Page 1 Mediant™ 600 & Mediant™ 1000 VoIP Media Gateways SIP Protocol User’s Manual Version 6.4 March 2012 Document # LTRT-83310...
Page 3: Table Of Contents
SIP User's Manual Contents Table of Contents Overview ......................17 Mediant 600 ......................17 Mediant 1000 ......................18 SIP Overview ......................20 Getting Started ......................21 Assigning the VoIP LAN IP Address ..............23 Using CLI ....................... 23 Using the Web Interface ..................25 Using BootP/TFTP Server ..................
Page 4 Mediant 600 & Mediant 1000 Web Login Authentication using Smart Cards ............70 Configuring Web and Telnet Access List ............... 70 Configuring RADIUS Settings ................72 CLI-Based Management ..................73 Configuring Telnet and SSH Settings ..............74 SNMP-Based Management ................75 Configuring SNMP Community Strings ..............
Page 5 SIP User's Manual Contents 10.4.1.6 State Column ..................121 10.4.2 Routing Table Configuration Summary and Guidelines ........121 10.4.3 Troubleshooting the Routing Table ...............122 10.5 Configuring QoS Settings ..................122 10.6 DNS ........................123 10.6.1 Configuring the Internal DNS Table ...............123 10.6.2 Configuring the Internal SRV Table ...............124 10.7 NAT (Network Address Translation) Support ............
Page 6 Mediant 600 & Mediant 1000 12.4.1 Answer Machine Detector (AMD) ................160 12.4.2 Configuring Automatic Gain Control (AGC) ............164 12.5 Configuring General Media Settings ..............165 12.6 Configuring Analog Settings ................. 166 12.7 Configuring DSP Templates ................. 166 12.8 DSP Channel Resources for Transcoding ............167 12.9 Configuring Media Realms ...................
Page 7 SIP User's Manual Contents 18.1.4 Configuring Digital Gateway Parameters ..............237 18.1.5 Tunneling Applications ...................238 18.1.5.1 TDM Tunneling ..................238 18.1.5.2 QSIG Tunneling ..................241 18.1.6 Advanced PSTN Configuration ................242 18.1.6.1 Release Reason Mapping ..............242 18.1.6.2 ISDN Overlap Dialing ................246 18.1.6.3 ISDN Non-Facility Associated Signaling (NFAS) ........
Page 8 Mediant 600 & Mediant 1000 18.6.2 Configuring Metering Tones ..................321 18.6.3 Configuring Charge Codes ..................323 18.6.4 Configuring FXO Settings ..................324 18.6.5 Configuring Authentication ..................325 18.6.6 Configuring Automatic Dialing ................326 18.6.7 Configuring Caller Display Information ..............327 18.6.8 Configuring Call Forward ..................328 18.6.9 Configuring Caller ID Permissions .................329...
Page 9 SIP User's Manual Contents 19.1.2.1 SAS Routing in Normal State ..............385 19.1.2.2 SAS Routing in Emergency State ............387 19.2 SAS Configuration ....................388 19.2.1 General SAS Configuration ...................388 19.2.1.1 Enabling the SAS Application ............... 388 19.2.1.2 Configuring Common SAS Parameters ..........388 19.2.2 Configuring SAS Outbound Mode .................391 19.2.3 Configuring SAS Redundant Mode ...............392 19.2.4 Configuring Gateway Application with SAS ............392...
Page 10 Mediant 600 & Mediant 1000 23 Software Upgrade .................... 483 23.1 Loading Auxiliary Files ..................483 23.1.1 Call Progress Tones File ..................486 23.1.1.1 Distinctive Ringing ................. 489 23.1.2 Prerecorded Tones File ..................491 23.1.3 Voice Prompts File ....................491 23.1.4 CAS Files .......................492 23.1.5 Dial Plan File ......................492...
Page 11 SIP User's Manual Contents Diagnostics ......................535 30 Configuring Syslog Settings ................537 31 Viewing Syslog Messages ................539 Appendices ......................541 A Configuration Parameters Reference ............543 Networking Parameters ..................543 A.1.1 Ethernet Parameters ....................543 A.1.2 Multiple Network Interfaces and VLAN Parameters ..........544 A.1.3 Static Routing Parameters ..................546 A.1.4...
Page 12 Mediant 600 & Mediant 1000 A.12 Gateway and IP-to-IP Parameters ............... 639 A.12.1 Fax and Modem Parameters .................639 A.12.2 DTMF and Hook-Flash Parameters ...............644 A.12.3 Digit Collection and Dial Plan Parameters .............649 A.12.4 Voice Mail Parameters ...................651 A.12.5 Supplementary Services Parameters ..............656 A.12.5.1 Caller ID Parameters ................
Page 13 SIP User's Manual Contents C.2.7 Diversion ........................792 C.2.8 Event ........................793 C.2.9 From ........................793 C.2.10 History-Info ......................794 C.2.11 Min-Se and Min-Expires ..................795 C.2.12 P-Asserted-Identity ....................796 C.2.13 P-Associated-Uri ....................796 C.2.14 P-Called-Party-Id ....................797 C.2.15 P-Charging-Vector ....................798 C.2.16 P-Preferred-Identity ....................798 C.2.17 Privacy ........................799 C.2.18 Proxy-Require ......................799 C.2.19 Reason........................800 C.2.20 Referred-By ......................801 C.2.21 Refer-To .........................801...
Page 14 Mediant 600 & Mediant 1000 C.7.14 Type ........................824 C.8 Actions and Types ....................824 C.9 Syntax ........................830 D DSP Templates ....................835 D.1 Analog Interfaces ....................835 D.2 Digital Interfaces ....................836 D.3 Media Processing Interfaces ................837 Selected Technical Specifications ..............839 Mediant 600 ......................
Page 15: Weee Eu Directive
SIP User's Manual Notices Notice This document describes the AudioCodes Mediant 600 and Mediant 1000 Voice-over-IP (VoIP) SIP media gateways. Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions.
Page 16: Related Documentation
The scope of this document does not fully cover security aspects for deploying the device in your environment. Security measures should be done in accordance with your organization’s security policies. For basic security guidelines, you can refer to AudioCodes Recommended Security Guidelines document. Note: Throughout this manual, unless otherwise specified, the term device refers to the Mediant 600 and Mediant 1000.
Page 17: Overview
SIP User's Manual 1. Overview Overview This section provides an overview of the Mediant 1000 and Mediant 600 media gateways. Mediant 600 The Mediant 600 (hereafter referred to as device) is a cost-effective, wireline Voice-over-IP (VoIP) Session Initiation Protocol (SIP)-based media gateway. It is designed to interface between Time-Division Multiplexing (TDM) and IP networks in enterprises, small and medium businesses (SMB), and CPE application service providers.
Page 18: Mediant 1000
Web browser (such as Microsoft™ Internet Explorer™). Mediant 1000 The Mediant 1000 (hereafter referred to as device) is a best-of-breed Voice-over-IP (VoIP) Session Initiation Protocol (SIP) Media Gateway, using field-proven, market-leading technology, implementing analog and digital cutting-edge technology. The device is...
Page 19 SIP User's Manual 1. Overview  Analog: The device's analog interface supports up to 24 analog ports (four ports per analog module) in various Foreign Exchange Office (FXO) or Foreign Exchange Station (FXS) configurations, supporting up to 24 simultaneous VoIP calls. The device supports up to six analog modules, each module providing four analog RJ-11 ports.
Page 20: Sip Overview
Mediant 600 & Mediant 1000 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 21: Getting Started
Part I Getting Started Before you can begin configuring your device, you need to access it with the default LAN IP address and change this IP address to suit your networking scheme. Once modified, you can then access the device using the new LAN IP address. This section describes how to perform this initialization process.
Page 22 Reader’s Notes...
Page 23: Assigning The Voip Lan Ip Address
Connect the RS-232 port of the device to the serial communication port on your computer. For more information, refer to the Hardware Installation Manual. Figure 2-1: Connecting to Serial Port for Initial Connectivity – Mediant 1000 Version 6.4 March 2012...
Page 24 Mediant 600 & Mediant 1000 Figure 2-2: Connecting to Serial Port for Initial Connectivity – Mediant 600 Establish a serial communication link with the device using a terminal emulator program (such as HyperTerminal) with the following communication port settings: •...
Page 25: Using The Web Interface
Connect one of the LAN ports of the device directly to the network interface of your computer, using a straight-through Ethernet cable. Figure 2-3: Connecting to LAN for Initial Connectivity – Mediant 1000 Figure 2-4: Connecting to LAN for Initial Connectivity – Mediant 600 Change the IP address and subnet mask of your computer to correspond with the default IP address and subnet mask of the device.
Page 26: 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 27 SIP User's Manual 2. Assigning the VoIP LAN IP Address Click the Add New Client icon. Figure 2-6: 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 28: Using The Fxs Voice Menu Guidance
Mediant 600 & Mediant 1000 Using the FXS Voice Menu Guidance You can assign an IP address that suits your networking scheme using a standard touch- tone telephone connected to one of the FXS ports. The voice menu can also be used to query and modify basic configuration parameters.
Page 29 Configuration File Name Pattern Description http://aa.bb.cc.dd/config.ini Standard config.ini. https://aa.bb.cc.dd/config.ini Secure HTTP. The device's MAC address is appended to the file http://aa.bb.cc.dd/audiocodes/<MAC>.ini name (e.g., http://36.44.0.6/audiocodes/00908f012300.ini). http://aa.bb.cc.dd:8080/config.ini HTTP on port 8080. http://aa.bb.cc.dd:1400/config.ini HTTP on port 1400. Generating configuration per IP/MAC address http://aa.bb.cc.dd/cgi-...
Page 30 Mediant 600 & Mediant 1000 The following is an example perl CGI script, suitable for most Apache-based HTTP servers for generating configuration dynamically per pattern #6 above. Copy this script to /var/www/cgi-bin/acconfig.cgi on your Apache server and edit it as required: #!/usr/bin/perl use CGI;...
Page 31: Management Tools
The CLI is used only for debugging. • If you use AudioCodes BootP/TFTP utility to assign an IP address to the device (see Using BootP/TFTP Server on page 26), you can also in the same process load a firmware file (.cmp) and a configuration ini file (.ini file).
Page 32 Reader’s Notes...
Page 33: 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 34: Accessing The Web Interface
Mediant 600 & Mediant 1000 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.
Page 35: 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 36: Toolbar Description
Mediant 600 & Mediant 1000 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 482).
Page 37: 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 38: Displaying Navigation Tree In Basic And Full View
Mediant 600 & Mediant 1000 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 &...
Page 39: 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 40: Working With Configuration Pages
Mediant 600 & Mediant 1000 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 41 SIP User's Manual 3. Web-Based Management 3.1.6.2.1 Displaying Basic and Advanced Parameters Some pages provide you with an Advanced Parameter List / Basic Parameter List toggle button that allows you to show or hide advanced parameters (in addition to displaying the basic parameters). This button is located on the top-right corner of the page and has two states: ...
Page 42: Modifying And Saving Parameters
Mediant 600 & Mediant 1000 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.
Page 43: Entering Phone Numbers
SIP User's Manual 3. Web-Based Management  To save configuration changes on a page to the device's volatile memory (RAM), do one of the following:  On the toolbar, click the Submit button.  At the bottom of the page, click the Submit button.
Page 44: Working With Tables
Mediant 600 & Mediant 1000 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 45 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 46 Mediant 600 & Mediant 1000 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 ...
Page 47 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 48: Searching For Configuration Parameters
Mediant 600 & Mediant 1000 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 49: 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 50 Mediant 600 & Mediant 1000 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 51: 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 52: Editing A Scenario
Mediant 600 & Mediant 1000 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 53: 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 54: Loading A Scenario To The Device
Mediant 600 & Mediant 1000 3.1.8.5 Loading a Scenario to the Device Instead of creating a Scenario, you can load a Scenario file (data file) from your PC to the device.  To load a Scenario to the device: On the Navigation bar, click the Scenarios tab; the Scenario appears in the Navigation tree.
Page 55: 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 56: Creating A Login Welcome Message
Mediant 600 & Mediant 1000 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 57: 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 58: Logging Off The Web Interface
Mediant 600 & Mediant 1000 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 66.
Page 59: Using The Home Page
 On the toolbar, click the Home icon. Figure 3-31: Home Page of Mediant 600 Figure 3-32: Home Page of Mediant 1000 Note: The displayed number and type of telephony interfaces depends on the device's hardware configuration. In addition to the color-coded status information depicted on the graphical display of the...
Page 60 Mediant 600 & Mediant 1000 device houses any of these analog modules)  Firmware Version: software version currently running on the device  Protocol Type: signaling protocol currently used by the device (i.e. SIP)  Gateway Operational State: operational state of the device: •...
Page 61 Power Supply Unit 1 status icon (applicable only to Mediant 1000):  (green): Power supply is operating  (red): Power supply failure or no power supply unit installed Power Supply Unit 2 status indicator (applicable only to Mediant 1000). See Item #11 for a description. Version 6.4 March 2012...
Page 62: Assigning A Port Name
Mediant 600 & Mediant 1000 3.2.1 Assigning a Port Name The Home page allows you to assign an arbitrary name or a brief description to each port. This description appears as a tooltip when you move your mouse over the port.
Page 63: Viewing Analog Port Information
SIP User's Manual 3. Web-Based Management 3.2.3 Viewing Analog Port Information The Home page allows you to view detailed information on a specific FXS or FXO analog port such as RTP/RTCP and voice settings.  To view detailed port information: Click the port for which you want to view port settings;...
Page 64: Replacing Modules
When only one module is available, removal of the module causes the device to reset. • Before inserting a module into a previously empty slot, you must power down the device. Note: This section is applicable only to Mediant 1000. SIP User's Manual Document #: LTRT-83310...
Page 65 SIP User's Manual 3. Web-Based Management  To replace a module: Remove the module by performing the following: In the Home page, click the title of the module that you want to replace; the Remove Module button appears: Figure 3-35: Remove Module Button Click the Remove Module button;...
Page 66: Configuring Web User Accounts
Mediant 600 & Mediant 1000 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 67 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 3-38: 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 68 Mediant 600 & Mediant 1000 To change the password of an account, perform the following: In the field 'Current Password', enter the current password. In the fields 'New Password' and 'Confirm New Password', enter the new password (maximum of 19 case-sensitive characters).
Page 69: Configuring Web Security Settings
SIP User's Manual 3. Web-Based Management Notes: • For security, it's recommended that you change the default user name and password. • A Web user with access level 'Security Administrator' can change all attributes of all the Web user accounts. Web users with an access level other than 'Security Administrator' can only change their own password and user name.
Page 70: Web Login Authentication Using Smart Cards
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 71 SIP User's Manual 3. Web-Based Management To add an authorized IP address, in the 'Add an authorized IP address' field, enter the required IP address, and then click Add New Entry; the IP address you entered is added as a new entry to the Web & Telnet Access List table. Figure 3-41: Web &...
Page 72: Configuring Radius Settings
Mediant 600 & Mediant 1000 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 541. ...
Page 73: 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 74: Configuring Telnet And Ssh Settings
Mediant 600 & Mediant 1000 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 554.  To define Telnet and SSH: Open the Telnet/SSH Settings page (Configuration tab >...
Page 75: 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 76: Configuring Snmp Trap Destinations
Mediant 600 & Mediant 1000 Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 482. 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 77: 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 78: Configuring Snmp V3 Users
Mediant 600 & Mediant 1000 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 79 SIP User's Manual 5. SNMP-Based Management Parameter Description Privacy Key Privacy key. Keys can be entered in the form of a text password or [SNMPUsers_PrivKey] long hex string. Keys are always persisted as long hex strings and keys are localized. Group The group with which the SNMP v3 user is associated.
Page 80 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 81: 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 82 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 83: Ini File-Based Management
 Web interface (see 'Backing Up and Loading Configuration File' on page 503)  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 84: Configuring Ini File Table Parameters
Mediant 600 & Mediant 1000 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 85: 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 86: 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 87: General System Settings
Part III General System Settings This part provides general system configurations.
Page 88 Reader’s Notes...
Page 89: 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 90 Mediant 600 & Mediant 1000 Open the Certificates page (Configuration tab > System menu > Certificates). Figure 8-1: Certificates Page SIP User's Manual Document #: LTRT-83310...
Page 91 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 92: Loading A Private Key
Mediant 600 & Mediant 1000 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 93: 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 94: Self-Signed Certificates
Mediant 600 & Mediant 1000 Self-Signed Certificates The device is shipped with an operational, self-signed server certificate. The subject name for this default certificate is 'ACL_nnnnnnn', where nnnnnnn denotes the serial number of the device. However, this subject name may not be appropriate for production and can be changed while still using self-signed certificates.
Page 95: Date And Time
SIP User's Manual 9. Date and Time Date and Time The date and time of the device can be configured manually or it can be obtained automatically from a Simple Network Time Protocol (SNTP) server. Configuring Manual Date and Time The date and time of the device can be configured manually.
Page 96 Mediant 600 & Mediant 1000 of reference called the Universal Time Coordinate (UTC). The time offset that the NTP client uses is configurable using the ini file parameter NTPServerUTCOffset, or via an SNMP MIB object (refer to the Product Reference Manual).
Page 97 SIP User's Manual 9. Date and Time 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. •...
Page 98 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 99: Voip Configuration
Part IV VoIP Configuration This part describes the VoIP configurations.
Page 100 Reader’s Notes...
Page 101: Network
SIP User's Manual 10. Network Network This section describes the network-related configuration. 10.1 Ethernet Interface Configuration The device's Ethernet connection can be configured (using the ini file parameter EthernetPhyConfiguration) for one of the following modes:  Manual mode: • 10Base-T Half-Duplex or 10Base-T Full-Duplex •...
Page 102: Ethernet Interface Redundancy
Mediant 600 & Mediant 1000 10.2 Ethernet Interface Redundancy The device supports Ethernet redundancy by providing two Ethernet ports, located on the CPU module. The Ethernet port redundancy feature is enabled using the ini file parameter MIIRedundancyEnable. By default, this feature is disabled.
Page 103 SIP User's Manual 10. Network Notes: • For more information and examples of network interfaces configuration, see 'Network Configuration' on page 106. • When adding more than one interface, ensure that you enable VLANs using the 'VLAN Mode' (VlANMode) parameter. •...
Page 104 Mediant 600 & Mediant 1000 Under the 'Multiple Interface Settings' group, click the Multiple Interface Table button; a confirmation message box appears: Figure 10-2: Confirmation Message for Accessing the Multiple Interface Table Click OK to confirm; the 'Multiple Interface Table page appears: In the 'Add Index' field, enter the desired index number for the new interface, and then click Add Index;...
Page 105 SIP User's Manual 10. Network Parameter Description  [5] Media + Control = Media and Call Control applications.  [6] OAMP + Media + Control = All application types are allowed on the interface. Note: For valid configuration guidelines, see 'Multiple Interface Table Configuration Summary and Guidelines' on page 112.
Page 106: Network Configuration
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: VLAN ID Defines the VLAN ID for each interface. Incoming traffic with this [InterfaceTable_VlanID] VLAN ID is routed to the corresponding interface and outgoing traffic from that interface is tagged with this VLAN ID.
Page 107: Multiple Network Interfaces And Vlans
SIP User's Manual 10. Network 10.3.1.1 Multiple Network Interfaces and VLANs A need often arises to have logically separated network segments for various applications (for administrative and security reasons). This can be achieved by employing Layer-2 VLANs and Layer-3 subnets. Figure 10-3: Multiple Network Interfaces The figure depicts a typical configuration featuring in which the device is configured with three network interfaces for:...
Page 108 Mediant 600 & Mediant 1000 Index Prefix Default VLAN Application Interface IP Address Interface Name Mode Length Gateway Control IPv4 10.32.174.50 0.0.0.0 ControlIF Media IPv4 10.33.174.50 10.33.0.1 Media1IF Media IPv4 10.34.174.50 0.0.0.0 Media2IF Media IPv4 10.35.174.50 10.35.0.1 Media3IF Media IPv4 10.36.174.50...
Page 109 SIP User's Manual 10. Network Each interface must have its own address space. Two interfaces may not share the same address space, or even part of it. The IP address should be configured as a dotted-decimal notation. For IPv4 interfaces, the prefix length values range from 0 to 30. OAMP Interface Address when Booting using BootP/DHCP: When booting using BootP/DHCP protocols, an IP address is obtained from the server.
Page 110 Mediant 600 & Mediant 1000 10.3.1.1.2.4 Interface Name Column This column allows the configuration of a short string (up to 16 characters) to name this interface. This name is displayed in management interfaces (Web, CLI, and SNMP) and is used in the Media Realm table. This column must have a unique value for each interface (no two interfaces can have the same name) and must not be left blank.
Page 111 SIP User's Manual 10. Network 10.3.1.1.3.4 Quality of Service Parameters The device allows you to specify values for Layer-2 and Layer-3 priorities, by assigning values to the following service classes:  Network Service class – network control traffic (ICMP, ARP) ...
Page 112 Mediant 600 & Mediant 1000 Application Traffic / Network Types Class-of-Service (Priority)  (EnableNTPasOAM): Control: Premium control   Management: Bronze OAMP  Control NFSServers_VlanType in the Gold NFSServers table 10.3.1.1.3.5 Assigning NTP Services to Application Types NTP applications can be associated with different application types (OAMP or Control) in different setups.
Page 113 SIP User's Manual 10. Network rules may be specified in the Routing table ('Configuring the IP Routing Table' on page 118).  The Interface Name column may have up to 16 characters. This column allows the user to name each interface with an easier name to associate the interface with. This column must have a unique value to each interface and must not be left blank.
Page 114: Setting Up Voip Networking
Mediant 600 & Mediant 1000  An IPv4 interface was defined with "Interface Type" different than "IPv4 Manual" (10).  Two interfaces have the exact VLAN ID value while VLANs are enabled.  Two interfaces have the same name. ...
Page 115 SIP User's Manual 10. Network This ini file shows the following:  A Multiple Interface table with a single interface (192.168.85.14/16, OAMP, Media and Control applications are allowed) and a default gateway (192.168.0.1).  A Routing table is configured with two routing rules, directing all traffic for subnet 201.201.0.0/16 to 192.168.0.2, and all traffic for subnet 202.202.0.0/16 to 192.168.0.3.
Page 116 Mediant 600 & Mediant 1000 InterfaceTable 0 = 6, 10, 192.168.85.14, 16, 192.168.0.1, 1, myInterface, , , ; [\InterfaceTable] ; Routing Table Configuration: [ StaticRouteTable ] FORMAT StaticRouteTable_Index = StaticRouteTable_InterfaceName, StaticRouteTable_Destination, StaticRouteTable_PrefixLength, StaticRouteTable_Gateway, StaticRouteTable_Description; StaticRouteTable 0 = 0, 201.201.0.0, 16, 192.168.0.2, ;...
Page 117 MediaCntrl2 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 118: Configuring The Ip Routing Table
Mediant 600 & Mediant 1000 MediaCntrl1,,,; InterfaceTable 2 = 5, 10, 200.200.86.14, 24, 200.200.86.1, 202, MediaCntrl2,,,; [\InterfaceTable] ; VLAN related parameters: VlanMode = 1 VlanNativeVlanId = 1 ; Routing Table Configuration: [ StaticRouteTable ] FORMAT StaticRouteTable_Index = StaticRouteTable_InterfaceName, StaticRouteTable_Destination, StaticRouteTable_PrefixLength, StaticRouteTable_Gateway, StaticRouteTable_Description;...
Page 119 SIP User's Manual 10. Network Click Add New Entry; the new routing rule is added to the IP routing table. To delete a routing rule from the table, select the 'Delete Row' check box corresponding to the required routing rule, and then click Delete Selected Entries. Notes: •...
Page 120: Routing Table Columns
Mediant 600 & Mediant 1000 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.
Page 121: Interface Column
SIP User's Manual 10. Network 10.4.1.4 Interface Column This column defines the interface index (in the Multiple Interface table) from which the gateway address is reached. 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.
Page 122: Troubleshooting The Routing Table
Mediant 600 & Mediant 1000 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 123: 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 482). 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 124: Configuring The Internal Srv Table
Mediant 600 & Mediant 1000  To configure the internal DNS table: Open the Internal DNS Table page (Configuration tab > VoIP menu > Network submenu > DNS submenu > Internal DNS Table). 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 125: 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 126: Stun
Mediant 600 & Mediant 1000 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). Furthermore, the initiating device must notify the receiving device where to send the media.
Page 127: 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 543). 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 128 Mediant 600 & Mediant 1000  To add remote NFS file systems: Open the Application Settings page (Configuration tab > System menu > Application Settings). Under the NFS Settings group, click the NFS Table button; the NFS Settings page appears.
Page 129 SIP User's Manual 10. Network Table 10-15: NFS Settings Parameters Parameter Description Index The row index of the remote file system. The valid range is 1 to 16. Host Or IP The domain name or IP address of the NFS server. If a domain name is provided, a DNS server must be configured.
Page 130: Robust Receipt Of Media Streams
Mediant 600 & Mediant 1000 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.
Page 131: Security
SIP User's Manual 11. Security Security This section describes the VoIP security-related configuration. 11.1 Configuring Firewall Settings The device provides an internal firewall, allowing you (the security administrator) to define network traffic filtering rules. You can add up to 50 ordered firewall rules. The access list provides the following firewall rules: ...
Page 132 Mediant 600 & Mediant 1000 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.
Page 133 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 134 Mediant 600 & Mediant 1000 Parameter Description Packet Size Defines the maximum allowed packet size. [AccessList_Packet_Size] The valid range is 0 to 65535. Note: When filtering fragmented IP packets, this field relates to the overall (re-assembled) packet size, and not to the size of each fragment.
Page 135: 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 541.  To configure the general security parameters: Open the General Security Settings page (Configuration tab >...
Page 136 Mediant 600 & Mediant 1000  To configure IP Security Proposals: Open the ‘IP Security Proposals Table page (Configuration tab > VoIP menu > Security submenu > IPSec Proposal Table). Figure 11-2: IP Security Proposals Table In the figure above, four proposals are defined.
Page 137: 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 138 Mediant 600 & Mediant 1000 Parameter Name Description Authentication Method Selects the method used for peer authentication during IKE [IPsecSATable_AuthenticationM main mode. ethod]  [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 139 SIP User's Manual 11. Security Parameter Name Description IPSec SA Lifetime (Kbs) Determines the maximum volume of traffic (in kilobytes) for [IPsecSATable_Phase2SaLifetim which the negotiated IPSec SA (Quick mode) is valid. After this eInKB] specified volume is reached, the SA is re-negotiated. The default value is 0 (i.e., the value is ignored).
Page 140 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 141: 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 541.
Page 142: Silence Suppression (Compression)
Mediant 600 & Mediant 1000 12.1.2 Silence Suppression (Compression) Silence suppression (compression) is a method for conserving bandwidth on VoIP calls by not sending packets when silence is detected. The device uses its VAD feature to detect periods of silence in the voice channel during an established call. When silence is detected, it stops sending packets in the channel.
Page 143: Fax And Modem Capabilities
SIP User's Manual 12. Media 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 144)  Fax and modem transport modes (see 'Fax/Modem Transport Modes' on page 144) ...
Page 144: Fax/Modem Operating Modes
Mediant 600 & Mediant 1000 12.2.1 Fax/Modem Operating Modes The device supports two modes of operation:  Fax/modem negotiation that is not performed during the establishment of the call.  Voice-band data (VBD) mode for V.152 implementation (see 'V.152 Support' on page 150): fax/modem capabilities are negotiated between the device and the remote endpoint at the establishment of the call.
Page 145: Fax / Modem Transport Mode
SIP User's Manual 12. Media Although this is a proprietary redundancy scheme, it should not create problems when working with other T.38 decoders. 12.2.2.1.1 Switching to T.38 Mode using SIP Re-INVITE In the Switching to T.38 Mode using SIP Re-INVITE mode, upon detection of a fax signal the terminating device negotiates T.38 capabilities using a Re-INVITE message.
Page 146: Fax Fallback
Mediant 600 & Mediant 1000  Dynamic Jitter Buffer Minimum Delay = 40  Dynamic Jitter Buffer Optimization Factor = 13 After a few seconds upon detection of fax V.21 preamble or super G3 fax signals, the device sends a second Re-INVITE enabling the echo canceller (the echo canceller is disabled only on modem transmission).
Page 147: 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 148: Fax / Modem Transparent With Events Mode
Mediant 600 & Mediant 1000 The Cisco gateway must include the following definition: "modem passthrough nse payload-type 100 codec g711alaw". To configure NSE mode, perform the following configurations:  IsFaxUsed = 0  FaxTransportMode = 2  NSEMode = 1 ...
Page 149: Rfc 2833 Ans Report Upon Fax/Modem Detection
SIP User's Manual 12. Media  BellModemTransportType = 0  Additional configuration parameters: • CodersGroup • DJBufOptFactor • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission. Instead, use the modes Bypass (see 'Fax/Modem Bypass Mode' on page 146) or Transparent with Events (see 'Fax / Modem Transparent with Events Mode' on page 148) for modem.
Page 150: Relay Mode For T.30 And V.34 Faxes
Mediant 600 & Mediant 1000  V23ModemTransportType = 2  V22ModemTransportType = 2 Configure the following parameters to use bypass mode for V.34 faxes and T.38 for T.30 faxes:  FaxTransportMode = 1 (Relay)  V34ModemTransportType = 2 (Modem bypass) ...
Page 151: Fax Transmission Behind Nat
SIP User's Manual 12. Media Instead of using VBD transport mode, the V.152 implementation can use alternative relay fax transport methods (e.g., fax relay over IP using T.38). The preferred V.152 transport method is indicated by the SDP ‘pmft’ attribute. Omission of this attribute in the SDP content means that VBD mode is the preferred transport mechanism for voice-band data.
Page 152: Configuring Rtp/Rtcp Settings
Mediant 600 & Mediant 1000 12.3 Configuring RTP/RTCP Settings The RTP/RTCP Settings page configures the Real-Time Transport Protocol (RTP) and Real-Time Transport (RTP) Control Protocol (RTCP) parameters. For a detailed description of the parameters appearing on this page, refer to 'Configuration Parameters Reference' on page 541.
Page 153: Configuring Dynamic Jitter Buffer Operation
SIP User's Manual 12. Media 12.3.1 Configuring 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 154: Comfort Noise Generation
Mediant 600 & Mediant 1000 The procedure below describes how to configure the jitter buffer using the Web interface.  To configure jitter buffer using the Web interface: Open the RTP/RTCP Settings page (Configuration tab > VoIP menu > Media submenu >...
Page 155 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 156: Configuring Rfc 2833 Payload
Mediant 600 & Mediant 1000 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 157: Configuring Rtp Multiplexing (Throughpacket)
The device's RTP Multiplexing (ThroughPacket™) feature is AudioCodes proprietary method for aggregating RTP streams from several channels when the device operates with another AudioCodes device. This feature reduces the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP headers and reduces the packet/data transmission rate.
Page 158: Configuring Rtp Base Udp Port
Mediant 600 & Mediant 1000 12.3.5 Configuring RTP Base UDP Port You can configure the range of UDP ports for RTP, RTCP, and T.38. The UDP port range can be configured using media realms in the Media Realm table, allowing you to assign different port ranges (media realms) to different interfaces.
Page 159: Configuring Rtp Control Protocol Extended Reports (Rtcp Xr)
SIP User's Manual 12. Media 12.3.6 Configuring RTP Control Protocol Extended Reports (RTCP XR) RTP Control Protocol Extended Reports (RTCP XR) is a VoIP management control that defines a set of metrics containing information for assessing VoIP call quality and for diagnosing problems.
Page 160: Configuring Ip Media Settings
Mediant 600 & Mediant 1000 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 541.
Page 161 SIP User's Manual 12. Media you can configure AMD per call, based on the called number or Trunk Group. This is achieved by defining the AMD parameters for a specific IP Profile (IPProfile parameter) and then assigning the IP Profile to a Trunk Group in the Inbound IP Routing table (PSTNPrefix parameter).
Page 162: Mediant 600 & Mediant
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 163 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: 30 Type= AMD SubType= AUTOMATA The device then detects the start of voice (i.e., the greeting message of the answering machine), and then sends the following to the Application server: INFO sip:sipp@172.22.2.9:5060 SIP/2.0...
Page 164: Configuring Automatic Gain Control (Agc)
Mediant 600 & Mediant 1000 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 Via: SIP/2.0/UDP 172.22.168.249;branch=z9hG4bKac482466515 Max-Forwards: 70 From: sut <sip:3000@172.22.168.249:5060>;tag=1c419779142 To: sipp <sip:sipp@172.22.2.9:5060>;tag=1...
Page 165: Configuring General Media Settings
SIP User's Manual 12. Media The procedure below describes how to configure AGC using the Web interface:  To configure AGC using the Web interface: Open the IPMedia Settings page (Configuration tab > VoIP menu > Media submenu > IPMedia Settings). Configure the following parameters: •...
Page 166: Configuring Analog Settings
Mediant 600 & Mediant 1000 12.6 Configuring Analog Settings The Analog Settings page allows you to configure various analog parameters. For a detailed description of the parameters appearing on this page, see 'Configuration Parameters Reference' on page 541. This page also selects the type (USA or Europe) of FXS and/or FXO coefficient information.
Page 167: Dsp Channel Resources For Transcoding
SIP User's Manual 12. Media Notes: • You must either use the ‘DSP Templates page or the DSPVersionTemplateNumber parameter to select the DSP template, not both. The DSP Templates page must be used only when two concurrent DSP templates are required; the DSPVersionTemplateNumber parameter must be used only when a single template is used.
Page 168 Mediant 600 & Mediant 1000 Notes: • If the device is installed with all three MPM modules, no other telephony interface module can be installed in the device. • It is recommended not to install an MPM module in Sot #6 as this reduces the available channels for transcoding to 20 (instead of 40).
Page 169: Configuring Media Realms
195) or SRDs (in the SRD table - see 'Configuring SRD Table' on page 191). 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 171.
Page 170 Mediant 600 & Mediant 1000 Table 12-3: Media Realm Table Parameter Descriptions Parameter Description Index Defines the required table index number. [CpMediaRealm_Index] Media Realm Name Defines an arbitrary, identifiable name for the Media Realm. [CpMediaRealm_MediaRealmName] The valid value is a string of up to 40 characters.
Page 171: Configuring Media Security
SIP User's Manual 12. Media 12.10 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 541.  To configure media security: Open the Media Security page (Configuration tab >...
Page 172 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 174.
Page 173 SIP User's Manual 12. Media MOS value changes by 0.1 (hysteresis) to 3.3 or 3.5, the device sends a report to the SEM indicating this change. If the value changes to 3.3, it sends a yellow state (i.e., medium quality); if the value changes to 3.5, it sends a green state. Configure the parameters as required.
Page 174: Configuring Server For Media Quality Of Experience
Mediant 600 & Mediant 1000 12.12 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 175: 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), enabling call routing decisions based on information stored on a third-party LDAP server (or Microsoft’s Active Directory™...
Page 176: Ad-Based Tel-To-Ip Routing In Microsoft Lync
Mediant 600 & Mediant 1000  To configure the LDAP server parameters: Open the LDAP Settings page (Configuration tab > VoIP menu > Services submenu > LDAP Settings). Figure 9: LDAP Settings Page The read-only 'LDAP Server Status' field displays one of the following possibilities: •...
Page 177 SIP User's Manual 13. Services configuration parameters listed in the table below are used to configure the query attribute keys that defines the AD attribute that you wish to query in the AD: Parameters for Configuring Query Attribute Key Parameter Queried User Domain (Attribute) in AD Query Result Example MSLDAPPBXNumAttribute...
Page 178 Mediant 600 & Mediant 1000 The device adds unique prefix keywords to the query results in order to identify the query type (i.e., IP domain). These prefixes are used as the prefix destination number value in the Outbound IP Routing table to denote the IP domains: •...
Page 179: Configuring Ad-Based Routing Rules
SIP User's Manual 13. Services The flowchart below summarizes the device's process for querying the AD and routing the call based on the query results: Figure 10: LDAP Query Flowchart 13.1.2.2 Configuring AD-Based Routing Rules The procedure below describes how to configure Tel-to-IP routing based on LDAP queries. ...
Page 180 Mediant 600 & Mediant 1000 Open the Advanced Parameters page (Configuration tab > VoIP menu > SIP Definitions submenu > Advanced Parameters). Figure 11: LDAP Parameters for Microsoft Lync Server 2010 Configure the LDAP attribute names as desired. Configure AD-based Tel-to-IP routing rules: Open the Outbound IP Routing Table page (Configuration tab >...
Page 181 SIP User's Manual 13. Services  Rule #1: Sends call to private telephone line (at 10.33.45.60) upon successful AD query result for the private attribute.  Rule #2: Sends call to IP PBX (at 10.33.45.65) upon successful AD query result for the PBX attribute.
Page 182: Least Cost Routing
Mediant 600 & Mediant 1000 13.2 Least Cost Routing This section provides a description of the device's least cost routing (LCR) feature and how to configure it. 13.2.1 Overview The LCR feature enables the device to choose the outbound IP destination routing rule based on lowest call cost.
Page 183 SIP User's Manual 13. Services  Example 1: This example uses two different Cost Groups for routing local calls and international calls: Two Cost Groups are configured as shown below: Table 13-2: Configured Cost Groups for Local and International Calls Cost Group Connection Cost Minute Cost...
Page 184 Mediant 600 & Mediant 1000 • Index 3 - no Cost Group is assigned, but as the Default Cost parameter is set to Min, it is selected as the cheapest route • Index 4 - Cost Group "B" is only second-matched rule (Index 1 is the first) ...
Page 185: Configuring Lcr
SIP User's Manual 13. Services 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 185. Configure Cost Groups - see 'Configuring Cost Groups' on page 187.
Page 186 Mediant 600 & Mediant 1000 Parameter Description LCR Call Length Defines the average call duration (in minutes) and is used to calculate [RoutingRuleGroups_LCR the variable portion of the call cost. This is useful, for example, when AverageCallLength] the average call duration spans over multiple time bands. The LCR is...
Page 187: Configuring Cost Groups
SIP User's Manual 13. Services 13.2.2.2 Configuring Cost Groups The procedure below describes how to configure Cost Groups. Cost Groups are defined with a fixed call connection cost and a call rate (charge per minute). Once configured, you can configure Time Bands for each Cost Group. Up to 10 Cost Groups can be configured. ...
Page 188: Configuring Time Bands For Cost Groups
Mediant 600 & Mediant 1000 13.2.2.3 Configuring Time Bands for Cost Groups The procedure below describes how to configure Time Bands for a Cost Group. The time band defines the day and time range for which the time band is applicable (e.g., from Saturday 05:00 to Sunday 24:00) as well as the fixed call connection charge and call rate per minute for this interval.
Page 189: Assigning Cost Groups To Routing Rules
SIP User's Manual 13. Services Parameter Description Connection Cost Defines the call connection cost during this time band. [CostGroupTimebands_ConnectionCost] This is added as a fixed charge to the call. The valid value range is 0-65533. The default is 0. Note: The entered value must be a whole number (i.e., not a decimal).
Page 190 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 191: 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 169).
Page 192 Mediant 600 & Mediant 1000  To configure SRDs: Open the SRD Settings page (Configuration tab > VoIP menu > Control Network submenu > SRD Table). From the 'SRD Index' drop-down list, select an index for the SRD, and then configure it according to the table below.
Page 193: Configuring Sip Interface Table
SIP User's Manual 14. Control Network Parameter Description this value becomes invalid in the SRD table.  For configuring Media Realms, see 'Configuring Media Realms' on page 169. 14.2 Configuring SIP Interface Table The SIP Interface Table page allows you to configure up to 32 SIP signaling interfaces, referred to as SIP Interfaces.
Page 194 Mediant 600 & Mediant 1000 Table 14-2: SIP Interface Table Parameters Parameter Description Network Interface Defines the Control-type IP network interface that you want to [SIPInterface_NetworkInter associate with the SIP Interface. This value string must be identical face] (including case-sensitive) to that configured in the 'Interface Name' in the Multiple Interface table (see 'Configuring IP Interface Settings' on page 102).
Page 195: Configuring Ip Groups
SIP User's Manual 14. Control Network 14.3 Configuring IP Groups The IP Group Table page allows you to create up to 32 logical IP entities called IP Groups. An IP Group is an entity with a set of definitions such as a Proxy Set ID (see 'Configuring Proxy Sets Table' on page 200), which represents the IP address of the IP Group.
Page 196 Mediant 600 & Mediant 1000  To configure IP Groups: Open the IP Group Table page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). Configure the IP group parameters according to the table below. Click Submit to apply your changes.
Page 197 SIP User's Manual 14. Control Network Parameter Description destination is obtained from this entry, and a SIP request is sent to the destination. The device supports up to 600 registered users. The device also supports NAT traversal for the SIP clients that are behind NAT.
Page 198 Mediant 600 & Mediant 1000 Parameter Description The SRD (defined in Configuring SRD Table on page 191) associated [IPGroup_SRD] with the IP Group. The default is 0. Note: For this parameter to take effect, a device reset is required. Media Realm Assigns a Media Realm to the IP Group.
Page 199 SIP User's Manual 14. Control Network Parameter Description Refer-To header in the REFER message or Contact header in the 3xx response (default).  [1] Proxy = Sends a new INVITE to the Proxy. Note: Applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0.
Page 200: Configuring Proxy Sets Table
Mediant 600 & Mediant 1000 Parameter Description Serving IP Group ID If configured, INVITE messages initiated from the IP Group are sent to [IPGroup_ServingIPGroup] this Serving IP Group (range 1 to 9). In other words, the INVITEs are sent to the address defined for the Proxy Set associated with this Serving IP Group.
Page 201 SIP User's Manual 14. Control Network  To add Proxy servers: Open the Proxy Sets Table page (Configuration tab > VoIP menu > Control Network submenu > Proxy Sets Table). Figure 14-1: Proxy Sets Table Page From the 'Proxy Set ID' drop-down list, select an ID for the desired group. Configure the Proxy parameters according to the following table.
Page 202 Mediant 600 & Mediant 1000 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 203 SIP User's Manual 14. Control Network Parameter Description REGISTER messages. If set to 'Using Options', the SIP OPTIONS message is sent every user-defined interval (configured by the parameter ProxyKeepAliveTime). If set to 'Using Register', the SIP REGISTER message is sent every user-defined interval (configured by the RegistrationTime parameter).
Page 204: Configuring Nat Translation Per Ip Interface
Mediant 600 & Mediant 1000 Parameter Description FQDN should be configured as a Proxy IP address. The Random Weights Load Balancing is not used in the following scenarios:  The Proxy Set includes more than one Proxy IP address. ...
Page 205 SIP User's Manual 14. Control Network Uses the StaticNATIP parameter to define one NAT IP address for all interfaces. Uses the NATTranslation parameter to define NAT per interface. If NAT is not configured (by any of the above-mentioned methods), the device sends the packet according to its IP address defined in the Multiple Interface table.
Page 206: Multiple Sip Signaling And Media Interfaces Using Srds
Mediant 600 & Mediant 1000 14.6 Multiple SIP Signaling and Media Interfaces using SRDs The device supports the configuration of multiple, logical SIP signaling interfaces and media (RTP) interfaces. Multiple SIP and media interfaces allows you to:  Separate SIP and media traffic between different applications (i.e., SAS, Gateway\IP- to-IP) ...
Page 207 SIP User's Manual 14. Control Network Figure 14-3: Configuring SRDs and Assignment Typically, an SRD is defined per group of SIP UAs (e.g., proxies, IP phones, application servers, gateways, softswitches) that communicate with each other. This provides these entities with VoIP services that reside on the same Layer-3 network (must be able to communicate without traversing NAT devices and must not have overlapping IP addresses).
Page 208 Mediant 600 & Mediant 1000 The figure below illustrates a typical scenario for implementing multiple SIP signaling interfaces. In this example, different SIP signaling interfaces and RTP traffic interfaces are assigned to Network 1 (ITSP A) and Network 2 (ITSP B).
Page 209 SIP User's Manual 14. Control Network 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: Figure 14-4: Multiple SIP Signaling/RTP Interfaces Example Version 6.4 March 2012...
Page 210 Mediant 600 & Mediant 1000 Note that only the steps specific to multiple SIP signaling/RTP configuration are described in detail in the procedure below.  To configure multiple SIP signaling and RTP interfaces: Configure Trunk Group ID #1 in the Trunk Group Table page (Configuration tab >...
Page 211 SIP User's Manual 14. Control Network Configure the SIP Interfaces in the SIP Interface Table page (Configuration tab > VoIP menu > Control Network submenu > SIP Interface Table): Figure 14-8: Defining SIP Interfaces Configure Proxy Sets in the Proxy Sets Table page (Configuration tab > VoIP menu >...
Page 212 Mediant 600 & Mediant 1000 Configure IP Groups in the IP Group Table page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). The figure below configures IP Group for ITSP A. Do the same for ITSP B but for Index 2 with SRD 1 and Media Realm to "Realm2".
Page 213: Enabling Applications
This page displays the application only if the device is installed with the relevant Software Upgrade Key supporting the application (see 'Loading Software Upgrade Key' on page 497). • The IP2IP application is applicable only to Mediant 1000. • For configuring the SAS application, see 'Stand-Alone Survivability (SAS) Application' on page 381.
Page 214 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 215: 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 216: Configuring Coder Groups
Mediant 600 & Mediant 1000  To configure the device's coders: Open the Coders page (Configuration tab > VoIP menu > Coders And Profiles submenu > Coders). Figure 16-1: Coders Page From the 'Coder Name' drop-down list, select the required coder.
Page 217: 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 218 Mediant 600 & Mediant 1000 Note: You can also configure Tel Profiles using the ini file table parameter TelProfile (see 'Configuration Parameters Reference' on page 541).  To configure Tel Profiles: Open the Tel Profile Settings page (Configuration tab > VoIP menu > Coders And Profiles submenu >...
Page 219: Configuring Ip Profiles
SIP User's Manual 16. Coders and Profiles coders common to both are used. The order of the coders is determined by the preference. Configure the parameters as required. For more information on each parameter, refer to the description of the "global" parameter. From the 'Coder Group' drop-down list, select the Coder Group (see 'Configuring Coder Groups' on page 216) or the device's default coder (see 'Configuring Coders' on page 215) to which you want to assign the Tel Profile.
Page 220 Mediant 600 & Mediant 1000  To configure IP Profiles: Open the IP Profile Settings page (Configuration tab > VoIP menu > Coders And Profiles submenu > IP Profile Settings). From the 'Profile ID' drop-down list, select the IP Profile index.
Page 221 SIP User's Manual 16. Coders and Profiles 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 222 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 223: 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 541.
Page 224: Configuring Advanced Parameters
Mediant 600 & Mediant 1000 Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 482. 17.2 Configuring Advanced Parameters The Advanced Parameters page allows you to configure advanced SIP control parameters.
Page 225: Configuring Account Table
SIP User's Manual 17. SIP Definitions Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 482. 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).
Page 226 Mediant 600 & Mediant 1000 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 227 SIP User's Manual 17. SIP Definitions Parameter Description Group’ table is used instead. This parameter can be up to 49 characters. Register Enables registration. [Account_Register]  [0] No = Don't register  [1] Yes = Enables registration When enabled, the device sends REGISTER requests to the Serving IP Group.
Page 228: Configuring Proxy And Registration Parameters
Mediant 600 & Mediant 1000 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 541.
Page 229 SIP User's Manual 17. SIP Definitions Configure the parameters as required. Click Submit to apply your changes. Click the Register or Un-Register buttons save your changes register/unregister the device to a Proxy/Registrar. Instead of registering the entire device, you can register specific entities (FXS/FXO endpoints, Trunk Groups, BRI endpoints, and Accounts), by using the Register button located on the page in which these entities are configured.
Page 230 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 231: 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 232 Mediant 600 & Mediant 1000  To modify the CAS state machine parameters: Open the CAS State Machine page (Configuration tab > VoIP menu > PSTN submenu > CAS State Machines). Figure 18-1: CAS State Machine Page Ensure that the trunk is inactive. The trunk number displayed in the 'Related Trunks' field must be green.
Page 233 SIP User's Manual 18. GW and IP to IP Parameter Description DTMF Max Detection Time Detects digit maximum on time (according to DSP [CasStateMachineDTMFMaxOnDetecti detection information event) in msec units. onTime] The value must be a positive value. The default value is - 1 (use value from CAS state machine).
Page 234: Configuring Trunk Settings
Mediant 600 & Mediant 1000 18.1.3 Configuring Trunk Settings The Trunk Settings page allows you to configure the device's trunks. This includes selecting the PSTN protocol and configuring related parameters. Some parameters can be configured when the trunk is in service, while others require you to take the trunk out of service (by clicking the Stop button).
Page 235 SIP User's Manual 18. GW and IP to IP On the top of the page, a bar with Trunk number icons displays the status of each trunk, according to the following color codes: • Grey: Disabled • Green: Active • Yellow: RAI alarm (also appears when you deactivate a Trunk by clicking the Deactivate button) •...
Page 236 Mediant 600 & Mediant 1000 Click the Apply Trunk Settings button to apply the changes to the selected trunk (or click Apply to All Trunks to apply the changes to all trunks); the Stop Trunk button replaces Apply Trunk Settings and the ‘Trunk Configuration State’ displays 'Active'.
Page 237: Configuring Digital Gateway Parameters
SIP User's Manual 18. GW and IP to IP 18.1.4 Configuring Digital Gateway Parameters The Digital Gateway Parameters page allows you to configure miscellaneous digital parameters. For a description of these parameters, see 'Configuration Parameters Reference' on page 541.  To configure the digital gateway parameters: Open the Digital Gateway Parameters page (Configuration tab >...
Page 238: Tunneling Applications
Mediant 600 & Mediant 1000 18.1.5 Tunneling Applications This section discusses the device's support for VoIP tunneling applications. 18.1.5.1 TDM Tunneling The device's TDM Tunneling feature allows you to tunnel groups of digital trunk spans or timeslots (B-channels) over the IP network. TDM Tunneling utilizes the device's internal...
Page 239 SIP User's Manual 18. GW and IP to IP Note: For TDM over IP, the parameter CallerIDTransportType must be set to '0' (disabled), i.e., transparent. Below is an example of ini files for two devices implementing TDM Tunneling for four E1 spans.
Page 240 Mediant 600 & Mediant 1000 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 ProtocolType_0 = 5 ProtocolType_1 = 5 ProtocolType_2 = 5 ProtocolType_3 = 5 ;Channel selection by Phone number.
Page 241: Qsig Tunneling
SIP User's Manual 18. GW and IP to IP 18.1.5.1.1 DSP Pattern Detector For TDM tunneling applications, you can use the DSP pattern detector feature to initiate the echo canceller at call start. The device can be configured to support detection of a specific one-byte idle data pattern transmitted over digital E1/T1 timeslots.
Page 242: Advanced Pstn Configuration
Mediant 600 & Mediant 1000  Mid-call communication: After the SIP connection is established, all QSIG messages are encapsulated in SIP INFO messages.  Call tear-down: The SIP connection is terminated once the QSIG call is complete. The Release Complete message is encapsulated in the SIP BYE message that terminates the session.
Page 243 SIP User's Manual 18. GW and IP to IP  Receiving Reason header: If a call is disconnected from the IP side and the SIP message includes the Reason header, it is sent to the Tel side according to the following logic: •...
Page 244 Mediant 600 & Mediant 1000 ISDN Release Description Description Reason Response Resource unavailable Service unavailable QoS unavailable 503* Service unavailable Facility not subscribed 503* Service unavailable Incoming calls barred within CUG Forbidden Bearer capability not authorized Forbidden Bearer capability not presently available...
Page 245 SIP User's Manual 18. GW and IP to IP ISDN Release Description Description Reason Response Protocol error Server internal error Interworking unspecified Server internal error * Messages and responses were created because the ‘ISUP to SIP Mapping’ draft doesn’t specify their cause code mapping. 18.1.6.1.3 Fixed Mapping of SIP Response to ISDN Release Reason The following table describes the mapping of SIP response to ISDN release reason.
Page 246: Isdn Overlap Dialing
Mediant 600 & Mediant 1000 ISDN Release SIP Response Description Description Reason Server internal error Temporary failure Not implemented Network out of order Bad gateway Network out of order Service unavailable Temporary failure Server timeout Recovery on timer expiry 505*...
Page 247 SIP User's Manual 18. GW and IP to IP The device stops collecting digits (from the ISDN) upon the following scenarios:  The device receives a Sending Complete IE in the ISDN Setup or Information messages, indicating no more digits. ...
Page 248: Isdn Non-Facility Associated Signaling (Nfas)
Mediant 600 & Mediant 1000 18.1.6.3 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. This channel is called the D-channel and usually resides on timeslot # 24.
Page 249 SIP User's Manual 18. GW and IP to IP Notes: • Usually the Interface Identifier is included in the Q.931 Setup/Channel Identification IE only on T1 trunks that doesn’t contain the D-channel. Calls initiated on B-channels of the Primary T1 trunk, by default, don’t contain the Interface Identifier.
Page 250: Redirect Number And Calling Name (Display)
Mediant 600 & Mediant 1000 18.1.6.3.3 Creating an NFAS-Related Trunk Configuration The procedures for creating and deleting an NFAS group must be performed in the correct order, as described below.  To create an NFAS Group: If there’s a backup (‘secondary’) trunk for this group, it must be configured first.
Page 251: 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 252 Mediant 600 & Mediant 1000 Parameter Description From Trunk Defines the starting physical Trunk number in the Trunk Group. The [TrunkGroup_FirstTrunkId] number of listed Trunks depends on the device's hardware configuration. Note: This parameter is applicable only to PRI and BRI modules.
Page 253: Configuring Trunk Group Settings
SIP User's Manual 18. GW and IP to IP Parameter Description [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 254 Mediant 600 & Mediant 1000 REGISTER sip:SipGroupName SIP/2.0 Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac862428454 From: <sip:101@GatewayName>;tag=1c862422082 To: <sip:101@GatewayName> Call-ID: 9907977062512000232825@10.33.37.78 CSeq: 3 REGISTER Contact: <sip:101@10.33.37.78>;expires=3600 Expires: 3600 User-Agent: Sip-Gateway/v.6.00A.008.002 Content-Length: 0 Table 18-7: Trunk Group Settings Parameters Parameter Description Trunk Group ID The Trunk Group ID that you want to configure.
Page 255 SIP User's Manual 18. GW and IP to IP Parameter Description endpoints from being registered by assigning them to a Trunk Group and configuring the Trunk Group registration mode to 'Don't Register'.  [5] Per Account = Registrations are sent (or not) to an IP Group, according to the settings in the Account table (see 'Configuring Account Table' on page 225).
Page 256: Manipulation
Mediant 600 & Mediant 1000 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 541.
Page 257 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 258 Mediant 600 & Mediant 1000  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 259 SIP User's Manual 18. GW and IP to IP Parameter Description Inbound IP Routing Table'. If not used (i.e., any IP Group), simply leave the field empty. Notes:  The value -1 indicates that this field is ignored in the rule. ...
Page 260: Configuring Redirect Number Ip To Tel
Mediant 600 & Mediant 1000 Parameter Description Web: NPI The Numbering Plan Indicator (NPI) assigned to this entry. EMS: Number Plan  [0] Unknown (default)  [9] Private  [1] E.164 Public  [-1] Not Configured = value received from PSTN/IP is used Notes: ...
Page 261 SIP User's Manual 18. GW and IP to IP Notes: • You can also configure the Redirect Number IP to Tel table using the ini file parameter RedirectNumberMapIp2Tel (see 'Number Manipulation Parameters' on page 750). • If the characteristics Destination Prefix, Redirect Prefix, and/or Source Address match the incoming SIP message, manipulation is performed according to the configured manipulation rule.
Page 262: Configuring Redirect Number Tel To Ip
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Suffix to Add The number or string that you want added to the end of the telephone number. For example, if you enter '00' and the phone number is 1234, the new number is 123400.
Page 263 SIP User's Manual 18. GW and IP to IP Notes: • Redirect Tel-to-IP manipulation is not done if the device copies the received destination number to the outgoing SIP redirect number, as enabled by the CopyDest2RedirectNumber parameter. • You can also configure the Redirect Number Tel to IP table using the ini file parameter RedirectNumberMapTel2Ip (see 'Number Manipulation Parameters' on page 750).
Page 264: Mapping Npi/Ton To Sip Phone-Context
Mediant 600 & Mediant 1000 Parameter Description the field empty. Notes:  The value -1 indicates that it is ignored in the rule.  This parameter is applicable only to the IP-to-IP application. Web/EMS: Destination Destination (called) telephone number prefix. An asterisk (*) represents Prefix any number.
Page 265 SIP User's Manual 18. GW and IP to IP  To configure the Phone-Context tables: Open the Phone Context Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations submenu > Phone Context). Figure 18-8: Phone Context Table Page Configure the Phone Context table according to the table below.
Page 266: Numbering Plans And Type Of Number
Mediant 600 & Mediant 1000 Parameter Description  [2] Level 1 Regional  [3] PSTN Specific  [4] Level 0 Regional (Local)  If you selected E.164 Public as the NPI, you can select one of the following:  [0] Unknown ...
Page 267: Configuring Release Cause Mapping
SIP User's Manual 18. GW and IP to IP For NI-2 and DMS-100 ISDN variants, the valid combinations of TON and NPI for calling and called numbers include (Plan/Type):  0/0 - Unknown/Unknown  1/1 - International number in ISDN/Telephony numbering plan ...
Page 268: Sip Calling Name Manipulations
Mediant 600 & Mediant 1000 In the 'Release Cause Mapping from ISDN to SIP' group, map different Q.850 Release Causes to SIP Responses. In the 'Release Cause Mapping from SIP to ISDN' group, map different SIP Responses to Q.850 Release Causes.
Page 269: Manipulating Number Prefix
SIP User's Manual 18. GW and IP to IP Notes: • Each message can be manipulated twice - once for the source leg manipulation rules and once in the destination leg (source and destination IP Groups). • Unknown SIP parts can only be added or removed. •...
Page 270: Routing
Mediant 600 & Mediant 1000 The first seven digits from the left are removed from the original number, by entering "7" in the 'Stripped Digits From Left' field. Figure 18-10: Prefix to Add Field with Notation In this configuration, the following manipulation process occurs: 1) the prefix is calculated, 020215 in the example;...
Page 271: Configuring Outbound Ip Routing Table
SIP User's Manual 18. GW and IP to IP 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 272 Mediant 600 & Mediant 1000 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 273 SIP User's Manual 18. GW and IP to IP 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 274 Mediant 600 & Mediant 1000  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). The figure above displays the following outbound IP routing rules: •...
Page 275 SIP User's Manual 18. GW and IP to IP Click Submit to apply your changes. To save the changes to flash memory, see 'Saving Configuration' on page 482. Table 18-13: Outbound IP Routing Table Parameters Parameter Description Web/EMS: Tel to IP Determines whether to route received calls to an IP destination before or after Routing Mode manipulation of the destination number.
Page 276 Mediant 600 & Mediant 1000 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) denotes 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 787.
Page 277 SIP User's Manual 18. GW and IP to IP Parameter Description address. However, if both parameters are configured in this table, the INVITE message is sent only to the IP Group (and not the defined IP address).  If the destination IP Group is of type USER, the device searches for a match between the Request-URI (of the received INVITE) to an AOR registration record in the device's database.
Page 278 Mediant 600 & Mediant 1000 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 279: Configuring Inbound Ip Routing Table
SIP User's Manual 18. GW and IP to IP 18.4.3 Configuring Inbound IP Routing Table The Inbound IP Routing Table page allows you to configure up to 24 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).
Page 280 Mediant 600 & Mediant 1000  To configure inbound IP routing rules: Open the Inbound IP Routing Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing submenu > IP to Trunk Group Routing).
Page 281 SIP User's Manual 18. GW and IP to IP Parameter Description Source Host Prefix The From URI host name prefix of the incoming SIP INVITE message. If this routing rule is not required, leave the field empty. Notes:  The asterisk (*) wildcard can be used to denote any prefix. ...
Page 282: Mapping Pstn Release Cause To Sip Response
Mediant 600 & Mediant 1000 Parameter Description 225). 18.4.4 Mapping PSTN Release Cause to SIP Response The device's FXO interface interoperates between the SIP network and the PSTN/PBX. This interoperability includes the mapping of PSTN/PBX Call Progress Tones to SIP 4xx or 5xx responses for IP-to-Tel calls.
Page 283: Alternative Routing For Tel-To-Ip Calls
SIP User's Manual 18. GW and IP to IP  DNS Resolution: When a host name (FQDN) is used (instead of an IP address) for the IP destination, it is resolved into an IP address by a DNS server. The device checks network connectivity and QoS of the resolved IP address.
Page 284: Alternative Routing Based On Sip Responses
Mediant 600 & Mediant 1000 alternative destination. For more information on the IP connectivity methods and on viewing IP connectivity status, see 'IP Destinations Connectivity Feature' on page 282. The table below shows an example of alternative routing where the device uses an available alternative routing rule in the Outbound IP Routing table to re-route the initial, unsuccessful Tel-to-IP call routing destination.
Page 285 SIP User's Manual 18. GW and IP to IP Depending on configuration, the alternative routing is done using one of the following entities:  Outbound IP Routing Rules: You can configure up to two alternative routing rules in the table. If the initial, main routing rule destination is unavailable, the device searches the table (starting from the top) for the next call matching rule (e.g., destination phone number), and if available attempts to re-route the call to the IP destination configured for this alternative routing rule.
Page 286: Pstn Fallback
Mediant 600 & Mediant 1000 The steps for configuring alternative Tel-to-IP routing based on SIP response codes are summarized below.  To configure alternative Tel-to-IP routing based on SIP response codes: Enable alternative routing based responses, setting RedundantRoutingMode parameter to 1 (for using the Outbound IP Routing table) or 2 (for using the Proxy Set).
Page 287: Alternative Routing For Ip-To-Tel Calls
SIP User's Manual 18. GW and IP to IP 18.4.7 Alternative Routing for IP-to-Tel Calls The device supports alternative IP-to-Tel call routing, as described in this section. 18.4.7.1 Alternative Routing to Trunk upon Q.931 Call Release Cause Code You can configure the device to do alternative IP-to-Tel call routing based on the received ISDN Q.931 cause code.
Page 288: Alternative Routing To Ip Destination Upon Busy Trunk
Mediant 600 & Mediant 1000 The steps for configuring alternative Trunk Group routing based on Q.931 cause codes are summarized below.  To configure alternative Trunk Group routing based on Q.931 cause codes: Set the RedundantRoutingMode parameter to 1 so that the device uses the Inbound IP Routing table for alternative routing.
Page 289 SIP User's Manual 18. GW and IP to IP Note: You can also configure the Forward on Busy Trunk Destination table using the ini file parameter table ForwardOnBusyTrunkDest.  To configure Forward on Busy Trunk Destination rules: Open the Forward on Busy Trunk Destination page (Configuration tab > VoIP menu >...
Page 290: Dtmf And Supplementary
Mediant 600 & Mediant 1000 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. For a description of the parameters appearing on this page, see 'Configuration Parameters Reference' on page 541.
Page 291: Configuring Supplementary Services
SIP User's Manual 18. GW and IP to IP 18.5.2 Configuring Supplementary Services The Supplementary Services page is used to configure parameters associated with supplementary services. For a description of the parameters appearing on this page, see 'Configuration Parameters Reference' on page 541. The procedure below describes how to access the Supplementary Services page and configure the supplementary services parameters.
Page 292 Mediant 600 & Mediant 1000  To configure supplementary services parameters: Open the Supplementary Services page (Configuration tab > VoIP menu > GW and IP to IP submenu > DTMF & Supplementary submenu > Supplementary Services). SIP User's Manual Document #: LTRT-83310...
Page 293: Call Hold And Retrieve
SIP User's Manual 18. GW and IP to IP Configure the parameters as required. Click Submit to apply your changes, or click the Subscribe to MWI or Unsubscribe to MWI buttons to save your changes and to subscribe / unsubscribe to the MWI server.
Page 294 Mediant 600 & Mediant 1000 The device also supports "double call hold" for FXS interfaces where the called party, which has been placed on-hold by the calling party, can then place the calling party on hold as well and make a call to another destination. The flowchart below provides an example of...
Page 295: Bri Suspend And Resume
SIP User's Manual 18. GW and IP to IP A calls C and establishes a voice path. B places A on hold; B hears a Dial tone. B calls D and establishes a voice path. A ends call with C; A hears a Held tone. B ends call with D.
Page 296: Call Transfer
Mediant 600 & Mediant 1000 18.5.2.4 Call Transfer The device supports the following call transfer types:  Consultation Transfer (see 'Consultation Call Transfer' on page 296)  Blind Transfer (see 'Blind Call Transfer' on page 297) Notes: • Call transfer is initiated by sending REFER with REPLACES.
Page 297 SIP User's Manual 18. GW and IP to IP 18.5.2.4.2 Consultation Transfer for QSIG Path Replacement The device can interwork consultation call transfer requests for ISDN QSIG-to-IP calls. When the device receives a request for a consultation call transfer from the PBX, the device sends a SIP REFER message with a Replaces header to the SIP UA to transfer it to another SIP UA.
Page 298: Call Forward
Mediant 600 & Mediant 1000 Notes: • Manipulation using the ManipulateIP2PSTNReferTo parameter does not affect IP-to-Trunk Group routing rules. • Currently, the device does not support blind transfer for BRI interfaces. 18.5.2.5 Call Forward For digital interfaces: The device supports Call Deflection (ETS-300-207-1) for Euro ISDN...
Page 299 SIP User's Manual 18. GW and IP to IP This is important in that it notifies (audibly) the FXS endpoint user that a call forwarding service is currently being performed. The device generates a Call Forward Reminder ring burst to the FXS endpoint each time it receives a SIP NOTIFY message with a “reminder ring”...
Page 300: Call Waiting
Mediant 600 & Mediant 1000  Event: ua-profile  Content-Type: "application/simservs+xml"  Message body is the XML body containing the “dial-tone-pattern” set to "standard- condition-tone" (<ss:dial-tone-pattern>standard-condition-tone</ss:dial-tone-pattern>), which is the regular dial tone indication. Therefore, the special dial tone is valid until another SIP NOTIFY is received that instructs otherwise (as described above).
Page 301: Message Waiting Indication
SIP User's Manual 18. GW and IP to IP call waiting tone (several configurable short beeps) and (for Bellcore and ETSI Caller IDs) can view the Caller ID string of the incoming call. The calling party hears a Call Waiting Ringback Tone.
Page 302 Mediant 600 & Mediant 1000  EnableMWISubscription  MWIExpirationTime  SubscribeRetryTime  SubscriptionMode  CallerIDType (determines the standard for detection of MWI signals)  ETSIVMWITypeOneStandard  BellcoreVMWITypeOneStandard  VoiceMailInterface  EnableVMURI The device supports the following MWI features for its digital PSTN interfaces: ...
Page 303: Caller Id
SIP User's Manual 18. GW and IP to IP • Send MWI Interrogation message, use its result, and use the MWI Activate requests. 18.5.2.8 Caller ID This section discusses the device's Caller ID support. Note: The Caller ID feature is applicable only to FXS/FXO interfaces. 18.5.2.8.1 Caller ID Detection / Generation on the Tel Side By default, generation and detection of Caller ID to the Tel side is disabled.
Page 304 Mediant 600 & Mediant 1000 standard (using parameters CallerIDType, BellcoreCallerIDTypeOneSubStandard, and ETSICallerIDTypeOneSubStandard). Define the number of rings before the device starts the detection of caller ID (using the parameter RingsBeforeCallerID). Verify that the correct FXO coefficient type is selected (using the parameter CountryCoefficients), as the device is unable to recognize caller ID signals that are distorted.
Page 305: Three-Way Conferencing
The device supports the following conference modes (configured by the parameter 3WayConferenceMode):  Conferencing controlled by an external AudioCodes Conference (media) server: The Conference-initiating INVITE sent by the device uses the ConferenceID concatenated with a unique identifier as the Request-URI. This same Request-URI is set as the Refer-To header value in the REFER messages that are sent to the two remote parties.
Page 306: Emergency E911 Phone Number Services
Mediant 600 & Mediant 1000 Notes: • Three-way conferencing using an external conference server is supported only by FXS interfaces. • The on-board, three-way conference mode is not supported by Mediant 600. • Instead of using the flash-hook button to establish a three-way conference call, you can dial a user-defined hook-flash code (e.g., "*1"),...
Page 307 SIP User's Manual 18. GW and IP to IP 18.5.2.10.1 FXS Device Emulating PSAP using DID Loop-Start Lines The FXS device can be configured to emulate PSAP (using DID loop start lines), according to the Telcordia GR-350-CORE specification. Figure 18-17: FXS Device Emulating PSAP using DID Loop-Start Lines The call flow of an E911 call to the PSAP is as follows: The E911 tandem switch seizes the line.
Page 308 Mediant 600 & Mediant 1000 After the call is disconnected by the PSAP, the PSAP sends a SIP BYE to the FXS device, and the FXS device reverses the polarity of the line toward the tandem switch. The following parameters need to be configured: ...
Page 309 SIP User's Manual 18. GW and IP to IP The MF KP, ST, and STP digits are mapped as follows:  * for KP  # for ST  B for STP For example, if ANI and PANI are received, the SIP INVITE contains the following From header: From: <sip:*nnnnnnnnnnnn#*mmmmmmmmmm#@10.2.3.4>;tag=1c14 Note:...
Page 310 Mediant 600 & Mediant 1000 18.5.2.10.2 FXO Device Interworking SIP E911 Calls from Service Provider's IP Network to PSAP DID Lines The FXO device can interwork SIP emergency E911 calls from the Service Provider's IP network to the analog PSAP DID lines. The standards that define this interface include TR- TSY-000350 or Bellcore’s GR-350-Jun2003.
Page 311 SIP User's Manual 18. GW and IP to IP For supporting the E911 service, used the following configuration parameter settings:  Enable911PSAP = 1 (also forces the EnableDIDWink and EnableReversalPolarity)  HookFlashOption = 1 (generates the SIP INFO hookflash message) or 4 for RFC 2833 telephony event ...
Page 312 Mediant 600 & Mediant 1000  KP is for .  ST is for #.  STP is for B. The MF duration of all digits, except for the KP digit is 60 msec. The MF duration of the KP digit is 120 msec.
Page 313: Multilevel Precedence And Preemption
SIP User's Manual 18. GW and IP to IP a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv  Example (b): The detection of a Wink signal generates the following SIP INFO message: INFO sip:4505656002@192.168.13.40:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.13.2:5060 From: port1vega1 <sip:06@192.168.13.2:5060> To: <sip:4505656002@192.168.13.40:5060>;tag=132878796- 1040067870294 Call-ID: 0010-0016-D69A7DA8-1@192.168.13.2 CSeq:2 INFO...
Page 314 Mediant 600 & Mediant 1000 of devices and resources that are associated with an MLPP subscriber. When an MLPP subscriber that belongs to a particular domain places a precedence call to another MLPP subscriber that belongs to the same domain, MLPP service can preempt the existing call that the called MLPP subscriber is on for a higher- precedence call.
Page 315 SIP User's Manual 18. GW and IP to IP • The device performs a preemption of a B-channel for a Tel-to-IP outbound call request from the softswitch for which it has not received an answer response (e.g., Connect), and the following sequence of events occurs: The device sends a Q.931 DISCONNECT over the ISDN MLPP PRI to the partner switch to preempt the remote end instrument.
Page 316: Denial Of Collect Calls
Mediant 600 & Mediant 1000 18.5.2.12 Denial of Collect Calls You can configure the device to reject (disconnect) incoming Tel-to-IP collect calls and to signal this denial to the PSTN. This capability is required, for example, in the Brazilian telecommunication system to deny collect calls. When this feature is enabled upon rejecting the incoming call, the device sends a sequence of signals to the PSTN.
Page 317 SIP User's Manual 18. GW and IP to IP  To configure BRI supplementary services: Open the ISDN Supp Services Table page (Configuration tab > VoIP menu > GW and IP to IP submenu > Digital Gateway submenu > ISDN Supp Services). Figure 18-19: ISDN Supp Services Table Page Configure the parameters as described in the table below.
Page 318: Configuring Voice Mail Parameters
Mediant 600 & Mediant 1000 18.5.4 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 541. Notes: • The Voice Mail Settings page is available only for FXO and CAS interfaces.
Page 319: Advice Of Charge Services For Euro Isdn
SIP User's Manual 18. GW and IP to IP 18.5.5 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 320: Analog Gateway
Mediant 600 & Mediant 1000 18.6 Analog Gateway This section describes configuration of analog settings. Note: The Analog Gateway submenu appears only if the device is installed with an FXS or FXO module. 18.6.1 Configuring Keypad Features The Keypad Features page enables you to activate and deactivate the following features directly from the connected telephone's keypad: ...
Page 321: Configuring Metering Tones
SIP User's Manual 18. GW and IP to IP  To configure the keypad features Open the Keypad Features page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Keypad Features). Figure 18-20: Keypad Features Page Configure the keypad features as required.
Page 322: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 Notes: • The Metering Tones page is available only for FXS interfaces. • Charge Code rules can be assigned to routing rules in the Outbound IP Routing Table (see 'Configuring Outbound IP Routing Table' on page 271).
Page 323: Configuring Charge Codes
SIP User's Manual 18. GW and IP to IP 18.6.3 Configuring Charge Codes The Charge Codes Table page is used to configure the metering tones (and their time interval) that the FXS interfaces generate to the Tel side. To associate a charge code to an outgoing Tel-to-IP call, use the Outbound IP Routing Table'.
Page 324: Configuring Fxo Settings
Mediant 600 & Mediant 1000 18.6.4 Configuring FXO Settings The FXO Settings page allows you to configure the device's specific FXO parameters. For a description of these parameters, see 'Configuration Parameters Reference' on page 541. Note: The FXO Settings page is available only for FXO interfaces.
Page 325: Configuring Authentication
SIP User's Manual 18. GW and IP to IP 18.6.5 Configuring Authentication The Authentication page defines a user name and password for authenticating each device port. Authentication is typically used for FXS interfaces, but can also be used for FXO interfaces.
Page 326: Configuring Automatic Dialing
Mediant 600 & Mediant 1000 18.6.6 Configuring Automatic Dialing The Automatic Dialing page allows you to define a telephone number that is automatically dialed when an FXS or FXO port is used (e.g., off-hooked). Notes: • After a ring signal is detected on an 'Enabled' FXO port, the device initiates a call to the destination number without seizing the line.
Page 327: Configuring Caller Display Information
SIP User's Manual 18. GW and IP to IP 18.6.7 Configuring Caller Display Information The Caller Display Information page allows you to define a caller identification string (Caller ID) for FXS and FXO ports and enable the device to send the Caller ID information to IP when a call is made.
Page 328: Configuring Call Forward
Mediant 600 & Mediant 1000 18.6.8 Configuring Call Forward The Call Forwarding Table page allows you to forward (redirect) IP-to-Tel calls (using SIP 302 response) originally destined to specific device ports, to other device ports or to an IP destination.
Page 329: Configuring Caller Id Permissions
SIP User's Manual 18. GW and IP to IP Parameter Description Time for No Reply If you have set the 'Forward Type' for this port to 'No Answer', enter the Forward number of seconds the device waits before forwarding the call to the phone number specified.
Page 330: Configuring Call Waiting
Mediant 600 & Mediant 1000 18.6.10 Configuring Call Waiting The Call Waiting page allows you to enable or disable call waiting per device FXS port. Notes: • This page is applicable only to FXS interfaces. • Instead of using this page, you can enable or disable call waiting for all the device's ports, using the global call waiting parameter 'Enable Call Waiting' (see 'Configuring Supplementary Services' on page 291).
Page 331 SIP User's Manual 18. GW and IP to IP configured per FXS endpoint or for a range of FXS endpoints. Therefore, different tones can be played per FXS endpoint/s depending on the source and/or destination number of the received call. In addition, you can configure multiple entries with different source and/or destination prefixes and tones for the same FXS port.
Page 332: Fxs/Fxo Coefficient Types
Mediant 600 & Mediant 1000 18.6.12 FXS/FXO Coefficient Types The FXS Coefficient and FXO Coefficient types used by the device can be one of the following:  US line type of 600 ohm AC impedance and 40 V RMS ringing voltage for REN = 2 ...
Page 333 SIP User's Manual 18. GW and IP to IP number. In other words, the IP caller doesn't dial the PSTN number upon hearing a dial tone. One-stage dialing incorporates the following FXO functionality:  Waiting for Dial Tone: Enables the device to dial the digits to the Tel side only after detecting a dial tone from the PBX line.
Page 334 Mediant 600 & Mediant 1000 18.6.13.1.2 Two-Stage Dialing Two-stage dialing is when the IP caller is required to dial twice. The caller initially dials to the FXO device and only after receiving a dial tone from the PBX (via the FXO device), dials the destination telephone number.
Page 335: Fxo Operations For Tel-To-Ip Calls
SIP User's Manual 18. GW and IP to IP  DID Wink can be used for connection to EIA/TIA-464B DID Loop Start lines  Both FXO (detection) and FXS (generation) are supported 18.6.13.2 FXO Operations for Tel-to-IP Calls The FXO device provides the following FXO operating modes for Tel-to-IP calls: ...
Page 336 Mediant 600 & Mediant 1000 18.6.13.2.2 Collecting Digits Mode When automatic dialing is not defined, the device collects the digits. The SIP call flow diagram below illustrates the Collecting Digits Mode. Figure 18-27: Call Flow for Collecting Digits Mode 18.6.13.2.3...
Page 337: Call Termination On Fxo Devices
SIP User's Manual 18. GW and IP to IP 18.6.13.3 Call Termination on FXO Devices This section describes the device's call termination capabilities for its FXO interfaces:  Calls terminated by a PBX (see 'Call Termination by PBX' on page 337) ...
Page 338: Remote Pbx Extension Between Fxo And Fxs Devices
Mediant 600 & Mediant 1000 Note: The implemented disconnect method must be supported by the CO or PBX. 18.6.13.3.2 Call Termination before Call Establishment The device supports the following call termination methods before a call is established:  Call termination upon receipt of SIP error response (in Automatic Dialing mode):...
Page 339: Dialing From Remote Extension (Phone At Fxs)
SIP User's Manual 18. GW and IP to IP FXS device. The routing is transparent as if the telephone connected to the FXS device is directly connected to the PBX. The following is required:  FXO interfaces with ports connected directly to the PBX lines (shown in the figure below) ...
Page 340: Dialing From Pbx Line Or Pstn
Mediant 600 & Mediant 1000 18.6.14.2 Dialing from PBX Line or PSTN The procedure below describes how to dial from a PBX line (i.e., from a telephone directly connected to the PBX) or from the PSTN to the 'remote PBX extension' (i.e., telephone connected to the FXS interface).
Page 341: Call Waiting For Remote Extensions
SIP User's Manual 18. GW and IP to IP 18.6.14.4 Call Waiting for Remote Extensions When the FXO device detects a Call Waiting indication (FSK data of the Caller Id - CallerIDType2) from the PBX, it sends a proprietary INFO message, which includes the caller identification to the FXS device.
Page 342 Mediant 600 & Mediant 1000 In the Outbound IP Routing Table page (see 'Configuring Outbound IP Routing Table' on page 271), enter 20 for the destination phone prefix, and 10.1.10.2 for the IP address of the FXO device. Figure 18-30: FXS Tel-to-IP Routing Configuration...
Page 343: Fxo Gateway Configuration
SIP User's Manual 18. GW and IP to IP 18.6.14.6 FXO Gateway Configuration The procedure below describes how to configure the FXO interface (to which the PBX is directly connected).  To configure the FXO interface: In the Trunk Group Table page (see Configuring Trunk Group Table on page 251, assign the phone numbers 200 to 204 to the device’s FXO endpoints.
Page 344: Dialing Plan Features
Mediant 600 & Mediant 1000 18.7 Dialing Plan Features This section discusses various dialing plan features supported by the device:  Digit mapping (see 'Digit Mapping' on page 344)  External Dial Plan file containing dial plans (see 'External Dial Plan File' on page 345) ...
Page 345: External Dial Plan File
SIP User's Manual 18. GW and IP to IP Below is an example of a digit map pattern containing eight rules: DigitMapping = 11xS|00[1- 7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x|xx.T 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).
Page 346 Mediant 600 & Mediant 1000  The number of additional digits can include a numerical range in the format x-y.  Empty lines and lines beginning with a semicolon (";") are ignored. An example of a Dial Plan file with indices (in ini-file format before conversion to binary .dat) is shown below:...
Page 347: Modifying Isdn-To-Ip Calling Party Number
SIP User's Manual 18. GW and IP to IP 18.7.2.1 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 348: Dial Plan Prefix Tags For Ip-To-Tel Routing
Mediant 600 & Mediant 1000 18.7.3 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 349 SIP User's Manual 18. GW and IP to IP 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): •...
Page 350: Sip Call Routing Examples
Mediant 600 & Mediant 1000 18.8 SIP Call Routing Examples 18.8.1 SIP Call Flow Example The SIP call flow (shown in the following figure), describes SIP messages exchanged between two devices during a basic call. In this call flow example, device (10.8.201.158) with phone number ‘6000’...
Page 351  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 1000/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 352: Sip Message Authentication Example
Mediant 600 & Mediant 1000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15  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 1000/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 353 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 354 Mediant 600 & Mediant 1000 Next, the par called A2 needs to be evaluated: • The method type is ‘REGISTER’. • Using SIP protocol ‘sip’. • Proxy IP from ini file is ‘10.2.2.222’. • The equation to be evaluated is ‘REGISTER:sip:10.2.2.222’.
Page 355: Establishing A Call Between Two Devices
18. GW and IP to IP 18.8.3 Establishing a Call between Two Devices This section provides an example on configuring two AudioCodes' devices with FXS interfaces for establishing call communication. This setup enables the establishment of calls between telephones connected to the same device, and between the two devices.
Page 356: Trunk-To-Trunk Routing Example
Mediant 600 & Mediant 1000 18.8.4 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 357: Sip Trunking Between Enterprise And Itsps
SIP User's Manual 18. GW and IP to IP 18.8.5 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 358 Mediant 600 & Mediant 1000 configure two Proxy Sets and for each, enable Proxy Keep-Alive (using SIP OPTIONS) and 'round robin' load-balancing method: • Proxy Set #1 includes two IP addresses of the first ITSP (ITSP 1) - 10.33.37.77 and 10.33.37.79 - and using UDP.
Page 359 SIP User's Manual 18. GW and IP to IP In the Trunk Group Table page, enable the Trunks connected between the Enterprise's PBX and the device (Trunk Group ID #1), and between the local PSTN and the device (Trunk Group ID #2). Figure 18-39: Assigning Trunks to Trunk Group ID #1 In the Trunk Group Settings page, configure 'Per Account' registration for Trunk Group ID #1 (without serving IP Group)
Page 360: Ip-To-Ip Routing Application
Mediant 600 & Mediant 1000 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 361: 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 362: Proxy Sets
Mediant 600 & Mediant 1000 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 363: Inbound And Outbound Ip Routing Rules
SIP User's Manual 18. GW and IP to IP The device also supports the IP-to-IP call routing Survivability mode feature (refer to the figure below) for USER IP Groups. The device records (in its database) REGISTER messages sent by the clients of the USER IP Group. If communication with the Serving IP Group (e.g., IP-PBX) fails, the USER IP Group enters into Survivability mode in which the device uses its database for routing calls between the clients of the USER IP Group.
Page 364: Accounts
Mediant 600 & Mediant 1000 18.9.1.4 Accounts Accounts are used by the device to register to a Serving IP Group (e.g., an ITSP) on behalf of a Served IP Group (e.g., IP-PBX). This is necessary for ITSP's that require registration to provide services.
Page 365: Ip-To-Ip Routing Configuration Example
SIP User's Manual 18. GW and IP to IP 18.9.2 IP-to-IP Routing Configuration Example This section provides step-by-step procedures for configuring IP-to-IP call routing. These procedures are based on the setup example described below. In this example, the device serves as the communication interface between the enterprise's IP-PBX (located on the LAN) and the following network entities: ...
Page 366 Mediant 600 & Mediant 1000 The figure below provides an illustration of this example scenario: Figure 18-48: 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 367).
Page 367: Step 1: Enable The Ip-To-Ip Capabilities
SIP User's Manual 18. GW and IP to IP 18.9.2.1 Step 1: Enable the IP-to-IP Capabilities This step describes how to enable the device's IP-to-IP application.  To enable IP-to-IP capabilities: Open the Applications Enabling page (Configuration tab > VoIP menu > Applications Enabling submenu >...
Page 368: Step 3: Define A Trunk Group For The Local Pstn
Mediant 600 & Mediant 1000 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 369 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 18-51: 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 370: Step 5: Configure The Ip Groups
Mediant 600 & Mediant 1000 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 371 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 18-54: Defining IP Group 1 Define IP Group #2 for ITSP-B: From the 'Type' drop-down list, select SERVER.
Page 372 Mediant 600 & Mediant 1000 Contact User = name that is sent in the SIP Request Contact header for this IP Group (e.g., ITSP-B). Figure 18-55: Defining IP Group 2 Define IP Group #3 for the IP-PBX: From the 'Type' drop-down list, select SERVER.
Page 373: Step 6: Configure The Account Table
SIP User's Manual 18. GW and IP to IP Define IP Group #4 for the remote IP-PBX users: From the 'Type' drop-down list, select USER. In the 'Description' field, type an arbitrary name for the IP Group (e.g., IP-PBX). In the 'SIP Group Name' field, enter the host name that is used internal in the device's database for this IP Group (e.g., RemoteIPPBXusers).
Page 374: Step 7: Configure Ip Profiles For Voice Coders
Mediant 600 & Mediant 1000  To configure the Account table: Open the Account Table page (Configuration tab > VoIP menu > SIP Definitions submenu > Account Table). Figure 18-58: Defining Accounts for Registration Configure Account ID #1 for IP-PBX authentication and registration with ITSP-A: •...
Page 375 SIP User's Manual 18. GW and IP to IP Click Submit. Figure 18-59: Defining Coder Group ID 1 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.
Page 376: Step 8: Configure Inbound Ip Routing
Mediant 600 & Mediant 1000 Click Submit. Figure 18-61: Defining IP Profile ID 1 Configure Profile ID #2 for the ITSP's: From the 'Profile ID' drop-down list, select 2. From the 'Coder Group' drop-down list, select Coder Group 2. Click Submit.
Page 377: Step 9: Configure Outbound Ip Routing
SIP User's Manual 18. GW and IP to IP • 'Dest Phone Prefix': enter "9" for the dialing prefix for local calls. • 'Trunk Group ID': enter "1" to indicate that these calls are routed to the Trunk (belonging to Trunk Group #1) connected between the device and the local PSTN network.
Page 378 Mediant 600 & Mediant 1000 destined to the Japanese market, then they are routed to ITSP-B; for all other destinations, the calls are routed to ITSP-A. This configuration uses the IP Groups defined in 'Step 5: Configure the IP Groups' on page and IP Profiles defined in 'Step 7: Configure IP Profiles for Voice Coders' on page 374.
Page 379 SIP User's Manual 18. GW and IP to IP • 'Dest Phone Prefix' and 'Source Phone Prefix': enter the asterisk (*) symbol to indicate all destinations (besides Japan) and all sources respectively. • 'Dest IP Group ID': select 2 to indicate the destination IP Group to where the calls must be sent, i.e., to ITSP-A.
Page 380: Step 10: Configure Destination Phone Number Manipulation
Mediant 600 & Mediant 1000 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 381: Stand-Alone Survivability (Sas) Application
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application Stand-Alone Survivability (SAS) Application This section describes the Sand-Alone Survivability (SAS) application. 19.1 Overview The device's Stand-Alone Survivability (SAS) feature ensures telephony communication continuity (survivability) for enterprises using hosted IP services (such as IP Centrex) or IP- PBX in cases of failure of these entities.
Page 382: Sas Outbound Mode
Mediant 600 & Mediant 1000 19.1.1.1 SAS Outbound Mode This section describes the SAS outbound mode, which includes the following states:  Normal state (see 'Normal State' on page 382)  Emergency state (see 'Emergency State' on page 382) 19.1.1.1.1 Normal State In normal state, SAS receives REGISTER requests from the enterprise's UAs and forwards them to the external proxy (i.e., outbound proxy).
Page 383: Sas Redundant Mode
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application The figure below illustrates the operation of SAS outbound mode in emergency state: Figure 19-2: SAS Outbound Mode in Emergency State (Example) When emergency state is active, SAS continuously attempts to communicate with the external proxy, using keep-alive SIP OPTIONS.
Page 384 Mediant 600 & Mediant 1000 19.1.1.2.1 Normal State In normal state, the UAs register and operate directly with the external proxy. Figure 19-3: SAS Redundant Mode in Normal State (Example) 19.1.1.2.2 Emergency State If the UAs detect that their primary (external) proxy does not respond, they immediately register to SAS and start routing calls to it.
Page 385: Sas Routing
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.1.1.2.3 Exiting Emergency and Returning to Normal State Once the connection with the primary proxy is re-established, the following occurs:  UAs: switch back to operate with the primary proxy.  SAS: ignores REGISTER requests from the UAs, forcing the UAs to switch back to the primary proxy.
Page 386 Mediant 600 & Mediant 1000 The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the external proxy: Figure 19-6: Flowchart of INVITE from Primary Proxy in SAS Normal State SIP User's Manual...
Page 387: Sas Routing In Emergency State
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.1.2.2 SAS Routing in Emergency State The flowchart below shows the routing logic for SAS in emergency state: Figure 19-7: Flowchart for SAS Emergency State Version 6.4 March 2012...
Page 388: Sas Configuration
Mediant 600 & Mediant 1000 19.2 SAS Configuration SAS supports various configuration possibilities, depending on how the device is deployed in the network and the network architecture requirements. This section provides step-by- step procedures on configuring the SAS application, using the device's Web interface.
Page 389 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application  To configure common SAS settings: Open the SAS Configuration page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability). Define the port used for sending and receiving SAS messages. This can be any of the following port types: •...
Page 390 Mediant 600 & Mediant 1000 Figure 19-9: Configuring Common Settings In the 'SAS Proxy Set' field, enter the Proxy Set used for SAS. The SAS Proxy Set must be defined only for the following SAS modes: • Outbound mode: In SAS normal state, SAS forwards REGISTER and INVITE messages received from the UAs to the proxy servers defined in this Proxy Set.
Page 391: Configuring Sas Outbound Mode
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application From the 'Enable Proxy Keep Alive' drop-down list, select Using Options. This instructs the device to send SIP OPTIONS messages to the proxy for the keep- alive mechanism. Figure 19-10: Defining UAs' Proxy Server Click Submit to apply your settings.
Page 392: Configuring Sas Redundant Mode
Mediant 600 & Mediant 1000 19.2.3 Configuring SAS Redundant Mode This section describes how to configure the SAS redundant mode. These settings are in addition to the ones described in 'Configuring Common SAS Parameters' on page 388. Note: The VoIP CPEs (such as IP phones or residential gateways) need to be...
Page 393: Gateway With Sas Outbound Mode
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.2.4.1 Gateway with SAS Outbound Mode The procedure below describes how to configure the Gateway application with SAS outbound mode.  To configure Gateway application with SAS outbound mode: Define the proxy server address for the Gateway application: Open the Proxy &...
Page 394: Gateway With Sas Redundant Mode
Mediant 600 & Mediant 1000 Disable use of user=phone in SIP URL: Open the SIP General Parameters page (Configuration tab > VoIP menu > SIP Definitions submenu > General Parameters). From the 'Use user=phone in SIP URL' drop-down list, select No. This instructs the Gateway application not to use user=phone in the SIP URL and therefore, REGISTER and INVITE messages use SIP URI.
Page 395 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application In the second 'Proxy Address' field, enter the IP address and port of the device (in the format x.x.x.x:port). This is the same port as defined in the 'SAS Local UDP/TCP/TLS Port' field (see 'Configuring Common SAS Parameters' on page 388).
Page 396: Advanced Sas Configuration
Mediant 600 & Mediant 1000 19.2.5 Advanced SAS Configuration This section describes the configuration of advanced SAS features that can be optionally implemented in your SAS deployment:  Manipulating incoming SAS Request-URI user part of REGISTER message (see 'Manipulating URI user part of Incoming REGISTER' on page 396) ...
Page 397: Manipulating Destination Number Of Incoming Invite
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application After manipulation, SAS registers the user in its database as follows:  AOR: 976653434@10.33.4.226  Associated AOR: 3434@10.33.4.226 (after manipulation, in which only the four digits from the right of the URI user part are retained) ...
Page 398 Mediant 600 & Mediant 1000 In normal state, the numbers are not manipulated. In this state, SAS searches the number 552155551234 in its database and if found, it sends the INVITE containing this number to the UA.  To manipulate destination number in SAS emergency state: Open the SAS Configuration page (Configuration tab >...
Page 399 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application Parameter Description URI (either source or destination) to the rule configured in the row above (defined by the parameter ManipulatedURI). Manipulation Purpose Defines the purpose of the manipulation: [ManipulationPurpose]  [0] Normal = Inbound manipulations affect the routing input and source and/or destination number (default).
Page 400: Sas Routing Based On Sas Routing Table
Mediant 600 & Mediant 1000 Parameter Description Remove From Right Defines the number of digits to remove from the right of the user name [RemoveFromRight] prefix. For example, if you enter 3 and the user name is "john", the new user name is "j".
Page 401 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application  To configure the IP2IP Routing table for SAS: In the SAS Configuration page, click the SAS Routing Table button; the IP2IP Routing Table page appears. Figure 19-18: IP2IP Routing Page Add an entry and then configure it according to the table below. Click the Apply button to save your changes.
Page 402 Mediant 600 & Mediant 1000 Parameter Description Manipulation'' on page 787. Destination Host Defines the host part of the incoming SIP dialog’s destination URI [IP2IPRouting_DestHost] (usually the Request-URI). If this rule is not required, leave the field empty. The asterisk (*) symbol can be used to denote any destination host.
Page 403 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application Parameter Description Type' is set to 'IP Group'. However, regardless of the settings of the parameter 'Destination Type', the IP Group is still used - only for determining the IP Profile or outgoing SRD. If neither IP Group nor SRD are defined in this table, the destination SRD is determined according to the source SRD associated with the Source IP Group (configured in the IP Group table, see...
Page 404: Blocking Calls From Unregistered Sas Users
Mediant 600 & Mediant 1000 Parameter Description input characteristics. Notes:  The alternative routing entry ([1] or [2]) must be defined in the next consecutive table entry index to the Route Row entry (i.e., directly below it). For example, if Index 4 is configured as a Route Row, Index 5 must be configured as the alternative route.
Page 405 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application Note: The port of the device is defined in the 'SIP UDP/TCP/TLS Local Port' field in the SIP General Parameters page (Configuration tab > VoIP menu > SIP Definitions > General Parameters). In the 'SAS Emergency Numbers' field, enter an emergency number in each field box.
Page 406: Adding Sip Record-Route Header To Sip Invite
Mediant 600 & Mediant 1000 19.2.5.6 Adding SIP Record-Route Header to SIP INVITE You can configure SAS to add the SIP Record-Route header to SIP requests (e.g. INVITE) received from enterprise UAs. SAS then sends the request with this header to the proxy.
Page 407: Viewing Registered Sas Users
SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 19.3 Viewing Registered SAS Users You can view all the users that are registered in the SAS registration database. This is displayed in the 'SAS Registered Users page, as described in 'Viewing SAS Registered Users' on page 520.
Page 408 Mediant 600 & Mediant 1000 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).
Page 409 SIP User's Manual 19. Stand-Alone Survivability (SAS) Application 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 410 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 411: Configuring The Ip Media Parameters
'Configuration Parameters Reference' on page 541. Note: This page is applicable only to Mediant 1000. This page appears only if your device is installed with the relevant Software Upgrade Key (see 'Loading Software Upgrade Key' on page 497).
Page 412: Conference Server
Mediant 600 & Mediant 1000 The device conference, transcoding, announcement and media server applications can be used separately, each on a different platform, or all on the same device. The SIP URI name in the INVITE message is used to identify the resource (media server, conference or announcement) to which the SIP session is addressed.
Page 413: Simple Conferencing (Netann)
SIP User's Manual 20. Configuring the IP Media Parameters 20.1.1.1 Simple Conferencing (NetAnn) 20.1.1.1.1 SIP Call Flow A SIP call flow for simple conferencing is shown below: 20.1.1.1.2 Creating a Conference The device creates a conference call when the first user joins the conference. To create a conference, the Application Server sends a regular SIP INVITE message to the device.
Page 414 Mediant 600 & Mediant 1000 (indicating that the requested Media Service is a Conference) and a Unique Conference Identifier (identifying a specific instance of a conference). INVITE sip: conf100@audiocodes.com SIP/2.0 By default, a request to create a conference reserves three resources on the device. It is possible to reserve a larger number of resources in advance by adding the number of required participants to the User Part of the Request-URI.
Page 415: Advanced Conferencing (Mscml)
SIP User's Manual 20. Configuring the IP Media Parameters 20.1.1.1.5 PSTN Participants Adding PSTN participants is done by performing a loopback from the IP side (the device's IP address is configured in the Outbound IP Routing Table). If the destination phone number in the incoming call from the PSTN is equal to the Conference Service Identifier and Unique Conference Identifier, the participant joins the conference.
Page 416 Mediant 600 & Mediant 1000 20.1.1.2.2 Joining a Conference To join an existing conference, the Application Server sends a SIP INVITE message with the same Request-URI as the one that created the conference. The INVITE message may include a <configure_leg> MSCML request body. If not included, defaults are used for that leg attributes.
Page 417 SIP User's Manual 20. Configuring the IP Media Parameters Table 20-1: MSCML Conferencing with Personalized Mixes Participant Team Members Mixmode Hears Supervisor “supervisor” Agent Private Customer and Agent Agent “agent” Supervisor Full Customer and Supervisor Customer “customer” Full Agent This scenario is established as follows: Conference is created on the control leg with <configure_conference>.
Page 418 Mediant 600 & Mediant 1000 20.1.1.2.4 Applying Media Services on a Conference The Application Server can issue a Media Service request (<play>, <playcollect>, or <playrecord>) on either the Control Leg or a specific Participant Leg. For a Participant Leg, all three requests are applicable. For the Control Leg, the <playcollect> is not applicable as there is no way to collect digits from the whole conference.
Page 419 SIP User's Manual 20. Configuring the IP Media Parameters 20.1.1.2.5 Active Speaker Notification After an advanced conference is established, the Application Server can subscribe to the device to receive notifications of the current set of active speakers in a conference at any given moment.
Page 420: Conference Call Flow Example
Mediant 600 & Mediant 1000 20.1.1.3 Conference Call Flow Example The call flow, shown in the following figure, describes SIP messages exchanged between the device (10.8.58.4) and three conference participants (10.8.29.1, 10.8.58.6 and 10.8.58.8). SIP MESSAGE 1: 10.8.29.1:5060 -> 10.8.58.4:5060 INVITE sip:conf100@10.8.58.4;user=phone SIP/2.0...
Page 421 Via: SIP/2.0/UDP 10.8.29.1;branch=z9hG4bKacRHmJhMj From: <sip:100@10.8.8.10>;tag=1c352329022 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c222574568 Call-ID: 1792526528qlax@10.8.29.1 CSeq: 1 INVITE Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40A.010.006 Content-Length: 0 SIP MESSAGE 3: 10.8.58.4:5060 -> 10.8.29.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.29.1;branch=z9hG4bKacRHmJhMj From: <sip:100@10.8.8.10>;tag=1c352329022 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c222574568 Call-ID: 1792526528qlax@10.8.29.1 CSeq: 1 INVITE Contact: <sip:10.8.58.4>...
Page 422 Mediant 600 & Mediant 1000 SIP MESSAGE 4: 10.8.29.1:5060 -> 10.8.58.4:5060 ACK sip:10.8.58.4 SIP/2.0 Via: SIP/2.0/UDP 10.8.29.1;branch=z9hG4bKacbUrWtRo Max-Forwards: 70 From: <sip:100@10.8.8.10>;tag=1c352329022 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c222574568 Call-ID: 1792526528qlax@10.8.29.1 CSeq: 1 ACK Contact: <sip:100@10.8.29.1> Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40A.010.006 Content-Length: 0 SIP MESSAGE 5: 10.8.58.6:5060 -> 10.8.58.4:5060 INVITE sip:conf100@10.8.58.4;user=phone SIP/2.0...
Page 423 SIP User's Manual 20. Configuring the IP Media Parameters Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40A.010.006 Content-Length: 0 SIP MESSAGE 7: 10.8.58.4:5060 -> 10.8.58.6:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.58.6;branch=z9hG4bKacfowEuut From: <sip:600@10.8.8.10>;tag=1c201038291 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c1673415884 Call-ID: 1008914574iYgW@10.8.58.6 CSeq: 1 INVITE Contact: <sip:conf100@10.8.58.4> Supported: em,timer,replaces,path...
Page 424 Mediant 600 & Mediant 1000 Contact: <sip:800@10.8.58.8> Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.4.60A.005.009 Content-Type: application/sdp Content-Length: 236 o=AudiocodesGW 558246 666026 IN IP4 10.8.58.8 s=Phone-Call c=IN IP4 10.8.58.8 t=0 0 m=audio 6000 RTP/AVP 4 96 a=rtpmap:4 g723/8000 a=fmtp:4 annexa=no a=rtpmap:96 telephone-event/8000...
Page 425 From: <sip:800@10.8.58.8>;tag=1c2419012378 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c3203015250 Call-ID: 150852731NDDC@10.8.58.8 CSeq: 1 ACK Contact: <sip:800@10.8.58.8> Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40A.010.006 Content-Length: 0 SIP MESSAGE 13: 10.8.58.8:5060 -> 10.8.58.4:5060 BYE sip:conf100@10.8.58.4 SIP/2.0 Via: SIP/2.0/UDP 10.8.58.8;branch=z9hG4bKackSIyGww Max-Forwards: 70 From: <sip:800@10.8.58.8>;tag=1c2419012378 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c3203015250 Call-ID: 150852731NDDC@10.8.58.8 CSeq: 2 BYE Contact: <sip:800@10.8.58.8>...
Page 426: Announcement Server
Mediant 600 & Mediant 1000 CSeq: 2 BYE Contact: <sip:600@10.8.58.6> Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40A.010.006 Content-Length: 0 SIP MESSAGE 16: 10.8.58.4:5060 -> 10.8.58.6:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.58.6;branch=z9hG4bKacQypxnvl From: <sip:600@10.8.8.10>;tag=1c201038291 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c1673415884 Call-ID: 1008914574iYgW@10.8.58.6 CSeq: 2 BYE Contact: <sip:conf100@10.8.58.4>...
Page 427: Mscml Interface
SIP User's Manual 20. Configuring the IP Media Parameters 20.1.2.1.2 Playing using HTTP/NFS Streaming To play a single announcement via HTTP or NFS streaming, the Application Server (or any SIP user agent) sends a regular SIP INVITE message with SIP URI that includes the NetAnn Announcement Identifier name.
Page 428 Mediant 600 & Mediant 1000 The following figure illustrates standard MSCML application architecture: The architecture comprises the following components:  device: Operating independently, the device controls and allocates its processing resources to match each application’s requirements. Its primary role is to handle requests from the Application server for playing announcements and collecting digits.
Page 429 SIP INVITE message with a SIP URI that includes the MSCML Identifier name. For example: INVITE sip:ivr@audiocodes.com SIP/2.0 The left part of the SIP URI includes the MSCML Identifier string ‘ivr’, which can be configured using the ini file (parameter MSCMLID) or Web interface (see 'Configuring the IPmedia Parameters' on page 411).
Page 430 Mediant 600 & Mediant 1000 Server User’s Manual (LTRT-971xx).  baseurl: defines a URL address that functions as a prefix to all audio segment URLs in the Prompt block. The Prompt block contains references to one or more audio segments. The following audio segment types are available: ...
Page 431 SIP User's Manual 20. Configuring the IP Media Parameters You can upload a bundle to the device using one of the following methods:  Loading an ini file as described above, and then resetting the device (hard reset). Optionally, you can configure parameters through Web interface or using SNMP, and then burn parameters to flash and reset the device through Web or SNMP (soft reset).
Page 432 Mediant 600 & Mediant 1000 20.1.2.2.4 Playing Announcements and Collecting Digits The <PlayCollect> request is used to play an announcement to the caller and to then collect entered DTMF digits. The play part of the <PlayCollect> request is identical to the <Play>...
Page 433 SIP User's Manual 20. Configuring the IP Media Parameters An example is shown below of an MSCML <PlayCollect> Request that includes a sequence with variables and an MGCP digit map: <?xml version="1.0" encoding="utf-8"?> <MediaServerControl version="1.0"> <request> <playcollect id="6379" barge="NO" returnkey="#"> <prompt>...
Page 434 Mediant 600 & Mediant 1000  endsilence: defines how long the device waits after speech has ended to stop the recording. This parameter may take an integer value in milliseconds.  duration: the total time in milliseconds for the entire recording. Once this time expires, recording stops and a response is generated.
Page 435 SIP User's Manual 20. Configuring the IP Media Parameters 20.1.2.2.7 Relevant Parameters The following parameters (described in 'IP Media Parameters' on page 770) are used to configure the MSCML:  AmsProfile = 1 (mandatory)  AASPackagesProfile = 3 (mandatory)  VoiceStreamUploadMethod = 1 (mandatory) ...
Page 436: Voice Streaming
Mediant 600 & Mediant 1000 Below is an example: <?xml version="1.0"?> <MediaServerControl version="1.0"> <request> <configure_leg> <subscribe> <events> <signal type="amd" report="yes"/> </events> </subscribe> </configure_leg> </request> </MediaServerControl> <?xml version="1.0"?> <MediaServerControl version="1.0"> <notification> <signal type="amd" subtype="voice"/> </notification> </MediaServerControl> 20.1.2.3 Voice Streaming The voice streaming layer provides you with the ability to play and record different types of files while using an NFS or HTTP server.
Page 437 SIP User's Manual 20. Configuring the IP Media Parameters 20.1.2.3.1.5 Using Proprietary Scripts You may use cgi or servlet scripts released with the version for recording to a remote HTTP server using the POST or PUT method. 20.1.2.3.1.6 Dynamic HTTP URLs Voice streaming supports dynamic HTTP URLs.
Page 438 Mediant 600 & Mediant 1000 20.1.2.3.1.8 Basic Record You may record a .wav, .au or .raw files to a remote server using G.711 coders. Note: This feature is relevant for both NFS and HTTP. 20.1.2.3.1.9 Remove DTMF Digits at End of Recording You may configure a recording to remove the DTMF received at the end, indicating an end of a recording.
Page 439 SIP User's Manual 20. Configuring the IP Media Parameters respond. By default, the value is set to 0 and not used - instead, the number of retransmissions is derived from the server response timeout parameter and the current Recovery Time Objective (RTO) of the system. These parameters may be configured using the ini file, Web interface, or SNMP.
Page 440 Mediant 600 & Mediant 1000 20.1.2.3.2.2 Recording a File The table below lists the device's support of channel coders and file coders for recording a file. Table 20-4: Coder Combinations - Recording a File File File Type Coder Channel Coder...
Page 441 SIP User's Manual 20. Configuring the IP Media Parameters Coder .wav file .raw file .au file AMR (Rate 5.15) AMR (Rate 5.9) AMR (Rate 6.7) AMR (Rate 7.4) AMR (Rate 7.95) AMR (Rate 10.2) AMR (Rate 12.2) QCELP (Rate 8) QCELP (Rate 13) 20.1.2.3.5 HTTP Recording Configuration The HTTP record method (PUT or POST) is configured using the following offline ini...
Page 442 • For further details, see 'Configuring the NFS Settings' on page 127. 20.1.2.3.7 Supported HTTP Servers The following is a list of HTTP servers that are known to be compatible with AudioCodes voice streaming under Linux™:  Apache: cgi scripts are used for recording and supporting dynamic URLs.
Page 443 MaxSpareThreads ThreadsPerChild MaxRequestsPerChild 16384 </IfModule> 20.1.2.3.8 Supporting NFS Servers The table below lists the NFS servers that are known to be compatible with AudioCodes Voice Streaming functionality. Table 20-6: Compatible NFS Servers Operating System Server Versions Solaris™ 5.8 and 5.9...
Page 444 If the systems administrator wishes to use a default other than AUTH_SYS in the nfssec.conf file, then you should add "sec=sys" to each line in the dfstab file that is to be shared with an AudioCodes system. For example: > cat /etc/dfs/dfstab...
Page 445 20.1.2.3.8.2 Linux-Based NFS Servers The AudioCodes device uses local UDP ports that are outside of the range of 0..IPPORT_RESERVED(1024). Therefore, when configuring a remote file system to be accessed by an AudioCodes device, use the insecure option in the /etc/exports file. The insecure option allows the nfs daemon to accept mount requests from ports outside of this range.
Page 446 Mediant 600 & Mediant 1000 20.1.2.3.9 Common Troubleshooting Always inspect the Syslog for any problem you may encounter; in many cases, the cause appears there. Table 20-7: Troubleshooting Problem Probable Cause Corrective Action General Voice Streaming Problems Attempts to perform voice streaming...
Page 447: Announcement Call Flow Example
The call flow, shown in the following figure, describes SIP messages exchanged between the device (10.33.24.1) and a SIP client (10.33.2.40) requesting to play local announcement #1 (10.8.25.17) using AudioCodes proprietary method. SIP MESSAGE 1: 10.33.2.40:5060 -> 10.33.24.1:5060 Version 6.4...
Page 448 Mediant 600 & Mediant 1000 INVITE sip:annc@10.33.24.1;play=http://10.3.0.2/hello.wav;repeat=2 SIP/2.0 Via: SIP/2.0/UDP 10.33.2.40;branch=z9hG4bKactXhKPQT Max-Forwards: 70 From: <sip:103@10.33.2.40>;tag=1c2917829348 To: <sip:annc@10.33.24.1> Call-ID: 1414622340oZZq@10.33.2.40 CSeq: 1 INVITE Contact: <sip:103@10.33.2.40> Supported: em,100rel,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-4.0 GA/v.4.0 GA Content-Type: application/sdp Content-Length: 215 o=AudiocodesGW 377662 728960 IN IP4 10.33.41.52 s=Phone-Call c=IN IP4 10.33.41.52...
Page 449 From: <sip:103@10.33.2.40>;tag=1c2917829348 To: <sip:annc@10.33.24.1>;tag=1c1528117157 Call-ID: 1414622340oZZq@10.33.2.40 CSeq: 1 ACK Contact: <sip:103@10.33.2.40> Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-4.0 GA/v.4.0 GA Content-Length: 0 SIP MESSAGE 5: 10.33.24.1:5060 -> 10.33.2.40:5060 BYE sip:103@10.33.2.40 SIP/2.0 Via: SIP/2.0/UDP 10.33.24.1;branch=z9hG4bKacFhtFbFR Max-Forwards: 70 From: <sip:annc@10.33.24.1>;tag=1c1528117157 To: <sip:103@10.33.2.40>;tag=1c2917829348 Call-ID: 1414622340oZZq@10.33.2.40 CSeq: 1 BYE Contact: <sip:10.33.24.1>...
Page 450: Voice Xml Interpreter
(i.e., text-to-speech or TTS). Its major goal is to bring the advantages of Web-based development and content delivery to interactive voice response applications. Notes: • VoiceXML is applicable only to Mediant 1000. • Currently, ASR and TTS are not supported. 20.1.3.1 Features VoiceXML offers the following features: ...
Page 451: Proprietary Extensions
To provide the functionality intended by the VXML specification and to extend the functionality of the VXML specification, some proprietary extensions have been included in the AudioCodes VXML Interpreter. These extensions are discussed in the following sections and are intended to enable a VXML script to make use of the advanced audio capabilities provided by the device.
Page 452 Mediant 600 & Mediant 1000 20.1.3.4.1 Record As the device doesn't provide the ability to record ‘on-board’, it is necessary to record a caller’s speech by streaming the audio to either an external NFS server. There are two additional attributes for the VXML <record> element that can be used to specify the off- board file name as well as the streaming mechanism for recording speech.
Page 453 This reference directs the VXML software to play the audio segment marked with identifier '123'. Using this method of access, the advanced audio structures defined by the AudioCodes Audio Provisioning Server (APS) can be referenced. While these various structures are outside the scope of the current document, they include sets, sequences, and multi- language variables.
Page 454 Mediant 600 & Mediant 1000 Say-as Token Variable Variable Subtype Variable Input Note Type Format the grammar rules of the language. duration duration None supported The input is up Duration is always to 10 digits, with announced as hours, the value minutes, and seconds.
Page 455 VXML doesn't define any capability for passing a value to a variable, therefore, the AudioCodes VXML Interpreter provides an extension to support this capability.
Page 456 User’s Guide. 20.1.3.4.3 Language Identifier Support The AudioCodes resident VXML engine supports language identifiers as specified by RFC 3066. However, when accessing audio resident on the device using the proprietary extensions described earlier, the country code portion of the identifier is ignored.
Page 457: Combining
The VXML specification supports multiple <audio> elements nested within other elements such as prompts. An example demonstrating this functionality which includes the AudioCodes extensions is useful to show how multiple components can be combined to create a single announcement. The following example shows how an announcement can be constructed that says “Welcome to Acme Corporation.
Page 458 Mediant 600 & Mediant 1000 20.1.3.7.1 VoiceXML Supported Elements and Attributes Table 20-10: VoiceXML Supported Elements and Attributes Element Parameter Max Size Shadow Variable Status Comments <assign> name expr <audio> The AudioCodes audio element has proprietary extensions in addition to attributes from the standard to support on-board audio variables.
Page 459 SIP User's Manual 20. Configuring the IP Media Parameters Element Parameter Max Size Shadow Variable Status Comments messageexpr fetchaudio fetchtimeout fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <clear> namelist 4 * 32 <disconnect> <else> <elseif> cond <enumerate>...
Page 460 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments name$.interpretation numeric name$.confidence <filled> mode namelist 4 * 32 <form> scope enum <goto> next expr nextitem expritem fetchaudio fetchtimeout fetchhint maxage maxstale <grammar> version For voice grammars, this is passed to the speech recognition engine.
Page 461 SIP User's Manual 20. Configuring the IP Media Parameters Element Parameter Max Size Shadow Variable Status Comments type in text-to-speech. weight numeric For voice grammars, this is passed to the speech recognition engine. fetchtimeout Voice grammars are maintained on the speech recognition server, not on device, thus this set of attributes that control caching of grammar doesn't apply.
Page 462 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments fetchtimeout fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <log> label expr <menu> scope enum dtmf true/false accept It's not obvious how to instruct the speech recognition engine that approximate matches are acceptable.
Page 463 SIP User's Manual 20. Configuring the IP Media Parameters Element Parameter Max Size Shadow Variable Status Comments fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <option> dtmf accept It's not obvious how to instruct the speech recognition engine that approximate matches are acceptable.
Page 464 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments beep true/false Requires that a user-defined tone be added to the system. For an example, Example of UDT ‘beep’ see ' Tone Definition' on page 472. Refer to the Auxilary Files section for additional details regarding creating user-defined tones.
Page 465 SIP User's Manual 20. Configuring the IP Media Parameters Element Parameter Max Size Shadow Variable Status Comments maxstale <subdialog> Playing a prompt from a sub-dialog element is not supported in this release. name expr cond namelist 4 * 32 srcexpr method enum enctype...
Page 466 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments message messageexpr <transfer> Name Expr Cond Dest Only numbers. destexpr Bridge Only blind transfer supported (false). type Only blind transfer supported (blind). connecttimeout maxtime transferaudio Aaiexpr name$.duration name$.inputmode...
Page 467 SIP User's Manual 20. Configuring the IP Media Parameters Element Parameter Max Size Shadow Variable Status Comments connecttimeout maxtime transferaudio Aaiexpr name$.duration name$.inputmode name$.utterance <value> expr <var> name expr <vxml> 20.1.3.7.2 SRGS and SSML Support Note that elements associated with either the Speech Recognition Grammar Specification (SRGS) or Speech Synthesis Markup Language (SSML) are used to control the behavior of a remote speech engine for either speech recognition or text-to-speech.
Page 468 Mediant 600 & Mediant 1000 Platform Properties Status Equivalent ini file parameter or Notes DTMF Recognizer Interdigittimeout VxmlInterDigitTimeout Termtimeout VxmlTermTimeout. Note that the system default is not 0 as directed in the specification for the protocol, but 3 seconds. This is to ensure digit collection functions correctly.
Page 469 SIP User's Manual 20. Configuring the IP Media Parameters Platform Properties Status Equivalent ini file parameter or Notes Universals Universal grammars and behaviors such as help, cancel, and exit. Default is none. Maxnbest Size of last result array 20.1.3.7.4 VoiceXML Variables and Events Table 20-12: VoiceXML Variables and Events Variable/Event Name Status...
Page 470 20.1.3.7.5 ECMAScript Support The following table describes the ECMAScript support that the AudioCodes resident VXML engine provides. As shown in the example below, all operands and operators in an expression must be separated by one or more ECMAScript whitespace characters.
Page 471 SIP User's Manual 20. Configuring the IP Media Parameters Table 20-13: ECMAScript Support Operand/Operator Examples Status Note Whitespace chars tab, vertical tab, form feed, and space Arithmetic Operators +, ++, -, --, *, /, % Logical Operators &&, ||, ! Assignment Operators =, +=, -=, *=, /=, %=, &=, ^=, |=, <<=, >>=,...
Page 472: Example Of Udt 'Beep' Tone Definition
Mediant 600 & Mediant 1000 20.1.3.8 Example of UDT ‘beep’ Tone Definition The following is an example definition for ‘beep’ tone used for the <record> element: #record beep tone [CALL PROGRESS TONE #1] Tone Type=202 Low Freq [Hz]=430 High Freq [Hz]=0...
Page 473: Transcoding Using Third-Party Call Control
SIP User's Manual 21. Transcoding using Third-Party Call Control Transcoding using Third-Party Call Control The device supports transcoding using a third-party call control Application server. This support is provided by the following:  Using RFC 4117 (see 'Using RFC 4117' on page 473) ...
Page 474: Using Rfc 4240 - Netann 2-Party Conferencing
Mediant 600 & Mediant 1000 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 475 SIP User's Manual 21. Transcoding using Third-Party Call Control The figure below illustrates an example of implementing an Application server: Version 6.4 March 2012...
Page 476 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 477: Maintenance
Part V Maintenance This part describes the maintenance procedures.
Page 478 Reader’s Notes...
Page 479: 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 480 Mediant 600 & Mediant 1000  To reset the device: Open the Maintenance Actions page (see 'Basic Maintenance' on page 477). Under the 'Reset Configuration' group, from the 'Burn To FLASH' drop-down list, select one of the following options: •...
Page 481: 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 482: Saving Configuration
Mediant 600 & Mediant 1000 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 483: Software Upgrade
Voice Prompts by the device during operation. For more information on VP files, see Voice Prompts File on page 491. Note: This file is applicable only to Mediant 1000. Call Progress This is a region-specific, telephone exchange-dependent file that contains the Tones Call Progress Tones (CPT) levels and frequencies for the device.
Page 484: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 File Description User Info The User Information file maps PBX extensions to IP numbers. This file can be used to represent PBX extensions as IP phones in the global 'IP world'. For more information on the User Info file, see 'User Information File' on page 494.
Page 485 SIP User's Manual 23. 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 486: Call Progress Tones File
Mediant 600 & Mediant 1000 You can also load auxiliary files using an ini file that is loaded to the device with BootP. Each auxiliary file has a specific ini file parameter that specifies the name of the auxiliary file that you want to load to the device with the ini file. For a description of these ini file parameters, see Auxiliary and Configuration Files Parameters on page 781.
Page 487 SIP User's Manual 23. 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 488 Mediant 600 & Mediant 1000 • Third Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the third cadence on-off cycle. Can be omitted if there isn't a third cadence. • Fourth Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the fourth cadence on-off cycle.
Page 489: Distinctive Ringing
SIP User's Manual 23. Software Upgrade 23.1.1.1 Distinctive Ringing Distinctive Ringing is applicable only to FXS interfaces. Using the Distinctive Ringing section of the Call Progress Tones auxiliary file, you can create up to 16 Distinctive Ringing patterns. Each ringing pattern configures the ringing tone frequency and up to four ringing cadences.
Page 490 Mediant 600 & Mediant 1000 An example of a ringing burst definition is shown below: #Three ringing bursts followed by repeated ringing of 1 sec on and 3 sec off. [NUMBER OF DISTINCTIVE RINGING PATTERNS] Number of Ringing Patterns=1 [Ringing Pattern #0]...
Page 491: Prerecorded Tones File
 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 483). 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 492: Cas Files
BootP/TFTP utility) after the device is reset. The device can be provided with a professionally recorded English (U.S.) VP file. Note: Voice Prompts are applicable only to Mediant 1000.  To generate and load the VP file: Prepare one or more voice files using standard utilities.
Page 493 SIP User's Manual 23. Software Upgrade prefix tags. For more information, see Dial Plan Prefix Tags for IP-to-Tel Routing on page 348. The Dial Plan file is first created using a text-based editor (such as Notepad) and saved with the file extension .ini. This ini file is then converted to a binary file (.dat) using the DConvert utility (refer to the Product Reference Manual).
Page 494: User Information File
Mediant 600 & Mediant 1000 23.1.6 User Information File The User Information file is a text-based file that can be used for mapping PBX extensions connected to the device to "global" IP numbers. The User Information file can be loaded to the device by using one of the following methods: ...
Page 495: Amd Sensitivity File
SIP User's Manual 23. Software Upgrade An example of a User Information file is shown in the figure below: Figure 23-1: 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>...
Page 496 Mediant 600 & Mediant 1000 The following example shows an xml file with two parameter suites:  Parameter Suite 0 with 6 sensitivity levels,  Parameter Suite 2 with 3 sensitivity levels. <AMDSENSITIVITY> <PARAMETERSUIT> <PARAMETERSUITID>0</PARAMETERSUITID>  <NUMBEROFLEVELS>8</NUMBEROFLEVELS>...
Page 497: Loading Software Upgrade Key
You can load a Software Upgrade Key using one of the following management tools:  Web interface  BootP/TFTP configuration utility (see Loading via BootP/TFTP on page 499)  AudioCodes’ EMS (refer to EMS User’s Manual or EMS Product Description) Version 6.4 March 2012...
Page 498 Mediant 600 & Mediant 1000 Warning: Do not modify the contents of the Software Upgrade Key file. Note: The Software Upgrade Key is an encrypted key.  To load a Software Upgrade Key: Open the Software Upgrade Key Status page (Maintenance tab > Software Update menu >...
Page 499: Loading Via Bootp/Tftp
Verify that the content of the file has not been altered. 23.2.1 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 500: Software Upgrade Wizard
(see 'Basic Maintenance' on page 477). Notes: • You can get the latest software files from AudioCodes Web site at http://www.audiocodes.com/downloads. • Before upgrading the device, it is recommended that you save a copy of the device's configuration settings (i.e., ini file) to your PC.
Page 501 SIP User's Manual 23. Software Upgrade  To load files using the Software Upgrade Wizard: Stop all traffic on the device using the Graceful Lock feature (refer to the warning bulletin above). Open the Software Upgrade wizard, by performing one of the following: •...
Page 502 Mediant 600 & Mediant 1000 Click the Next button; the wizard page for loading an ini file appears. You can now perform one of the following: • Load a new ini file: Click Browse, navigate to the ini file, and then click Send File;...
Page 503: Backing Up And Loading Configuration File
SIP User's Manual 23. Software Upgrade 23.4 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 504 Mediant 600 & Mediant 1000  To load the ini file: Click the Browse button, navigate to the folder in which the ini file is located, select the file, and then click Open; the name and path of the file appear in the field beside the Browse button.
Page 505: Restoring Factory Defaults
SIP User's Manual 24. Restoring Factory Defaults Restoring Factory Defaults You can restore the device's configuration to factory defaults using one of the following methods:  Using the CLI (see 'Restoring Defaults using CLI' on page 505)  Using the hardware Reset button (see Restoring Defaults using Hardware Reset Button on page 506) ...
Page 506: Restoring Defaults Using Hardware Reset Button
Mediant 600 & Mediant 1000 24.2 Restoring Defaults using Hardware Reset Button The device's hardware Reset pinhole button can be used to reset the device to default settings.  To restore default settings using the hardware Reset button:  With a paper clip or any other similar pointed object, press and hold down the Reset button (located on the CPU module) for at least 12 seconds (but no more than 25 seconds).
Page 507: Status, Performance Monitoring And Reporting
Part VI Status, Performance Monitoring and Reporting This part describes the status and performance monitoring procedures.
Page 508 Reader’s Notes...
Page 509: 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 510: Viewing Ethernet Port Information
Mediant 600 & Mediant 1000 25.2 Viewing Ethernet Port Information The Ethernet Port Information page displays read-only information on the Ethernet port connections. This includes information such as activity status, duplex mode, and speed. Notes: • The Ethernet Port Information page also be accessed from the Home page (see 'Using the Home Page' on page 59).
Page 511: Carrier-Grade Alarms
SIP User's Manual 26. 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 26.1 Viewing Active Alarms The Active Alarms page displays a list of currently active alarms.
Page 512: Viewing Alarm History
Mediant 600 & Mediant 1000 26.2 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 27. 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 27.1 Viewing Trunk Utilization The Trunk Utilization page...
Page 514 Mediant 600 & Mediant 1000 For more graph functionality, see the following table: Table 27-1: Additional Graph Functionality for Trunk Utilization Button Description Add button Displays additional trunks in the graph. Up to five trunks can be displayed simultaneously in the graph. To view another trunk, click this button and then from the new 'Trunk' drop-down list, select the required trunk.
Page 515: Viewing Mos Per Media Realm
SIP User's Manual 27. Performance Monitoring 27.2 Viewing MOS per Media Realm The MOS Per Media Realm page displays statistics on Media Realms (configured in 'Configuring Media Realms' on page 169). This page provides two graphs:  Upper graph: displays the Mean Opinion Score (MOS) quality in RTCP data per selected Media Realm.
Page 516 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 517: Voip Status
SIP User's Manual 28. VoIP Status VoIP Status This section describes how to view the following VoIP status and statistics:  IP network interface - see 'Viewing Active IP Interfaces' on page  Performance - see 'Viewing Performance Statistics' on page ...
Page 518: Viewing Call Counters
Mediant 600 & Mediant 1000 28.3 Viewing Call Counters The IP to Tel Calls Count page and Tel to IP Calls Count page provide you with statistical information on incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. The statistical information is updated according to the release reason that is received after a call is terminated (during the same time as the end-of-call Call Detail Record or CDR message is sent).
Page 519 SIP User's Manual 28. VoIP Status Counter Description Calls due to No Answer' counter. The rest of the release reasons increment the 'Number of Failed Calls due to Other Failures' counter. Percentage of The percentage of established calls from attempted calls. Successful Calls (ASR) Number of Calls Indicates the number of calls that failed as a result of a busy line.
Page 520: Viewing Sas Registered Users
Mediant 600 & Mediant 1000 28.4 Viewing SAS Registered Users The SAS/SBC Registered Users page displays a list of registered SAS users recorded in the device's database.  To view registered users:  Open the SAS/SBC Registered Users page (Status & Diagnostics tab > VoIP Status menu >...
Page 521: Viewing Registration Status
SIP User's Manual 28. VoIP Status Table 28-3: Call Routing Status Parameters Parameter Description  Proxy/GK = Proxy server is used to route calls. Call-Routing Method  Routing Table = The Outbound IP Routing Table is used to route calls. ...
Page 522: Viewing Ip Connectivity
Mediant 600 & Mediant 1000 • Group Name: name of the served group, if applicable • Status: indicates whether or not the group is registered ("Registered" or "Unregistered")  BRI Phone Number Status: • Phone Number: phone number of BRI endpoint •...
Page 523 SIP User's Manual 28. VoIP Status Table 28-4: IP Connectivity Parameters Column Name Description IP Address The IP address can be one of the following:  IP address defined as the destination IP address in the Outbound IP Routing Table'. ...
Page 524 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 525: Reporting Information To External Party
SIP User's Manual 29. Reporting Information to External Party Reporting Information to External Party 29.1 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 526 Mediant 600 & Mediant 1000 Field Name Description TrmSd Initiator of call release (IP, Tel, or Unknown) TrmReason Termination reason (see 'Release Reasons in CDR' on page 527) Fax transaction during call InPackets Number of incoming packets Number of outgoing packets...
Page 527: Release Reasons In Cdr
SIP User's Manual 29. 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 29.1.2 Release Reasons in CDR The possible reasons for call termination which is represented in the CDR field TrmReason are listed below: ...
Page 528 Mediant 600 & Mediant 1000  "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 529: Supported Radius Attributes
SIP User's Manual 29. Reporting Information to External Party  "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"  "GWAPP_RECOVERY_ON_TIMER_EXPIRY"  "GWAPP_PROTOCOL_ERROR_UNSPECIFIED"...
Page 530 Mediant 600 & Mediant 1000 Attribute Attribute Value Purpose Example Number Name Format Time Stop Acc The call’s originator: H323-Call- Answer, Start Acc Answering (IP) or String Origin Originate etc Stop Acc Originator (PSTN) H323-Call- Protocol type or family Start Acc...
Page 531 SIP User's Manual 29. Reporting Information to External Party Attribute Attribute Value Purpose Example Number Name Format Number of packets sent Numeric Stop Acc during the call Physical port type of Start Acc device on which the call is String Asynchronous Stop Acc active...
Page 532: Event Notification Using X-Detect Header
Mediant 600 & Mediant 1000 29.2 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 533 SIP User's Manual 29. Reporting Information to External Party Table 29-4: Special Information Tones (SITs) Reported by the device Special Description First Tone Second Tone Third Tone Information Frequency Frequency Frequency Tones (SITs) Duration Duration Duration Name (Hz) (ms) (Hz) (ms) (Hz) (ms)
Page 534: Querying Device Channel Resources Using Sip Options
Mediant 600 & Mediant 1000 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 535: Diagnostics
Part VII Diagnostics This part describes the diagnostics procedures.
Page 536 Reader’s Notes...
Page 537: Configuring Syslog Settings
SIP User's Manual 30. 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 563. For viewing Syslog messages in the Web interface, see Viewing Syslog Messages on page 539.
Page 538 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 539: 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 540 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 541: Appendices
Part VIII Appendices This part includes various appendices.
Page 542 Reader’s Notes...
Page 543: A Configuration Parameters Reference
SIP User's Manual A. Configuration Parameters Reference Configuration Parameters Reference The device's configuration parameters, default values, and their descriptions are documented in this section. Parameters and values enclosed in square brackets ([...]) represent the ini file parameters and their enumeration values; parameters not enclosed in square brackets represent their corresponding Web interface and/or EMS parameters.
Page 544: Multiple Network Interfaces And Vlan Parameters
Mediant 600 & Mediant 1000 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: IP Network Interfaces and VLAN Parameters Parameter Description Multiple Interface Table Web: Multiple Interface Table...
Page 545 SIP User's Manual A. Configuration Parameters Reference Parameter Description  To configure multiple IP interfaces in the Web interface and for a detailed description of the table's parameters, see 'Configuring IP Interface Settings' on page 102).  For a description of configuring ini file table parameters, see 'Configuring ini File Table Parameters' on page 84.
Page 546: Static Routing Parameters
Mediant 600 & Mediant 1000 Parameter Description parameter to a value other than any VLAN ID in the table. [EnableNTPasOAM] Defines the application type for NTP services.  [1] = OAMP (default)  [0] = Control. Note: For this parameter to take effect, a device reset is required.
Page 547: Quality Of Service Parameters
SIP User's Manual A. Configuration Parameters Reference A.1.4 Quality of Service Parameters The Quality of Service (QoS) parameters are described in the table below. The device allows you to specify values for Layer-2 and Layer-3 priorities by assigning values to the following service classes: ...
Page 548: Nat And Stun Parameters
Mediant 600 & Mediant 1000 Parameter Description determined by the following (according to priority):  IPDiffServ value in the selected IP Profile (IPProfile parameter).  PremiumServiceClassMediaDiffServ. Web: Control Premium QoS Defines the DiffServ value for Premium Control CoS EMS: Premium Service Class Control Diff Serv...
Page 549 SIP User's Manual A. Configuration Parameters Reference Parameter Description [STUNServerPrimaryIP] 0.0.0.0. Note: For this parameter to take effect, a device reset is required. Web: STUN Server Secondary Defines the IP address of the secondary STUN server. The valid range is the legal IP addresses. The default value is EMS: Secondary Server IP 0.0.0.0.
Page 550: Nfs Parameters
Mediant 600 & Mediant 1000 Parameter Description effect (i.e., the parameter DisableNAT is set to 0).  For information on RTP Multiplexing, see RTP Multiplexing (ThroughPacket) on page 157. [EnableUDPPortTranslation] Enables UDP port translation.  [0] = Disables UDP port translation (default).
Page 551: Dns Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description  To avoid terminating current calls, a row must not be deleted or modified while the device is currently accessing files on the remote NFS file system.  The combination of host/IP and Root Path must be unique for each index in the table.
Page 552: Dhcp Parameters
Mediant 600 & Mediant 1000 Parameter Description FORMAT SRV2IP_Index = SRV2IP_InternalDomain, SRV2IP_TransportType, SRV2IP_Dns1, SRV2IP_Priority1, SRV2IP_Weight1, SRV2IP_Port1, SRV2IP_Dns2, SRV2IP_Priority2, SRV2IP_Weight2, SRV2IP_Port2, SRV2IP_Dns3, SRV2IP_Priority3, SRV2IP_Weight3, SRV2IP_Port3; [\SRV2IP] For example: SRV2IP 0 = SrvDomain,0,Dnsname1,1,1,500,Dnsname2,2,2,501,$$,0,0,0; Notes:  This parameter can include up to 10 indices.
Page 553: Ntp And Daylight Saving Time Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: DHCP Speed Factor Defines the DHCP renewal speed. [DHCPSpeedFactor]  [0] = Disable  [1] = Normal (default)  [2] to [10] = Fast When set to 0, the DHCP lease renewal is disabled. Otherwise, the renewal time is divided by this factor.
Page 554: Management Parameters
Mediant 600 & Mediant 1000 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...
Page 555 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Deny Authentication Timer Defines the time (in seconds) that login to the Web interface is [DenyAuthenticationTimer] denied for a user that has reached maximum login attempts as defined by the DenyAccessOnFailCount parameter. Only after this time expires can the user attempt to login from the same IP address.
Page 556: Telnet Parameters
Mediant 600 & Mediant 1000 Parameter Description [ResetWebPassword] Determines whether the device resets the username and password of the primary and secondary accounts to their default settings.  [0] = Password and username retain their values (default).  [1] = Password and username are reset.
Page 557: Snmp Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Telnet Server TCP Port Defines the port number for the embedded Telnet server. EMS: Server Port The valid range is all valid port numbers. The default port is 23. [TelnetServerPort] Web: Telnet Server Idle Defines the timeout (in minutes) for disconnection of an idle Telnet Timeout...
Page 558 Mediant 600 & Mediant 1000 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.
Page 559 SIP User's Manual A. Configuration Parameters Reference Parameter Description 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 560: Serial Parameters
Mediant 600 & Mediant 1000 A.2.5 Serial Parameters The RS-232 serial parameters are described in the table below. Table A-14: Serial Parameters Parameter Description Enables the device's RS-232 (serial) port. [DisableRS232]  [0] = Enabled (default)  [1] = Disabled The RS-232 serial port can be used to change the networking parameters and view error/notification messages.
Page 561: Debugging And Diagnostics Parameters
SIP User's Manual A. Configuration Parameters Reference Debugging and Diagnostics Parameters This subsection describes the device's debugging and diagnostic parameters. A.3.1 General Parameters The general debugging and diagnostic parameters are described in the table below. Table A-15: General Debugging and Diagnostic Parameters Parameter Description EMS: Enable Diagnostics...
Page 562 Mediant 600 & Mediant 1000 Parameter Description  FXS Port 1 of each FXS module FXS Port 1 connects to the POTS (Lifeline) phone as well as to the PSTN / PBX, using a splitter cable.  [0] = Lifeline is activated upon power outage (default).
Page 563: Syslog, Cdr And Debug Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description The FXS interface initiates the signaling by grounding of the TIP lead. 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...
Page 564 Mediant 600 & Mediant 1000 Parameter Description parameter EnableSyslog is set to 1). Web/EMS: CDR Report Determines whether Call Detail Records (CDR) are sent to the Syslog Level server and when they are sent. [CDRReportLevel]  [0] None = CDRs are not used (default).
Page 565 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [18] = local use 2 (local2)  [19] = local use 3 (local3)  [20] = local use 4 (local4)  [21] = local use 5 (local5)  [22] = local use 6 (local6) ...
Page 566: Resource Allocation Indication Parameters
Mediant 600 & Mediant 1000 A.3.3 Resource Allocation Indication Parameters The Resource Allocation Indication (RAI) parameters are described in the table below. Table A-17: RAI Parameters Parameter Description Enables RAI alarm generation if the device's busy endpoints [EnableRAI] exceed a user-defined threshold.
Page 567 SIP User's Manual A. Configuration Parameters Reference Parameter Description  (default). [5] = 8 DHCP packets   [4] = 10 BootP retries, 30 sec. [6] = 9 DHCP packets   [5] = 20 BootP retries, 60 sec. [7] = 10 DHCP packets ...
Page 568: Security Parameters
Mediant 600 & Mediant 1000 Security Parameters This subsection describes the device's security parameters. A.4.1 General Parameters The general security parameters are described in the table below. Table A-19: General Security Parameters Parameter Description Web: Voice Menu Defines the password for accessing the device's voice menu, used for Password configuring and monitoring the device.
Page 569: Https Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description For example: AccessList 10 = mgmt.customer.com, , , 32, 0, 80, tcp, 1, OAMP, 0, 0, 0, allow; AccessList 22 = 10.4.0.0, , , 16, 4000, 9000, any, 0, , 0, 0, 0, block; In the example above, Rule #10 allows traffic from the host ‘mgmt.customer.com’...
Page 570 Mediant 600 & Mediant 1000 Parameter Description Encryption” Software Upgrade Key is enabled. Web: HTTP Authentication Mode Determines the authentication mode used for the Web interface. EMS: Web Authentication Mode  [0] Basic Mode = Basic authentication (clear text) is used [WebAuthMode] (default).
Page 571: Srtp Parameters
SIP User's Manual A. Configuration Parameters Reference A.4.3 SRTP Parameters The Secure Real-Time Transport Protocol (SRTP) parameters are described in the table below. Table A-21: SRTP Parameters Parameter Description Web: Media Security Enables Secure Real-Time Transport Protocol (SRTP). EMS: Enable Media Security ...
Page 572 Mediant 600 & Mediant 1000 Parameter Description  [1] = Enabled - the answer crypto line contains (or excludes) an MKI value according to the selected crypto line in the offer. For example, assume that the device receives an INVITE containing...
Page 573: Tls Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description  [RTCPEncryptionDisableT [1] Disable A.4.4 TLS Parameters The Transport Layer Security (TLS) parameters are described in the table below. Table A-22: TLS Parameters Parameter Description Web/EMS: TLS Version Determines the supported versions of SSL/TLS (Secure Socket [TLSVersion] Layer/Transport Layer Security.
Page 574 Mediant 600 & Mediant 1000 Parameter Description and the SubjectAltName is marked as ‘critical’, the TLS connection is not established. If DNSName is used, the certificate can also use wildcards (‘*’) to replace parts of the domain name. If the SubjectAltName is not marked as ‘critical’ and there is no match, the CN value of the SubjectName field is compared with the parameter TLSRemoteSubjectName.
Page 575: Ssh Parameters
SIP User's Manual A. Configuration Parameters Reference A.4.5 SSH Parameters Secure Shell (SSH) parameters are described in the table below. Table A-23: SSH Parameters Parameter Description Web/EMS: Enable SSH Server Enables the device's embedded SSH server. [SSHServerEnable]  [0] Disable (default) ...
Page 576: Ipsec Parameters
Mediant 600 & Mediant 1000 A.4.6 IPSec Parameters The Internet Protocol security (IPSec) parameters are described in the table below. Table A-24: IPSec Parameters Parameter Description IPSec Parameters Web: Enable IP Security Enables IPSec on the device. EMS: IPSec Enable ...
Page 577: Ocsp Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description [ IPsecProposalTable ] FORMAT IPsecProposalTable_Index = IPsecProposalTable_EncryptionAlgorithm, IPsecProposalTable_AuthenticationAlgorithm, IPsecProposalTable_DHGroup; [ \IPsecProposalTable ] For example: IPsecProposalTable 0 = 3, 2, 1; IPsecProposalTable 1 = 2, 2, 1; In the example above, two proposals are defined: ...
Page 578: Radius Parameters
Mediant 600 & Mediant 1000 RADIUS Parameters The RADIUS parameters are described in the table below. For supported RADIUS attributes, see 'Supported RADIUS Attributes' on page 529. Table A-26: RADIUS Parameters Parameter Description Web: Enable RADIUS Access Enables the RADIUS application.
Page 579: Sip Media Realm Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description [DefaultAccessLevel] (authentication) response doesn't include an access level attribute. The valid range is 0 to 255. The default value is 200 (Security Administrator'). Web: Local RADIUS Password Determines the device's mode of operation regarding the timer Cache Mode (configured by the parameter RadiusLocalCacheTimeout) that [RadiusLocalCacheMode]...
Page 580: Quality Of Experience Reporting
Mediant 600 & Mediant 1000 Parameter Description [\CpMediaRealm] For example, CpMediaRealm 1 = Mrealm1, Voice, , 6600, 20, 6790, , 1; CpMediaRealm 2 = Mrealm2, Voice, , 6800, 10, 6890; , 0; Notes:  For this parameter to take effect, a device reset is required.
Page 581: Control Network Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Media Realm > Quality of Experience Table Web: Media Realm > This table configures Quality of Experience parameters per Media Quality Of Experience Realm. [QOERules] [ QOERules ] FORMAT QOERules_Index = QOERules_MediaRealmIndex, QOERules_RuleIndex, QOERules_MonitoredParam, QOERules_Profile, QOERules_GreenYellowThreshold, QOERules_GreenYellowHystersis, QOERules_YellowRedThreshold, QOERules_YellowRedHystersis;...
Page 582 Mediant 600 & Mediant 1000 Parameter Description Mediant 600.  For a detailed description of the ini file table's parameters and for configuring this table using the Web interface, see 'Configuring IP Groups' on page 195.  For configuring ini file table parameters, see 'Configuring ini File Table Parameters' on page 84.
Page 583 SIP User's Manual A. Configuration Parameters Reference Parameter Description [\Account] For example: Account 1 = 1, -1, 1, user, 1234, acl, 1, ITSP1; Notes:  This table can include up to 32 indices (where 1 is the first index).  The parameter Account_ApplicationType is not applicable.
Page 584 Mediant 600 & Mediant 1000 Parameter Description  [IsFallbackUsed] [1] Enable = The Outbound IP Routing Table is used when Proxy servers are unavailable. When the device falls back to the Outbound IP Routing Table', it continues scanning for a Proxy. When the device locates an active Proxy, it switches from internal routing back to Proxy routing.
Page 585 SIP User's Manual A. Configuration Parameters Reference Parameter Description If set to A-Record [0], no NAPTR or SRV queries are performed. If set to SRV [1] and the Proxy/Registrar IP address parameter, Contact/Record-Route headers, or IP address defined in the Routing tables contain a domain name, an SRV query is performed.
Page 586 Mediant 600 & Mediant 1000 Parameter Description Note: This parameter is applicable only to digital interfaces. Web/EMS: Use Gateway Determines whether the device uses its IP address or gateway name in Name for OPTIONS keep-alive SIP OPTIONS messages. [UseGatewayNameForOp ...
Page 587 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [2] Full = Caches all challenges from the proxies. Note: Challenge Caching is used with all proxies and not only with the active one. Proxy IP Table Web: Proxy IP Table This parameter table configures the Proxy Set table with Proxy Set IDs, EMS: Proxy IP each with up to five Proxy server IP addresses (or fully qualified domain...
Page 588 Mediant 600 & Mediant 1000 Parameter Description 'Configuring Proxy Sets Table' on page 200.  For configuring ini file table parameters, see 'Configuring ini File Table Parameters' on page 84. Registrar Parameters Web: Enable Registration Enables the device to register to a Proxy/Registrar server.
Page 589 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Re-registration Defines the re-registration timing (in percentage). The timing is a Timing [%] percentage of the re-register timing set by the Registrar server. EMS: Time Divider The valid range is 50 to 100. The default value is 50. [RegistrationTimeDivider For example: If this parameter is set to 70% and the Registration Expires time is 3600, the device re-sends its registration request after...
Page 590 Mediant 600 & Mediant 1000 Parameter Description Web: Gateway Defines the user name that is used in the From and To headers in SIP Registration Name REGISTER messages. If no value is specified (default) for this EMS: Name parameter, the UserName parameter is used instead.
Page 591 SIP User's Manual A. Configuration Parameters Reference Parameter Description Authorization Header registration (REGISTER) requests sent by the device. [EmptyAuthorizationHea  [0] Disable (default) der]  [1] Enable The Authorization header carries the credentials of a user agent (UA) in a request to a server. The sent REGISTER message populates the Authorization header with the following parameters: ...
Page 592: Network Application Parameters
Mediant 600 & Mediant 1000 Parameter Description and its transport type is set to TCP or TLS. The device first sends a SIP OPTION message to establish the TCP/TLS connection and if it receives any SIP response, it continues sending the CRLF keep-alive sequences.
Page 593 SIP User's Manual A. Configuration Parameters Reference Parameter Description SIPInterface_TCPPort, SIPInterface_TLSPort, SIPInterface_SRD; [\SIPInterface] For example: SIPInterface 0 = Voice, 2, 5060, 5060, 5061, 1; SIPInterface 1 = Voice, 2, 5070, 5070, 5071, 2; SIPInterface 2 = Voice, 0, 5090, 5000, 5081, 2; Notes: ...
Page 594: General Sip Parameters
Mediant 600 & Mediant 1000 Parameter Description Uses an external STUN server (STUNServerPrimaryIP parameter) to assign a NAT address for all interfaces. Uses the StaticNATIP parameter to define one NAT IP address for all interfaces. Uses the NATTranslation parameter to define NAT per interface.
Page 595 SIP User's Manual A. Configuration Parameters Reference Parameter Description rule in the Outbound IP Routing table that reroutes the call to an alternative IP Group. You also need to add a rule to the Reason for Alternative Routing table to initiate an alternative rule for Tel-to- IP calls using cause 805.
Page 596 Mediant 600 & Mediant 1000 Parameter Description  The Supported and Required headers contain the '100rel' tag.  The device sends PRACK messages if 180/183 responses are received with '100rel' in the Supported or Required headers. Web/EMS: Enable Early Digital: Enables the device to send a 18x response with SDP instead...
Page 597 SIP User's Manual A. Configuration Parameters Reference Parameter Description device did not send a 183 with an SDP and it receives an Alert without PI, the device sends a 180 (without SDP). If it receives an Alert with PI it sends a 183with an SDP. When disabled, the device sends additional 18x responses as a result of receiving Alerting and Progress messages, regardless of whether or not a 18x response was already sent.
Page 598 Mediant 600 & Mediant 1000 Parameter Description  [1] UPDATE = Uses UPDATE messages. Notes:  The device can receive session-timer refreshes using both methods.  The UPDATE message used for session-timer is excluded from the SDP body. [RemoveToTagInFailureRe Determines whether the device removes the ‘to’ header tag from final sponse] SIP failure responses to INVITE transactions.
Page 599 SIP User's Manual A. Configuration Parameters Reference Parameter Description  Silence Compression = Off  Echo Canceller Non-Linear Processor Mode = Off  Dynamic Jitter Buffer Minimum Delay = 40  Dynamic Jitter Buffer Optimization Factor = 13  If the device initiates a fax session using G.711 (option 2 and possibly 3), a 'gpmd' attribute is added to the SDP in the following format: ...
Page 600 Mediant 600 & Mediant 1000 Parameter Description  This feature can be used only if the remote party supports T.38 fax relay; otherwise, the fax fails. Web: SIP Transport Type Determines the default transport layer for outgoing SIP calls initiated EMS: Transport Type by the device.
Page 601 SIP User's Manual A. Configuration Parameters Reference Parameter Description used. Persistent TCP connection ensures less network traffic due to fewer setting up and tearing down of TCP connections and reduced latency on subsequent requests due to avoidance of initial TCP handshake. For TLS, persistent connection may reduce the number of costly TLS handshakes to establish security associations, in addition to the initial TCP connection set up.
Page 602 Mediant 600 & Mediant 1000 Parameter Description last entry, and concatenates a new destination to it (if an additional request is sent). The order of the reasons is as follows: Q.850 Reason SIP Reason SIP Response code  Upon receiving the final response (success or failure), the device searches for a Redirect reason in the History-Info (i.e., 3xx/4xx SIP...
Page 603 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [3] Hotline = Interworks the hotline "Off Hook Indicator" parameter between SIP and ISDN:  For IP-to-ISDN calls: - The device interworks the SIP tgrp=hotline parameter (received in INVITE) to ISDN Setup with the Off Hook Indicator IE of “Voice”, and “Speech”...
Page 604 Mediant 600 & Mediant 1000 Parameter Description contain the 'tgrp' parameter or if the Trunk Group number is not defined, then the Inbound IP Routing Table is used for routing the call. Below is an example of an INVITE Request-URI with the 'tgrp' parameter, indicating that the IP call should be routed to Trunk Group INVITE sip:200;tgrp=7;trunk-...
Page 605 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 1000/v.6.40.010.006 The maximum string length is 50 characters.
Page 606 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: SDP Session Defines the value of the Owner line ('o' field) in outgoing SDP Owner messages. [SIPSDPSessionOwner] The valid range is a string of up to 39 characters. The default value is 'AudiocodesGW'.
Page 607 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web/EMS: Retry-After Time Defines the time (in seconds) used in the Retry-After header when a 503 (Service Unavailable) response is generated by the device. [RetryAfterTime] The time range is 0 to 3,600. The default value is 0. Web/EMS: Fake Retry After Determines whether the device, upon receipt of a SIP 503 response [sec]...
Page 608 Mediant 600 & Mediant 1000 Parameter Description  The "From" and "Pai2" values are not case-sensitive.  Once a URL is selected, all the calling party parameters are set from this header. If P-Asserted-Identity is selected and the Privacy header is set to 'id', the calling number is assumed restricted.
Page 609 SIP User's Manual A. Configuration Parameters Reference Parameter Description in the Outbound IP Routing table. Web/EMS: Enable Reason Enables the usage of the SIP Reason header. Header  [0] Disable [EnableReasonHeader]  [1] Enable (default) Web/EMS: Gateway Name Defines a name for the device (e.g., device123.com'). [SIPGatewayName] Notes: ...
Page 610 Mediant 600 & Mediant 1000 Parameter Description identify their SIP Trunking customers by their source phone number or IP address, reflected in the From header of the SIP INVITE. Therefore, even customers blocking their Caller ID can be identified by the service provider. Typically, if the device receives a call with blocked Caller ID from the PSTN side (e.g., Trunk connected to a...
Page 611 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Enable X-Channel Determines whether the SIP X-Channel header is added to SIP Header messages for providing information on the physical Trunk/B-channel EMS: X Channel Header on which the call is received or placed. [XChannelHeader] ...
Page 612 Mediant 600 & Mediant 1000 Parameter Description  Call] [1] = Use Transparent coder for data calls (according to RFC 4040). The Transparent' coder can be used on data calls. When the device receives a Setup message from the ISDN with 'TransferCapabilities = data', it can initiate a call using the coder 'Transparent' (even if the coder is not included in the coder list).
Page 613 SIP User's Manual A. Configuration Parameters Reference Parameter Description 'Release Reason Mapping' on page 242. Web: Enable Microsoft Enables the modification of the called and calling number for numbers Extension received with Microsoft's proprietary "ext=xxx" parameter in the SIP [EnableMicrosoftExt] INVITE URI user part.
Page 614 Mediant 600 & Mediant 1000 Parameter Description  EMS: Comfort Noise [0] Disable Generation  [1] Enable (default) [ComfortNoiseNegotiation] The use of CN is indicated by including a payload type for CN on the media description line of the SDP. The device can use CN with a codec whose RTP time stamp clock rate is 8,000 Hz (G.711/G.726).
Page 615 SIP User's Manual A. Configuration Parameters Reference Parameter Description regret timeout (configured by the parameter RegretTime). Therefore, the device notifies the far-end that the call has been re-answered.  [0] Disable (default)  [1] Enable This parameter is typically implemented for incoming IP-to-Tel collect calls to the FXS port.
Page 616 Mediant 600 & Mediant 1000 Parameter Description remote IP address is determined according to the Outbound IP Routing table (Prefix parameter). The port is the same port as the local RTP port (configured by the BaseUDPPort parameter and the channel on which the call is received).
Page 617 SIP User's Manual A. Configuration Parameters Reference Parameter Description Note: When not configured (i.e., default), the SITQ850Cause parameter is used. Web/EMS: SIT Q850 Cause Defines the Q.850 cause value specified in the SIP Reason header For RO that is included in a 4xx response when SIT-RO (Reorder - System [SITQ850CauseForRO] Busy Special Information Tone) is detected from the PSTN for IP-to- Tel calls.
Page 618 Mediant 600 & Mediant 1000 Parameter Description complementary optional behavior.  For Digital interfaces: The Busy Out behavior varies between different protocol types.  For Digital interfaces: The Busy-Out condition can also be applied to a specific Trunk Group. If there is no connectivity to the Serving...
Page 619 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Number of RTX Before Defines the number of retransmitted INVITE/REGISTER messages Hot-Swap before the call is routed (hot swap) to another Proxy/Registrar. EMS: Proxy Hot Swap Rtx The valid range is 1 to 30. The default value is 3. [HotSwapRtx] Note: This parameter is also used for alternative routing using the Outbound IP Routing Table'.
Page 620: Coders And Profile Parameters
Mediant 600 & Mediant 1000 Parameter Description  [0] Use Current Condition = The condition entered in this row must be matched in order to perform the defined action (default).  [1] Use Previous Condition = The condition of the rule configured directly above this row must be used in order to perform the defined action.
Page 621 SIP User's Manual A. Configuration Parameters Reference Parameter Description [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = g711Alaw64k, 20, 0, 255, 0; CodersGroup0 1 = eg711Ulaw, 10, 0, 71, 0; CodersGroup0 2 = eg711Ulaw, 10, 0, 71, 0; [ \CodersGroup0 ] [ CodersGroup1 ] FORMAT CodersGroup1_Index = CodersGroup1_Name,...
Page 622 Mediant 600 & Mediant 1000 Parameter Description 20 (default) 4.75 [0], Dynamic (0- Disable [0] [Amr] 5.15 [1], 127) Enable [1] 5.90 [2], 6.70 [3], 7.40 [4], 7.95 [5], 10.2 [6], 12.2 [7] (default) QCELP 20 (default), 40, 60, Always...
Page 623 SIP User's Manual A. Configuration Parameters Reference Parameter Description IP Profile Table Web: IP Profile This parameter table configures the IP Profile table. Each IP Profile ID Settings includes a set of parameters (which are typically configured separately using EMS: Protocol their individual "global"...
Page 624 Mediant 600 & Mediant 1000 Parameter Description IpProfile_IsFaxUsed Fax Signaling Method IsFaxUsed IpProfile_JitterBufMi Dynamic Jitter Buffer DJBufMinDelay nDelay Minimum Delay IpProfile_JitterBufO Dynamic Jitter Buffer DJBufOptFactor ptFactor Optimization Factor IpProfile_IPDiffServ RTP IP DiffServ PremiumServiceClassMed iaDiffServ IpProfile_SigIPDiffS Signaling DiffServ PremiumServiceClassCont rolDiffServ IpProfile_SCE...
Page 625 SIP User's Manual A. Configuration Parameters Reference Parameter Description IpProfile_RxDTMFO Declare RFC 2833 in RxDTMFOption ption IpProfile_EnableHol Enable Hold EnableHold IpProfile_InputGain Input Gain InputGain IpProfile_VoiceVolu Voice Volume VoiceVolume IpProfile_AddIEInSe Add IE In SETUP AddIEinSetup IpProfile_MediaIPVe Media IP Version MediaIPVersionPreference rsionPreference Preference Transcoding Mode TranscodingMode...
Page 626 Mediant 600 & Mediant 1000 Parameter Description V32ModemTransportType, and V34ModemTransportType.  IP Profiles can also be used when operating with a Proxy server (set the parameter AlwaysUseRouteTable to 1).  The following parameters are not applicable: IsDTMFUsed (deprecated), SBCExtensionCodersGroupID, TranscodingMode...
Page 627 SIP User's Manual A. Configuration Parameters Reference Parameter Description TelProfile_CodersGr Coder Group CodersGroup0 oupID TelProfile_IsFaxUse Fax Signaling Method IsFaxUsed TelProfile_JitterBuf Dynamic Jitter Buffer DJBufMinDelay MinDelay Minimum Delay TelProfile_JitterBuf Dynamic Jitter Buffer DJBufOptFactor OptFactor Optimization Factor RTP IP DiffServ PremiumServiceClassMed TelProfile_IPDiffSer iaDiffServ TelProfile_SigIPDiff Signaling DiffServ...
Page 628: Channel Parameters
Mediant 600 & Mediant 1000 Parameter Description TelProfile_EnableVo Enable Voice Mail iceMailDelay Delay TelProfile_DialPlanI Dial Plan Index DialPlanIndex ndex TelProfile_Enable91 Enable 911 PSAP Enable911PSAP 1PSAP TelProfile_SwapTelT Swap Tel To IP SwapTEl2IPCalled&Callin oIpPhoneNumbers Phone Numbers gNumbers TelProfile_EnableA Enable AGC EnableAGC TelProfile_ECNlpMo...
Page 629 SIP User's Manual A. Configuration Parameters Reference Parameter Description 'Configuring Tel Profiles' on page 217). Web: Voice Volume Defines the voice gain control (in decibels). This parameter sets EMS: Volume (dB) the level for the transmitted (IP-to-Tel/PSTN) signal. [VoiceVolume] The valid range is -32 to 31 dB. The default value is 0 dB. Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see 'Configuring IP Profiles' on page 219) and per Tel Profile, using the TelProfile parameter (see...
Page 630 Mediant 600 & Mediant 1000 Parameter Description  If EnableSilenceCompression is 2 and IsCiscoSCEMode is 0: 'annexb=yes'.  If EnableSilenceCompression is 2 and IsCiscoSCEMode is 1: 'annexb=no'. Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see 'Configuring IP Profiles' on page 219).
Page 631: Coder Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Note: This parameter can also be configured per Tel Profile, using the TelProfile parameter (see 'Configuring Tel Profiles' on page 217). [EchoCancellerAggressiveNLP] Enables the Aggressive NLP at the first 0.5 second of the call. ...
Page 632 Mediant 600 & Mediant 1000 Parameter Description [DspTemplates] FORMAT DspTemplates_Index = DspTemplates_DspTemplateNumber, DspTemplates_DspResourcesPercentage; [\DspTemplates] For example, to load DSP Template 1 to 50% of the DSPs, and DSP Template 2 to the remaining 50%, the table is configured as follows: DspTemplates 0 = 1, 50;...
Page 633: Dtmf Parameters
SIP User's Manual A. Configuration Parameters Reference A.11.3 DTMF Parameters The dual-tone multi-frequency (DTMF) parameters are described in the table below. Table A-35: 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 634: Rtp, Rtcp And T.38 Parameters
Mediant 600 & Mediant 1000 A.11.4 RTP, RTCP and T.38 Parameters The RTP, RTCP and T.38 parameters are described in the table below. Table A-36: RTP/RTCP and T.38 Parameters Parameter Description Web: Dynamic Jitter Buffer Defines the minimum delay (in msec) for the Dynamic Jitter Minimum Delay Buffer.
Page 635 SIP User's Manual A. Configuration Parameters Reference Parameter Description parameter RFC2198PayloadType. a=rtpmap:<PT> RED/8000 Where <PT> is the payload type as defined by RFC2198PayloadType. The device sends the INVITE message with "a=rtpmap:<PT> RED/8000" and responds with a 18x/200 OK and "a=rtpmap:<PT> RED/8000" in the SDP. Notes: ...
Page 636: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 Parameter Description [EnableDetectRemoteMACCha Determines whether the device changes the RTP packets nge] according to the MAC address of received RTP packets and according to Gratuitous Address Resolution Protocol (GARP) messages.  [0] = Nothing is changed.
Page 637 SIP User's Manual A. Configuration Parameters Reference Parameter Description  This parameter can also be configured per IP Profile, using the IPProfile parameter.  For more information on RTP multiplexing, see RTP Multiplexing (ThroughPacket) on page 157. Web: RTP Multiplexing Local Defines the local (source) UDP port for outgoing multiplexed RTP UDP Port packets, for RTP multiplexing.
Page 638 Mediant 600 & Mediant 1000 Parameter Description [VQMonEnable] summary-13.  [0] Disable = Disable (default)  [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 (number EMS: GMin of frames).
Page 639: Gateway And Ip-To-Ip Parameters
SIP User's Manual A. Configuration Parameters Reference 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-37: Fax and Modem Parameters Parameter Description Web: Fax Transport Mode Determines the fax transport mode used by the device.
Page 640 Mediant 600 & Mediant 1000 Parameter Description Web: V.34 Modem Transport Determines the V.90/V.34 modem transport type. Type  [0] Disable = Disable (Transparent) EMS: V34 Transport  [1] Enable Relay = N/A [V34ModemTransportType  [2] Enable Bypass = (default) ...
Page 641 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Fax Relay Max Rate Defines the maximum rate (in bps) at which fax relay messages are (bps) transmitted (outgoing calls). EMS: Relay Max Rate  [0] 2400 = 2.4 kbps [FaxRelayMaxRate] ...
Page 642 Mediant 600 & Mediant 1000 Parameter Description EMS: Basic Packet Interval Defines the basic frame size used during fax/modem bypass sessions. [FaxModemBypassBasicR  [0] = Determined internally (default) TPPacketInterval]  [1] = 5 msec (not recommended)  [2] = 10 msec ...
Page 643 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: T38 Use RTP Port Defines the port (with relation to RTP port) for sending and receiving [T38UseRTPPort] T.38 packets.  [0] = Use the RTP port +2 to send/receive T.38 packets (default). ...
Page 644: Dtmf And Hook-Flash Parameters
Mediant 600 & Mediant 1000 Parameter Description set to 1 (T.38 Relay) or 3 (Fax Fallback). Enables fax transmission of T.38 “no-signal” packets to the terminating [T38FaxSessionImmediate fax machine. Start]  [0] Disable (default)  [1] Enable This is used for transmission from fax machines (connected to the device) located inside a Network Address Translation (NAT).
Page 645 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [5] INFO (Lucent) = Sends proprietary SIP INFO message with Hook-Flash indication. The device sends the INFO message as follows: Content-Type: application/hook-flash Content-Length: 11 signal=hf  [6] INFO (NetCentrex) = Sends proprietary SIP INFO message with Hook-Flash indication.
Page 646 Mediant 600 & Mediant 1000 Parameter Description hook-flash, set this parameter to 550.  This parameter can also be configured per Tel Profile, using the TelProfile parameter. DTMF Parameters EMS: Use End of DTMF Determines when the detection of DTMF events is notified.
Page 647 SIP User's Manual A. Configuration Parameters Reference Parameter Description SDPs. Sends DTMF packets using RFC 2833 payload type according to the payload type in the received SDP. Expects to receive RFC 2833 packets with the same payload type as configured by the parameter RFC2833PayloadType. Removes DTMF digits in transparent mode (as part of the voice stream).
Page 648 Mediant 600 & Mediant 1000 Parameter Description 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. For example, if the called number is '1007766p100', the device places a call with 1007766 as the destination number, then after the call is answered it waits 1.5...
Page 649: Digit Collection And Dial Plan Parameters
SIP User's Manual A. Configuration Parameters Reference A.12.3 Digit Collection and Dial Plan Parameters The digit collection and dial plan parameters are described in the table below. Table A-39: Digit Collection and Dial Plan Parameters Parameter Description Web/EMS: Dial Plan Index Defines the Dial Plan index to use in the external Dial Plan file.
Page 650 Mediant 600 & Mediant 1000 Parameter Description 99|998, then the digit collection is terminated after the first two 9 digits are received. Therefore, the second rule of 998 can never be matched. But when the digit map is 99s|998, then after dialing the first two 9 digits, the device waits another two seconds within which the caller can enter the digit 8.
Page 651: Voice Mail Parameters
SIP User's Manual A. Configuration Parameters Reference A.12.4 Voice Mail Parameters The voice mail parameters are described in the table below. For more information on the Voice Mail application, refer to the CPE Configuration Guide for Voice Mail. Table A-40: Voice Mail Parameters Parameter Description Web/EMS: Voice Mail...
Page 652 Mediant 600 & Mediant 1000 Parameter Description [WaitForBusyTime] Defines the time (in msec) that the device waits to detect busy and/or reorder tones. This feature is used for semi-supervised PBX call transfers (i.e., the LineTransferMode parameter is set to 2).
Page 653 SIP User's Manual A. Configuration Parameters Reference Parameter Description generates an additional hook-flash toward the FXO line to restore connection to the original call.  E1/T1 CAS: The device performs a supervised transfer to the PBX. The device performs a CAS wink, waits a user-defined time (configured by the WaitForDialTime parameter), and then dials the Refer-To number.
Page 654 Mediant 600 & Mediant 1000 Parameter Description [MWISubscribeIPGroupID] Defines the IP Group ID used when subscribing to an MWI server. The 'The SIP Group Name' field value of the IP Group table is used as the Request-URI host name in the outgoing MWI SIP SUBSCRIBE message.
Page 655 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Forward on No Reason Defines the digit pattern used by the PBX to indicate 'call forward Digit Pattern (Internal) with no reason' when the original call is received from an internal EMS: Digit Pattern Forward extension.
Page 656: Supplementary Services Parameters
Mediant 600 & Mediant 1000 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-41: Caller ID Parameters Parameter Description Web: Caller ID Permissions Table EMS: SIP Endpoints >...
Page 657 SIP User's Manual A. Configuration Parameters Reference Parameter Description  IsCidRestricted =  [0] Allowed = sends the defined caller ID string when a Tel-to- IP call is made using the corresponding device port (default).  [1] Restricted = does not send the defined caller ID string. ...
Page 658 Mediant 600 & Mediant 1000 Parameter Description  [17] Standard Denmark = Caller ID and MWI  [18] Standard India  [19] Standard Brazil Notes:  Typically, the Caller ID signals are generated/detected between the first and second rings. However, sometimes the Caller ID is detected before the first ring signal (in such a scenario, configure the parameter RingsBeforeCallerID to 0).
Page 659 SIP User's Manual A. Configuration Parameters Reference Parameter Description cpc=payphone Payphone user Notes:  This parameter is applicable only to FXS interfaces.  For this parameter to be enabled, you must also set the parameter EnableCallingPartyCategory to 1. [EnableCallerIDTypeTwo] Disables the generation of Caller ID type 2 when the phone is off- hooked.
Page 660 Mediant 600 & Mediant 1000 Parameter Description includes the value 'id' ('Privacy: id'). Otherwise, for allowed Caller ID, 'Privacy: none' is used. If Caller ID is restricted (received from Tel or configured in the device), the From header is set to <anonymous@anonymous.invalid>.
Page 661: Call Waiting Parameters
SIP User's Manual A. Configuration Parameters Reference Call Waiting Parameters A.12.5.2 The call waiting parameters are described in the table below. Table A-42: Call Waiting Parameters Parameter Description Web/EMS: Enable Call Waiting Determines whether Call Waiting is enabled. [EnableCallWaiting]  [0] Disable = Disable the Call Waiting service.
Page 662 Mediant 600 & Mediant 1000 Parameter Description initiates the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received.  Port = Port number.  Module = Module number. For example: CallWaitingPerPort 0 = 0,1,1; (call waiting disabled for Port 1 of Module 1) CallWaitingPerPort 1 = 1,1,2;...
Page 663: Call Forwarding Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description ToneIndex parameter table.  Playing the call waiting tone according to the parameter “CallWaitingTone#' of a SIP INFO message. The device plays the tone received in the 'play tone CallWaitingTone#' parameter of an INFO message plus the value of this parameter minus 1.
Page 664 Mediant 600 & Mediant 1000 Parameter Description device's port to which the call was originally routed. The format of this parameter is as follows: [FwdInfo] FORMAT FwdInfo_Index = FwdInfo_Type, FwdInfo_Destination, FwdInfo_NoReplyTime, FwdInfo_Module, FwdInfo_Port; [\FwdInfo] Where,  Type = the scenario for forwarding the call: ...
Page 665: Message Waiting Indication Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description  [1] Enable Web: AS Subscribe Defines the IP Group ID that contains the Application server for IPGroupID Subscription. [ASSubscribeIPGroupID] The valid value range is 1 to 8. The default is -1 (i.e., not configured). Web: NRT Retry Defines the Retry period (in seconds) for Dialog subscription if a Subscription Time...
Page 666 Mediant 600 & Mediant 1000 Parameter Description [EnableMWISubscription] parameter MWIServerIP address). Note: To configure whether the device subscribes per endpoint or per the entire device, use the parameter SubscriptionMode. Web: MWI Server IP Defines the MWI server's IP address. If provided, the device Address subscribes to this IP address.
Page 667: Call Hold Parameters
SIP User's Manual A. Configuration Parameters Reference Call Hold Parameters A.12.5.5 The call hold parameters are described in the table below. Table A-45: Call Hold Parameters Parameter Description Web/EMS: Enable Hold For digital interfaces: Enables interworking of the Hold/Retrieve [EnableHold] supplementary service from PRI to SIP.
Page 668: Call Transfer Parameters
Mediant 600 & Mediant 1000 Parameter Description or a waiting call when the phone is returned to on-hook position.  [0] = (default) The reminder ring feature is active. In other words, if a call is on hold or there is a call waiting, and the phone is changed from offhook to onhook, the phone rings (for a duration defined by the CHRRTimeout parameter) to "remind"...
Page 669 SIP User's Manual A. Configuration Parameters Reference Parameter Description interfaces) number from the originally dialed number. Web: Transfer Prefix IP 2 Tel Defines the prefix that is added to the destination number received [XferPrefixIP2Tel] in the SIP Refer-To header (for IP-to-Tel calls). This parameter is applicable to FXO/CAS blind transfer modes, i.e., LineTransferMode = 1, 2 or 3, and TrunkTransferMode = 1 or 3 (for CAS).
Page 670: Three-Way Conferencing Parameters
Mode is used. EMS: 3 Way Mode  [0] AudioCodes Media Server = The Conference-initiating INVITE [3WayConferenceMode] (sent by the device) uses the ConferenceID concatenated with a unique identifier as the Request-URI. This same Request-URI is set as the Refer-To header value in the REFER messages that are sent to the two remote parties.
Page 671: Emergency Call Parameters
Conference-initiating INVITE that is sent to the media server when Enable3WayConference is set to 1. When using the Mediant 1000 Media Processing Module (MPM): To join a conference, the INVITE URI must include the Conference ID string, preceded by the number of the participants in the conference, and terminated by a unique number.
Page 672: Call Cut-Through Parameters
Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Emergency Defines a list of “emergency” numbers. Numbers For FXS: When one of these numbers is dialed, the outgoing INVITE [EmergencyNumbers] message includes the SIP Priority and Resource-Priority headers. If the user places the phone on-hook, the call is not disconnected.
Page 673: Automatic Dialing Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description  [1] Enabled When enabled, this feature operates as follows: A Tel-to-IP call is established (connected) by the device for a B- channel. The device receives a SIP BYE (i.e., IP side ends the call) and plays a reorder tone to the PSTN side for the duration set by the CutThroughTimeForReOrderTone parameter.
Page 674: Direct Inward Dialing Parameters
Mediant 600 & Mediant 1000 Parameter Description (configured by the parameter HotLineToneDuration), the destination phone number is automatically dialed.  Module = Module number (where 1 denotes the module in Slot 1).  Port = Port number (where 1 denotes the Port 1 of the module).
Page 675 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web/EMS: Delay Before Defines the time interval (in msec) between detection of off-hook and DID Wink generation of a DID Wink. [DelayBeforeDIDWink] The valid range is 0 to 1,000. The default value is 0. Note: This parameter is applicable only to FXS interfaces.
Page 676: Mlpp Parameters
Mediant 600 & Mediant 1000 MLPP Parameters A.12.5.12 The Multilevel Precedence and Preemption (MLPP) parameters are described in the table below. Table A-52: MLPP Parameters Parameter Description Web/EMS: Call Priority Mode Enables priority calls handling. [CallPriorityMode]  [0] Disable = Disable (default).
Page 677 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [4] 4 = IMMEDIATE  [6] 6 = FLASH  [8] 8 = FLASH-OVERRIDE  [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 678 Mediant 600 & Mediant 1000 Parameter Description [MLPPDefaultServiceDomain] domain in the SIP Resource-Priority header in outgoing (Tel-to-IP calls) INVITE messages. This parameter is used in conjunction with the parameter SIPDefaultCallPriority. 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.
Page 679 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web/EMS: RTP DSCP for Defines the RTP DSCP for MLPP Routine precedence call level. MLPP Routine The valid range is -1 to 63. The default is -1. [MLPPRoutineRTPDSCP] 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).
Page 680: Isdn Bri Parameters
Mediant 600 & Mediant 1000 ISDN BRI Parameters A.12.5.13 The automatic dialing upon off-hook parameters are described in the table below. Table A-53: Automatic Dialing Parameters Parameter Description Web: ISDN Supp Services Table [ISDNSuppServ] This parameter table defines BRI phone extension numbers per BRI port and configures various ISDN supplementary services per BRI endpoint.
Page 681: Tty/Tdd Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Call Forward on No Reply Defines the prefix code for activating Call Forward on No Reply sent [SuppServCodeCFNR] to the softswitch. The valid value is a string. The default is an empty string. Note: The string must be enclosed in single apostrophe (e.g., ‘*72’).
Page 682: Pstn Parameters
Mediant 600 & Mediant 1000 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-55: General PSTN Parameters Parameter Description Web/EMS: Protocol Type Defines the PSTN protocol for all the Trunks. To configure the...
Page 683 SIP User's Manual A. Configuration Parameters Reference Parameter Description - HKT.  [21] E1 QSIG = ECMA 143 QSIG over E1  [22] E1 TNZ = ISDN PRI protocol for Telecom New Zealand (similar to ETSI)  [23] T1 QSIG = ECMA 143 QSIG over T1 ...
Page 684 Mediant 600 & Mediant 1000 Parameter Description If an outgoing call from the device to ISDN is not answered during this timeout, the call is released. The valid range is 10 to 240. The default value is 50. Notes: ...
Page 685 SIP User's Manual A. Configuration Parameters Reference Parameter Description [ClockMaster_x] Same as the description for parameter ClockMaster, but for a specific Trunk ID (where x denotes the Trunk ID and 0 is the first Trunk). Web/EMS: Line Code Selects B8ZS or AMI for T1 spans, and HDB3 or AMI for E1 spans. [LineCode] ...
Page 686 Mediant 600 & Mediant 1000 Parameter Description  For this parameter to take effect, a device reset is required.  When the device is locked from the Web interface, this parameter changes to 0.  To define the administrative state per trunk, use the TrunkAdministrativeState parameter.
Page 687: Tdm Bus And Clock Timing Parameters
SIP User's Manual A. Configuration Parameters Reference TDM Bus and Clock Timing Parameters A.12.6.2 The TDM Bus parameters are described in the table below. Table A-56: TDM Bus and Clock Timing Parameters Parameter Description TDM Bus Parameters Web/EMS: PCM Law Select Determines the type of pulse-code modulation (PCM) companding [PCMLawSelect] algorithm law in input and output TDM bus.
Page 688 Mediant 600 & Mediant 1000 Parameter Description Web: TDM Bus Fallback Clock Determines the fallback clock source on which the device Source synchronizes in the event of a clock failure. EMS: TDM Bus Fallback Clock  [4] Network (default) [TDMBusFallbackClock] ...
Page 689: Cas Parameters
SIP User's Manual A. Configuration Parameters Reference CAS Parameters A.12.6.3 The Common Channel Associated (CAS) parameters are described in the table below. Note that CAS is not applicable to BRI interfaces. Table A-57: CAS Parameters Parameter Description Web: CAS Transport Type Determines the ABCD signaling transport type over IP.
Page 690 Mediant 600 & Mediant 1000 Parameter Description DialPlanFileName = 'DialPlan_USA.dat' CASTrunkDialPlanName_5 = 'AT_T' [CASFileName_x] Defines the CAS file name (e.g., 'E_M_WinkTable.dat') that defines the CAS protocol, where x denotes the CAS file ID (0-7). It is possible to define up to eight different CAS files by repeating this parameter.
Page 691 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Generate Inter Digit Time Generates digit off-time (in msec). [CASStateMachineGenerateIn The value must be a positive value. The default value is -1. terDigitTime] Web: DTMF Max Detection Detects digit maximum on time (according to DSP detection Time information event) in msec units.
Page 692: Isdn Parameters
Mediant 600 & Mediant 1000 ISDN Parameters A.12.6.4 The ISDN parameters are described in the table below. Table A-58: ISDN Parameters Parameter Description Web: ISDN Termination Side Determines the ISDN termination side. EMS: Termination Side  [0] User side = ISDN User Termination Equipment (TE) side...
Page 693 SIP User's Manual A. Configuration Parameters Reference Parameter Description D-channel to control multiple PRI interfaces. Notes:  For this parameter to take effect, a device reset is required.  This parameter is applicable only to T1 ISDN protocols.  For more information on NFAS, see 'ISDN Non-Facility Associated Signaling (NFAS)' on page 248.
Page 694 Mediant 600 & Mediant 1000 Parameter Description (answer) message on incoming Tel calls.  [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 695 SIP User's Manual A. Configuration Parameters Reference Parameter Description unknown/unrecognized Facility IE. Otherwise, the Q.931 message that contains the unknown Facility IE is rejected. Note: This option is applicable only to ISDN variants where a complete ASN1 decoding is performed on Facility IE. ...
Page 696 Mediant 600 & Mediant 1000 Parameter Description maintenance. Notes:  To configure the device to support several ISDNIBehavior features, enter a summation of the individual feature values. For example, to support both [512] and [2048] features, set the parameter ISDNIBehavior is set to 2560 (i.e., 512 + 2048).
Page 697 SIP User's Manual A. Configuration Parameters Reference Parameter Description more flexible application control on the UUI. When this bit is not set (default behavior), CC implements the UUI-protocol as specified in the ETS 300-403 standards for Implicit Service 1.  [65536] GTD5 TBCT = CC implements the VERIZON-GTD-5 Switch variant of the TBCT Supplementary Service, as specified in FSD 01-02-40AG Feature Specification Document...
Page 698: Isdn And Cas Interworking Parameters
Mediant 600 & Mediant 1000 Parameter Description several ISDNOutCallsBehavior features, add the individual feature values. For example, to support both [2] and [16] features, set ISDNOutCallsBehavior = 18 (i.e., 2 + 16). [ISDNOutCallsBehavior_x] Same as the description for parameter ISDNOutCallsBehavior, but for a specific trunk ID.
Page 699 SIP User's Manual A. Configuration Parameters Reference Parameter Description the device sends its value (i.e. date and time) in the ISDN Connect Date / Time IE. If the 200 OK does not include this header, the device uses its internal, local date and time for the Date / Time IE, which it adds to the sent ISDN Connect message.
Page 700 Mediant 600 & Mediant 1000 Parameter Description MinOverlapDigitsForRouting parameter.  When option [2] is configured, even if SIP 4xx responses are received during this ISDN overlap receiving, the device does not release the call.  The MaxDigits parameter can be used to limit the length of the collected number for ISDN overlap dialing (if Sending Complete is not received).
Page 701 SIP User's Manual A. Configuration Parameters Reference Parameter Description messages are tunneled using SIP INFO, and ISDN Disconnect/Release message is tunneled using SIP BYE messages. The raw data from the ISDN is inserted into a proprietary SIP header (X-ISDNTunnelingInfo) or a dedicated message body (application/isdn) in the SIP messages.
Page 702 Mediant 600 & Mediant 1000 Parameter Description the QSIGTunnelingMode parameter.  Tunneling according to ECMA-355 is applicable to all ISDN variants (in addition to the QSIG protocol).  For more information on QSIG tunneling, see 'QSIG Tunneling' on page 241.
Page 703 SIP User's Manual A. Configuration Parameters Reference Parameter Description values are mapped to the outgoing SIP INVITE message's ‘isub’ parameter in accordance with RFC 4715. [IgnoreISDNSubaddress] Determines whether the device ignores the Subaddress from the incoming ISDN Called and Calling numbers when sending to IP.
Page 704 Mediant 600 & Mediant 1000 Parameter Description Web: Digital Out-Of-Service Determines the method for setting digital trunks to Out-Of- Behavior Service state per trunk. EMS: Digital OOS Behavior For  [-1] Not Configured = Use the settings of the Trunk Value DigitalOOSBehavior parameter for per device (default).
Page 705 SIP User's Manual A. Configuration Parameters Reference Parameter Description For example: CauseMapISDN2SIP 0 = 50,480; CauseMapISDN2SIP 0 = 6,406; When a Release Cause is received (from the PSTN side), the device searches this mapping table for a match. If the Q.850 Release Cause is found, the SIP response assigned to it is sent to the IP side.
Page 706 Mediant 600 & Mediant 1000 Parameter Description  [0] = Format: X-UserToUser (default).  [1] = Format: User-to-User with Protocol Discriminator (pd) attribute. User-to- User=3030373435313734313635353b313233343b3834;pd =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.
Page 707 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [1] PI = 1; [8] PI = 8: Sends a 183 response to IP. EMS: Connect On Progress Ind Enables the play of announcements from IP to PSTN without [ConnectOnProgressInd] the need to answer the Tel-to-IP call.
Page 708 Mediant 600 & Mediant 1000 Parameter Description Note: To enable this feature, the parameter ISDNDuplicateQ931BuffMode must be set to 1. [CallReroutingMode] Determines whether ISDN call rerouting (call forward) is performed by the PSTN instead of by the SIP side. This call forwarding is based on Call Deflection for Euro ISDN (ETS-300- 207-1) and QSIG (ETSI TS 102 393).
Page 709 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: IPMedia Detectors Enables the device's DSP detectors. EMS: DSP Detectors Enable  [0] = Disable (default). [EnableDSPIPMDetectors]  [1] = Enable. Notes:  For this parameter to take effect, a device reset is required. ...
Page 710 Mediant 600 & Mediant 1000 Parameter Description INFO to User Information, SIP 18x to Alerting, and SIP BYE to Disconnect. Notes:  The interworking of ISDN User-to-User IE to SIP INFO is applicable only to the Euro ISDN, QSIG, and 4ESS PRI variants.
Page 711 SIP User's Manual A. Configuration Parameters Reference Parameter Description The valid values of this parameter are described below:  [0] = Not supported (default).  [1] = Supports CAS NFA DMS-100 transfer. When a SIP REFER message is received, the device performs a Blind Transfer by executing a CAS Wink, waits for an acknowledged Wink from the remote side, dials the Refer-to number to the switch, and then releases the call.
Page 712 Mediant 600 & Mediant 1000 Parameter Description IP to Tel Routing table is the same Trunk Group as the original call, then the device performs the in-band DTMF transfer; otherwise, the device sends the INVITE according to regular transfer rules.
Page 713 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [0] = (Default) ISDN Transfer Capability for data calls is 64k unrestricted (data).  [1] = ISDN Transfer Capability for data calls is determined according to the ISDNTransferCapability parameter. Web: ISDN Transfer On Connect This parameter is used for the ECT/TBCT/RLT/Path EMS: Send ISDN Transfer On Replacement ISDN transfer methods.
Page 714 Mediant 600 & Mediant 1000 Parameter Description A places B on hold, and calls C; C answers the call. A performs a call transfer (the transfer is done internally by the PBX); B and C are connected to one another.
Page 715: Answer And Disconnect Supervision Parameters
SIP User's Manual A. Configuration Parameters Reference A.12.8 Answer and Disconnect Supervision Parameters The answer and disconnect supervision parameters are described in the table below. Table A-60: Answer and Disconnect Parameters Parameter Description Web: Answer Supervision Enables the sending of SIP 200 OK upon detection of speech, EMS: Enable Voice Detection fax, or modem.
Page 716 Mediant 600 & Mediant 1000 Parameter Description BrokenConnectionEventTimeout parameter.  This feature is applicable only if the RTP session is used without Silence Compression. If Silence Compression is enabled, the device doesn't detect a broken RTP connection.  During a call, if the source IP address (from where the RTP packets are received) is changed without notifying the device, the device filters these RTP packets.
Page 717 SIP User's Manual A. Configuration Parameters Reference Parameter Description according to packet count (FarEndDisconnectSilenceMethod is set to 1).  For this parameter to take effect, a device reset is required. [BrokenConnectionDuringSilen Enables the generation of the BrokenConnection event during a silence period if the channel’s NoOp feature is enabled (using the parameter NoOpEnable) and if the channel stops receiving NoOp RTP packets.
Page 718 Mediant 600 & Mediant 1000 Parameter Description Typically, if the RJ-11 cabling is connected correctly (without crossing, Tip to Tip, Ring to Ring), the Tip line is positive compared to the Ring line. In this case, set this parameter to 0.
Page 719: Tone Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description  For this parameter to take effect, a device reset is required. EMS: Current Disconnect Defines the duration (in msec) of the current disconnect pulse. Duration The range is 200 to 1500. The default is 900. [CurrentDisconnectDuration] Notes: ...
Page 720 Mediant 600 & Mediant 1000 Parameter Description hold. The Held tone must be configured in the CPT or PRT file. Note: This parameter is applicable only to the IP-to-IP application (enabled using the parameter EnableIP2IPApplication). Web/EMS: Dial Tone Defines the duration (in seconds) that the dial tone is played (for digital Duration [sec] interfaces, to an ISDN terminal).
Page 721 SIP User's Manual A. Configuration Parameters Reference Parameter Description [TimeForReorderTone] The valid range is 0 to 254. The default is 0 seconds. Typically, after playing a Reorder/Busy tone for the specified duration, the device starts playing an Offhook Warning tone. For Digital: Defines the duration (in seconds) that the CAS device plays a Busy or Reorder Tone before releasing the line.
Page 722 Mediant 600 & Mediant 1000 Parameter Description the PSTN. This is applicable only if the call is released from the IP [Busy Here (486) or Not Found (404)] before it reaches the Connect state; otherwise, the Disconnect message is sent immediately and no tones are played.
Page 723 SIP User's Manual A. Configuration Parameters Reference Parameter Description SIP183Behaviour is set to 1). If SIP183Behaviour is set to 1 (183 is handled the same way as a 180 with SDP), the device sends an Alert message with PI = 8 without playing a ringback tone. ...
Page 724 Mediant 600 & Mediant 1000 Parameter Description SDP message is received, the device cuts through the voice channel and doesn't play the ringback tone.  Digital Interfaces: Plays according to 'Early Media'. If a SIP 180 response is received and the voice channel is already open (due...
Page 725 SIP User's Manual A. Configuration Parameters Reference Parameter Description signaling to the calling party to open a voice channel to hear the played ringback tone. Notes:  To enable the device to send a 183/180+SDP responses, set the EnableEarlyMedia parameter to 1. ...
Page 726: Tone Detection Parameters
Mediant 600 & Mediant 1000 Parameter Description (default is 0):  Ringing tone index = index in the CPT file for playing the ring tone.  Call Waiting tone index = priority index + FirstCallWaitingToneID(*). For example, if you want to select the...
Page 727 SIP User's Manual A. Configuration Parameters Reference Parameter Description To disconnect IP-to-ISDN calls when a SIT tone is detected, the following parameters must be configured:  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...
Page 728: Metering Tone Parameters
Mediant 600 & Mediant 1000 Metering Tone Parameters A.12.9.3 The metering tone parameters are described in the table below. Table A-63: 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 729: Telephone Keypad Sequence Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description For example: ChargeCode 1 = 7,30,1,14,20,2,20,15,1,0,60,1; ChargeCode 2 = 5,60,1,14,20,1,0,60,1; ChargeCode 3 = 0,60,1; ChargeCode 0 = 6, 3, 1, 12, 2, 1, 18, 5, 2, 0, 2, 1; Notes:  The parameter can include up to 25 indices (i.e., up to 25 different metering rules can be defined).
Page 730 Mediant 600 & Mediant 1000 Parameter Description Hook Flash Parameters Web: Flash Keys Sequence Determines the hook-flash key sequence for FXS interfaces. Style  [0] 0 = Flash hook (default) - only the phone's Flash button is [FlashKeysSequenceStyle] used, according to the following scenarios: ...
Page 731 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Do Not Disturb Defines the keypad sequence to activate the Do Not Disturb option EMS: CF Do Not Disturb (immediately reject incoming calls). [KeyCFDoNotDisturb] To activate the required forward method from the telephone: Dial the user-defined sequence number on the keypad;...
Page 732: General Fxo Parameters
Mediant 600 & Mediant 1000 Parameter Description Web: Deactivate Defines the keypad sequence that de-activate the reject anonymous EMS: Reject Anonymous Call call option. After the sequence is pressed, a confirmation tone is Deact heard. [KeyRejectAnonymousCallD eact] [RejectAnonymousCallPerP This parameter table determines whether the device rejects incoming ort] anonymous calls on FXS interfaces.
Page 733 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [0] = FXO line current limit is disabled (default).  [1] = FXO loop current is limited to a maximum of 60 mA. Note: For this parameter to take effect, a device reset is required. [FXONumberOfRings] Defines the number of rings before the device's FXO interface answers a call by seizing the line.
Page 734 Mediant 600 & Mediant 1000 Parameter Description Web: Time to Wait before For digital interfaces: Defines the delay after hook-flash is Dialing [msec] generated and until dialing begins. Applies to call transfer (i.e., the EMS: Time Before Dial parameter TrunkTransferMode is set to 3) on CAS protocols.
Page 735: Fxs Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: FXO Double Answer Enables the FXO Double Answer feature, which rejects [EnableFXODoubleAnswer] (disconnects) incoming Tel (FXO)-to-IP collect calls and signals (informs) this call denial to the PSTN.  [0] Disable (default) ...
Page 736 Mediant 600 & Mediant 1000 Parameter Description  For configuring this table in the Web interface, see Configuring Trunk Group Table on page 251.  For a description of ini file table parameters, see 'Configuring ini File Table Parameters' on page 84.
Page 737 SIP User's Manual A. Configuration Parameters Reference Parameter Description starts ascending again.  [2] Ascending = Selects the lowest available channel in the Trunk Group and if unavailable, selects the next higher channel.  [3] Cyclic Descending = Selects the next available channel in descending cyclic order.
Page 738 Mediant 600 & Mediant 1000 Parameter Description the call is sent to that channel. If the number is not located or the channel is unavailable (e.g., busy), the device searches, in ascending order, for the next available channel in the Trunk Group. If located, the call is sent to that channel.
Page 739 SIP User's Manual A. Configuration Parameters Reference Parameter Description Source Number and also as the Tel Display Name, and Presentation is set to Allowed (0). If no Display Name is received from IP, the IP Source Number is used as the Tel Source Number and Presentation is set to Restricted (1).
Page 740 Mediant 600 & Mediant 1000 Parameter Description [PSTNPrefix] FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix, PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix, PstnPrefix_SourceAddress, PstnPrefix_ProfileId, PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix, PstnPrefix_SrcHostPrefix; [\PSTNPrefix] For example: PstnPrefix 0 = 100, 1, 200, *, 0, 2, , ; PstnPrefix 1 = *, 2, *, , 1, 3, acl, joe;...
Page 741 SIP User's Manual A. Configuration Parameters Reference Parameter Description defined in the Proxy Set table. All other incoming calls are rejected.  [2] Secure All calls = The device accepts SIP calls only from IP addresses (in dotted-decimal notation format) that are defined in the Outbound IP Routing Table table or Proxy Set table, and rejects all other incoming calls.
Page 742 Mediant 600 & Mediant 1000 Parameter Description Web: Add CIC Determines whether to add the Carrier Identification Code (CIC) as [AddCicAsPrefix] a prefix to the destination phone number for IP-to-Tel calls.  [0] No (default)  [1] Yes When this parameter is enabled, the cic parameter in the incoming SIP INVITE can be used for IP-to-Tel routing decisions.
Page 743: Ip Connectivity Parameters
SIP User's Manual A. Configuration Parameters Reference A.12.14 IP Connectivity Parameters The IP connectivity parameters are described in the table below. Table A-68: IP Connectivity Parameters Parameter Description Web: Alt Routing Tel to IP Mode Determines the IP connectivity event(s) reason for triggering EMS: Alternative Routing Mode alternative routing.
Page 744: Alternative Routing Parameters
Mediant 600 & Mediant 1000 A.12.15 Alternative Routing Parameters The alternative routing parameters are described in the table below. Table A-69: 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 745 SIP User's Manual A. Configuration Parameters Reference Parameter Description FORMAT AltRouteCauseTel2IP_Index = AltRouteCauseTel2IP_ReleaseCause; [\AltRouteCauseTel2IP] For example: AltRouteCauseTel2IP 0 = 486; (Busy Here) AltRouteCauseTel2IP 1 = 480; (Temporarily Unavailable) AltRouteCauseTel2IP 2 = 408; (No Response) Notes:  This parameter can include up to 5 indices. ...
Page 746: Alternative Routing Parameters
Mediant 600 & Mediant 1000 Parameter Description alternative route is used. This tone is played for a user- defined time (configured by the parameter AltRoutingToneDuration).  For configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page 84.
Page 747 SIP User's Manual A. Configuration Parameters Reference Parameter Description AltRouteCauseTEL2IP (Reasons for Alternative Routing table). Web: Enable Alt Routing Tel to IP Enables the Alternative Routing feature for Tel-to-IP calls. EMS: Enable Alternative Routing  [0] Disable = Disables the Alternative Routing feature [AltRoutingTel2IPEnable] (default).
Page 748 Mediant 600 & Mediant 1000 Parameter Description Web: Alt Routing Tel to IP Keep Defines the time interval (in seconds) between SIP OPTIONS Alive Time Keep-Alive messages used for the IP Connectivity application. EMS: Alternative Routing Keep The valid range is 5 to 2,000,000. The default value is 60.
Page 749 SIP User's Manual A. Configuration Parameters Reference Parameter Description  For configuring ini file table parameters, see 'Configuring ini File Table Parameters' on page Reasons for Alternative IP-to-Tel Routing Table Web: Reasons for Alternative IP-to- This parameter table configures call failure reason values Tel Routing received from the PSTN side (in Q.931 presentation).
Page 750: Number Manipulation Parameters
Mediant 600 & Mediant 1000 Parameter Description unavailable: ForwardOnBusyTrunkDest 1 = 2, 112@10.13.4.12:5060;transport=tcp; When configured with user@host, the original destination number is replaced by the user part. Notes:  The maximum number of indices (starting from 1) depends on the maximum number of Trunk Groups.
Page 751 SIP User's Manual A. Configuration Parameters Reference Parameter Description Use EndPoint Number As Calling Enables the use of the B-channel number as the calling number Number Tel2IP (sent in the From field of the INVITE) instead of the number [UseEPNumAsCallingNumTel2IP received in the Q.931 Setup message, for Tel-to-IP calls.
Page 752 Mediant 600 & Mediant 1000 Parameter Description Web: Set IP-to-TEL Redirect Defines the redirect reason for IP-to-Tel calls. If redirect Reason (diversion) information is received from the IP, the redirect [SetIp2TelRedirectReason] reason is set to the value of this parameter before the device sends it on to the Tel.
Page 753 SIP User's Manual A. Configuration Parameters Reference Parameter Description Web: Send Screening Indicator to Overrides the screening indicator of the calling party's number ISDN for IP-to-Tel ISDN calls. EMS: Screening Indicator To ISDN  [-1] Not Configured = Not configured (interworking from IP to [ScreeningInd2ISDN] ISDN) (default).
Page 754 Mediant 600 & Mediant 1000 Parameter Description outgoing INVITE message excludes the redirect number that was used to replace the calling number. The replacement is done only if a redirect number is present in the incoming Tel call.  [2] = Manipulation is done on the new calling party number...
Page 755 SIP User's Manual A. Configuration Parameters Reference Parameter Description Number Type of Numbering (TON) are added to the Calling Number for EMS: Add NPI And TON As Prefix Tel-to-IP calls. To Calling Number  [0] No = Do not change the Calling Number (default). [AddNPIandTON2CallingNumber ...
Page 756 Mediant 600 & Mediant 1000 Parameter Description  [0] No = Don't change numbers (default). Web/EMS: Swap Redirect and Called Numbers  [1] Yes = Incoming ISDN call that includes a redirect number [SwapRedirectNumber] (sometimes referred to as 'original called number') uses the redirect number instead of the called number.
Page 757 SIP User's Manual A. Configuration Parameters Reference Parameter Description CallingNameMapIp2Tel_Suffix2Add; [ \CallingNameMapIp2Tel ] Calling Name Manipulations Tel-to-IP Table [CallingNameMapTel2Ip] Configures rules for manipulating the calling name (caller ID) for Tel-to-IP calls. This can include modifying or removing the calling name. [ CallingNameMapTel2Ip ] FORMAT CallingNameMapTel2Ip_Index = CallingNameMapTel2Ip_DestinationPrefix,...
Page 758 Mediant 600 & Mediant 1000 Parameter Description  '*' (asterisk): represents any number between 0 and 255. For example, 10.8.8.* represents addresses between 10.8.8.0 and 10.8.8.255.  The following parameteris not applicable: IsPresentationRestricted.  To configure manipulation of destination numbers for IP-to- Tel calls using the Web interface, see 'Configuring Number Manipulation Tables' on page 256).
Page 759 SIP User's Manual A. Configuration Parameters Reference Parameter Description  The following parameters are not applicable: SourceAddress and IsPresentationRestricted.  To configure manipulation of destination numbers for Tel-to- IP calls using the Web interface, see 'Configuring the Number Manipulation Tables' on page 256). ...
Page 760 Mediant 600 & Mediant 1000 Parameter Description parameter). This enables you to configure only a few manipulation rules for complex number manipulation requirements (that generally require many rules).  [0] = Disable (default)  [1] = Enable Source Phone Number Manipulation for Tel-to-IP Calls Table...
Page 761 SIP User's Manual A. Configuration Parameters Reference Parameter Description  9,0 = Private, Unknown  9,1 = Private, Level 2 Regional  9,2 = Private, Level 1 Regional  9,3 = Private, PISN Specific  9,4 = Private, Level 0 Regional (local) ...
Page 762 Mediant 600 & Mediant 1000 Parameter Description manipulation has been performed on it. Redirect Number Tel-to-IP Table Web: Redirect Number Tel -> IP This parameter table manipulates the redirect number for Tel-to- EMS: Redirect Number Map Tel to IP calls. The manipulated Redirect Number is sent in the SIP Diversion, History-Info, or Resource-Priority headers.
Page 763: Ldap Parameters
SIP User's Manual A. Configuration Parameters Reference Parameter Description PhoneContext_Ton, PhoneContext_Context; [\PhoneContext] For example: PhoneContext 0 = 0,0,unknown.com PhoneContext 1 = 1,1,host.com PhoneContext 2 = 9,1,na.e164.host.com Notes:  This parameter can include up to 20 indices.  Several entries with the same NPI-TON or Phone-Context are allowed.
Page 764 Mediant 600 & Mediant 1000 Parameter Description For example: LDAPBindDN = "CN=Search user,OU=Labs,DC=OCSR2,DC=local" Web: LDAP Search Dn Defines the search DN for LDAP search requests. This is [LDAPSearchDN] the top DN of the subtree where the search is performed. This parameter is mandatory for the search.
Page 765: Least Cost Routing Parameters
SIP User's Manual A. Configuration Parameters Reference A.12.19 Least Cost Routing Parameters The Least Cost Routing parameters are described in the table below. Table A-73: LCR Parameters Parameter Description Web: Routing Rule Groups This parameter table enables the LCR feature and configures the Table average call duration and default call cost.
Page 766 Mediant 600 & Mediant 1000 Parameter Description calls internal to the local network or outgoing to PSTN. Note: For this parameter to take effect, a device reset is required. Web: SAS Local SIP UDP Port Defines the local UDP port for sending and receiving SIP EMS: Local SIP UDP messages for SAS.
Page 767 SIP User's Manual A. Configuration Parameters Reference Parameter Description [SASProxySet] that are served by the SAS application. The valid range is 0 to 5. The default value is 0 (i.e., default Proxy Set). Web: Redundant SAS Proxy Set Defines the Proxy Set (index number) used in SAS Emergency EMS: Redundant Proxy Set mode for fallback when the user is not found in the Registered [RedundantSASProxySet]...
Page 768 Mediant 600 & Mediant 1000 Parameter Description a Proxy), and enters the registrations in its SAS database.  [4] Use Routing Table only in Normal mode = The device uses the IP-to-IP Routing table to route IP-to-IP SAS calls only when in SAS Normal mode (and is unavailable when SAS is in Emergency mode).
Page 769 SIP User's Manual A. Configuration Parameters Reference Parameter Description SAS Registration Manipulation Table Web: SAS Registration This parameter table configures the SAS Registration Manipulation Manipulation table. This table is used by the SAS application to EMS: Stand-Alone Survivability manipulate the SIP Request-URI user part of incoming INVITE [SASRegistrationManipulation] messages and of incoming REGISTER request AoR (To header), before saving it to the registered users database.
Page 770: Ip Media Parameters
[0] = Disable (default)  [1] = Enable Notes:  This parameter is applicable only to Mediant 1000.  For this parameter to take effect, a device reset is required. [IPmediaChannels] This ini file parameter table defines the number of DSP channels that are "borrowed"...
Page 771 IPMediaChannels 1 = 1, 15; IPMediaChannels 2 = 2, 10; Notes:  This parameter is applicable only to Mediant 1000.  The value of DSPChannelsReserved must be in multiples of 5 (since the reservation is done per DSP device and not per DSP channel).
Page 772 Mediant 600 & Mediant 1000 Parameter Description The valid value can be up to 16 characters. The default is "callingnumber". Note: The APS server support must be enabled to support this feature. Below are the relevant ini file parameter settings: ...
Page 773 SIP User's Manual A. Configuration Parameters Reference Parameter Description [AMSAllowUrlAsAlias] Determines whether or not play requests for remote URLs are first verified with local audio segments to determine if any have an alias matching for the URL. If a match is found, the corresponding local audio segment is played.
Page 774 Mediant 600 & Mediant 1000 Parameter Description Note: For this parameter to take effect, a device reset is required. Web: End of Record Trim Defines the maximum amount (in milliseconds) of audio to remove [cpEndOfRecordCutTime] from the end of a recording. This is used to remove the DTMF signals generated by the end user for terminating the record.
Page 775 SIP User's Manual A. Configuration Parameters Reference Parameter Description  [15] 15 = 6.00 dB/sec  [16] 16 = 7.00 dB/sec  [17] 17 = 8.00 dB/sec  [18] 18 = 9.00 dB/sec  [19] 19 = 10.00 dB/sec  [20] 20 = 11.00 dB/sec ...
Page 776 Mediant 600 & Mediant 1000 Parameter Description IPProfile parameter (see Configuring IP Profiles on page 219). Web: Answer Machine Defines the AMD detection sensitivity level of the selected AMD Detector Sensitivity Level Parameter Suite. [AMDSensitivityLevel] The valid value range is 0 (for best detection of an answering machine) to 15 (for best detection of a live call).
Page 777 SIP User's Manual A. Configuration Parameters Reference Parameter Description EMS: Time Out Defines the timeout (in msec) between receiving Connect messages [AMDTimeout] from the ISDN and sending AMD results. The valid range is 1 to 30,000. The default is 2,000 (i.e., 2 seconds). Web/EMS: AMD Beep Determines the AMD beep detection mode.
Page 778 Mediant 600 & Mediant 1000 Parameter Description Web: Enable Pattern Detector Enables the Pattern Detector (PD) feature. [EnablePatternDetector]  [0] Disable (default)  [1] Enable [PDPattern] Defines the patterns that can be detected by the Pattern Detector. The valid range is 0 to 0xFF.
Page 779 SIP User's Manual A. Configuration Parameters Reference Parameter Description The default value is 0 (i.e., don't set this parameter on recognition attempt). Note: For this parameter to take effect, a device reset is required. [VxmlInterDigitTimeout] Defines the inter-digit timeout value (in msec) used when DTMF is received.
Page 780 Mediant 600 & Mediant 1000 Parameter Description [VxmlSystemInputModes] Determines which inputs are valid for grammars.  [0] = DTMF is valid (default)  [1] = Voice is valid  [2] = Both are valid Note: For this parameter to take effect, a device reset is required.
Page 781: Auxiliary And Configuration Files Parameters
Prompts. Notes:  For this parameter to take effect, a device reset is required.  This parameter is applicable only to Mediant 1000.  For more information on this file, see Voice Prompts File on page 491. Web/EMS: Prerecorded Tones...
Page 782 Mediant 600 & Mediant 1000 Parameter Description Web: CAS File Defines the CAS file name (e.g., 'E_M_WinkTable.dat'), which EMS: Trunk Cas Table Index defines the CAS protocol (where x denotes the CAS file ID 0 to 7). [CASFileName_x] It is possible to define up to eight different CAS files by repeating this parameter.
Page 783: Automatic Update Parameters
SIP User's Manual A. Configuration Parameters Reference A.15.2 Automatic Update Parameters The automatic update of software and configuration files parameters are described in the table below. Table A-77: Automatic Update of Software and Configuration Files Parameters Parameter Description General Automatic Update Parameters [AutoUpdateCmpFile] Enables the Automatic Update mechanism for the cmp file.
Page 784  The maximum length of the URL address is 99 characters.  This parameter is applicable only to Mediant 1000. [CasFileURL] Defines the name of the CAS file and the path to the server (IP address or FQDN) on which it is located.
Page 785 SIP User's Manual A. Configuration Parameters Reference Parameter Description [TLSRootFileUrl] Defines the name of the TLS trusted root certificate file and the URL from where it can be downloaded. Note: For this parameter to take effect, a device reset is required. Defines the name of the TLS certificate file and the URL from where [TLSCertFileUrl] it can be downloaded.
Page 786 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 787: B Dialing Plan Notation For Routing And Manipulation
SIP User's Manual B. Dialing Plan Notation for Routing and Manipulation Dialing Plan Notation for Routing and Manipulation The device supports flexible dialing plan notations for denoting the prefix and/or suffix source and/or destination numbers and SIP URI user names in the routing and manipulation tables.
Page 788 Mediant 600 & Mediant 1000 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 denote a one-digit number starting with 2, 3, 4, 5, or 6: ...
Page 789: Csip Message Manipulation Syntax
SIP User's Manual C. SIP Message Manipulation Syntax SIP Message Manipulation Syntax This section provides a detailed description on the support and syntax for configuring SIP message manipulation rules. For configuring message manipulation rules, see the parameter MessageManipulations. Actions The actions that can be done on SIP message manipulation in the Message Manipulations table are listed in the table below.
Page 790: Accept-Language
Mediant 600 & Mediant 1000 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...
Page 791: Contact
SIP User's Manual C. SIP Message Manipulation Syntax Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes String Read Only Below is a header manipulation example: Rule: Add a proprietary header to all INVITE messages using the data in the Call-id header: MessageManipulations 0 = 1, invite, , header.Xitsp-abc, 0, header.call-id, 0;...
Page 792: Diversion
Mediant 600 & Mediant 1000 Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Integer Read Only Type String Read Only Below is a header manipulation example: If the Cseq number is 1, then modify the user in the Contact header to fred.
Page 793: Event
SIP User's Manual C. SIP Message Manipulation Syntax MessageManipulations 1 = 1, invite, , header.Diversion.reason, 2, '1', 0; Diversion: <tel:+101>;reason=user- Result: busy;screen=no;privacy=off;counter=1 Example 3 Rule: The URL in the Diversion header is modified to that which is contained in the header URL: MessageManipulations 2 = 1, invite, , header.Diversion.URL, 2, header.from.url, 0;...
Page 794: History-Info
Mediant 600 & Mediant 1000 Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Name String Read/Write Param Param Read/Write String Read Only URL Structure (refer to Read/Write 'URL' on page 815) Below are header manipulation examples:...
Page 795: Min-Se And Min-Expires
SIP User's Manual C. SIP Message Manipulation Syntax 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: History-Info:sip:UserA@audc.example.com;index=2 History-Info: <sip:UserA@audc.mydomain.com;index=3> Example 2 Rule: Delete an unwanted History-Info header from the message:...
Page 796: P-Asserted-Identity
Mediant 600 & Mediant 1000 C.2.12 P-Asserted-Identity An example of the header is shown below: P-Asserted-Identity: Jane Doe <sip:567@itsp.com> The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types...
Page 797: P-Called-Party-Id
SIP User's Manual C. SIP Message Manipulation Syntax 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; P-Associated-URI:<sip:admin@10.132.10.108> Result: Example 2 Rule: Modify the user portion of the URL in the header to 'alice': MessageManipulations 5 = 1, register.response, ,header.P-Associated-URI.url.user, 2, 'alice', 0;...
Page 798: P-Charging-Vector
Mediant 600 & Mediant 1000 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 Sub Types...
Page 799: Privacy
SIP User's Manual C. SIP Message Manipulation Syntax P-Preferred-Identity: "Alice Biloxi" Result: <sip:fluffy@abc.com> C.2.17 Privacy An example of the header is shown below: Privacy: none The header properties are shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types...
Page 800: Reason
Mediant 600 & Mediant 1000 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 801: Referred-By
SIP User's Manual C. SIP Message Manipulation Syntax 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 802: Remote-Party-Id
Mediant 600 & Mediant 1000 Keyword Sub Types Attributes Below are header manipulation examples: Add a basic header: Exam Rule ple 1 MessageManipulations 0 = 1, any, ,header.Refer-to, 0, '<sip:referto@referto.com>', 0; Refer-To: <sip:referto@referto.com> ult: Add a Refer-To header with URI headers:...
Page 803: Request-Uri
SIP User's Manual C. SIP Message Manipulation Syntax 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 804: Require
Mediant 600 & Mediant 1000 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;...
Page 805: Resource-Priority
SIP User's Manual C. SIP Message Manipulation Syntax 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 , 0; Require: em,replaces,early-session, privacy Result: C.2.25 Resource-Priority An example of the header is shown below: Resource-Priority: wps.3 The header properties are shown in the table below:...
Page 806: Server Or User-Agent
Mediant 600 & Mediant 1000 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...
Page 807: Session-Expires
SIP User's Manual C. SIP Message Manipulation Syntax 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 808: Subject
Mediant 600 & Mediant 1000 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...
Page 809: C.2.32 To
SIP User's Manual C. SIP Message Manipulation Syntax 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 810: Unsupported
Mediant 600 & Mediant 1000 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...
Page 811: Warning
SIP User's Manual C. SIP Message Manipulation Syntax Keyword Sub Types Attributes Port Integer Read Only TransportType Enum TransportType (see Read Only 'TransportType' on page 824) 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 812: Unknown Header
Mediant 600 & Mediant 1000 C.2.36 Unknown Header An Unknown header is a SIP header that is not included in this list of supported headers. An example of the header is shown below: MYEXP: scooby, doo, goo, foo The header properties are shown in the table below:...
Page 813: Structure Definitions
SIP User's Manual C. SIP Message Manipulation Syntax Structure Definitions C.3.1 Event Structure The Event structure is used in the Event header (see 'Event' on page 793). Table C-2: Event Structure Keyword Sub Types Attributes EventPackage Enum Event Package (see Read/Write 'Event Package' on page 818)
Page 814: Reason Structure
Mediant 600 & Mediant 1000 Keyword Sub Types HEADER Boolean SESSION Boolean USER Boolean CRITICAL Boolean IDENTITY Boolean HISTORY Boolean C.3.5 Reason Structure This structure is applicable to the Reason header (see 'Reason' on page 800). Table C-6: Reason Structure...
Page 815: Url
SIP User's Manual C. SIP Message Manipulation Syntax C.3.7 This structure is applicable to the following headers:  Contact (see 'Contact' on page 791)  Diversion (see 'Diversion' on page 792)  From (see 'From' on page 793)  P-Asserted-Identity (see 'P-Asserted-Identity' on page 796) ...
Page 816: Random Type
Mediant 600 & Mediant 1000 Random Type Manipulation rules can include random strings and integers. An example of a manipulation rule using random values is shown below: MessageManipulations 4 = 1, Invite.Request, , Header.john, 0, rand.string.56.A.Z, 0; In this example, a header called "john" is added to all INVITE messages received by the device and a random string of 56 characters containing characters A through Z is added to the header.
Page 817: Copying Information Between Messages Using Variables
SIP User's Manual C. SIP Message Manipulation Syntax Copying Information between Messages using Variables You can use variables in SIP message manipulation rules to copy specific information from one message to another. Information from one message is copied to a variable and then information from that variable is copied to any subsequent message.
Page 818: Enum Definitions
Mediant 600 & Mediant 1000 Enum Definitions C.7.1 AgentRole These ENUMs are applicable to the Server or User-Agent headers (see 'Server or User- Agent' on page 806). Table C-9: Enum Agent Role AgentRole Value Client Server C.7.2 Event Package These ENUMs are applicable to the Server or User-Agent (see 'Server or User-Agent' on page 806) and Event (see 'Event' on page 793) headers.
Page 819: Mlpp Reason Type
SIP User's Manual C. SIP Message Manipulation Syntax C.7.3 MLPP Reason Type These ENUMs are applicable to the MLPP Structure (see 'MLPP' on page 813). 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 802).
Page 820: Privacy
Mediant 600 & Mediant 1000 C.7.6 Privacy These ENUMs are applicable to the Remote-Party-Id (see 'Remote-Party-Id' on page 802) and Diversion (see 'Diversion' on page 792) 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 792).
Page 821 SIP User's Manual C. SIP Message Manipulation Syntax 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...
Page 822 Mediant 600 & Mediant 1000 Reason Value UNKNOWN_RESOURCE_PRIORITY_417 BAD_EXTENSION_420 EXTENSION_REQUIRED_421 SESSION_INTERVAL_TOO_SMALL_422 SESSION_INTERVAL_TOO_SMALL_423 ANONYMITY_DISALLOWED_433 UNAVAILABLE_480 TRANSACTION_NOT_EXIST_481 LOOP_DETECTED_482 TOO_MANY_HOPS_483 ADDRESS_INCOMPLETE_484 AMBIGUOUS_485 BUSY_486 REQUEST_TERMINATED_487 NOT_ACCEPTABLE_HERE_488 BAD_EVENT_489 REQUEST_PENDING_491 UNDECIPHERABLE_493 SECURITY_AGREEMENT_NEEDED_494 SERVER_INTERNAL_ERROR_500 NOT_IMPLEMENTED_501 BAD_GATEWAY_502 SERVICE_UNAVAILABLE_503 SERVER_TIME_OUT_504 VERSION_NOT_SUPPORTED_505 MESSAGE_TOO_LARGE_513 PRECONDITION_FAILURE_580 BUSY_EVERYWHERE_600 DECLINE_603 DOES_NOT_EXIST_ANYWHERE_604 NOT_ACCEPTABLE_606 SIP User's Manual...
Page 823: Reason (Remote-Party-Id)
SIP User's Manual C. SIP Message Manipulation Syntax C.7.9 Reason (Remote-Party-Id) These ENUMs are applicable to the Remote-Party-Id header (see 'Remote-Party-Id' on page 802). 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 807).
Page 824: Transporttype
Mediant 600 & Mediant 1000 C.7.13 TransportType These ENUMs are applicable to the URL Structure (see 'URL' on page 815) and the Via header (see 'Via' on page 810). Table C-21: Enum TransportType TransportType Value SCTP C.7.14 Type These ENUMs are applicable to the URL Structure (see 'URL' on page 815).
Page 825 SIP User's Manual C. SIP Message Manipulation Syntax Element Comman Command Value Type Remarks Type d Type contains String Returns true if the string given is found in the parameter value. !contains String Returns true if the string given is not found in the parameter value.
Page 826 Mediant 600 & Mediant 1000 Element Comman Command Value Type Remarks Type d Type String Returns true if a header not equals to the value. The header element must not be a list. *Header contains String Returns true if the header contains the string.
Page 827 SIP User's Manual C. SIP Message Manipulation Syntax Element Comman Command Value Type Remarks Type d Type Parameter Match String Returns true if the header’s parameter’s value equals to the value. Parameter String Returns true if the header’s parameter’s value not equals to the value.
Page 828 Mediant 600 & Mediant 1000 Element Comman Command Value Type Remarks Type d Type String Match String Returns true if the string element equals to the value. String Returns true if the string element not equals to the value. contains...
Page 829 SIP User's Manual C. SIP Message Manipulation Syntax Element Comman Command Value Type Remarks Type d Type Action Modify Boolean Sets the Boolean element to the value. Boolean – can be either 0 or 1. Attribute Match Integer Returns true if the attribute element equals to the value.
Page 830: Syntax
Mediant 600 & Mediant 1000 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>...
Page 831 SIP User's Manual C. SIP Message Manipulation Syntax • header.john exists • header.john exists AND header.to.host !contains 'john' • header.from.user == '100' OR header.from.user == '102' OR header.from.user == '300' match-type Description: Comparison to be made Syntax: ♦ equals ♦ not equals ♦...
Page 832 Mediant 600 & Mediant 1000 ♦ fifth Via header sub-element Description: Header's element Syntax: sub-element-name Examples: ♦ user ♦ host sub-element-param Description: Header's element Syntax: sub-element-name [.sub-element-param-name ] Example: ♦ header.from.param.expires sub-element-param-name Description: Header's parameter name - relevant only to parameter sub-...
Page 833 SIP User's Manual C. SIP Message Manipulation Syntax ♦ user refers to Contact user in IP Group ♦ host refers to Group Name in IP Group table ♦ type refers to Type field in IP Group table ♦ refers to IP Group ID (used to identify source or destination IP Group) string Description: String...
Page 834 Mediant 600 & Mediant 1000 Reader's Notes SIP User's Manual Document #: LTRT-83310...
Page 835: D Dsp Templates
The maximum number of channels on any form of analog, digital and MPM modules assembly is 120. • For additional DSP templates, contact your AudioCodes representative. Analog Interfaces The DSP templates for analog interfaces are shown in the table below.
Page 836: Digital Interfaces
Mediant 600 & Mediant 1000 Digital Interfaces The DSP templates for digital interfaces are shown in the table below. Table D-2: DSP Firmware Templates for Digital Interfaces DSP Template 0 or 10 1 or 11 2 or 12 5 or 15...
Page 837: Media Processing Interfaces
SIP User's Manual D. DSP Templates Media Processing Interfaces The DSP templates for the media processing interfaces (i.e., MPM module) are shown in the table below. Notes: • The MPM module DSP templates are applicable only to Mediant 1000. • Assembly of the MPM module in Slot #6 enables DSP conferencing capabilities.
Page 838 Mediant 600 & Mediant 1000 Table D-3: DSP Firmware Templates for MPM Module DSP Template 0 or 10 1 or 11 2 or 12 5 or 15 6 or 16 Assembly Slot no. Supplementary Capabilities Number of Channels Yes Yes...
Page 839: E Selected Technical Specifications
ETSI/EURO, ANSI NI2, DMS-100, 5ESS, VN3, VN4, VN6 QSIG (Basic Call and Supplementary Services) and other variants Control & Management Control Protocols Operations & Management AudioCodes Element Management System Embedded HTTP Web Server Telnet SNMP V2/V3 Remote configuration and TFTP, HTTP, HTTPS, DHCP and BootP, RADIUS, Syslog (for...
Page 840: Mediant 1000
EN6600-3-2: 2000 + A2: 2005, EN6600-3-3: 1995 + A1: 2001 EN60950-1:2001, A11: 2004) Environmental ETS 300019-2-1 Storage T1.2 Specifications Mediant 1000 The table below lists the main technical specifications of the Mediant 1000. Table E-2: Mediant 1000 Functional Specifications Function Specification Interfaces Modularity and Capacity Voice interface: Equipped with 6 Slots that can host voice modules.
Page 841 Loop Start, Ground Start Control & Management Control Protocols SIP, MSCML Operations & Management AudioCodes Element Management System Embedded HTTP Web Server Telnet SNMP V2, V3 Remote configuration and software download via TFTP, HTTP, HTTPS, DHCP and BootP, RADIUS, Syslog (for events, alarms and...
Page 842 Mediant 600 & Mediant 1000 Function Specification Security IPSec, HTTPS, TLS (SIPS), SSL, Web access list, RADIUS login and SRTP2 Hardware Specifications Power Supply Single universal power supply 100-240V 50-60 Hz 1.5A max., optional redundant power supply Physical 1U high, 19-inch wide...
Page 843 SIP User's Manual E. Selected Technical Specifications Reader's Notes Version 6.4 March 2012...
Page 844 User's Manual Ver. 6.4 www.audiocodes.com...

This manual is also suitable for:

Mediant 600

AudioCodes Mediant 1000 User Manual

1 Overview

Getting Started

2 Assigning the Voip LAN IP Address

Management Tools

3 Web-Based Management

4 CLI-Based Management

5 SNMP-Based Management

6 EMS-Based Management

7 INI File-Based Management

General System Settings

8 Configuring Certificates

9 Date and Time

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

Quick Links

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Summary of Contents for AudioCodes Mediant 1000