table A-55: DS3 Parameters Parameter Description DS3 Settings Table [DS3Config] This parameter table configures the DS3 (T3) interfaces.
Page 694: Sdh/Sonet Parameters
[2] Down = Administrative status Down (currently not supported) SDH/SONET Parameters A.12.6.6 The SDH/SONET parameters are described in the table below. Note: SDH/SONET interface is applicable only to Mediant 3000 with TP-6310. Table A-56: SDH/SONET Parameters Parameter Description Web/EMS: Transmission Type Defines the PSTN transmission type for the device.
Page 695 SIP User's Manual 32. Viewing Syslog Messages Parameter Description parameter SDHFbrGrp_Mapping_Type to 0 (i.e., VT1.5) and ProtocolType for DS1. Notes:  For this parameter to take effect, a device reset is required.  This parameter is relevant only when the parameter TDMBusType is set to acFRAMERS (2), PSTNTransmissionType is set to SONET/SDH (1), and SdhFbrGrp_SdhSonetMode is not set to UNKNOWN (0).
Page 696 Notes:  For this parameter to take effect, a device reset is required.  This parameter is applicable only to Mediant 3000/TP-6310.  This parameter is relevant only when the parameter TDMBusType is set to acFRAMERS (2), and PSTNTransmissionType is set to Optical SONET or SDH Transmission type (1).
Page 697: Isdn And Cas Interworking Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description [SonetSdhMediumCircuitIdentifi Defines the SDH/SONET circuit name. Note: For this parameter to take effect, a device reset is required. [SDHPmEnable] Enables performance monitoring for the fiber group.  [0] = Disables performance monitoring. ...
Page 698 Mediant 3000 Parameter Description SIP 484 Address Incomplete response in order to maintain the current dialog session and receive additional digits from subsequent INVITEs. Note: When IP-to-Tel overlap dialing is enabled, to send ISDN Setup messages without the Sending Complete IE, the ISDNOutCallsBehavior parameter must be set to USER SENDING COMPLETE (2).
Page 699 SIP User's Manual 32. Viewing Syslog Messages Parameter Description only accepts digits from ISDN Info messages.  [0] Don't Mute (default)  [1] Mute DTMF in Overlap Dialing = The device ignores in- band DTMF digits received during ISDN overlap dialing (disables the DTMF in-band detector).
Page 700 Mediant 3000 Parameter Description Web/EMS: Enable QSIG Tunneling Enables QSIG tunneling-over-SIP for all calls. This is according [EnableQSIGTunneling] to IETF Internet-Draft draft-elwell-sipping-qsig-tunnel-03 and ECMA-355 and ETSI TS 102 345.  [0] Disable = Disable (default).  [1] Enable = Enable QSIG tunneling from QSIG to SIP and vice versa.
Page 701 SIP User's Manual 32. Viewing Syslog Messages Parameter Description same phone number (ISDN Calling and Called numbers) for interworking between ISDN and SIP networks.  [0] = ASCII - IA5 format that allows up to 20 digits. Indicates that the 'isub' parameter value needs to be encoded using ASCII characters (default) ...
Page 702 Mediant 3000 Parameter Description SIP/2.0 then the device maps this called number to the destination number of the ISDN Setup message, and the destination subaddress in this ISDN Setup message remains empty.  If the called number (that appears in the user part of the Request-URI) does not start with zero, for example, INVITE sip:5694564@host.domain:user=phone SIP/2.0...
Page 703 SIP User's Manual 32. Viewing Syslog Messages Parameter Description ProgressIndicator2ISDN_ID is configured differently).  If the parameter LocalISDNRBSource is set to 0, the device doesn't play an RBT and an Alert message (without PI) is sent to the ISDN. In this case, the PBX/PSTN plays the RBT to the originating terminal by itself.
Page 704 Mediant 3000 Parameter Description Trunk_ID] below).  [1] Service = Sends ISDN In or Out of Service (only for ISDN protocols that support Service message).  [2] D-Channel = Takes D-Channel down or up (ISDN only).  [3] Alarm = Sends or clears PSTN AIS Alarm (ISDN and CAS).
Page 705 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Release Cause is found, the SIP response assigned to it is sent to the IP side. If no match is found, the default static mapping is used. Notes:  This parameter can appear up to 12 times. ...
Page 706 Mediant 3000 Parameter Description =4. (This format is according to IETF Internet-Draft draft- johnston-sipping-cc-uui-04.)  [2] = Format: User-to-User with encoding=hex at the end and pd embedded as the first byte. User-to- User=043030373435313734313635353b313233343b3834; encoding=hex. Where "04" at the beginning of this message is the pd.
Page 707 SIP User's Manual 32. Viewing Syslog Messages Parameter Description before an ISDN Connect message is received.  [0] = Connect message isn't sent after SIP 183 Session Progress message is received (default).  [1] = Connect message is sent after SIP 183 Session Progress message is received.
Page 708 Mediant 3000 Parameter Description forwarding is based on Call Deflection for Euro ISDN (ETS-300- 207-1) and QSIG (ETSI TS 102 393).  [0] Disable (default)  [1] Enable = Enables ISDN call rerouting. When the device sends the INVITE message to the remote SIP entity and...
Page 709 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  For this parameter to take effect, a device reset is required.  The device's Software Upgrade Key must contain the 'IPMDetector' DSP option.  When enabled (1), the number of available channels is reduced.
Page 710 Mediant 3000 Parameter Description ISDNGeneralCCBehavior must be set to 16384. [Enable911LocationIdIP2Tel] Enables interworking of Emergency Location Identification from SIP to PRI.  [0] = Disabled (default)  [1] = Enabled When enabled, the From header received in the SIP INVITE is translated into the following ISDN IE's: ...
Page 711 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Note: A specific NFA CAS table is required.  [2] = Supports ISDN transfer - Release Link Trunk (RLT) (DMS-100), Two B Channel Transfer (TBCT) (NI2), Explicit Call Transfer (ECT) (EURO ISDN), and Path Replacement (QSIG).
Page 712 Mediant 3000 Parameter Description SIP NOTIFY with 200 OK message; otherwise, the device sends a NOTIFY with 4xx message.  [6] = Supports AT&T toll free out-of-band blind transfer for trunks configured with the 4ESS ISDN protocol. AT&T courtesy transfer is a supplementary service which enables a user (e.g., user "A") to transform an established call...
Page 713 SIP User's Manual 32. Viewing Syslog Messages Parameter Description call is in Connect state. Note: For RLT ISDN transfer (TrunkTransferMode = 2 and ProtocolType = 14 DMS-100), this parameter must be set to 1. [ISDNTransferCompleteTimeout] Defines the timeout (in seconds) for determining ISDN call transfer (ECT, RLT, or TBCT) failure.
Page 714: Answer And Disconnect Supervision Parameters
Mediant 3000 Parameter Description from the P-Asserted-Identity and Privacy headers. [CASSendHookFlash] Enables sending Wink signal toward CAS trunks.  [0] = Disable (default).  [1] = Enable. If the device receives a mid-call SIP INFO message with flashhook event body (as shown below) and this parameter is set to 1, the device generates a wink signal toward the CAS trunk.
Page 715 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Send Digit Pattern on Defines a digit pattern to send to the Tel side after a SIP 200 OK is Connect received from the IP side. The digit pattern is a user-defined DTMF EMS: Connect Code sequence that is used to indicate an answer signal (e.g., for billing).
Page 716 Mediant 3000 Parameter Description  iod] This parameter is applicable only for DSP templates 2 and 3.  For this parameter to take effect, a device reset is required. Web: Silence Detection Method Determines the silence detection method. [FarEndDisconnectSilenceMe ...
Page 717: Tone Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Notes:  This parameter is applicable only to CAS protocols.  This parameter can also be configured per Tel Profile, using the TelProfile parameter. A.12.9 Tone Parameters This subsection describes the device's tone parameters. Telephony Tone Parameters A.12.9.1 The telephony tone parameters are described in the table below.
Page 718 Mediant 3000 Parameter Description (for CAS channels) parameter. Web: Play Ringback Tone to Enables the play of the ringback tone (RBT) to the Tel side and determines the method for playing the RBT. It applies to all trunks EMS: Play Ring Back Tone To that are not configured by the parameter PlayRBTone2Trunk.
Page 719: Tone Detection Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description EMS: Play RBT On ISDN or Explicit Call Transfer (ECT) call transfers to the originator when Transfer the second leg receives an ISDN Alerting or Progress message. [PlayRBTOnISDNTransfer]  [0] Don't Play (default). ...
Page 720 Mediant 3000 Parameter Description  SITDetectorEnable = 1  UserDefinedToneDetectorEnable = 1  ISDNDisconnectOnBusyTone = 1 (applicable for Busy, Reorder and SIT tones) Another parameter for handling the SIT tone is SITQ850Cause, which determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when a SIT tone is detected on an IP-to-Tel call.
Page 721: Metering Tone Parameters
SIP User's Manual 32. Viewing Syslog Messages Metering Tone Parameters A.12.9.3 The metering tone parameters are described in the table below. Table A-61: 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 722: Trunk Groups And Routing Parameters
Mediant 3000 A.12.10 Trunk Groups and Routing Parameters The routing parameters are described in the table below. Table A-62: 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 723 SIP User's Manual 32. Viewing Syslog Messages Parameter Description 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. MWI Activate requests are interworked to SIP NOTIFY MWI messages.
Page 724 Mediant 3000 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...
Page 725 SIP User's Manual 32. Viewing Syslog Messages Parameter Description 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. If no Display Name is received from the Tel side, the Tel Source Number is used as the IP Source Number and also as the IP Display Name.
Page 726 Mediant 3000 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,,;...
Page 727 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  For available notations for depicting a range of multiple numbers, see 'Dialing Plan Notation for Routing and Manipulation' on page 775.  For a description on using ini file table parameters, see 'Configuring ini File Table Parameters' on page 88.
Page 728 Mediant 3000 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 729: Alternative Routing Parameters
SIP User's Manual 32. Viewing Syslog Messages A.12.11 Alternative Routing Parameters The alternative routing parameters are described in the table below. Table A-63: 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 730 Mediant 3000 Parameter Description Web: Alt Routing Tel to IP Determines the method used by the device for periodically Connectivity Method querying the connectivity status of a destination IP address. EMS: Alternative Routing  [0] ICMP Ping (default) = Internet Control Message Protocol Telephone to IP Connection (ICMP) ping messages.
Page 731 SIP User's Manual 32. Viewing Syslog Messages Parameter Description only when a Proxy is not used.  When there is no response to an INVITE message (after INVITE retransmissions), the device issues an internal 408 'No Response' implicit release reason. ...
Page 732: Number Manipulation Parameters
Mediant 3000 Parameter Description [\ForwardOnBusyTrunkDest] For example, the below configuration forwards IP-to-Tel calls to destination user “112” at host IP address 10.13.4.12, port 5060, using transport protocol TCP, if Trunk Group ID 2 is unavailable: ForwardOnBusyTrunkDest 1 = 2, 112@10.13.4.12:5060;transport=tcp;...
Page 733 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Tel2IP Default Redirect Reason Determines the default redirect reason for Tel-to-IP calls [Tel2IPDefaultRedirectReason] when no redirect reason (or “unknown”) exists in the received Q931 ISDN Setup message. The device includes this default redirect reason in the SIP History-Info header of the outgoing INVITE.
Page 734 Mediant 3000 Parameter Description Web: Set TEL-to-IP Redirect Reason Defines the redirect reason for Tel-to-IP calls. If redirect [SetTel2IpRedirectReason] (diversion) information is received from the Tel, the redirect reason is set to the value of this parameter before the device sends it on to the IP.
Page 735 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Web: Copy Destination Number to Determines whether the device copies the received Redirect Number ISDNcalled number to the outgoing SIP Diversion header for EMS: Copy Dest to Redirect Number Tel-to-IP calls (even if a Redirecting Number IE is not [CopyDest2RedirectNumber] received in the ISDN Setup message).
Page 736 Mediant 3000 Parameter Description  [1] Yes = Add Trunk Group ID as prefix to called number. Notes:  This option can be used to define various routing rules.  To use this feature, you must configure the Trunk Group IDs (see Configuring Trunk Group Table on page 277).
Page 737 SIP User's Manual 32. Viewing Syslog Messages Parameter Description [RemovePrefix] Routing Table' - see 'Configuring Inbound IP Routing Table' on page 306) from a telephone number for an IP- to-Tel call before forwarding it to Tel. For example: To route an incoming IP-to-Tel call with destination number 21100, the Inbound IP Routing Table' is scanned for a matching prefix.
Page 738 Mediant 3000 Parameter Description Calling Name Manipulations IP-to-Tel Table [CallingNameMapIp2Tel] Configures rules for manipulating the calling name (caller ID) in the received SIP message for IP-to-Tel calls. This can include modifying or removing the calling name. The format of this ini file parameter table is as follows:...
Page 739 SIP User's Manual 32. Viewing Syslog Messages Parameter Description NumberMapIp2Tel 0 = 01,034,10.13.77.8,$$,0,$$,2,$$,667,$$; NumberMapIp2Tel 1 = 10,10,1.1.1.1,255,255,3,0,5,100,$$,255; Notes:  This table parameter can include up to 100 indices.  The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 740 Mediant 3000 Parameter Description NumberMapTel2Ip_LeaveFromRight, NumberMapTel2Ip_Prefix2Add, NumberMapTel2Ip_Suffix2Add, NumberMapTel2Ip_IsPresentationRestricted, NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_ SrcIPGroupID; [\NumberMapTel2Ip] For example: NumberMapTel2Ip 0 = 01,$$,*,0,0,2,$$,$$,971,$$,$$,$$,$$; NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; Notes:  This table parameter can include up to 120 indices (0- 119).  The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 741 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Notes:  This table parameter can include up to 120 indices.  The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.  If the called and calling numbers match the DestinationPrefix, SourcePrefix, and/or SourceAddress conditions, then the RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add,...
Page 742 Mediant 3000 Parameter Description 22,03,$$,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.  If the called and calling numbers match the...
Page 743 SIP User's Manual 32. Viewing Syslog Messages Parameter Description The format of this parameter is as follows: [RedirectNumberMapIp2Tel] FORMAT RedirectNumberMapIp2Tel_Index = RedirectNumberMapIp2Tel_DestinationPrefix, RedirectNumberMapIp2Tel_RedirectPrefix, RedirectNumberMapIp2Tel_SourceAddress, RedirectNumberMapIp2Tel_NumberType, RedirectNumberMapIp2Tel_NumberPlan, RedirectNumberMapIp2Tel_RemoveFromLeft, RedirectNumberMapIp2Tel_RemoveFromRight, RedirectNumberMapIp2Tel_LeaveFromRight, RedirectNumberMapIp2Tel_Prefix2Add, RedirectNumberMapIp2Tel_Suffix2Add, RedirectNumberMapIp2Tel_IsPresentationRestricted; [\RedirectNumberMapIp2Tel] For example: RedirectNumberMapIp2Tel 1 = *, 88, *, 1, 1, 2, 0, 255, 9, , 255;...
Page 744 Mediant 3000 Parameter Description  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.  If the table's matching characteristics rule (i.e.,...
Page 745: Ldap Parameters
SIP User's Manual 32. Viewing Syslog Messages A.12.13 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 191. Table A-65: LDAP Parameters Parameter Description...
Page 746: Least Cost Routing Parameters
Mediant 3000 Parameter Description Web: MS LDAP MOBILE Number Defines the name of the attribute that represents the user attribute name Mobile number in the Microsoft AD database. [MSLDAPMobileNumAttributeName] The valid value is a string of up to 49 characters. The default is "mobile".
Page 747: Sbc Parameters
SIP User's Manual 32. Viewing Syslog Messages A.13 SBC Parameters The SBC parameters are described in the table below. Table A-67: SBC Parameters Parameter Description Web: Enable SBC Enables the Session Border Control (SBC) application. EMS: Enable SBC  [0] Disable (default) [EnableSBCApplication] ...
Page 748 Mediant 3000 Parameter Description Web: Keep original user in Determines whether the device replaces the Contact user with a Register unique Contact user in the outgoing message in response to a [SBCKeepContactUserinRegi REGISTER request. ster]  [0] Disable = (default) The device replaces the original Contact user with a unique Contact user, for example: ...
Page 749 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [1] = The device changes the URI in the Contact header of the received SIP 3xx response to its own URI and adds a special user prefix ("T~&R_”), which is then sent to the FEU. The FEU then sends a new INVITE to the device, which the device then sends to the correct destination.
Page 750 Mediant 3000 Parameter Description Web: Lifetime of the nonce in Defines the lifetime (in seconds) that the current nonce is valid for seconds server-based authentication. The device challenges a message [AuthNonceDuration] that attempts to use a server nonce beyond this period. This parameter is used to provide replay protection (i.e., ensures that...
Page 751 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Time] terminated by the device). When set to 0, the device uses the value set by the SBCUserRegistrationTime parameter for the device's response. The valid range is 0 to 2,000,000 seconds. The default is 0. Web: SBC GRUU Mode Determines the Globally Routable User Agent (UA) URI (GRUU) support, according to RFC 5627.
Page 752 Mediant 3000 Parameter Description Web: SBC Direct Media Enables the No Media Anchoring feature (i.e., direct media) for all [SBCDirectMedia] SBC calls. No Media Anchoring uses SIP signaling capabilities without handling the RTP/SRTP (media) flow between remote SIP user agents (UA). The RTP packets do not traverse the device, instead, the two SIP UAs establish a direct RTP/SRTP flow between one another.
Page 753 SIP User's Manual 32. Viewing Syslog Messages Parameter Description belong to the same SRD is configurable only (in this case). Web: Transcoding Mode Defines the voice transcoding mode (media negotiation) between [TranscodingMode] the two SBC legs for the SBC application. ...
Page 754 Mediant 3000 Parameter Description [IpProfile_SBCHistoryInfoMo more information on interworking of the History-Info and Diversion headers, see 'Interworking SIP Diversion and History-Info Headers' on page 382.  [0] Don't Care = History-Info header is not handled. (default)  [1] Add = Diversion header converted to a History-Info header.
Page 755 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  For more information on the Allowed Coders feature, see 'Coder Restrictions Control' on page 376. SBC Preferences Mode Determines the order of the Extension coders (coders added if [SBCPreferencesMode] there are no common coders between SDP offered coders and Allowed coders) and Allowed coders (defined in the Allowed Coders Group table) in the outgoing SIP message (in the SDP).
Page 756 Mediant 3000 Parameter Description the coders of the selected Coders Group ID (SBCFaxCodersGroupID, then the device uses this coder. If no match exists, the device uses the first listed coder of the matched coders between the incoming offer coders (from the calling "fax") and the coders of the selected Coders Group ID.
Page 757 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Allowed Audio Coders Table Web: Allowed Audio Coders This parameter table allows you to define up to 5 Allowed Coders CLI: configure voip > sbc Groups, each with up to 10 coders. The Allowed Coders Group allowed-coders-group determines the coders that can be used for a specific SBC leg.
Page 758 Mediant 3000 Parameter Description MessagePolicy_BodyListType, MessagePolicy_BodyList; [/MessagePolicy] Classification Table Web: Classification Table This parameter table configures the Classification table. This table EMS: SBC Classification classifies the incoming SIP INVITE to a Source IP Group. The CLI: configure voip > sbc...
Page 759 SIP User's Manual 32. Viewing Syslog Messages Parameter Description Notes:  This table can include up to 120 indices (where 0 is the first index).  For a specific routing rule to be effective, the matching characteristics must match. If no matching rule is located, the call is rejected.
Page 760 Mediant 3000 Parameter Description IPInboundManipulation_LeaveFromRight, IPInboundManipulation_Prefix2Add, IPInboundManipulation_Suffix2Add; [\IPInboundManipulation] For example: IPInboundManipulation 1 = 0, 0, 0, -1, *, abc, *, *, 0, 0, 0, 255, , ; Notes:  This table can include up to 100 indices.  For SIP URI host name (source and destination) manipulations, you can also use the IP Group table.
Page 761: Standalone Survivability Parameters
SIP User's Manual 32. Viewing Syslog Messages Parameter Description Destination IP Groups respectively.  For a detailed description of the table's individual parameters and for configuring the table using the Web interface, see 'Configuring IP-to-IP Outbound Manipulations' on page 414. ...
Page 762 Mediant 3000 Parameter Description Web: SAS Local SIP TLS Port Defines the local TLS port used to send/receive SIP messages EMS: Local SIP TLS Port for the SAS application. The SIP entities in the local network [SASLocalSIPTLSPort] need to send the registration requests to this port. When forwarding the requests to the proxy ('Normal Mode'), this port serves as the source port.
Page 763 SIP User's Manual 32. Viewing Syslog Messages Parameter Description the top-most Via header and retains the original Contact header. Thus, the top-most Via header and the Contact header point to different hosts.  [1] = Enable - the device changes the Contact header so that it points to the SAS host and therefore, the top-most Via header and the Contact header point to the same host.
Page 764 Mediant 3000 Parameter Description Up to four emergency numbers can be defined, where each number can be up to four digits. [SASEmergencyPrefix] Defines a prefix that is added to the Request-URI user part of the INVITE message that is sent by the device's SAS agent when in Emergency mode to the default gateway or to any other destination (using the IP2IP Routing table).
Page 765 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  You can only configure one index entry.  For a detailed description of the individual parameters in this table and for configuring this table using the Web interface, see 'Manipulating Destination Number of Incoming INVITE' on page 459.
Page 766: Ip Media Parameters
Mediant 3000 A.15 IP Media Parameters The IP media parameters are described in the table below. Table A-69: IP Media Parameters Parameter Description Web: Number of Media Channels Defines the number of DSP channels that are allocated for EMS: Media Channels various functionality (, IP-to-IP sessions).
Page 767 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [19] 19 = 10.00 dB/sec  [20] 20 = 11.00 dB/sec  [21] 21 = 12.00 dB/sec  [22] 22 = 13.00 dB/sec  [23] 23 = 14.00 dB/sec  [24] 24 = 15.00 dB/sec ...
Page 768 Mediant 3000 Parameter Description AMDSensitivityLevel parameter.  This parameter can also be configured per IP Profile, using the IPProfile parameter (see Configuring IP Profiles on page 236). Web: Answer Machine Detector Defines the AMD detection sensitivity level of the selected Sensitivity Level AMD Parameter Suite.
Page 769 SIP User's Manual 32. Viewing Syslog Messages Parameter Description using the IPProfile parameter (see Configuring IP Profiles on page 236). [AMDMaxPostGreetingSilenceTime] Defines the maximum duration of silence from after the greeting time is over (defined by AMDMaxGreetingTime) until the AMD decision. Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see Configuring IP Profiles on page 236).
Page 770 Mediant 3000 Parameter Description Pattern Detection Parameters Note: For an overview on the pattern detector feature for TDM tunneling, see DSP Pattern Detector on page 260. Web: Enable Pattern Detector Enables the Pattern Detector (PD) feature. [EnablePatternDetector]  [0] Disable (default) ...
Page 771: Auxiliary And Configuration Files Parameters
SIP User's Manual 32. Viewing Syslog Messages A.16 Auxiliary and Configuration Files Parameters This subsection describes the device's auxiliary and configuration files parameters. A.16.1 Auxiliary/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 772: Automatic Update Parameters
Mediant 3000 Parameter Description Web: Dial Plan File Defines the name (and path) of the Dial Plan file (defining dial EMS: Dial Plan File Name plans). This file should be constructed using the DConvert utility [DialPlanFileName] (refer to the Product Reference Manual).
Page 773 SIP User's Manual 32. Viewing Syslog Messages Parameter Description  [2] = Check CRC for individual lines. Use this option when the HTTP server scrambles the order of lines in the provided ini file. [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.
Page 774 Mediant 3000 Parameter Description [CasFileURL] Defines the name of the CAS file and the path to the server (IP address or FQDN) on which it is located. For example: http://server_name/file, https://server_name/file. Note: The maximum length of the URL address is 99 characters.
Page 775: B Dialing Plan Notation For Routing And Manipulation
SIP User's Manual 32. Viewing Syslog Messages 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 776 Mediant 3000 Notation Description must have the same number of digits. For example, (23-34) is correct, 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: ...
Page 777: Csip Message Manipulation Syntax
SIP User's Manual 32. Viewing Syslog Messages 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 'Configuring Message Manipulations' on page 408. Actions The actions that can be done on SIP message manipulation in the Message Manipulations table are listed in the table below.
Page 778: Accept-Language
Mediant 3000 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 779: Call-Id
SIP User's Manual 32. Viewing Syslog Messages 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 780: Cseq
Mediant 3000 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 781: Event
SIP User's Manual 32. Viewing Syslog Messages 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)' on page...
Page 782: From
Mediant 3000 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: Header Level Action Delete Modify List Entries Operations Supported Keyword...
Page 783: Min-Se And Min-Expires
SIP User's Manual 32. Viewing Syslog Messages 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 Result:...
Page 784: P-Asserted-Identity
Mediant 3000 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- Expires.time, 2, "'700'", 0;...
Page 785: P-Called-Party-Id
SIP User's Manual 32. Viewing Syslog Messages Keyword Sub Types Attributes Param Param Read/Write URL Structure (see 'URL' Read/Write on page 802) 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 786: P-Charging-Vector
Mediant 3000 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; orig-ioi=home1.net The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword...
Page 787: Privacy
SIP User's Manual 32. Viewing Syslog Messages P-Preferred-Identity: "Cullen Jennings" Result: <sip:fluffy@abc.com> Example 2 Rule: Modify the display name in the P-Preferred-Identity header: MessageManipulations 2 = 1, any, , header.P-Preferred- Identity.name, 2, "'Alice Biloxi'", 0; P-Preferred-Identity: "Alice Biloxi" Result: <sip:fluffy@abc.com> C.2.17 Privacy An example of the header is shown below: Privacy: none...
Page 788: Reason
Mediant 3000 Below are header manipulation examples: Example 1 Rule: Add a Proxy-Require header to the message: 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: MessageManipulations 2 = 1, any, , header.Proxy-...
Page 789: Referred-By
SIP User's Manual 32. Viewing Syslog Messages 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. If the cause is 0, then the text string (see Example 3) is generated from the reason number.
Page 790: Remote-Party-Id
Mediant 3000 Keyword Sub Types Attributes Below are header manipulation examples: Add a basic header: Example 1 Rule: 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,...
Page 791: Request-Uri
SIP User's Manual 32. Viewing Syslog Messages 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; Remote-Party-ID: Result: <sip:999@10.132.10.108>;party=calling;npi=0;ton=0 Example 2 Rule: Create a Remote-Party-Id header using the url in the From header using the + operator to concatenate strings: MessageManipulations 0 = 1, Invite, ,header.REMOTE- PARTY-ID, 0, "'<'+header.from.url +'>' +...
Page 792: Require
Mediant 3000 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'", header.REMOTE-PARTY-ID.url, 2, 'sip:3200@110.18.5.41;tusunami=0', 0; Remote-Party-ID: Result: <sip:3200@110.18.5.41;tusunami=0>;party=calling;npi=...
Page 793: Resource-Priority
SIP User's Manual 32. Viewing Syslog Messages MessageManipulations 0 = 0, invite, , header.require.earlymedia, 0, “1” , 0; Require: em,replaces,early-session, early-media Result: Example 4 Rule: Set the privacy options tag in the Require header: MessageManipulations 0 = 0, invite, , header.require.privacy, 0, “1”...
Page 794: Server Or User-Agent
Mediant 3000 C.2.27 Server or User-Agent An example of the header is shown below: User-Agent: Sip Message Generator V1.0.0.5 The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types...
Page 795: Session-Expires
SIP User's Manual 32. Viewing Syslog Messages Service-Route:sip:itsp.com;lr Result: Service-Route: <sip:HSP.HOME.EXAMPLE.COM;lr> Example 3 Rule: Modify the Service-Route header in list entry 0: MessageManipulations 4 = 1, Invite, ,header.service- route.0.serviceroute, 2, "'<sip:home.itsp.com;lr>'", 0; Service-Route:sip:home.itsp.com;lr Result: Service-Route: <sip:itsp.com;lr> C.2.29 Session-Expires An example of the header is shown below: Session-Expires: 480 The header properties are shown in the table below: Header Level Action...
Page 796: Subject
Mediant 3000 C.2.30 Subject An example of the header is shown below: Subject: A tornado is heading our way! The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types...
Page 797: C.2.32 To
SIP User's Manual 32. Viewing Syslog Messages C.2.32 To An example of the header is shown below: To: <sip:101@10.132.10.128;user=phone> The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Name String...
Page 798: Unsupported
Mediant 3000 C.2.33 Unsupported An example of the header is shown below: Unsupported: 100rel The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Capabilities SIPCapabilities Struct Read/Write...
Page 799: Warning
SIP User's Manual 32. Viewing Syslog Messages Keyword Sub Types Attributes Port Integer Read Only TransportType Enum TransportType (see Read Only 'TransportType' on page 811) Below is a header manipulation example: Rule: Check the transport type in the first Via header and if it's set to UDP, then modify the From header's URL: MessageManipulations 0 = 1, Invite.request, "header.VIA.0.transporttype == '0'", header.from.url, 2,...
Page 800: Structure Definitions
Mediant 3000 Keyword Sub Types Attributes Below are header manipulation examples: Example 1 Rule: Add a custom header to all messages: MessageManipulations 0 = 1, , , header.myExp, 0, "'scooby, doo, goo, foo'", 0; MYEXP: scooby, doo, goo, foo Result:...
Page 801: Host
SIP User's Manual 32. Viewing Syslog Messages C.3.2 Host The host structure is applicable to the URL structure (see 'URL' on page 802) and the Via header (see 'Via' on page 798). Table C-3: Host Structure Keyword Sub Types Port Short Name String...
Page 802: Sipcapabilities
Mediant 3000 C.3.6 SIPCapabilities This structure is applicable to the following headers:  Supported (see 'Supported' on page 796)  Require (see 'Require' on page 792)  Proxy-Require (see 'Proxy-Require' on page 787)  Unsupported (see 'Unsupported' on page 798)
Page 803: Random Type
SIP User's Manual 32. Viewing Syslog Messages Keyword Sub Types Host Host Structure (see 'Host' on page 801) MHost Structure UserPhone Boolean LooseRoute Boolean User String TransportType Enum Transport (see 'TransportType' on page 811) Param Param 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;...
Page 804: Wildcarding For Header Removal
Mediant 3000 Wildcarding for Header Removal The device supports the use of the "*" wildcard character to remove headers. The "*" character may only appear at the end of a string. For example, "X-*" is a valid wildcard request, but "X-*ID" is not.
Page 805: Enum Definitions
SIP User's Manual 32. Viewing Syslog Messages  Example 2: • Store a value in a global variable: Stores the Priority header of the INVITE with ‘company’ in the host part of the From header: MessageManipulations 0 = 0, Invite.Request, header.from.url.host == ‘company’, var.global.1, 2, header.priority, 0;...
Page 806: Mlpp Reason Type
Mediant 3000 Package Value START_CWT STOP_CWT UA_PROFILE LINE_SEIZE C.7.3 MLPP Reason Type These ENUMs are applicable to the MLPP Structure (see 'MLPP' on page 801). 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 790).
Page 807: Privacy
SIP User's Manual 32. Viewing Syslog Messages C.7.6 Privacy These ENUMs are applicable to the Remote-Party-Id (see 'Remote-Party-Id' on page 790) and Diversion (see 'Diversion' on page 780) 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 780).
Page 808 Mediant 3000 Reason Value REFER 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 UNSUPPORTED_MEDIA_415 UNSUPPORTED_URI_SCHEME_416...
Page 809 SIP User's Manual 32. Viewing Syslog Messages Reason Value 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 Version 6.4 November 2011...
Page 810: Reason (Remote-Party-Id)
Mediant 3000 C.7.9 Reason (Remote-Party-Id) These ENUMs are applicable to the Remote-Party-Id header (see 'Remote-Party-Id' on page 790). 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 795).
Page 811: Transporttype
SIP User's Manual 32. Viewing Syslog Messages C.7.13 TransportType These ENUMs are applicable to the URL Structure (see 'URL' on page 802) and the Via header (see 'Via' on page 798). Table C-21: Enum TransportType TransportType Value SCTP C.7.14 Type These ENUMs are applicable to the URL Structure (see 'URL' on page 802).
Page 812 Mediant 3000 Element Command Command Value Type Remarks Type Type equals to the value. "!=" String Returns true if the body's content not equals to the value. "contains" String Returns true if the string given is found in the body's content.
Page 813 SIP User's Manual 32. Viewing Syslog Messages Element Command Command Value Type Remarks Type Type "Remove" Removes the header from the message, if the header is part of a list only that header is removed. "Add" String Adds a new header to the end of the list.
Page 814 Mediant 3000 Element Command Command Value Type Remarks Type Type "!=" String Returns true if the header's structure's value not equals to the *Structure value. The string given must be able to be parsed to the structure. Action Modify String Sets the header's structure to the value.
Page 815 SIP User's Manual 32. Viewing Syslog Messages Element Command Command Value Type Remarks Type Type "!=" Boolean Returns true if the Boolean element not equals to the value. Boolean – can be either "0" or "1". Action "Modify" Boolean Sets the Boolean element to the value.
Page 816: Syntax
Mediant 3000 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 "." message-role Examples: •...
Page 817 SIP User's Manual 32. Viewing Syslog Messages match-condition: Description: matching criteria for the rule Syntax: ( message-element / param ) SWS match-type [SWS value] * [ SWS logical- expression SWS match-condition ] Examples: • header.from.user == 100 • header.contact.header-param.expires > 3600 •...
Page 818 Mediant 3000 message-element: Description: element in the message Syntax: ( "header" / "body" ) "." message-element-name [ "." header-index ] * [ "." ( sub-element / sub-element-param ) ] Examples: • Header.from • Header.via.2.host • Header.contact.header-param.expires • Header.to.uri-param.user-param • Body.application/dtmf-relay message-element-name Description: name of the message's element - "/"...
Page 819 SIP User's Manual 32. Viewing Syslog Messages Examples: ♦ param.ipg. src.user ♦ param.ipg.dst.host ♦ param.ipg.src.type ♦ param.call.src.user param-sub-element Description: determines whether the param being accessed is a call or an IP Group Syntax: ( "call" / "IPG" ) Examples: ♦ call –...
Page 820 Mediant 3000 action-type: Description: action to be performed on the element Syntax: ( "modify" / "add-prefix" / "remove-prefix" / "add-suffix" / "remove-suffix" / "add" / "remove" ) Examples: • "modify" – sets the element to the new value (all element types) •...
Page 821: D Dsp Templates
Mediant 3000 full chassis – see 'Mediant 3000 Full Chassis' on page  Mediant 3000 with 16 E1 / 21 T1 – see 'Mediant 3000 16 E1 / 21 T1' on page  Mediant 3000 with single T3 – see 'Mediant 3000 with Single T3' on page ...
Page 822: Mediant 3000 Full Chassis
Mediant 3000 Mediant 3000 Full Chassis The DSP templates for a full chassis configuration are shown in the table below. Table D-1: DSP Firmware Templates for Mediant 3000 Full DSP Template Supplementary Capabilities Number of Channels 2016 2016 1764 1260...
Page 823: Mediant 3000 16 E1 / 21 T1
SIP User's Manual 32. Viewing Syslog Messages Mediant 3000 16 E1 / 21 T1 The DSP templates for Mediant 3000 16 E1 / 21 T1 are shown in the table below. Notes: • For each IP-to-IP transcoding call, two DSP channels are required.
Page 824 Mediant 3000 DSP Template Supplementary Capabilities Number of Channels iLBC MS GSM MS-RTA (NB) MS-RTA (WB) T.38 Version 3 SIP User's Manual Document #: LTRT-89712...
Page 825: Mediant 3000 With Single T3
SIP User's Manual 32. Viewing Syslog Messages Mediant 3000 with Single T3 The DSP templates for Mediant 3000 with a single T3 interface are shown in the table below. Table D-3: DSP Firmware Templates for Mediant 3000 with Single T3...
Page 826: Dsp Template Mix Feature For Mediant 3000
Mediant 3000 DSP Template Mix Feature for Mediant 3000 The device can operate (and be loaded) with up to two DSP templates. The channel capacity per DSP template is approximately 50%, with alignment to the number of DSP's present in the device.
Page 827: E Selected Technical Specifications
SIP User's Manual 32. Viewing Syslog Messages Selected Technical Specifications The section lists the technical specifications of the Mediant 3000. Note: All specifications in this document are subject to change without prior notice. Table E-1: Technical Specifications Function Specification Trunk & Channel Capacity Note: Channel capacity depends on configuration settings.
Page 828 T3/DS3 PSTN Up to three Mini-SMB T3/DS3 (44.736 Mbps, 75 Ohm coax) interfaces. Supports depopulated configuration providing a single DS3. Note: Applicable only to Mediant 3000 with TP-6310 blade. RS-232 One RS-232 serial interface for serial configuration Voice and Tone ...
Page 829 FORWARD (RFC 4292), IP-MIB (RFC 4293), NOTIFICATION-LOG-MIB (RFC 3014), RTCPXR-MIB, RTP-MIB (RFC 2959), SNMP-FRAMEWORK- MIB (RFC 3411), SNMPv2-TC, SONET-MIB (RFC 3592), TCP-MIB (RFC 4022), UDP-MIB (RFC 4113), and many other AudioCodes' proprietary MIBs Configuration and run-time monitoring using a Web browser...
Page 830 E1/T1 Interface Two 100-pin SCSI connectors for Trunks 1-25 and 43-67. Two 68-pin SCSI connectors for Trunks 26-42 and 68-84. Note: Applicable only to Mediant 3000 with TP-8410. STM-1/OC-3 Interface Two fiber optical 155.54-Mbps SFP modules (1+1 redundancy). The SFP modules accept twin single-mode fiber optic cables terminated with LC- type connectors (not supplied).
Page 831 SIP User's Manual 32. Viewing Syslog Messages Function Specification Power AC Power Input Power input range 100-240 VAC at a nominal 50/60 Hz line frequency. AC Power Supply Depends on installed blades and configuration: Voltages and Power  TP-6310 OC-3/STM-1 Simplex: Consumption (Typical) ...
Page 832 19-inch rack mounting, shelf mounting, or desktop mounting Mounting Full blade hot-swap supported for media processing blades according to Hot Swap PICMG 2.1. Host Interface Via Packet interface using AudioCodes’ proprietary TPNCP or standard control protocols Regulatory Compliance Telecommunication FCC part 68, TBR4 and TBR13 Standards ...
Page 833 SIP User's Manual 32. Viewing Syslog Messages Reader's Notes Version 6.4 November 2011...
Page 834 User's Manual Ver. 6.4 www.audiocodes.com...

AudioCodes Mediant 3000 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 Configuration

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 Session Border Controller

Quick Links

Related Manuals for AudioCodes Mediant 3000

Summary of Contents for AudioCodes Mediant 3000