Page 1 User's Manual Version 6.0 Document #: LTRT-83306 March 2010...
Page 3: Table Of Contents
SIP User's Manual Contents Table of Contents Overview ......................21 Mediant 1000 ....................... 21 Mediant 600 ......................22 SIP Overview ....................... 24 Configuration Concepts ................... 25 Web-Based Management ................. 27 Getting Acquainted with the Web Interface ............28 3.1.1 Computer Requirements ..................28 3.1.2...
Page 4 Mediant 600 & Mediant 1000 3.3.2.5 Configuring the General Media Settings..........76 3.3.2.6 Configuring the Analog Settings .............. 76 3.3.2.7 Configuring Media Security ..............77 3.3.3 PSTN Settings ......................77 3.3.3.1 Configuring the CAS State Machines ............77 3.3.3.2 Configuring the Trunk Settings ..............80 3.3.4...
Page 5 SIP User's Manual Contents INI File Configuration ..................221 INI File Format ....................221 4.1.1 Configuring Individual ini File Parameters ............222 4.1.2 Configuring ini File Table Parameters..............222 4.1.3 General ini File Formatting Rules ................. 224 Modifying an ini File ................... 225 Secured Encoded ini File ...................
Page 6 Mediant 600 & Mediant 1000 6.3.5 BootP Parameters ....................270 Security Parameters ..................271 6.4.1 General Parameters ....................271 6.4.2 HTTPS Parameters ....................272 6.4.3 SRTP Parameters ....................274 6.4.4 TLS Parameters ....................275 6.4.5 SSH Parameters ....................276 6.4.6 IPSec Parameters ....................
Page 7 SIP User's Manual Contents 6.18.4 LDAP Parameters ....................450 6.19 Channel Parameters ..................451 6.19.1 Voice Parameters ....................451 6.19.2 Coder Parameters ....................454 6.19.3 Fax and Modem Parameters ................455 6.19.4 DTMF Parameters ....................461 6.19.5 RTP, RTCP and T.38 Parameters ................ 462 6.20 Auxiliary and Configuration Files Parameters ............
Page 8 Mediant 600 & Mediant 1000 9.4.2 FXO Device Interworking SIP E911 Calls from Service Provider's IP Network to PSAP DID Lines ......................... 534 Routing Based on LDAP Active Directory Queries ..........538 9.5.1 LDAP Overview ..................... 538 9.5.2 AD-Based Tel-to-IP Routing in Microsoft OCS 2007 Environment ....... 539 Configuring DTMF Transport Types ..............
Page 9 SIP User's Manual Contents 9.11.5 Trunk-to-Trunk Routing Example ................586 9.11.6 SIP Trunking between Enterprise and ITSPs ............587 9.12 Mapping PSTN Release Cause to SIP Response ..........590 9.13 Querying Device Channel Resources using SIP OPTIONS ....... 590 9.14 Answer Machine Detector (AMD) ............... 591 9.15 Event Notification using X-Detect Header ............
Page 10 Mediant 600 & Mediant 1000 11.4.3 Creating an NFAS-Related Trunk Configuration ..........635 11.5 Redirect Number and Calling Name (Display) ............ 637 11.6 Automatic Gain Control (AGC) ................637 12 Tunneling Applications .................. 639 12.1 TDM Tunneling ....................639 12.1.1 DSP Pattern Detector.................... 642 12.2 QSIG Tunneling ....................
Page 11 13.3.8 Example of UDT ‘beep’ Tone Definition ..............711 13.3.9 Limitations and Restrictions .................. 711 14 SIP Software Package ..................713 15 Selected Technical Specifications ..............715 15.1 Mediant 1000 ..................... 715 15.2 Mediant 600 ....................... 717 Version 6.0 March 2010...
Page 12 Figure 3-26: Mediant 600 Home Page....................54 Figure 3-27: Mediant 1000 Home Page....................54 Figure 3-28: Shortcut Menu (Example, Mediant 1000) ................56 Figure 3-29: Text Box (Example, Mediant 1000) ................... 56 Figure 3-30: Shortcut Menu (Example, Mediant 1000) ................57 Figure 3-31: Selecting Port Settings from Shortcut Menu ..............
Page 13 SIP User's Manual Contents Figure 3-58: Web & Telnet Access List Table ..................86 Figure 3-59: Firewall Settings Page ....................... 87 Figure 3-60: Certificates Signing Request Page ................... 90 Figure 3-61: IKE Table Listing Loaded Certificate Files ................ 91 Figure 3-62: General Security Settings Page ..................94 Figure 3-63: IP Security Proposals Table ....................
Page 14 Mediant 600 & Mediant 1000 Figure 3-116: SNMP Community Strings Page ................... 192 Figure 3-117: SNMP V3 Setting Page ....................193 Figure 3-118: SNMP Trusted Managers ....................195 Figure 3-119: Regional Settings Page ....................196 Figure 3-120: Maintenance Actions Page.................... 197 Figure 3-121: Reset Confirmation Message Box .................
Page 15 SIP User's Manual Contents Figure 9-18: Defining Coder Group ID 1 ....................510 Figure 9-19: Defining Coder Group ID 2 ....................510 Figure 9-20: Defining IP Profile ID 1 ....................511 Figure 9-21: Defining Inbound IP Routing Rules ................. 512 Figure 9-22: Defining Outbound IP Routing Rules ................
Page 16 Mediant 600 & Mediant 1000 List of Tables Table 3-1: Description of Toolbar Buttons ..................... 31 Table 3-2: ini File Parameter for Welcome Login Message ..............51 Table 3-3: Description of the Areas of the Home Page ................. 54 Table 3-4: Color-Coding Status for Trunk's Channels ................58 Table 3-5: Multiple Interface Table Parameters Description ..............
Page 17 SIP User's Manual Contents Table 6-17: BootP Parameters ......................270 Table 6-18: General Security Parameters ................... 271 Table 6-19: HTTPS Parameters ......................272 Table 6-20: SRTP Parameters ......................274 Table 6-21: TLS Parameters ....................... 275 Table 6-22: SSH Parameters ....................... 276 Table 6-23: IPSec Parameters ......................
Page 18 Table 13-19: VoiceXML Variables and Events ..................708 Table 13-20: ECMAScript Support ...................... 710 Table 14-1: Software Package ......................713 Table 15-1: Mediant 1000 Functional Specifications ................715 Table 15-2: Mediant 600 Functional Specifications ................717 SIP User's Manual Document #: LTRT-83306...
Page 19: Weee Eu Directive
SIP User's Manual Notices Notice This document describes the AudioCodes Mediant 1000 and Mediant 600 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 20: Related Documentation
Mediant 600 Installation Manual CPE Configuration Guide for IP Voice Mail Note: The term device refers to the Mediant 1000 and Mediant 600 gateways . Note: Before configuring the device, ensure that it is installed correctly as instructed in the device's Installation Manual .
Page 21: Overview
SIP User's Manual 1. Overview Overview This section provides an overview of the Mediant 1000 and Mediant 600 media gateways. 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.
Page 22: Mediant 600
Mediant 600 & Mediant 1000 • Depending on configuration, the device can provide IP Media channels at the expense of PSTN channels. These channels may be used for Media Server applications.  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.
Page 23 SIP User's Manual 1. Overview The device supports the following interfaces:  Up to two E1/T1/J1 spans (including fractional E1/T1)  Up to eight ISDN Basic Rate Interface (BRI) interfaces  Up to four FXO interfaces (RJ-11 ports) - for connecting analog lines of an enterprise's PBX or the PSTN to the IP network ...
Page 24: 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 25: Configuration Concepts
SIP User's Manual 2. Configuration Concepts Configuration Concepts You can configure the device, using the following management tools:  The device's HTTP-based Embedded Web Server (Web interface), using any standard Web browser (described in ''Web-based Management'' on page  A configuration ini file loaded to the device (refer to ''ini File Configuration'' on page ...
Page 26 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83306...
Page 27: Web-Based Management
SIP User's Manual 3. Web-Based Management Web-Based Management The device's Embedded Web Server (Web interface ) provides FCAPS (fault management, configuration, accounting, performance, and security) functionality. The Web interface allows you to remotely configure your device for quick-and-easy deployment, including uploading of software (*.cmp), configuration (*.ini), and auxiliary files, and resetting the device.
Page 28: Getting Acquainted With The Web Interface
Mediant 600 & Mediant 1000 Getting Acquainted with the Web Interface This section describes the Web interface with regards to its graphical user interface (GUI) and basic functionality. 3.1.1 Computer Requirements To use the device's Web interface, the following is required: ...
Page 29: Figure 3-1: Enter Network Password Screen
SIP User's Manual 3. Web-Based Management  To access the Web interface: Open a standard Web browser application. In the Web browser's Uniform Resource Locator (URL) field, specify the device's IP address (e.g., http://10.1.10.10); the Web interface's 'Enter Network Password' dialog box appears, as shown in the figure below Figure 3-1: Enter Network Password Screen In the 'User Name' and 'Password' fields, enter the case-sensitive, user name and...
Page 30: Areas Of The Gui
Mediant 600 & Mediant 1000 3.1.3 Areas of the GUI The figure below displays the general layout of the Graphical User Interface (GUI) of the Web interface: Figure 3-2: Main Areas of the Web Interface GUI The Web GUI is composed of the following main areas: ...
Page 31: Toolbar
SIP User's Manual 3. Web-Based Management 3.1.4 Toolbar The toolbar provides command buttons for quick-and-easy access to frequently required commands, as described in the table below: Table 3-1: Description of Toolbar Buttons Icon Button Description Name Submit Applies parameter settings to the device (refer to '' Saving Configuration'' on page...
Page 32: Navigation Tree
Mediant 600 & Mediant 1000 3.1.5 Navigation Tree The Navigation tree, located in the Navigation pane, displays the menus (pertaining to the menu tab selected on the Navigation bar) used for accessing the configuration pages. The Navigation tree displays a tree-like structure of menus. You can easily drill-down to the required page item level to open its corresponding page in the Work pane.
Page 33: Displaying Navigation Tree In Basic And Full View
SIP User's Manual 3. Web-Based Management  To navigate to a page: Navigate to the required page item, by performing the following: • Drilling-down using the plus signs to expand the menus and submenus • Drilling-up using the minus signs to collapse the menus and submenus Select the required page item;...
Page 34: Showing / Hiding The Navigation Pane
Mediant 600 & Mediant 1000 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 page with a table that's wider than the Work pane and to view the all the columns, you need to use scroll bars.
Page 35: Viewing Parameters
SIP User's Manual 3. Web-Based Management  To open a configuration page in the Work pane: On the Navigation bar, click the required tab: • Configuration (refer to ''Configuration Tab'' on page 60) • Management (refer to ''Management Tab'' on page 188) •...
Page 36: Figure 3-7: Toggling Between Basic And Advanced Page View
Mediant 600 & Mediant 1000 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).
Page 37: Figure 3-8: Expanding And Collapsing Parameter Groups
SIP User's Manual 3. Web-Based Management Notes: • When the Navigation tree is in 'Full' mode (refer to ''Navigation Tree'' page ), configuration pages display all their parameters (i.e., the 'Advanced Parameter List' view is displayed). • If a page contains only basic parameters, the Basic Parameter List button is not displayed.
Page 38: Modifying And Saving Parameters
Mediant 600 & Mediant 1000 3.1.6.3 Modifying and Saving Parameters When you change parameter values on a page, the Edit symbol appears to the right of these parameters. This is especially useful for indicating the parameters that you have currently modified (before applying the changes). After you save your parameter modifications (refer to the procedure described below), the Edit symbols disappear.
Page 39: Entering Phone Numbers
SIP User's Manual 3. Web-Based Management If you enter an invalid parameter value (e.g., not in the range of permitted values) and then click Submit , a message box appears notifying you of the invalid value. In addition, the parameter value reverts to its previous value and is highlighted in red, as shown in the figure below: Figure 3-10: Value Reverts to Previous Valid Value 3.1.6.4...
Page 40: Figure 3-11: Adding An Index Entry To A Table
Mediant 600 & Mediant 1000  To add an entry to a table: In the 'Add Index' field, enter the desired index entry number, and then click Add Index ; an index entry row appears in the table: Figure 3-11: Adding an Index Entry to a Table Click Apply to save the index entry.
Page 41: Searching For Configuration Parameters
SIP User's Manual 3. Web-Based Management  To organize the index entries in ascending, consecutive order:  Click Compact ; the index entries are organized in ascending, consecutive order, starting from index 0. For example, if you added three index entries 0, 4, and 6, then the index entry 4 is re-assigned index number 1 and the index entry 6 is re-assigned index number 2.
Page 42: Working With Scenarios
Mediant 600 & Mediant 1000 In the 'Search' field, enter the parameter name or sub-string of the parameter name that you want to search. If you have performed a previous search for such a parameter, instead of entering the required string, you can use the 'Search History' drop-down list to select the string (saved from a previous search).
Page 43: Creating A Scenario
SIP User's Manual 3. Web-Based Management 3.1.8.1 Creating a Scenario The Web interface allows you to create one Scenario with up to 20 configuration pages, as described in the procedure below:  To create a Scenario: On the Navigation bar, click the Scenarios tab;...
Page 44: Figure 3-15: Creating A Scenario
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 3-15: Creating a Scenario Repeat steps 5 through 8 to add additional Steps (i.e., pages).
Page 45: 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 46: 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 47: Saving A Scenario To A Pc
SIP User's Manual 3. Web-Based Management • Add or Remove Parameters: In the Navigation tree, select the required Step; the corresponding page opens in the Work pane. To add parameters, select the check boxes corresponding to the desired parameters; to remove parameters, clear the check boxes corresponding to the parameters that you want removed.
Page 48: Loading A Scenario To The Device
Mediant 600 & Mediant 1000  To save a Scenario to a PC: On the Navigation bar, click the Scenarios tab; the Scenario appears in the Navigation tree. Click the Get/Send Scenario File button (located at the bottom of the Navigation tree);...
Page 49: Deleting A Scenario
SIP User's Manual 3. Web-Based Management Click the Browse button, and then navigate to the Scenario file stored on your PC. Click the Send File button. Notes: • You can only load a Scenario file to a device that has an identical hardware configuration setup to the device in which it was created.
Page 50: Exiting Scenario Mode
Mediant 600 & Mediant 1000 Click OK ; the Scenario is deleted and the Scenario mode closes. Note: You can also delete a Scenario using the following alternative methods: • Loading an empty dat file (refer to ''Loading a Scenario to the Device'' page •...
Page 51: Creating A Login Welcome Message
SIP User's Manual 3. Web-Based Management 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 device's Web interface. The ini file table parameter WelcomeMessage allows you to create the Welcome message. Up to 20 lines of character strings can be defined for the message.
Page 52: Getting Help
Mediant 600 & Mediant 1000 3.1.10 Getting Help The Web interface provides you with context-sensitive Online Help. The Online Help provides you with brief descriptions of most of the parameters you'll need to successfully configure the device. The Online Help provides descriptions of parameters pertaining to the currently opened page.
Page 53: Logging Off The Web Interface
SIP User's Manual 3. Web-Based Management 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 detailed information on the Web User Accounts, refer to User Accounts.  To log off the Web interface: On the toolbar, click the Log Off button;...
Page 54: Using The Home Page
 On the toolbar, click the Home icon; the 'Home' page is displayed. Figure 3-26: Mediant 600 Home Page Figure 3-27: Mediant 1000 Home Page Note: The displayed number and type of modules depends on the device's hardware configuration. The table below describes the areas of the 'Home' page.
Page 55 SIP User's Manual 3. Web-Based Management Item # Description Module slot number (1 to 26). Module type: FXS, FXO, DIGITAL (i.e., E1/T1), BRI, IPMEDIA. Module status icon:  (green): Module has been inserted or is correctly configured.  (gray): Module was removed. 'Reserved' is displayed alongside the module's name.
Page 56: Assigning A Port Name
 (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). Refer to Item #10 for an explanation. 3.2.1 Assigning a Port Name The 'Home' page allows you to assign an arbitrary name or a brief description to each port.
Page 57: Viewing Analog Port Information
Reset Channel ; the channel is changed to inactive (i.e., the port icon is displayed in grey). Figure 3-30: Shortcut Menu (Example, Mediant 1000) 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.
Page 58: Viewing Trunks' Channels
Mediant 600 & Mediant 1000 3.2.4 Viewing Trunks' Channels The 'Home' page allows you to drill-down to view a detailed status of the channels pertaining to a trunk In addition, you can also view the trunk's configuration.  To view a detailed status of a trunk's channels: In the Home page, click the desired trunk of whose status you want to view;...
Page 59: Replacing Modules
• Before inserting a module into a previously empty slot, you must power down the device. Note: This section is applicable only to Mediant 1000.  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...
Page 60: Configuration Tab
Mediant 600 & Mediant 1000 Click OK to confirm removal; after a few seconds, the module is software- removed, the module status icon turns to grey, and the name of the module is suffixed with the word 'Reserved': Figure 3-37: Removed Module...
Page 61: Configuring The Multiple Interface Table
SIP User's Manual 3. Web-Based Management  IP Routing Table (refer to ''Configuring the IP Routing Table'' on page  QoS Settings (refer to ''Configuring the QoS Settings'' on page 3.3.1.1 Configuring the Multiple Interface Table The 'Multiple Interface Table' page allows you to configure up to 16 logical network interfaces, each with its own IP address, unique VLAN ID (if enabled), interface name, and application type permitted on the interface: ...
Page 62: Figure 3-39: Ip Settings Page
Mediant 600 & Mediant 1000  To configure the multiple IP interface table: Open the 'IP Settings' page (Configuration tab > Network Settings menu > IP Settings Figure 3-39: IP Settings Page Under the 'Multiple Interface Settings' group, click the Multiple Interface Table button;...
Page 63: Table 3-5: Multiple Interface Table Parameters Description
SIP User's Manual 3. Web-Based Management Table 3-5: Multiple Interface Table Parameters Description Parameter Description Table parameters Index Index of each interface. The range is 0 to 15. Note: Each interface index must be unique. Web: Application Type Types of applications that are allowed on the specific EMS: Application Types interface.
Page 64 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Prefix Length Defines the Classless Inter-Domain Routing (CIDR)-style [InterfaceTable_PrefixLength] representation of a dotted decimal subnet notation. The CIDR-style representation uses a suffix indicating the number of bits which are set in the dotted decimal format (e.g.
Page 65 SIP User's Manual 3. Web-Based Management Parameter Description Web/EMS: Interface Name Defines a string (up to 16 characters) to name this interface. [InterfaceTable_InterfaceName] This name is displayed in management interfaces (Web , CLI and SNMP) for better readability (and has no functional use) as well as the 'SIP Media Realm' table (refer to ''Configuring Media Realms'' on page...
Page 66: Configuring The Application Settings
Mediant 600 & Mediant 1000 3.3.1.2 Configuring the Application Settings The 'Application Settings' page is used for configuring various application parameters such as Network Time Protocol (NTP), daylight saving time, and Telnet. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page ...
Page 67: Configuring The Nfs Settings
SIP User's Manual 3. Web-Based Management 3.3.1.3 Configuring the NFS Settings Network File System (NFS) enables the device to access a remote server's shared files and directories, and to handle them as if they're located locally. You can configure up to 16 different NFS file systems.
Page 68: Configuring The Ip Routing Table
Mediant 600 & Mediant 1000 Table 3-6: 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 69: Figure 3-44: Ip Routing Table Page
SIP User's Manual 3. Web-Based Management  To configure static IP routing: Open the 'IP Routing Table' page (Configuration tab > Network Settings menu > IP Routing Table page item). Figure 3-44: IP Routing Table Page In the 'Add a new table entry' group, add a new static routing rule according to the parameters described in the table below.
Page 70: Configuring The Qos Settings
Mediant 600 & Mediant 1000 Parameter Description 'Multiple Interface Table' page (refer to ''Configuring the Multiple Interface Table'' on page Metric The maximum number of times a packet can be [RoutingTableHopsCountColumn] forwarded (hops) between the device and destination (typically, up to 20).
Page 71: Media Settings
SIP User's Manual 3. Web-Based Management 3.3.2 Media Settings The Media Settings menu allows you to configure the device's channel parameters. This menu contains the following items:  Voice Settings (refer to '' Configuring the Voice Settings'' on page  Fax/Modem/CID Settings (refer to “Configuring the Fax/Modem/CID Settings”...
Page 72: Configuring The Voice Settings
Mediant 600 & Mediant 1000 3.3.2.1 Configuring the Voice Settings The 'Voice Settings' page is used for configuring various voice parameters such as voice volume, silence suppression, and DTMF transport type. For a detailed description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page ...
Page 73: Configuring The Fax/Modem/Cid Settings
SIP User's Manual 3. Web-Based Management 3.3.2.2 Configuring the Fax/Modem/CID Settings The 'Fax/Modem/CID Settings' page is used for configuring fax, modem, and Caller ID (CID) parameters. For a detailed description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page ...
Page 74: Configuring The Rtp/Rtcp Settings
Mediant 600 & Mediant 1000 3.3.2.3 Configuring the RTP/RTCP Settings The 'RTP/RTCP Settings' page allows you to configure 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...
Page 75: Configuring The Ip Media Settings
SIP User's Manual 3. Web-Based Management 3.3.2.4 Configuring the 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, refer to ''Configuration Parameters Reference'' on page ...
Page 76: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 3.3.2.5 Configuring the General Media Settings The 'General Media Settings' page allows you to configure various media parameters. For a detailed description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page ...
Page 77: Configuring Media Security
SIP User's Manual 3. Web-Based Management 3.3.2.7 Configuring Media Security The 'Media Security' page allows you to configure media security. For a detailed description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page  To configure media security: Open the 'Media Security' page (Configuration tab >...
Page 78: Figure 3-53: Cas State Machine Page
Mediant 600 & Mediant 1000  To modify the CAS state machine parameters: Open the ‘CAS State Machine' page (Configuration tab > PSTN Settings menu > CAS State Machines page item). Figure 3-53: CAS State Machine Page Ensure that the trunk is inactive. The trunk number displayed in the 'Related Trunks' field must be green.
Page 79: Table 3-8: Cas State Machine Parameters Description
SIP User's Manual 3. Web-Based Management Table 3-8: CAS State Machine Parameters Description Parameter Description Generate Digit On Time Generates digit on-time (in msec). [CasStateMachineGenerateDigitOnTime] The value must be a positive value. The default value is -1 (use value from CAS state machine). Generate Inter Digit Time Generates digit off-time (in msec).
Page 80: Configuring The Trunk Settings
Mediant 600 & Mediant 1000 3.3.3.2 Configuring the 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 81: Figure 3-55: Trunk Scroll Bar
SIP User's Manual 3. Web-Based Management 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 82: Security Settings
Mediant 600 & Mediant 1000 Configure the desired trunk parameters. 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’...
Page 83: Configuring The Web User Accounts
SIP User's Manual 3. Web-Based Management 3.3.4.1 Configuring the 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 84: Figure 3-56: Web User Accounts Page (For Users With 'Security Administrator' Privileges)
Mediant 600 & Mediant 1000  To change the Web user accounts attributes: Open the 'Web User Accounts' page (Configuration tab > Security Settings menu > Web User Accounts page item). Figure 3-56: WEB User Accounts Page (for Users with 'Security Administrator' Privileges)
Page 85: Configuring The Web And Telnet Access List
SIP User's Manual 3. Web-Based Management In the fields 'New Password' and 'Confirm New Password', enter the new password (maximum of 19 case-sensitive characters). Click Change Password ; if you are currently logged into the Web interface with this account, the 'Enter Network Password' dialog box appears, requesting you to enter the new password.
Page 86: Configuring The Firewall Settings
Mediant 600 & Mediant 1000 To add an authorized IP address, in the 'Add a New Authorized IP Address' field, enter the required IP address, and then click Add New Address ; the IP address you entered is added as a new entry to the 'Web & Telnet Access List' table.
Page 87: Figure 3-59: Firewall Settings
SIP User's Manual 3. Web-Based Management For each packet received on the network interface, the table is scanned from the top down until a matching rule is found. This rule can either deny (block) or permit (allow) the packet. Once a rule in the table is located, subsequent rules further down the table are ignored. If the end of the table is reached without a match, the packet is accepted.
Page 88: Table 3-11: Internal Firewall Parameters
Mediant 600 & Mediant 1000  To edit a rule: In the 'Edit Rule' column, select the rule that you want to edit. Modify the fields as desired. Click the Apply button to save the changes. To save the changes to flash memory, refer to ''...
Page 89: Configuring The Certificates
SIP User's Manual 3. Web-Based Management Parameter Description Protocol The protocol type (e.g., UDP, TCP, ICMP, ESP or 'Any'), or the IANA [AccessList_Protocol] protocol number (in the range of 0 (Any) to 255). Note: This field also accepts the abbreviated strings 'SIP' and 'HTTP'. Specifying these strings implies selection of the TCP or UDP protocols, and the appropriate port numbers as defined on the device.
Page 90: Figure 3-60: Certificates Signing Request
Mediant 600 & Mediant 1000 Open the ‘Certificates Signing Request' page (Configuration tab > Security Settings menu > Certificates page item). Figure 3-60: Certificates Signing Request Page In the 'Subject Name' field, enter the DNS name, and then click Generate CSR textual certificate signing request that contains the SSL device identifier is displayed.
Page 91: Figure 3-61: Ike Table Listing Loaded Certificate Files
SIP User's Manual 3. Web-Based Management Notes: • The certificate replacement process can be repeated when necessary (e.g., the new certificate expires). • It is possible to use the IP address of the device (e.g., 10.3.3.1) instead of a qualified DNS name in the Subject Name. This is not recommended since the IP address is subject to changes and may not uniquely identify the device.
Page 92 Mediant 600 & Mediant 1000 3.3.4.4.2 Client Certificates By default, Web servers using SSL provide one-way authentication. The client is certain that the information provided by the Web 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 93 SIP User's Manual 3. Web-Based Management 3.3.4.4.3 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 94: Configuring The General Security Settings
Mediant 600 & Mediant 1000 3.3.4.5 Configuring the 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 ...
Page 95: Configuring The Ip Security Proposal Table
SIP User's Manual 3. Web-Based Management 3.3.4.6 Configuring the IP Security Proposal Table The 'IP Security Proposals Table' page is used to configure Internet Key Exchange (IKE) with up to four proposal settings. Each proposal defines an encryption algorithm, an authentication algorithm, and a Diffie-Hellman group identifier.
Page 96: Configuring The Ip Security Associations Table
Mediant 600 & Mediant 1000 Parameter Name Description Diffie Hellman Group Determines the length of the key created by the [IPsecProposalTable_DHGroup] DH protocol for up to four proposals. For the file parameter, X depicts the proposal number (0 to 3).
Page 97: Table 3-14: Ip Security Associations Table Configuration Parameters
SIP User's Manual 3. Web-Based Management Click Apply ; the rule is applied on-the-fly. To save the changes to flash memory, refer to '' Saving Configuration'' on page Table 3-14: IP Security Associations Table Configuration Parameters Parameter Name Description Operational Mode Defines the IPSec mode of operation.
Page 98 Mediant 600 & Mediant 1000 Parameter Name Description Protocol Defines the protocol type to which this configuration applies. [IPsecSATable_Protocol] Standard IP protocol numbers, as defined by the Internet Assigned Numbers Authority (IANA) should be used, for example:  0 = Any protocol (default) ...
Page 99: Protocol Configuration
SIP User's Manual 3. Web-Based Management Parameter Name Description Remote Prefix Length Defines the prefix length of the Remote Subnet IP Address [IPsecSATable_RemoteSubnetPre parameter (in bits). The prefix length defines the subnet class fixLength] of the remote network. A prefix length of 16 corresponds to a Class B subnet (255.255.0.0);...
Page 100: Configuring Media Realms
Mediant 600 & Mediant 1000 3.3.5.1 Configuring Media Realms The 'SIP Media Realm Table' page allows you to define a pool of up to 16 media interfaces, termed Media Realms. This table allows you to divide a Media-type interface (defined in the...
Page 101 SIP User's Manual 3. Web-Based Management Parameter Description IPv4 Interface Name Associates the IPv4 interface to the Media Realm. [CpMediaRealm_IPv4IF] Note: The name of this interface must be exactly as configured in the 'Multiple Interface' table (InterfaceTable parameter). Port Range Start Defines the starting port for the range of Media interface [CpMediaRealm_PortRangeStart] UDP ports.
Page 102: Enabling Applications
Software Upgrade Key supporting the application (refer to ''Loading a Software Upgrade Key'' on page • The IP2P application is applicable only to Mediant 1000. • For enabling an application, a device reset is required.  To enable an application: Open the 'Applications Enabling' page (Configuration tab >...
Page 103: Figure 3-67: Trunk Group Table
SIP User's Manual 3. Web-Based Management  To configure the Trunk Group Table: Open the 'Trunk Group Table' page (Configuration tab > Protocol Configuration menu > Trunk Group submenu > Trunk Group page item). Figure 3-67: Trunk Group Table Page Configure the Trunk Group according to the table below Click the Submit...
Page 104 Mediant 600 & Mediant 1000 Parameter Description Phone Number The telephone number that is assigned to the channel. For a [TrunkGroup_FirstPhoneNumber] range of channels, enter only the first telephone number. Subsequent channels are assigned the next consecutive telephone number. For example, if you enter 400 for channels 1 to 4, then channel 1 is assigned phone number 400, channel 2 is assigned phone number 401, and so on.
Page 105: Figure 3-68: Trunk Group Settings
SIP User's Manual 3. Web-Based Management Note: You can also configure the 'Trunk Group Settings' table using the ini file table parameter TrunkGroupSettings (refer to ''Number Manipulation and Routing Parameters'' on page  To configure the Trunk Group Settings table: Open the 'Trunk Group Settings' page (Configuration tab >...
Page 106: Table 3-17: Trunk Group Settings Parameters
Mediant 600 & Mediant 1000 Table 3-17: Trunk Group Settings Parameters Parameter Description Trunk Group ID The Trunk Group ID that you want to configure. [TrunkGroupSettings_TrunkGrou pId] Channel Select Mode The method for which IP-to-Tel calls are assigned to channels [TrunkGroupSettings_ChannelSel pertaining to a Trunk Group.
Page 107 SIP User's Manual 3. Web-Based Management Parameter Description  If the device is configured globally (ChannelSelectMode) to register Per Endpoint, and a channels Group comprising four channels is configured to register Per Gateway, the device registers all channels except the first four channels. The channels Group of these four channels sends a single registration request.
Page 108: Protocol Definition
Mediant 600 & Mediant 1000 3.3.5.4 Protocol Definition The Protocol Definition submenu allows you to configure the main SIP protocol parameters. This submenu contains the following page items:  SIP General Parameters (refer to '' SIP General Parameters'' on page ...
Page 109: Figure 3-69: Sip General Parameters
SIP User's Manual 3. Web-Based Management Figure 3-69: SIP General Parameters Page Configure the parameters as required. Click the Submit button to save your changes. To save the changes to flash memory, refer to '' Saving Configuration'' on page Version 6.0 March 2010...
Page 110: Application Network Setting
Mediant 600 & Mediant 1000 3.3.5.4.2 Configuring DTMF and Dialing Parameters 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, refer to ''Configuration Parameters Reference'' on page ...
Page 111: Figure 3-71: Srd Table
SIP User's Manual 3. Web-Based Management (associated with a Media Realm) interfaces for multiple Layer-3 networks.  Ability to operate with multiple gateway customers that may reside either in the same or in different Layer-3 networks as the device. This allows separation of signaling traffic between different customers.
Page 112: Figure 3-72: Srd Table
Mediant 600 & Mediant 1000 3.3.5.5.2 Configuring the SIP Interface Table The 'SIP Interface Table' page allows you to configure up to six SIP Interfaces. A SIP Interface represents a SIP SIP signaling interface (IPv4), which is a combination of ports (UDP, TCP, and TLS) associated with a specific IP address and an SRD ID.
Page 113: Proxies, Registration, Ip Groups
SIP User's Manual 3. Web-Based Management Parameter Description  GW/IP2IP (default) = IP-to-IP routing application and regular gateway functionality  SAS = Stand-Alone Survivability (SAS) application UDP Port Determines the listening and source UDP port. [SIPInterface_UDPPort] The valid range is 1 to 65534. The default is 5060. Note: This port must be outside of the RTP port range.
Page 114: Figure 3-73: Ip Group
Mediant 600 & Mediant 1000 Notes: • When working with multiple IP Groups, the default Proxy server should not be used (i.e., the parameter IsProxyUsed must be set to 0). • You can also configure the IP Groups table using the ini file table...
Page 115: Table 3-20: Ip Group Parameters
SIP User's Manual 3. Web-Based Management Table 3-20: IP Group Parameters Parameter Description Common Parameters Type The IP Group can be defined as one of the following types: [IPGroup_Type]  SERVER = used when the destination address (configured by the Proxy Set) of the IP Group (e.g., ITSP, Proxy, IP- PBX, or Application server) is known.
Page 116 Mediant 600 & Mediant 1000 Parameter Description SIP Group Name The request URI host name used in INVITE and REGISTER [IPGroup_SIPGroupName] messages that are sent to this IP Group, or the host name in the From header of INVITE messages received from this IP Group.
Page 117 SIP User's Manual 3. Web-Based Management Parameter Description  [0] Routing Table = The device routes the call according to the 'Outbound IP Routing Table'.  [1] Serving IP Group = The device sends the SIP INVITE to the selected Serving IP Group. If no Serving IP Group is selected, the default IP Group is used.
Page 118 Mediant 600 & Mediant 1000 Parameter Description mode in which the device uses its database for routing calls between the clients (e.g., IP phones) of the USER-type IP Group. The RTP packets between the IP phones in Survivability mode always traverse through the device. In Survivability mode, the device is capable of receiving new registrations.
Page 119: Figure 3-74: Account Table
SIP User's Manual 3. Web-Based Management  To configure Accounts: Open the 'Account Table' page (Configuration tab > Protocol Configuration menu > Proxies, Registration, IP Groups submenu > Account Table page item). Figure 3-74: Account Table Page To add an Account, in the 'Add' field, enter the desired table row index, and then click .
Page 120 Mediant 600 & Mediant 1000 Parameter Description Serving IP Group The destination IP Group ID (defined in '' Configuring the IP [Account_ServingIPGroup] Groups'' on page ) to where the REGISTER requests (if enabled) are sent or Authentication is performed. The actual...
Page 121 SIP User's Manual 3. Web-Based Management Parameter Description Register Enables registration. [Account_Register]  No = Don't register  Yes = Enables registration When enabled, the device sends REGISTER requests to the Serving IP Group. In addition, to activate registration, you also need to set the parameter 'Registration Mode' to 'Per Account' in the 'Trunk Group Settings' table for the specific Trunk Group.
Page 122: Figure 3-75: Proxy & Registration
Mediant 600 & Mediant 1000 3.3.5.6.3 Configuring Proxy and Registration Parameters The 'Proxy & Registration' page allows you to configure parameters that are associated with Proxy and Registration. For a description of the parameters appearing on this page, refer to...
Page 123 SIP User's Manual 3. Web-Based Management Click the Proxy Set Table button to open the 'Proxy Sets Table' page to configure groups of proxy addresses. Alternatively, you can open this page from the Proxy Sets Table page item (refer to ''Configuring the Proxy Sets Table'' on page for a description of this page).
Page 124: Figure 3-76: Proxy Sets Table
Mediant 600 & Mediant 1000  To add Proxy servers and configure Proxy parameters: Open the 'Proxy Sets Table' page (Configuration tab > Protocol Configuration menu > Proxies, Registration, IP Groups submenu > Proxy Sets Table page item). Figure 3-76: Proxy Sets Table Page From the Proxy Set ID drop-down list, select an ID for the desired group.
Page 125 SIP User's Manual 3. Web-Based Management Parameter Description 'Trunk Group Settings' table.  According to the 'Outbound IP Routing Table' if the parameter PreferRouteTable is set to 1.  To the default Proxy. Typically, when IP Groups are used, there is no need to use the default Proxy, and all routing and registration rules can be configured using IP Groups and the Account tables (refer to ''Configuring the Account Table''...
Page 126 Mediant 600 & Mediant 1000 Parameter Description Web: Proxy Load Balancing Enables the Proxy Load Balancing mechanism per Proxy Set ID. Method  Disable = Load Balancing is disabled (default). EMS: Load Balancing Method  Round Robin = Round Robin.
Page 127: Coders And Profile Definitions
SIP User's Manual 3. Web-Based Management Parameter Description redundancy is used.  When this parameter is set to 'Using REGISTER', the homing redundancy mode is disabled.  When the active proxy doesn't respond to INVITE messages sent by the device, the proxy is tagged as 'offline'. The behavior is similar to a Keep-Alive (OPTIONS or REGISTER) failure.
Page 128 Mediant 600 & Mediant 1000 In addition, you can associate different Profiles per the device's channels. Each Profile contains a set of parameters such as coders, T.38 Relay, Voice and DTMF Gain, Silence Suppression, Echo Canceler, RTP DiffServ, Current Disconnect and more.
Page 129: Figure 3-77: Coders
SIP User's Manual 3. Web-Based Management  To configure the device's coders: Open the 'Coders' page (Configuration tab > Protocol Configuration menu > Coders And Profile Definitions submenu > Coders page item). Figure 3-77: Coders Page From the 'Coder Name' drop-down list, select the required coder. From the 'Packetization Time' drop-down list, select the packetization time (in msec) for the selected coder.
Page 130: Figure 3-78: Coder Group Settings
Mediant 600 & Mediant 1000 3.3.5.7.2 Configuring Coder Groups The 'Coder Group Settings' page provides a table for defining up to four different coder groups. These coder groups are used in the 'Tel Profile Settings' and 'IP Profile Settings' pages to assign different coders to Profiles. For each coder group, you can define up to ten coders, where the first coder (and its attributes) in the table takes precedence over the second coder, and so on.
Page 131 SIP User's Manual 3. Web-Based Management From the 'Packetization Time' drop-down list, select the packetization time (in msec) for the coder. The packetization time determines how many coder payloads are combined into a single RTP packet. From the 'Rate' drop-down list, select the bit rate (in kbps) for the coder you selected. In the 'Payload Type' field, if the payload type (i.e., format of the RTP payload) for the coder you selected is dynamic, enter a value from 0 to 120 (payload types of 'well- known' coders cannot be modified).
Page 132: Figure 3-79: Tel Profile Settings
Mediant 600 & Mediant 1000 3.3.5.7.3 Configuring Tel Profile The 'Tel Profile Settings' page allows you to define up to nine Tel Profiles. You can then assign these Tel Profiles to the device's channels (in the 'Trunk Group Table ' page), thereby applying different behaviors to different channels.
Page 133 SIP User's Manual 3. Web-Based Management From the 'Profile ID' drop-down list, select the Tel Profile identification number you want to configure. In the 'Profile Name' field, enter an arbitrary name that enables you to easily identify the Tel Profile. From the 'Profile Preference' drop-down list, select the priority of the Tel Profile, where '1' is the lowest priority and '20' is the highest.
Page 134: Figure 3-80: Ip Profile Settings
Mediant 600 & Mediant 1000  To configure the IP Profile settings: Open the 'IP Profile Settings' page (Configuration tab > Protocol Configuration menu > Coders And Profile Definitions submenu > IP Profile Settings Figure 3-80: IP Profile Settings Page From the 'Profile ID' drop-down list, select an identification number for the IP Profile.
Page 135: Sip Advanced Parameters
SIP User's Manual 3. Web-Based Management 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 136: Figure 3-81: Advanced Parameters
Mediant 600 & Mediant 1000  To configure the advanced general protocol parameters: Open the 'Advanced Parameters' page (Configuration tab > Protocol Configuration menu > SIP Advanced Parameters submenu > Advanced Parameters page item). Figure 3-81: Advanced Parameters Page SIP User's Manual...
Page 137 SIP User's Manual 3. Web-Based Management Configure the parameters as required. Click the Submit button to save your changes. To save the changes to flash memory, refer to '' Saving Configuration'' on page Version 6.0 March 2010...
Page 138: Figure 3-82: Supplementary Services
Mediant 600 & Mediant 1000 3.3.5.8.2 Configuring Supplementary Services The 'Supplementary Services' page is used to configure parameters that are associated with supplementary services. For a description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page .
Page 139: Figure 3-83: Metering Tones
SIP User's Manual 3. Web-Based Management 3.3.5.8.3 Configuring Metering Tones The FXS interfaces can generate 12/16 KHz metering pulses towards the Tel side (e.g., for connection to a payphone or private meter). Tariff pulse rate is determined according to an internal table.
Page 140: Figure 3-84: Charge Codes Table
Mediant 600 & Mediant 1000 3.3.5.8.4 Configuring the Charge Codes Table 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 141 SIP User's Manual 3. Web-Based Management 3.3.5.8.5 Configuring Keypad Features The 'Keypad Features' page enables you to activate and deactivate the following features directly from the connected telephone's keypad:  Call Forward (refer to ''Configuring Call Forward'' on page  Caller ID Restriction (refer to ''Configuring Caller Display Information'' on page ...
Page 142: Figure 3-85: Keypad Features
Mediant 600 & Mediant 1000  To configure the keypad features Open the 'Keypad Features' page (Configuration tab > Protocol Configuration menu > SIP Advanced Parameters submenu > Keypad Features page item). Figure 3-85: Keypad Features Page Configure the keypad features as required. For a description of these parameters,...
Page 143: Manipulation Tables
SIP User's Manual 3. Web-Based Management 3.3.5.9 Manipulation Tables The Manipulation Tables submenu allows you to configure number manipulation and mapping of NPI/TON to SIP messages. This submenu includes the following items:  General Settings (refer to '' Configuring General Settings'' on page ...
Page 144 Mediant 600 & Mediant 1000 3.3.5.9.2 Configuring the Number Manipulation Tables The device provides four number manipulation tables for incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly. For example, telephone number manipulation can be implemented for the following: ...
Page 145 SIP User's Manual 3. Web-Based Management Notes: • Number manipulation can occur before or after a routing decision is made. For example, you can route a call to a specific Trunk Group according to its original number, and then you can remove or add a prefix to that number before it is routed.
Page 146: Figure 3-87: Source Phone Number Manipulation Table For Tel-To-Ip Calls
Mediant 600 & Mediant 1000  To configure the Number Manipulation tables: Open the required 'Number Manipulation' page (Configuration tab > Protocol Configuration menu > Manipulation Tables submenu > Dest Number IP->Tel, Dest Number Tel->IP, Source Number IP->Tel, or Source Number Tel->IP page item);...
Page 147 SIP User's Manual 3. Web-Based Management Parameter Description Manipulation Table for Tel -> IP Calls' and 'Destination Phone Number Manipulation Table for Tel -> IP Calls' pages.  For IP-to-IP call routing, this parameter is not required (i.e., leave the field empty). Source IP Group The IP Group from where the IP-to-IP call originated.
Page 148 Mediant 600 & Mediant 1000 Parameter Description Web: Suffix to Add The number or string that you want added to the end of the telephone EMS: Prefix/Suffix To Add number. For example, if you enter '00' and the phone number is [_Suffix2Add] 1234, the new number is 123400.
Page 149: Figure 3-88: Reditect Number Ip To Tel
SIP User's Manual 3. Web-Based Management 3.3.5.9.3 Configuring Redirect Number IP to Tel The 'Redirect Number IP > Tel' page allows you to configure IP-to-Tel redirect number manipulation rules. This feature allows you to manipulate the value of the received SIP Diversion, Resource-Priority, or History-Info header, which is then added to the Redirecting Number Information Element (IE) in the ISDN Setup message, sent to the Tel side.
Page 150 Mediant 600 & Mediant 1000 Parameter Description Web: Stripped Digits From Number of digits to remove from the left of the telephone number Left prefix. For example, if you enter 3 and the phone number is 5551234, EMS: Remove From Left the new phone number is 1234.
Page 151: Figure 3-89: Redirect Number Tel To Ip
SIP User's Manual 3. Web-Based Management Parameter Description Web: NPI The Numbering Plan Indicator (NPI) assigned to this entry. EMS: Number Plan  Unknown (default) [_NumberPlan]  Private  E.164 Public  [-1] Not Configured = value received from PSTN/IP is used Note: For a detailed list of the available NPI/TON values, refer to ''Numbering Plans and Type of Number'' on page...
Page 152: Table 3-26: Redirect Number Tel To Ip Parameters Description
Mediant 600 & Mediant 1000 The figure below shows an example configuration in which the redirect prefix "555" is manipulated. According to the configured rule, if for example the number 5551234 is received, after manipulation the device sends the number to IP as 91234.
Page 153: Figure 3-90: Phone Context Table
SIP User's Manual 3. Web-Based Management Parameter Description  Allowed = sends Caller ID information when a call is made using these destination / source prefixes.  Restricted = restricts Caller ID information for these prefixes. Notes:  If 'Presentation' is set to 'Restricted' and 'Asserted Identity Mode' is set to 'P-Asserted', the From header in the INVITE message includes the following: From: 'anonymous' <sip: anonymous@anonymous.invalid>...
Page 154: Table 3-27: Phone-Context Parameters Description
Mediant 600 & Mediant 1000 Notes: • Several rows with the same NPI-TON or Phone-Context are allowed. In such a scenario, a Tel-to-IP call uses the first match. • Phone-Context '+' is a unique case as it doesn't appear in the Request- URI as a Phone-Context parameter.
Page 155: Table 3-28: Npi/Ton Values For Isdn Etsi
SIP User's Manual 3. Web-Based Management 3.3.5.9.6 Numbering Plans and Type of Number The IP-to-Tel destination or source number manipulation tables allow you to classify numbers by their Numbering Plan Indication (NPI) and Type of Number (TON). The device supports all NPI/TON classifications used in the standard. The list of ISDN ETSI NPI/TON values is shown in the following table: Table 3-28: NPI/TON Values for ISDN ETSI Description...
Page 156: Routing Tables
Mediant 600 & Mediant 1000 3.3.5.10 Routing Tables The Routing Tables submenu allows you to configure call routing rules. This submenu includes the following page items:  Alternative Routing (refer to '' Configuring Reasons for Alternative Routing'' on page ...
Page 157: Figure 3-91: Reasons For Alternative Routing
SIP User's Manual 3. Web-Based Management Notes: • To enable alternative routing using the IP-to-Tel routing table, configure the parameter RedundantRoutingMode to 1 (default). • The reasons for alternative routing for Tel-to-IP calls also apply for Proxies (if the parameter RedundantRoutingMode is set to 2). •...
Page 158: Figure 3-92: Routing General Parameters
Mediant 600 & Mediant 1000 3.3.5.10.2 Configuring General Routing Parameters The 'Routing General Parameters' page allows you to configure the general routing parameters. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page  To configure the general routing parameters: Open the 'Routing General Parameters' page (Configuration tab >...
Page 159 SIP User's Manual 3. Web-Based Management IP Group, the call is routed to the Proxy Set (IP address) associated with the IP Group. If the number dialed does not match these characteristics, the call is not made. When using a proxy server, you don't need to configure this table unless you require one of the following: ...
Page 160: Figure 3-93: Outbound Ip Routing Table
Mediant 600 & Mediant 1000 Notes: • If the alternative routing destination is the device itself, the call can be configured to be routed to the PSTN. This feature is referred to as 'PSTN Fallback'. For example, if poor voice quality occurs over the IP network, the call is rerouted through the legacy telephony system (PSTN).
Page 161: Table 3-29: Outbound Ip Routing Table Parameters
SIP User's Manual 3. Web-Based Management Table 3-29: Outbound IP Routing Table Parameters Parameter Description Web/EMS: Tel to IP Routing Determines whether to route received calls to an IP destination Mode before or after manipulation of the destination number. [RouteModeTel2IP] ...
Page 162 Mediant 600 & Mediant 1000 Parameter Description All calls matching all or any combination of the above characteristics are sent to the destination IP address defined below. Note: For alternative routing, additional entries of the same prefix can be configured.
Page 163 SIP User's Manual 3. Web-Based Management Parameter Description destination IP 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 164: Figure 3-94: Inbound Ip Routing Table
Mediant 600 & Mediant 1000 This table provides two main areas for defining a routing rule:  Matching Characteristics: user-defined characteristics of the incoming IP call are defined in this area. If the characteristics match a table entry, the rule is used to route the call.
Page 165: Table 3-30: Inbound Ip Routing Table Description
SIP User's Manual 3. Web-Based Management From the 'Routing Index' drop-down list, select the range of entries that you want to add. Configure the inbound IP routing rule according to the table below. Click the Submit button to save your changes. To save the changes so they are available after a power failure, refer to '' Saving Configuration''...
Page 166 Mediant 600 & Mediant 1000 Parameter Description Source IP Address The source IP address of an IP-to-Tel call (obtained from the [PstnPrefix_SourceAddress] Contact header in the INVITE message) that can be used for routing decisions. Notes:  You can configure from where the source IP address is obtained, using the parameter SourceIPAddressInput.
Page 167: Figure 3-95: Internal Dns Table
SIP User's Manual 3. Web-Based Management 3.3.5.10.5 Configuring the Internal DNS Table The 'Internal DNS Table' page, similar to a DNS resolution is used to translate up to 20 host (domain) names into IP addresses (e.g., when using the 'Outbound IP Routing Table'). Up to four different IP addresses can be assigned to the same host name, typically used for alternative routing (for Tel-to-IP call routing).
Page 168: Figure 3-96: Internal Srv Table
Mediant 600 & Mediant 1000 3.3.5.10.6 Configuring the Internal SRV Table The 'Internal SRV Table' page provides a table for resolving host names to DNS A- Records. Three different A-Records can be assigned to each host name. Each A-Record contains the host name, priority, weight, and port.
Page 169: Figure 3-97: Release Cause Mapping
SIP User's Manual 3. Web-Based Management 3.3.5.10.7 Configuring Release Cause Mapping The 'Release Cause Mapping' page consists of two groups that allow the device to map up to 12 different SIP Response Codes to Q.850 Release Causes and vice versa, thereby overriding the hard-coded mapping mechanism (described in ''Release Reason Mapping'' on page Note:...
Page 170: Figure 3-98: Forward On Busy Trunk Destination
Mediant 600 & Mediant 1000 3.3.5.10.8 Configuring Call Forward upon Busy Trunk The 'Forward on Busy Trunk Destination' page allows you to configure forwarding of IP-to- Tel calls to a different (alternative) IP destination, using SIP 3xx response, upon the following scenarios: ...
Page 171: Endpoint Settings
SIP User's Manual 3. Web-Based Management 3.3.5.11 Endpoint Settings The Endpoint Settings submenu allows you to configure analog (FXS/FXO) port-specific parameters. This submenu includes the following page items:  Authentication (refer to '' Configuring Authentication'' on page  Automatic Dialing (refer to ''Configuring Automatic Dialing'' on page ...
Page 172: Figure 3-99: Authentication
Mediant 600 & Mediant 1000  To configure the Authentication Table: Set the parameter 'Authentication Mode' (AuthenticationMode ) to 'Per Endpoint'. Open the 'Authentication' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu > Authentication page item). Figure 3-99: Authentication Page In the 'User Name' and 'Password' fields corresponding to a port, enter the user name and password respectively.
Page 173: Figure 3-100: Automatic Dialing
SIP User's Manual 3. Web-Based Management  To configure Automatic Dialing: Open the 'Automatic Dialing' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu > Automatic Dialing page item). Figure 3-100: Automatic Dialing Page In the 'Destination Phone Number' field corresponding to a port, enter the telephone number that you want automatically dialed.
Page 174: Figure 3-101: Caller Display Information
Mediant 600 & Mediant 1000 3.3.5.11.3 Configuring Caller Display Information The 'Caller Display Information' page allows you to enable the device to send Caller ID information to IP when a call is made. The called party can use this information for caller identification.
Page 175: Figure 3-102: Call Forward Table
SIP User's Manual 3. Web-Based Management 3.3.5.11.4 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 176: Figure 3-103: Caller Id Permissions
Mediant 600 & Mediant 1000 Parameter Description Forward to Phone The telephone number or URI (<number>@<IP address>) to where the Number call is forwarded. Note: If this field only contains a telephone number and a Proxy isn't used, the 'forward to' phone number must be specified in the 'Outbound...
Page 177: Figure 3-104: Caller Waiting
SIP User's Manual 3. Web-Based Management 3.3.5.11.6 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' (refer to ''Configuring Supplementary Services'' on page...
Page 178: Configuring Digital Gateway Parameters
Mediant 600 & Mediant 1000 3.3.5.12 Configuring Digital Gateway Parameters The 'Digital Gateway Parameters' page allows you to configure miscellaneous digital parameters. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page  To configure the digital gateway parameters: Open the 'Digital Gateway Parameters' page (Configuration tab >...
Page 179: Configuring The Ipmedia Parameters
The 'IP Media Settings' page allows you to configure the IP media parameters. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page Note: This page is applicable only to Mediant 1000. This page is Software Upgrade Key dependant (refer to ''Upgrading the Software Upgrade Key'' on page ...
Page 180: Sas Parameters
Mediant 600 & Mediant 1000 3.3.5.14 SAS Parameters The SAS submenu allows you to configure the SAS application. This submenu includes the Stand Alone Survivability item page (refer to ''Configuring Stand-Alone Survivability Parameters'' on page ), from which you can also access the 'IP2IP Routing Table' page...
Page 181: Figure 3-107: Sas Configuration
SIP User's Manual 3. Web-Based Management  To configure the Stand-Alone Survivability parameters: Open the 'SAS Configuration' page (Configuration tab > Protocol Configuration menu > SAS submenu > Stand Alone Survivability page item). Figure 3-107: SAS Configuration Page Configure the parameters as described in ''SIP Configuration Parameters'' on page Click the Submit button to apply your changes.
Page 182: Figure 3-108: Ip2Ip Routing
Mediant 600 & Mediant 1000 3.3.5.14.2 Configuring the IP2IP Routing Table (SAS) The 'IP2IP Routing Table' page allows you to configure up to 120 SAS routing rules (for Normal and Emergency modes). The device routes the SAS call (received SIP INVITE message) once a rule in this table is matched.
Page 183 SIP User's Manual 3. Web-Based Management Parameter Description Source Host The host part of the incoming SIP INVITE’s source URI [IP2IPRouting_SrcHost] (usually the From URI). If this rule is not required, leave the field empty. To denote any host name, use the asterisk (*) symbol.
Page 184: Configuring Tdm Bus Settings
Mediant 600 & Mediant 1000 Parameter Description Destination Address The destination IP address (or domain name, e.g., [IP2IPRouting_DestAddress] domain.com) to where the call is sent. Notes:  This parameter is applicable only if the parameter 'Destination Type' is set to 'Dest Address' [1].
Page 185: Advanced Applications
SIP User's Manual 3. Web-Based Management 3.3.7 Advanced Applications The Advanced Applications menu allows you to configure advanced SIP-based applications. This menu includes the following page items:  Voice Mail Settings (refer to Configuring Voice Mail Parameters on page 185) ...
Page 186: Configuring Radius Accounting Parameters
Mediant 600 & Mediant 1000  To configure the Voice Mail parameters: Open the 'Voice Mail Settings' page (Configuration tab > Advanced Applications menu > Voice Mail Settings page item). Figure 3-110: Voice Mail Settings Page Configure the parameters as required.
Page 187: Configuring Fxo Parameters
SIP User's Manual 3. Web-Based Management  To configure the RADIUS parameters: Open the ‘RADIUS Parameters' page (Configuration tab > Advanced Applications menu > RADIUS Parameters page item). Figure 3-111: RADIUS Parameters Page Configure the parameters as required. Click the Submit button to save your changes.
Page 188: Configuring Ldap Settings
Mediant 600 & Mediant 1000 Configure the parameters as required. Click the Submit button to save your changes. To save the changes to flash memory, refer to '' Saving Configuration'' on page 3.3.7.4 Configuring LDAP Settings The 'LDAP Settings' page is used for configuring the Lightweight Directory Access Protocol (LDAP) parameters.
Page 189: Management Tab
SIP User's Manual 3. Web-Based Management Management Tab The Management tab on the Navigation bar displays menus in the Navigation tree related to device management. These menus include the following:  Management Configuration (refer to '' Management Configuration'' on page ...
Page 190: Configuring The Management Settings
Mediant 600 & Mediant 1000 3.4.1.1 Configuring the Management Settings The 'Management Settings' page allows you to configure the device's management parameters. For detailed description on the SNMP parameters, refer to ''SNMP Parameters'' on page  To configure the management parameters: Open the 'Management Settings' page (Management tab >...
Page 191: Figure 3-115: Snmp Trap Destinations
SIP User's Manual 3. Web-Based Management • SNMP V3 Table: Click the arrow button to configure the SNMP V3 users (refer to ''Configuring SNMP V3 Table'' on page 193). • SNMP Trusted Managers: Click the arrow button to configure the SNMP Trusted Managers (refer to ''Configuring SNMP Trusted Managers'' on page 194).
Page 192: Figure 3-116: Snmp Community Strings
Mediant 600 & Mediant 1000 Parameter Description IP Address IP address of the remote host used as an SNMP [SNMPManagerTableIP_x] Manager. The device sends SNMP traps to these IP addresses. Enter the IP address in dotted-decimal notation, e.g., 108.10.1.255. Trap Port Defines the port number of the remote SNMP Manager.
Page 193: Figure 3-117: Snmp V3 Setting
SIP User's Manual 3. Web-Based Management To save the changes to flash memory, refer to ''Saving Configuration'' on page Note: To delete a community string, select the Delete check box corresponding to the community string that you want to delete, and then click Submit . Table 3-34: SNMP Community Strings Parameters Description Parameter Description...
Page 194: Table 3-35: Snmp V3 Users Parameters
Mediant 600 & Mediant 1000 Notes: • For a description of the web interface's table command buttons (e.g., Duplicate and Delete), refer to ''Working with Tables'' on page • You can also configure SNMP v3 users using the ini file table parameter...
Page 195: Figure 3-118: Snmp Trusted Managers
SIP User's Manual 3. Web-Based Management 3.4.1.1.4 Configuring SNMP Trusted Managers The 'SNMP Trusted Managers' page allows you to configure up to five SNMP Trusted Managers, based on IP addresses. By default, the SNMP agent accepts SNMP Get and Set requests from any IP address, as long as the correct community string is used in the request.
Page 196: Configuring The Regional Settings
Mediant 600 & Mediant 1000 3.4.1.2 Configuring the Regional Settings The 'Regional Settings' page allows you to define and view the device's internal date and time.  To configure the device's date and time: Open the 'Regional Settings' page (Management tab > Management Configuration menu >...
Page 197: Figure 3-120: Maintenance Actions
SIP User's Manual 3. Web-Based Management  To access the 'Maintenance Actions' page:  On the Navigation bar, click the Management tab, and then in the Navigation tree, select the Management Configuration menu, and then choose the Maintenance Actions page item. Figure 3-120: Maintenance Actions Page 3.4.1.3.1 Resetting the Device The 'Maintenance Actions' page allows you to remotely reset the device.
Page 198: Figure 3-121: Reset Confirmation Message Box
Mediant 600 & Mediant 1000 Click the Reset button; a confirmation message box appears, requesting you to confirm. Figure 3-121: Reset Confirmation Message Box Click OK to confirm device reset; if the parameter 'Graceful Option' is set to 'Yes' (in Step 3), the reset is delayed and a screen displaying the number of remaining calls and time is displayed.
Page 199: Figure 3-122: Device Lock Confirmation Message Box
SIP User's Manual 3. Web-Based Management Click the LOCK button; a confirmation message box appears requesting you to confirm device Lock. Figure 3-122: Device Lock Confirmation Message Box Click OK to confirm device Lock; if 'Graceful Option' is set to 'Yes', the lock is delayed and a screen displaying the number of remaining calls and time is displayed.
Page 200: Software Update
The voice announcement file contains a set of Voice Prompts (VP) that are played by the device during operation. 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 that the device uses.
Page 201: Figure 3-123: Load Auxiliary Files
SIP User's Manual 3. Web-Based Management Notes: • You can schedule automatic loading of updated auxiliary files using HTTP/HTTPS, FTP, or NFS (refer to the Product Reference Manual • For a detailed description on auxiliary files, refer to '' Auxiliary Configuration Files'' on page •...
Page 202: Loading A Software Upgrade Key
Mediant 600 & Mediant 1000 Repeat steps 2 through 3 for each file you want to load. To save the loaded auxiliary files to flash memory, refer to ''Saving Configuration'' page To reset the device (if you have loaded a Call Progress Tones file), refer to ''Resetting...
Page 203: Figure 3-124: Software Upgrade Key Status
SIP User's Manual 3. Web-Based Management The procedure below describes how to load a Software Upgrade Key to the device using the Web interface.  To load a Software Upgrade Key: Open the 'Software Upgrade Key Status' page (Management tab > Software Update menu >...
Page 204: Figure 3-125: Software Upgrade Key With Multiple S/N Lines
Mediant 600 & Mediant 1000 • Multiple S/N lines (as shown below): Figure 3-125: Software Upgrade Key with Multiple S/N Lines in the 'Send Upgrade Key file' field, click the Browse button and navigate to the folder in which the Software Upgrade Key text file is located on your PC.
Page 205: Software Upgrade Wizard
SIP User's Manual 3. Web-Based Management From the 'INI File' drop-down list, select the Software Upgrade Key file. Note that the device's cmp file must be specified in the 'Boot File' field. Configure the initial BootP/TFTP parameters as required, and then click Reset the device;...
Page 206: Figure 3-126: Start Software Upgrade Wizard Screen
Mediant 600 & Mediant 1000 Notes: • Before you can load an ini or any auxiliary file, you must first load a cmp file. • When you activate the wizard, the rest of the Web interface is unavailable. After the files are successfully loaded, access to the full Web interface is restored.
Page 207: Figure 3-127: End Process Wizard
SIP User's Manual 3. Web-Based Management Click one of the following buttons: • Reset; the device resets with the newly loaded cmp, utilizing the existing configuration and auxiliary files. • Next; the 'Load an ini File' wizard page opens. Note that as you progress by clicking Next , the relevant file name corresponding to the applicable Wizard page is highlighted in the file list on the left.
Page 208: Backing Up And Restoring Configuration
Mediant 600 & Mediant 1000 3.4.2.4 Backing Up and Restoring Configuration 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 209: Status & Diagnostics Tab
SIP User's Manual 3. Web-Based Management Status & Diagnostics Tab The Status & Diagnostics tab on the Navigation bar displays menus in the Navigation tree related to device operating status and diagnostics. These menus include the following:  Status & Diagnostics (refer to '' Status &...
Page 210: Viewing Ethernet Port Information
Mediant 600 & Mediant 1000  To activate the Message Log: Set the parameter 'Debug Level' (GwDebugLevel) to 7 (refer ''Configuring Advanced Parameter'' on page ). This parameter determines the Syslog logging level in the range 0 to 6, where 7 is the highest level.
Page 211: Viewing Active Ip Interfaces
SIP User's Manual 3. Web-Based Management  To view Ethernet port information:  Open the ‘Ethernet Port Information’ page (Status & Diagnostics tab > Status & Diagnostics menu > Ethernet Port Information page item). Figure 3-130: Ethernet Port Information Screen Table 3-37: Ethernet Port Information Parameters Parameter Description...
Page 212: Viewing Device Information
Mediant 600 & Mediant 1000 3.5.1.4 Viewing Device Information The 'Device Information' page displays the device's specific hardware and software product information. 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.
Page 213: Viewing Active Alarms
SIP User's Manual 3. Web-Based Management  To view performance statistics:  Open the 'Performance Statistics’ page (Status & Diagnostics tab > Status & Diagnostics menu > Performance Statistics page item) Figure 3-133: Performance Statistics Page  To reset the performance statistics to zero: ...
Page 214: Gateway Statistics
Mediant 600 & Mediant 1000 3.5.2 Gateway Statistics The Gateway Statistics menu allows you to monitor real-time activity such as IP connectivity information, call details and call statistics, including the number of call attempts, failed calls, fax calls, etc. This menu includes the following page items: ...
Page 215: Table 3-38: Call Counters Description
SIP User's Manual 3. Web-Based Management Table 3-38: Call Counters Description Counter Description Number of Attempted Indicates the number of attempted calls. It is composed of established Calls and failed calls. The number of established calls is represented by the 'Number of Established Calls' counter.
Page 216: Viewing Sas Registered Users
Mediant 600 & Mediant 1000 Counter Description GWAPP_SERVICE_NOT_IMPLEMENTED_UNSPECIFIED (79) reason. Number of Failed Calls Indicates the number of calls that failed due to unavailable resources or due to No Resources a device lock. The counter is incremented as a result of one of the following release reasons: ...
Page 217: Viewing Call Routing Status
SIP User's Manual 3. Web-Based Management 3.5.2.3 Viewing Call Routing Status The 'Call Routing Status' page provides you with information on the current routing method used by the device. This information includes the IP address and FQDN (if used) of the Proxy server with which the device currently operates.
Page 218: Viewing Registration Status
Mediant 600 & Mediant 1000 3.5.2.4 Viewing Registration Status The 'Registration Status' page displays whether the device, endpoints and SIP Accounts are registered to a SIP Registrar/Proxy server.  To view Registration status:  Open the 'Registration Status' page (Status & Diagnostics tab > Gateway Statistics menu >...
Page 219: Viewing Ip Connectivity
SIP User's Manual 3. Web-Based Management 3.5.2.5 Viewing IP Connectivity The 'IP Connectivity' page displays online, read-only network diagnostic connectivity information on all destination IP addresses configured in the 'Outbound IP Routing Table' page (refer to ''Configuring the Outbound IP Routing Table'' on page Notes: •...
Page 220 Mediant 600 & Mediant 1000 Column Name Description Connectivity The status of the IP address' connectivity according to the method in the Status 'Connectivity Method' field.  OK = Remote side responds to periodic connectivity queries.  Lost = Remote side didn't respond for a short period.
Page 221: Ini File Configuration
SIP User's Manual 4. INI File Configuration INI File Configuration The device can also be configured by loading an ini file containing user-defined parameters. The ini file can be loaded to the device using the following methods:  Web interface (refer to '' Backing Up and Restoring Configuration'' on page ...
Page 222: Configuring Individual Ini File Parameters
Mediant 600 & Mediant 1000 4.1.1 Configuring Individual ini File Parameters The format of individual ini file parameters includes an optional, subsection name (group name) to conveniently group similar parameters by their functionality. Following this line are the actual parameter settings. These format lines are shown below: [subsection name] ;...
Page 223 SIP User's Manual 4. INI File Configuration  Data line(s): Contain the actual values of the columns (parameters). The values are interpreted according to the Format line. • The first word of the Data line must be the table’s string name followed by the Index field.
Page 224: General Ini File Formatting Rules
Mediant 600 & Mediant 1000 For general ini file formatting rules, refer to ''General ini File Formatting Rules'' on page The table below displays an example of an ini file table parameter: [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce;...
Page 225: Modifying An Ini File
SIP User's Manual 4. INI File Configuration Modifying an ini File You can modify an ini file currently used by the device. Modifying an ini file instead of loading an entirely new ini file preserves the device's current configuration, including factory default values.
Page 226: Secured Encoded Ini File
Mediant 600 & Mediant 1000 Secured Encoded ini File The ini file contains sensitive information that is required for the functioning of the device. Typically, it is loaded to or retrieved from the device using TFTP or HTTP. These protocols are not secure and are vulnerable to potential hackers.
Page 227: Element Management System (Ems)
SIP User's Manual 5. Element Management System (EMS) Element Management System (EMS) This section provides a brief description on configuring various device configurations using AudioCodes Element Management System (EMS). The 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 228: Securing Ems-Device Communication
Mediant 600 & Mediant 1000 Securing EMS-Device Communication 5.2.1 Configuring IPSec Before you can configure the device through the EMS, you need to configure the secure communication protocol IPSec for communicating between the EMS and the device. Before you enable IPSec in the EMS, you must define the IPSec IKE pre-shared key in a secure manner.
Page 229: Changing Ssh Login Password
SIP User's Manual 5. Element Management System (EMS) IPsecSATable_IPsecMode, IPsecSATable_RemoteTunnelAddress, IPsecSATable_RemoteSubnetIPAddress, IPsecSATable_RemoteSubnetPrefixLength; IPsecSATable 1 = <IP address>, 0, <IKE password>, 0, 0, 0, 28800, 28800, 0, 0, 0, 0.0.0.0, 0.0.0.0, 16 [ \IPsecSATable ] EnableIPSec = 1 where: • <IKE password> is the password for the initial IKE pre-shared key. •...
Page 230: Adding The Device In Ems
Mediant 600 & Mediant 1000 Adding the Device in EMS Once you have defined the IPSec communication protocol for communicating between EMS and the device and configured the device's IP address (refer to the device's Installation Manual ), you can add the device in the EMS.
Page 231: Figure 5-3: Adding A Region
SIP User's Manual 5. Element Management System (EMS) In the MG Tree, right-click the Globe icon, and then click Add Region ; the Region dialog box appears. Figure 5-3: Adding a Region In the 'Region Name' field, enter a name for the Region (e.g., a geographical name), and then click OK ;...
Page 232: Configuring Trunks
Mediant 600 & Mediant 1000 Configuring Trunks This section describes the provisioning of trunks:  E1/T1Trunk configuration (refer to ''General Trunk Configuration'' on page  ISDN NFAS (refer to Configuring ISDN NFAS on page 233) 5.4.1 General Trunk Configuration This section describes how to provision a PSTN trunk.
Page 233: Configuring Isdn Nfas
SIP User's Manual 5. Element Management System (EMS) Figure 5-7: General Settings Screen From the 'Protocol Type' drop-down list, select the required protocol. From the 'Framing Method' drop-down list, select the required framing method. For E1, always set this parameter to Extended Super Frame. From the 'Clock Master' drop-down list, set the Clock Master to one of the following values: •...
Page 234: Figure 5-8: Ems Isdn Settings Screen
Mediant 600 & Mediant 1000 With NFAS it is possible to define a group of T1 trunks, called an NFAS group, in which a single D-channel carries ISDN signaling messages for the entire group. The NFAS group’s B-channels are used to carry traffic such as voice or data. The NFAS mechanism also enables definition of a backup D-channel on a different T1 trunk, to be used if the primary D-channel fails.
Page 235 SIP User's Manual 5. Element Management System (EMS) To apply the configured fields to multiple trunks, use the Profiles that appear on the lower part of the screen. Select the General Settings tab, and then configure each trunk in the group with the same values for the following parameters: •...
Page 236: Configuring Basic Sip Parameters
Mediant 600 & Mediant 1000 Configuring Basic SIP Parameters This section describes how to configure the device with basic SIP control protocol parameters using the EMS.  To configure basic SIP parameters: In the MG Tree, select the device that you want to configure; a graphical representation of the device is displayed in the main pane.
Page 237: Configuring Advanced Ipsec/Ike Parameters
SIP User's Manual 5. Element Management System (EMS) Select the Registration tab. Configure 'Is Register Needed' field: ♦ No = the device doesn't register to a Proxy/Registrar server (default). ♦ Yes = the device registers to a Proxy/Registrar server at power up and every user-defined interval (‘Registration Time’...
Page 238: Provisioning Sip Srtp Crypto Offered Suites
Mediant 600 & Mediant 1000 Select the IPSec Proposal tab; the 'IPSec Proposal' screen is displayed. Figure 5-10: IPSec Table Screen Select the button to add a new entry, and then click Yes at the confirmation prompt; a row is added to the table.
Page 239: Provisioning Sip Mlpp Parameters
SIP User's Manual 5. Element Management System (EMS) Open the 'Authentication & Security' screen (Configuration icon > SIP Protocol Definitions menu > Authentication & Security tab). Figure 5-11: Authentication & Security Screen From the 'SRTP Offered Suites' (SRTPofferedSuites) drop-down list, select one of the crypto suites.
Page 240: Configuring The Device To Operate With Snmpv3
Mediant 600 & Mediant 1000 Figure 5-12: MLPP Screen Configure the MLPP parameters as required. Note: If the following RTP DSCP parameters are set to “-1” (i.e., Not Configured, Default), the DiffServ value is set with the PremiumServiceClassMediaDiffserv global gateway parameter, or by using IP Profiles: MLPPRoutineRTPDSCP,...
Page 241: Configuring Snmpv3 Using Ssh
SIP User's Manual 5. Element Management System (EMS) 5.9.1 Configuring SNMPv3 using SSH The procedure below describes how to configure SNMPv3 using SSH.  To configure the device to operate with SNMPv3 via SSH: Open an SSH Client session (e.g. PuTTY), and then connect, using the default user name and password ("Admin"...
Page 242: Configuring Ems To Operate With A Pre-Configured Snmpv3 System
Mediant 600 & Mediant 1000 5.9.2 Configuring EMS to Operate with a Pre-configured SNMPv3 System The procedure below describes how to configure the device with a pre-configured SNMPv3.  To configure the EMS to operate with a pre-configured SNMPv3 system: In the MG Tree, select the required Region to which the device belongs, and then right-click the device.
Page 243: Configuring Snmpv3 To Operate With Non-Configured Snmpv3 System
SIP User's Manual 5. Element Management System (EMS) 5.9.3 Configuring SNMPv3 to Operate with Non-Configured SNMPv3 System The procedure below describes how to configure SNMPv3 using the EMS.  To configure the device to operate with SNMPv3 via EMS (to a non-configured System): In the MG Tree, select the required Region to which the device belongs;...
Page 244: Cloning Snmpv3 Users
Mediant 600 & Mediant 1000 5.9.4 Cloning SNMPv3 Users According to the SNMPv3 standard, SNMPv3 users on the SNMP Agent (on the device) cannot be added via the SNMP protocol, e.g. SNMP Manager (i.e., the EMS). Instead, new users must be defined by User Cloning. The SNMP Manager creates a new user according to the original user permission levels.
Page 245: Upgrading The Device's Software
SIP User's Manual 5. Element Management System (EMS) 5.11 Upgrading the Device's Software The procedure below describes how to upgrade the devices software (i.e., cmp file) using the EMS.  To upgrade the device's cmp file: From the Tools menu, choose Software Manager ;...
Page 246: Figure 5-18: Files Manager Screen
Mediant 600 & Mediant 1000 Select the cmp file, by performing the following: Ensure that the CMP File Only option is selected. In the 'CMP' field, click the browse button and navigate to the required cmp file; the software version number of the selected file appears in the 'Software Version' field.
Page 247: Configuration Parameters Reference
SIP User's Manual 6. 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 248: Multiple Ip Interfaces And Vlan Parameters
Mediant 600 & Mediant 1000 6.1.2 Multiple IP Interfaces and VLAN Parameters The IP network interfaces and VLAN parameters are described in the table below. Table 6-2: IP Network Interfaces and VLAN Parameters Parameter Description Web: Multiple Interface Table EMS: IP Interface Settings...
Page 249 SIP User's Manual 6. Configuration Parameters Reference Parameter Description with a temporary address for initial management and configuration while retaining the address to be used for deployment.  For configuring additional routing rules for other interfaces, use the 'Outbound IP Routing Table'. ...
Page 250 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Native VLAN ID Defines the VLAN ID to which untagged incoming traffic is [VLANNativeVLANID] assigned. Outgoing packets sent to this VLAN are sent only with a priority tag (VLAN ID = 0).
Page 251: Static Routing Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.1.3 Static Routing Parameters The static routing parameters are described in the table below. Table 6-3: Static Routing Parameters Parameter Description Static IP Routing Table Parameters You can define up to 50 static IP routing rules for the device. For example, you can define static routing rules for the OAMP and Control networks, since a default gateway is supported only for the Media traffic network.
Page 252: Quality Of Service Parameters
Mediant 600 & Mediant 1000 Parameter Description Web: Interface Specifies the interface (network type) to which the EMS: Interface Index routing rule is applied. [RoutingTableInterfacesColumn]  = OAMP (default).  = Media.  = Control. For detailed information on the network types, refer to...
Page 253 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Bronze Priority Defines the VLAN priority (IEEE 802.1p) for the EMS: Bronze Service Class Priority Bronze CoS content. [VLANBronzeServiceClassPriority] The valid range is 0 to 7. The default value is 2. Layer-3 Class of Service (TOS/DiffServ) Parameters For detailed information on IP QoS via Differentiated Services, refer to ''IP QoS via Differentiated Services (DiffServ)''...
Page 254: Nat And Stun Parameters
Mediant 600 & Mediant 1000 6.1.5 NAT and STUN Parameters The Network Address Translation (NAT) and Simple Traversal of UDP through NAT (STUN) parameters are described in the table below. Table 6-5: NAT and STUN Parameters Parameter Description STUN Parameters...
Page 255 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Note: For this parameter to take effect, a device reset is required. Web: NAT IP Address Global (public) IP address of the device to enable static NAT EMS: Static NAT IP Address between the device and the Internet.
Page 256: Nfs Parameters
Mediant 600 & Mediant 1000 6.1.6 NFS Parameters The Network File Systems (NFS) configuration parameters are described in the table below. Table 6-6: NFS Parameters Parameter Description [NFSBasePort] Start of the range of numbers used for local UDP ports used by the NFS client.
Page 257: Dns Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.1.7 DNS Parameters The Domain name System (DNS) parameters are described in the table below. Table 6-7: DNS Parameters Parameter Description Web: DNS Primary The IP address of the primary DNS server. Enter the IP address in Server IP dotted-decimal notation, for example, 10.8.2.255.
Page 258: 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 259: Ntp And Daylight Saving Time Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description  This parameter is a special 'Hidden' parameter. Once defined and saved in flash memory, its assigned value doesn't revert to its default even if the parameter doesn't appear in the ini file.
Page 260: Web And Telnet Parameters
Mediant 600 & Mediant 1000 Parameter Description Daylight Saving Time Parameters Web: Day Light Saving Time Determines whether to enable daylight saving time. EMS: Mode  Disable (default) [DayLightSavingTimeEnable]  Enable Web: Start Time Defines the date and time when daylight saving begins.
Page 261: Web Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description the user name and password are transmitted in clear text over the network. Therefore, it's recommended to set the parameter HTTPSOnly to 1 to force the use of HTTPS, since the transport is encrypted.
Page 262: Telnet Parameters
Mediant 600 & Mediant 1000 Parameter Description [ScenarioFileName] Defines the file name of the Scenario file to be loaded to the device. The file name must have the *.dat extension and can be up to 47 characters. For loading a Scenario using the Web interface, refer to Loading a Scenario to the Device on page 48.
Page 263: Debugging And Diagnostics Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Telnet Server Idle Defines the timeout (in minutes) for disconnection of an idle Telnet Timeout session. When set to zero, idle sessions are not disconnected. EMS: Server Idle Disconnect The valid range is any value. The default value is 0. [TelnetServerIdleDisconnec Note: For this parameter to take effect, a device reset is required.
Page 264 Mediant 600 & Mediant 1000 Parameter Description  [0] = Disable device's watch dog. [WatchDogStatus]  [1] = Enable device's watch dog (default). Note: For this parameter to take effect, a device reset is required. [LifeLineType] Defines the scenario upon which the Lifeline phone is activated.
Page 265: Syslog, Cdr And Debug Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description  For ground-start signaling, ensure that the FXO G module is installed (and not the regular FXO module) in the device's chassis.  For FXO ground-start signaling, ensure that the parameter EnableCurrentDisconnect is set to 1 and the parameter FXOBetweenRingTime is set to 300.
Page 266 Mediant 600 & Mediant 1000 Parameter Description Web: Enable Syslog Sends the logs and error message generated by the device to the EMS: Syslog enable Syslog server. [EnableSyslog]  Disable = Logs and errors are not sent to the Syslog server (default).
Page 267 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: CDR Report Determines whether Call Detail Records (CDR) are sent to the Syslog Level server and when they are sent. [CDRReportLevel]  None = CDRs are not used (default).  End Call = CDR is sent to the Syslog server at the end of each call.
Page 268: Remote Alarm Indication Parameters
Mediant 600 & Mediant 1000 Parameter Description  (7) 'Web Access List'  (8) 'Web User Accounts'  [NAA] Non Authorized Access = Attempt to access the Web interface with a false or empty user name or password.  [SPC]...
Page 269: Serial Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.3.4 Serial Parameters The RS-232 serial parameters are described in the table below. (Serial interface is mainly used for debugging and for SMDI Table 6-16: Serial Parameters Parameter Description [DisableRS232] Enables or disables the device's RS-232 port. ...
Page 270: Bootp Parameters
Mediant 600 & Mediant 1000 6.3.5 BootP Parameters The BootP parameters are described in the table below. The BootP parameters are special 'hidden' parameters. Once defined and saved in the device's flash memory, they are used even if they don't appear in the ini file.
Page 271: Security Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description  = Disable (default). [ExtBootPReqEnable]  = Enable extended information to be sent in BootP request. If enabled, the device uses the Vendor Specific Information field in the BootP request to provide device-related initial startup information such as blade type, current IP address, software version.
Page 272: Https Parameters
Mediant 600 & Mediant 1000 Parameter Description Web: Internal Firewall Parameters EMS: Firewall Settings [AccessList] This ini file table parameter configures the device's access list (firewall), which defines network traffic filtering rules. For each packet received on the network interface, the table is scanned from the top down until a matching rule is found.
Page 273 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: HTTPS Cipher String Defines the Cipher string for HTTPS (in OpenSSL cipher list [HTTPSCipherString] format). For the valid range values, refer to URL http://www.openssl.org/docs/apps/ciphers.html. The default value is ‘EXP’ (Export encryption algorithms). For example, use ‘ALL’...
Page 274: Srtp Parameters
Mediant 600 & Mediant 1000 6.4.3 SRTP Parameters The Secure Real-Time Transport Protocol (SRTP) parameters are described in the table below. Table 6-20: SRTP Parameters Parameter Description Web: Media Security Enables Secure Real-Time Transport Protocol (SRTP). EMS: Enable Media Security ...
Page 275: Tls Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Disable Encryption On On a secured RTP session, this parameter determines whether to Transmitted RTCP Packets enable encryption on transmitted RTCP packets. EMS: RTCP EncryptionDisable  Enable (default)  [1] Disable [RTCPEncryptionDisableTx] 6.4.4 TLS Parameters...
Page 276: Ssh Parameters
Mediant 600 & Mediant 1000 Parameter Description When a remote certificate is received and this parameter is not disabled, the value of SubjectAltName is compared with the list of available Proxies. If a match is found for any of the configured Proxies, the TLS connection is established.
Page 277: Ipsec Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description [SSHRequirePublicKey] Enables or disables RSA public keys for SSH.  = RSA public keys are optional if a value is configured for the parameter SSHAdminKey (default).  = RSA public keys are mandatory. Web/EMS: SSH Server Enables or disables the embedded Secure Shell (SSH) server.
Page 278 Mediant 600 & Mediant 1000 Parameter Description Notes:  Each row in the table refers to a different IP destination.  To support more than one Encryption/Authentication proposal, for each proposal specify the relevant parameters in the Format line. ...
Page 279: Ocsp Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.4.7 OCSP Parameters The Online Certificate Status Protocol (OCSP) parameters are described in the table below. Table 6-24: OCSP Parameters Parameter Description EMS: OCSP Enable Enables or disables certificate checking using OCSP. [OCSPEnable] ...
Page 280 Mediant 600 & Mediant 1000 Parameter Description Web: AAA Indications Determines the Authentication, Authorization and Accounting EMS: Indications (AAA) indications. [AAAIndications]  None = No indications (default).  Accounting Only = Only accounting indications are used. Web: Device Behavior Upon Defines the device's response upon a RADIUS timeout.
Page 281: Snmp Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: RADIUS VSA Vendor ID Defines the vendor ID that the device accepts when parsing a [RadiusVSAVendorID] RADIUS response packet. The valid range is 0 to 0xFFFFFFFF. The default value is 5003. Web: RADIUS VSA Access Defines the code that indicates the access level attribute in the Level Attribute...
Page 282 Mediant 600 & Mediant 1000 Parameter Description EMS: Keep Alive Trap Port The port to which the keep-alive traps are sent. [KeepAliveTrapPort] The valid range is 0 - 65534. The default is port 162. [SendKeepAliveTrap] When enabled, this parameter invokes the keep-alive trap and sends it every 9/10 of the time defined in the parameter defining NAT Binding Default Timeout.
Page 283 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [SNMPManagerTrapUser_x] This parameter can be set to the name of any configured SNMPV3 user to associate with this trap destination. This determines the trap format, authentication level, and encryption level. By default, the trap is associated with the SNMP trap community string.
Page 284: Sip Configuration Parameters
Mediant 600 & Mediant 1000 SIP Configuration Parameters This subsection describes the device's SIP parameters. 6.7.1 General SIP Parameters The general SIP parameters are described in the table below. Table 6-27: General SIP Parameters Parameter Description [SIPForceRport] Determines whether the device sends SIP responses to the UDP port from where SIP requests are received even if the 'rport' parameter is not present in the SIP Via header.
Page 285 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Note: You can also configure early SIP 183 response immediately upon receipt of an INVITE, using the EnableEarly183 parameter. Web/EMS: Enable Early 183 Determines whether the device sends a SIP 183 response with [EnableEarly183] SDP to the IP immediately upon receipt of an INVITE message (for IP-to-Tel calls).
Page 286 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Session Expires Determines the SIP method used for session-timer updates. Method  Re-INVITE = Uses Re-INVITE messages for session- [SessionExpiresMethod] timer updates (default).  [1] UPDATE = Uses UPDATE messages. Notes: ...
Page 287 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Fax Signaling Method Determines the SIP signaling method for establishing and EMS: Fax Used transmitting a fax session after a fax is detected. [IsFaxUsed]  No Fax = No fax negotiation using SIP signaling. Fax transport method is according to the parameter FaxTransportMode (default).
Page 288 Mediant 600 & Mediant 1000 Parameter Description Web: SIP TCP Local Port Local TCP port for SIP messages. EMS: TCP Local SIP Port The valid range is 1 to 65535. The default value is 5060. [TCPLocalSIPPort] Web: SIP TLS Local Port Local TLS port for SIP messages.
Page 289 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: SIP Destination Port SIP destination port for sending initial SIP requests. EMS: Destination Port The valid range is 1 to 65534. The default port is 5060. [SIPDestinationPort] Note: SIP responses are sent to the port specified in the Via header.
Page 290 Mediant 600 & Mediant 1000 Parameter Description Web: Enable History-Info Header Enables usage of the History-Info header. EMS: Enable History Info  Disable (default) [EnableHistoryInfo]  [1] Enable User Agent Client (UAC) Behavior:  Initial request: The History-Info header is equal to the Request-URI.
Page 291 SIP User's Manual 6. Configuration Parameters Reference Parameter Description  Send and Receive = The functionality of outgoing SIP messages is identical to the functionality described in option 1. In addition, for incoming SIP INVITEs, if the Request-URI includes a 'tgrp' parameter, the device routes the call according to that value (if possible).
Page 292 Mediant 600 & Mediant 1000 Parameter Description parameter for IP-to-Tel routing. If the received INVITE Request-URI does not 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.
Page 293 SIP User's Manual 6. Configuration Parameters Reference Parameter Description If the Registrar/Proxy supports GRUU, the REGISTER responses contain the 'gruu' parameter in each Contact header field. The Registrar/Proxy provides the same GRUU for the same AOR and instance-id in case of sending REGISTER again after expiration of the registration.
Page 294 Mediant 600 & Mediant 1000 Parameter Description Web: Multiple Packetization Time Determines whether the 'mptime' attribute is included in the Format outgoing SDP. EMS: Multi Ptime Format  None = Disabled (default) [MultiPtimeFormat]  PacketCable = includes the 'mptime' attribute in the...
Page 295 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Enable P-Associated- Determines the device usage of the P-Associated-URI header. URI Header This header can be received in 200 OK responses to [EnablePAssociatedURIHeader] REGISTER requests. When enabled, the first URI in the P- Associated-URI header is used in subsequent requests as the From/P-Asserted-Identity headers value.
Page 296 Mediant 600 & Mediant 1000 Parameter Description BYE in response to any additional SIP 2xx received from the Proxy within this timeout. Once this timeout elapses, the device ignores any subsequent SIP 2xx. The number of supported forking calls per channel is 20. In...
Page 297 SIP User's Manual 6. Configuration Parameters Reference Parameter Description This parameter may be useful, for example, for service providers who 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.
Page 298 Mediant 600 & Mediant 1000 Parameter Description message sets the 'rport' value of the response to the actual port from where the request was received. This method is used, for example, to enable the device to identify its port mapping outside a NAT.
Page 299 SIP User's Manual 6. Configuration Parameters Reference Parameter Description with SDP in response to an ISDN Alerting or it sends a 183 Session Progress message with SDP in response to only the first received ISDN Proceeding or Progress message after a call is placed to PBX/PSTN over the trunk.
Page 300 Mediant 600 & Mediant 1000 Parameter Description  For mapping SIP-to-Q.931 and Q.931-to-SIP release causes, refer to Configuring Release Cause Mapping on page 169.  For a list of SIP responses-Q.931 release cause mapping, refer to '' Release Reason Mapping''...
Page 301 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [TransparentCoderPresentation] Determines the format of the Transparent coder representation in the SDP.  [0] = clearmode (default)  [1] = X-CCD Web: Comfort Noise Generation Enables negotiation and usage of Comfort Noise (CN). Negotiation ...
Page 302 Mediant 600 & Mediant 1000 Parameter Description This parameter is typically implemented for incoming IP-to-Tel collect calls to the FXS port. If the FXS user does not wish to accept the collect call, the user disconnects the call by on- hooking the phone.
Page 303 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: SIT Q850 Cause For Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-IC [SITQ850CauseForIC] (Operator Intercept Special Information Tone) is detected from the PSTN for IP-to-Tel calls.
Page 304 Mediant 600 & Mediant 1000 Parameter Description  The IP Connectivity mechanism is enabled (using the parameter AltRoutingTel2IPEnable) and there is no connectivity to any destination IP address. Notes:  For Analog interfaces: The FXSOOSBehavior parameter determines the behavior of the FXS endpoints when a Busy Out or Graceful Lock occurs.
Page 305 SIP User's Manual 6. Configuration Parameters Reference Parameter Description  The third retransmission is sent after 2000 (2*1000) msec.  The fourth retransmission and subsequent retransmissions until SIPMaxRtx are sent after 4000 (2*2000) msec. Web: SIP T2 Retransmission The maximum interval (in msec) between retransmissions of Timer [msec] SIP messages.
Page 306: Network Application Parameters
Mediant 600 & Mediant 1000 6.7.2 Network Application Parameters The SIP network application parameters are described in the table below: Table 6-28: SIP Network Application Parameters Parameter Description Web: Default CP Media For a description of this parameter, refer to ''...
Page 307 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web :Signaling Routing Domain (SRD) Table EMS: SRD Table [SRD] This file table parameter configures the Signaling Routing Domain (SRD) table. The format of this parameter is as follows: [SRD] FORMAT SRD_Index = SRD_Name, SRD_MediaRealm; [\SRD] For example: SRD 1 = LAN_SRD, Mrealm1;...
Page 308: Ip Group, Proxy, Registration And Authentication Parameters
Mediant 600 & Mediant 1000 6.7.3 IP Group, Proxy, Registration and Authentication Parameters The proxy server, registration and authentication SIP parameters are described in the table below. Table 6-29: Proxy, Registration and Authentication SIP Parameters Parameter Description Web: IP Group Table EMS: Endpoints >...
Page 309 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Authentication Table EMS: SIP Endpoints > Authentication [Authentication] This ini file table parameter defines a user name and password for authenticating each device port. The format of this parameter is as follows: [Authentication] FORMAT Authentication_Index = Authentication_UserId, Authentication_UserPassword, Authentication_Module,...
Page 310 Mediant 600 & Mediant 1000 Parameter Description Web: Account Table EMS: SIP Endpoints > Account [Account] This ini file table parameter configures the Account table for registering and/or authenticating (digest) Trunk Groups or IP Groups (e.g., an IP-PBX) to a Serving IP Group (e.g., an Internet Telephony Service Provider - ITSP).
Page 311 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Redundancy Mode Determines whether the device switches back to the primary EMS: Proxy Redundancy Mode Proxy after using a redundant Proxy. [ProxyRedundancyMode]  Parking = device continues working with a redundant (now active) Proxy until the next failure, after which it works with the next redundant Proxy (default).
Page 312 Mediant 600 & Mediant 1000 Parameter Description Note: This option is applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0.  [2] Routing Table = Uses the Routing table to locate the destination and then sends a new INVITE to this destination.
Page 313 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Proxy DNS Query Type Enables the use of DNS Naming Authority Pointer (NAPTR) [ProxyDNSQueryType] and Service Record (SRV) queries to discover Proxy servers.  A-Record (default)   NAPTR If set to A-Record [0], no NAPTR or SRV queries are performed.
Page 314 Mediant 600 & Mediant 1000 Parameter Description SIPGatewayName. The device uses the OPTIONS request as a keep-alive message to its primary and redundant Proxies (i.e., the parameter EnableProxyKeepAlive is set to 1). Web/EMS: User Name User name used for Registration and Basic/Digest [UserName] authentication with a Proxy/Registrar server.
Page 315 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Proxy IP Table EMS: Proxy IP [ProxyIP] This file table parameter configures the Proxy Set table with up to six Proxy Set IDs, each with up to five Proxy server IP addresses (or fully qualified domain name/FQDN).
Page 316 Mediant 600 & Mediant 1000 Parameter Description  For configuring the Proxy Set IDs and their IP addresses, use the parameter ProxyIP.  For configuring the Proxy Set ID table using the Web interface and for a detailed description of the parameters of...
Page 317 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Registrar Transport Determines the transport layer used for outgoing SIP dialogs Type initiated by the device to the Registrar. [RegistrarTransportType]  [-1] Not Configured (default)    [2] TLS Note: When set to ‘Not Configured’, the value of the parameter SIPTransportType is used.
Page 318 Mediant 600 & Mediant 1000 Parameter Description Web: Gateway Registration Name Defines the user name that is used in the From and To EMS: Name headers in SIP REGISTER messages. If no value is specified [GWRegistrationName] (default) for this parameter, the UserName parameter is used instead.
Page 319: Voice Mail Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description REGISTER request. UA's should support this mechanism so that bindings can be removed before their expiration interval has passed. Use of the "*" Contact header field value allows a registering UA to remove all bindings associated with an address-of-record (AOR) without knowing their precise values.
Page 320 Mediant 600 & Mediant 1000 Parameter Description Web: Enable VoiceMail URI Enables or disables the interworking of target and cause for EMS: Enable VMURI redirection from Tel to IP and vice versa, according to RFC 4468. [EnableVMURI]  Disable = Disable (default).
Page 321 SIP User's Manual 6. Configuration Parameters Reference Parameter Description SMDI Parameters Web/EMS: Enable SMDI Enables Simplified Message Desk Interface (SMDI) interface on [SMDI] the device.  [0] Disable = Normal serial (default)  [1] Enable (Bellcore)  [2] Ericsson MD-110 ...
Page 322 Mediant 600 & Mediant 1000 Parameter Description Digit Patterns The following digit pattern parameters apply only to voice mail applications that use the DTMF communication method. For the available pattern syntaxes, refer to the CPE Configuration Guide for Voice Mail...
Page 323: Fax And Modem Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: External Call Digit Pattern Determines the digit pattern used by the PBX to indicate an EMS: Digit Pattern External Call external call. [DigitPatternExternalCall] The valid range is a 120-character string. Web: Disconnect Call Digit Determines a digit pattern that when received from the Tel side, Pattern...
Page 324 Mediant 600 & Mediant 1000 Parameter Description If the initial INVITE used to establish the voice call (not fax) was already sent, a CANCEL (if not connected yet) or a BYE (if already connected) is sent to tear down the voice call.
Page 325: Dtmf And Hook-Flash Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.7.6 DTMF and Hook-Flash Parameters The DTMF and hook-flash parameters parameters are described in the table below. Table 6-32: DTMF and Hook-Flash Parameters Parameter Description Hook-Flash Parameters Web/EMS: Hook-Flash Code For analog interfaces: Defines the digit pattern that when received [HookFlashCode] from the Tel side, indicates a Hook Flash event.
Page 326 Mediant 600 & Mediant 1000 Parameter Description Web: Min. Flash-Hook Defines the minimum time (in msec) for detection of a hook-flash Detection Period [msec] event. Detection is guaranteed for hook-flash periods of at least 60 EMS: Min Flash Hook Time msec (when setting the minimum time to 25).
Page 327 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Tx DTMF Option Determines a single or several preferred transmit DTMF [TxDTMFOption] negotiation methods.  Not Supported = No negotiation - DTMF digits are sent according to the parameters DTMFTransportType and RFC2833PayloadType (default).
Page 328 Mediant 600 & Mediant 1000 Parameter Description [DisableAutoDTMFMute] Enables/disables the automatic muting of DTMF digits when out-of- band DTMF transmission is used.  = Automatic mute is used (default).  = No automatic mute of in-band DTMF. When this parameter is set to 1, the DTMF transport type is set...
Page 329 SIP User's Manual 6. Configuration Parameters Reference Parameter Description For example (for FXS/FXO interfaces), the called number can be as follows: d1005, dpp699, p9p300. To add the 'd' and 'p' digits, use the usual number manipulation rules.  For analog interfaces: To use this feature with FXO interfaces, configure the device to operate in one-stage dialing mode.
Page 330: Digit Collection And Dial Plan Parameters
Mediant 600 & Mediant 1000 6.7.7 Digit Collection and Dial Plan Parameters The digit collection and dial plan parameters are described in the table below. Table 6-33: Digit Collection and Dial Plan Parameters Parameter Description Web/EMS: Dial Plan Index Determines the Dial Plan index to use in the external Dial Plan file.
Page 331: Coders And Profile Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Plan index), then the parameter DigitMapping is ignored.  For a detailed description of the digit mapping, refer to ''Digit Mapping'' on page Web: Max Digits in Phone Defines the maximum number of collected destination number digits that can be received (i.e., dialed) from the Tel side (analog) EMS: Max Digits in Phone...
Page 332 Mediant 600 & Mediant 1000 Parameter Description  Name = Coder name.  Ptime = Packetization time (ptime) - how many coder payloads are combined into a single RTP packet.  Rate = Packetization rate.  PayloadType = Identifies the format of the RTP payload.
Page 333 SIP User's Manual 6. Configuration Parameters Reference Parameter Description G.726 10, 20 16 [0] Dynamic Disable [0] [g726] (default), 30, (default), (0-127) Enable [1] 40, 50, 60, 80, 24 [1], Default 100, 120 32 [2], is 23 Disable [0] G.727 ADPCM 10, 20 16, 24, Dynamic...
Page 334 Mediant 600 & Mediant 1000 Parameter Description packetization time is assigned the default value.  The value of several fields is hard-coded according to common standards (e.g., payload type of G.711 U-law is always 0). Other values can be set dynamically. If no value is specified for a dynamic field, a default value is assigned.
Page 335 SIP User's Manual 6. Configuration Parameters Reference Parameter Description For example: IPProfile 0 = Sevilia, 1, 1, 0, 10, 10, 46, 40, 0, 0, 0, 0, 2, 0, 0, 0, 0, -1, 1, 0, 0, -1, 1, -1, -1, 1, 1, 0, 0, , -1, 4294967295, 0; Notes: ...
Page 336 Mediant 600 & Mediant 1000 Parameter Description [TelProfile] FORMAT TelProfile_Index = TelProfile_ProfileName, TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP, TelProfile_TimeForReorderTone, TelProfile_EnableDIDWink, TelProfile_IsTwoStageDial, TelProfile_DisconnectOnBusyTone, TelProfile_EnableVoiceMailDelay, TelProfile_DialPlanIndex, TelProfile_Enable911PSAP, TelProfile_SwapTelToIpPhoneNumbers, TelProfile_EnableAGC, TelProfile_ECNlpMode;...
Page 337: Supplementary Services Parameters
SIP User's Manual 6. Configuration Parameters Reference Supplementary Services Parameters This subsection describes the device's supplementary telephony services parameters. 6.8.1 Caller ID Parameters The caller ID parameters are described in the table below. Table 6-35: Caller ID Parameters Parameter Description Web: Caller ID Permissions Table EMS: SIP Endpoints >...
Page 338 Mediant 600 & Mediant 1000 Parameter Description From header. The format of this parameter is as follows: [CallerDisplayInfo] FORMAT CallerDisplayInfo_Index = CallerDisplayInfo_DisplayString, CallerDisplayInfo_IsCidRestricted, CallerDisplayInfo_Module, CallerDisplayInfo_Port; [\CallerDisplayInfo] Where,  DisplayString = Caller ID string (up to 18 characters).  IsCidRestricted = ...
Page 339 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Enable Caller ID Determines whether Caller ID is enabled. [EnableCallerID]  [0] Disable = Disable the Caller ID service (default).  [1] Enable = Enable the Caller ID service. If the Caller ID service is enabled, then for FXS interfaces, calling number and Display text (from IP) are sent to the device's port.
Page 340 Mediant 600 & Mediant 1000 Parameter Description Web: Enable FXS Caller ID Enables the interworking of Calling Party Category (cpc) code Category Digit For Brazil Telecom from SIP INVITE messages to FXS Caller ID first digit. [AddCPCPrefix2BrazilCallerID]  [0] Disable (default) ...
Page 341 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: Caller ID Timing Mode Determines when Caller ID is generated. [AnalogCallerIDTimingMode]  [0] = Caller ID is generated between the first two rings (default).  [1] = The device attempts to find an optimized timing to generate the Caller ID according to the selected Caller ID type.
Page 342: Call Waiting Parameters
Mediant 600 & Mediant 1000 Parameter Description restricted (received from Tel or configured in the device), the From header is set to <anonymous@anonymous.invalid>. The 200 OK response can contain the connected party CallerID - Connected Number and Connected Name. For example, if the call is answered by the device, the 200 OK response includes the P-Asserted-Identity with Caller ID.
Page 343 SIP User's Manual 6. Configuration Parameters Reference Parameter Description  For information on the Call Waiting feature, refer to ''Call Waiting'' on page  For information on the Call Progress Tones file, refer to Configuring the Call Progress Tones File. EMS: Send 180 For Call Waiting Determines the SIP response code for indicating Call Waiting.
Page 344 Mediant 600 & Mediant 1000 Parameter Description  For a description on using ini file table parameters, refer to Configuring ini File Table Parameters on page 222. Web: Number of Call Waiting Number of Call Waiting indications that are played to the called Indications telephone that is connected to the device for Call Waiting.
Page 345: Call Forwarding Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description INFO sip:06@192.168.13.2:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.13.40:5060;branch=z9hG4bK040066422630 From: <sip:4505656002@192.168.13.40:5060>;tag=1455352915 To: <sip:06@192.168.13.2:5060> Call-ID:0010-0008@192.168.13.2 CSeq:342168303 INFO Content-Length:28 Content-Type:application/broadsoft play tone CallWaitingTone1 6.8.3 Call Forwarding Parameters The call forwarding parameters are described in the table below. Table 6-37: Call Forwarding Parameters Parameter Description...
Page 346 Mediant 600 & Mediant 1000 Parameter Description  Destination = Telephone number or URI (<number>@<IP address>) to where the call is forwarded.  NoReplyTime = Timeout (in seconds) for No Reply. If you have set the Forward Type for this port to No Answer [3], enter the number of seconds the device waits before forwarding the call to the specified phone number.
Page 347: Message Waiting Indication Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.8.4 Message Waiting Indication Parameters The message waiting indication (MWI) parameters are described in the table below. Table 6-38: MWI Parameters Parameter Description Web: Enable MWI Enables Message Waiting Indication (MWI). EMS: MWI Enable ...
Page 348 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: MWI Server Transport Determines the transport layer used for outgoing SIP dialogs Type initiated by the device to the MWI server. [MWIServerTransportType]  [-1] Not Configured (default)    [2] TLS Note: When set to ‘Not Configured’, the value of the parameter...
Page 349: Call Hold Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.8.5 Call Hold Parameters The call hold parameters are described in the table below. Table 6-39: 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 350: Call Transfer Parameters
Mediant 600 & Mediant 1000 6.8.6 Call Transfer Parameters The call transfer parameters are described in the table below. Table 6-40: Call Transfer Parameters Parameter Description Web/EMS: Enable Transfer Determines whether call transfer is enabled. [EnableTransfer]  Disable = Disable the call transfer service.
Page 351: Three-Way Conferencing Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description [KeyBlindTransfer] Keypad sequence that activates blind transfer for Tel-to-IP calls. There are two possible scenarios:  Option 1: After this sequence is dialed, the current call is put on hold (using Re-INVITE), a dial tone is played to the B-channel, and then phone number collection starts.
Page 352: 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 353: Fxs Call Cut-Through Parameter
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Emergency Defines a list of numbers, which are defined as 'emergency Numbers numbers'. When one of these numbers is dialed, the outgoing INVITE [EmergencyNumbers] message includes the Priority and Resource-Priority headers. If the user sets the phone on-hook, the call is not disconnected, but instead a Hold Re-INVITE request is sent to the remote party.
Page 354: Automatic Dialing Parameters
Mediant 600 & Mediant 1000 6.8.10 Automatic Dialing Parameters The automatic dialing upon off-hook parameters are described in the table below. Table 6-44: Automatic Dialing Parameters Parameter Description Web: Automatic Dialing Table EMS: SIP Endpoints > Auto Dial [TargetOfChannel] This file table parameter defines telephone numbers that are automatically dialed when a specific FXS or FXO port is used (i.e.,...
Page 355: Direct Inward Dialing Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.8.11 Direct Inward Dialing Parameters The Direct Inward Dialing (DID) parameters are described in the table below. Table 6-45: DID Parameters Parameter Description Web/EMS: Enable DID Enables Direct Inward Dialing (DID) using Wink-Start signaling. Wink ...
Page 356: Mlpp Parameters
Mediant 600 & Mediant 1000 Parameter Description ''Configuring ini File Table Parameters'' on page [WinkTime] Defines the time (in msec) elapsed between two consecutive polarity reversals. This parameter can be used for DID signaling, for example, E911 lines to the Public Safety Answering Point (PSAP), according to the Bellcore GR-350-CORE standard (refer to the ini file parameter Enable911PSAP).
Page 357 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: MLPP DiffServ Defines the DiffServ value (differentiated services code EMS: Diff Serv point/DSCP) used in IP packets containing SIP messages that [MLPPDiffserv] are related to MLPP calls. This parameter defines DiffServ for incoming and outgoing MLPP calls with the Resource-Priority header.
Page 358 Mediant 600 & Mediant 1000 Parameter Description "000000". Note: This parameter is applicable only to the MLPP NI-2 ISDN variant with CallPriorityMode set to 1. EMS: E911 MLPP Behavior Defines the E911(or Emergency Telecommunication Services [E911MLPPBehavior] - ETS) MLPP Preemption mode: ...
Page 359: Standalone Survivability Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Immediate precedence call Immediate level. [MLPPImmediateRTPDSCP] The valid range is -1 to 63. The default is -1. 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 360 Mediant 600 & Mediant 1000 Parameter Description Web: SAS Default Gateway IP The default gateway used in SAS 'Emergency Mode'. When an EMS: Default Gateway IP incoming SIP INVITE is received and the destination Address-Of- [SASDefaultGatewayIP] Record is not included in the SAS database, the request is immediately sent to this default gateway.
Page 361 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Redundant SAS Proxy Set Determines the Proxy Set (index number) used in SAS EMS: Redundant Proxy Set Emergency mode for fallback when the user is not found in the [RedundantSASProxySet] Registered Users database.
Page 362 Mediant 600 & Mediant 1000 Parameter Description Web: SAS Emergency Numbers Defines emergency numbers for the device's SAS application. [SASEmergencyNumbers] When the device's SAS agent receives a SIP INVITE (from an IP phone) that includes one of the emergency numbers (in the SIP user part), it forwards the INVITE to the default gateway (configured by the parameter SASDefaultGatewayIP), i.e., the...
Page 363: Ip Media Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: SAS IP-to-IP Routing Table [IP2IPRouting] This file table parameter configures the IP-to-IP Routing table for SAS routing rules. The format of this parameter is as follows: [IP2IPRouting] FORMAT IP2IPRouting_Index = IP2IPRouting_SrcIPGroupID, IP2IPRouting_SrcUsernamePrefix, IP2IPRouting_SrcHost, IP2IPRouting_DestUsernamePrefix, IP2IPRouting_DestHost, IP2IPRouting_DestType, IP2IPRouting_DestIPGroupID,...
Page 364 [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 365 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [VoiceStreamUploadPostURI] Defines the URI used on the POST request to upload voice data from the media server to a Web server. Note: For this parameter to take effect, a device reset is required.
Page 366 Mediant 600 & Mediant 1000 Parameter Description Web: MSCML ID Media Server Control Markup Language (MSCML) identification [MSCMLID] string (up to 16 characters). To start an MSCML session, the application server sends a regular SIP INVITE message with a SIP URI that includes this string.
Page 367 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Conferencing Parameters Web/EMS: Conference ID Conference Identification string (up to 16 characters). [ConferenceID] The default value is ‘conf’. For example: ConferenceID = MyConference Note: 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 368 Mediant 600 & Mediant 1000 Parameter Description Automatic Gain Control (AGC) Parameters Web: Enable AGC Activates the AGC mechanism. The AGC mechanism adjusts EMS: AGC Enable the level of the received signal to maintain a steady [EnableAGC] (configurable) volume level.
Page 369 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: AGC Redirection Determines the AGC direction. EMS: Redirection  [0] 0 = AGC works on signals from the TDM side (default). [AGCRedirection]  [1] 1 = AGC works on signals from the IP side. Web: AGC Target Energy Determines the signal energy value (dBm) that the AGC EMS: Target Energy...
Page 370 Mediant 600 & Mediant 1000 Parameter Description Detector (AMD) on page 590. Notes:  For configuring higher sensitivity resolutions (i.e., greater than 7), set the parameter AMDSensitivityResolution to 1 (High), and then for ini file configuration use the parameter AMDDetectionSensitivityHighResolution to define the sensitivity level.
Page 371 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Energy Detector Parameters Enable Energy Detector Currently, not supported. [EnableEnergyDetector] Energy Detector Quality Factor Currently, not supported. [EnergyDetectorQualityFactor] Energy Detector Threshold Currently, not supported. [EnergyDetectorThreshold] Pattern Detection Parameters Note: For an overview on the pattern detector feature for TDM tunneling, refer to DSP Pattern Detector on page 642.
Page 372 Mediant 600 & Mediant 1000 Parameter Description [VxmlCompleteTimeout] Optional parameter that defines the amount of silence (in msec) to wait after speech grammar has been matched before reporting the match. The default value is 0 (i.e., don't set this parameter on recognition attempt).
Page 373 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [VxmlMaxPorts] Defines the number of channels in the system that can simultaneously run VXML scripts. The range is from 0 to the maximum number of channels in the system. This value can be used to ensure there are sufficient VXML resources for each call.
Page 374 Mediant 600 & Mediant 1000 Parameter Description [VxmlTermChar] Defines the default terminating digit for received DTMF. The default value is 35 (equivalent to ASCII '#'). Note: For this parameter to take effect, a device reset is required. [VxmlTermTimeout] Defines the time to wait before terminating received DTMF (in msec).
Page 375: Pstn Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description active in RTSP sessions. The range is 0 - 20.The default value is 10. Note: For this parameter to take effect, a device reset is required. 6.11 PSTN Parameters This subsection describes the device's PSTN parameters. 6.11.1 General Parameters The general PSTN parameters are described in the table below.
Page 376 Mediant 600 & Mediant 1000 Parameter Description  [14] T1 DMS100 ISDN = ISDN PRI protocol for the Nortel™ DMS switch.  [15] J1 TRANSPARENT  [16] T1 NTT ISDN = ISDN PRI protocol for the Japan - Nippon Telegraph Telephone (known also as INS 1500).
Page 377 SIP User's Manual 6. Configuration Parameters Reference Parameter Description  When both the parameters ISDNDmsTimerT310 and ISDNTimerT310 are configured, the value of the parameter ISDNTimerT310 prevails. [ISDNDMSTimerT310] Overrides the T310 timer for the DMS-100 ISDN variant. T310 defines the timeout between the receipt of a Proceeding message and the receipt of an Alerting/Connect message.
Page 378 Mediant 600 & Mediant 1000 Parameter Description  T1 FRAMING F72 = T1 72-Frame multiframe (SLC96)  [F] T1 FRAMING ESF CRC6 J2 = J1 Extended SuperFrame with CRC6 (Japan) Note: This parameter is not configurable for BRI interfaces; the device automatically uses the BRI framing method.
Page 379: Tdm Bus And Clock Timing Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description  [3] -22.5 dB Note: This parameter is applicable only to T1 trunks. [TDMHairPinning] Defines static TDM hair-pinning (cross-connection) performed at initialization. The connection is between trunks with an option to exclude a single B-Channel in each trunk.
Page 380 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Idle ABCD Pattern Defines the ABCD (CAS) Pattern that is applied to the CAS [IdleABCDPattern] signaling bus when the channel is idle. The valid range is 0x0 to 0xF. The default is -1 (i.e., default pattern is 0000).
Page 381 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: TDM Bus PSTN Auto Enables or disables the PSTN trunk Auto-Fallback Clock feature. FallBack Clock  Disable (default) = Recovers the clock from the E1/T1 line EMS: TDM Bus Auto Fall Back defined by the parameter TDMBusLocalReference.
Page 382: Cas Parameters
Mediant 600 & Mediant 1000 6.11.3 CAS Parameters The Common Channel Associated (CAS) parameters are described in the table below. Note that CAS is not applicable to BRI interfaces. Table 6-51: CAS Parameters Parameter Description Web: CAS Transport Type Controls the ABCD signaling transport type over IP.
Page 383 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Dial Plan The CAS Dial Plan name that is used on a specific trunk (where denotes the trunk ID). EMS: Dial Plan Name [CASTrunkDialPlanName_x] The range is up to 11 characters. For example, the below configures E1_MFCR2 trunk with a single protocol (Trunk 5): ProtocolType_5 = 7...
Page 384 Mediant 600 & Mediant 1000 Parameter Description [CASTablesNum] Indicates how many CAS protocol configurations files are loaded. The valid range is 1 to 8. Note: For this parameter to take effect, a device reset is required. CAS State Machines Parameters...
Page 385: Isdn Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.11.4 ISDN Parameters The ISDN parameters are described in the table below. Table 6-52: ISDN Parameters Parameter Description Web: ISDN Termination Side Selects the ISDN termination side. EMS: Termination Side  User side = ISDN User Termination Equipment (TE) [TerminationSide] side (default) ...
Page 386 Mediant 600 & Mediant 1000 Parameter Description NFAS Parameters Note: These parameters are not applicable to BRI interfaces. Web: NFAS Group Number Indicates the NFAS group number (NFAS member) for the EMS: Group Number selected trunk, where x depicts the Trunk ID.
Page 387 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: PI For Setup Message Determines whether and which Progress Indicator (PI) [PIForSetupMsg] information element (IE) is added to the sent ISDN Setup message. Some ISDN protocols such as NI-2 or Euro ISDN can optionally contain PI = 1 or PI = 3 in the Setup message.
Page 388 Mediant 600 & Mediant 1000 Parameter Description  Network provided, Network provided: the first calling number is used  Network provided, User provided: the second one is used  User provided, Network provided: the first one is used  User provided, user provided: the first one is used...
Page 389 SIP User's Manual 6. Configuration Parameters Reference Parameter Description  [65536] EXPLICIT PRES SCREENING = The calling party number (octet 3a) is always present even when presentation and screening are at their default. Note: This option is applicable only to ETSI, NI-2, and 5ESS.
Page 390 Mediant 600 & Mediant 1000 Parameter Description Web: General Call Control Behavior Bit-field for determining several general CC behavior EMS: General CC Behavior options. To select the options, click the arrow button, and [ISDNGeneralCCBehavior] then for each required option, select 1 to enable. The default is 0 (i.e., disable).
Page 391 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Service, as specified in FSD 01-02-40AG Feature Specification Document from Verizon. Otherwise, TBCT is implemented as specified in GR-2865-CORE specification (default behavior). Note: When using the ini file to configure the device to support several ISDNGeneralCCBehavior features, add the individual feature values.
Page 392: Isdn And Cas Interworking Parameters
Mediant 600 & Mediant 1000 Parameter Description format of the DLCI field must be used. Note: When using the ini file to configure the device to support several ISDNOutCallsBehavior features, add the individual feature values. For example, to support both [2] and [16] features, set ISDNOutCallsBehavior = 18 (i.e., 2 +...
Page 393 SIP User's Manual 6. Configuration Parameters Reference Parameter Description received all the ISDN signaling messages containing parts of the called number, and only then it sends a SIP INVITE with the entire called number in the Request-URI.  [2] Through SIP = Interworking of ISDN Overlap Dialing to SIP, based on RFC 3578.
Page 394 Mediant 600 & Mediant 1000 Parameter Description [ConnectedNumberType] Defines the Numbering Type of the ISDN Q.931 Connected Number IE that the device sends in the Connect message to the ISDN (for Tel-to-IP calls). This is interworked from the P- Asserted-Identity header in SIP 200 OK.
Page 395 SIP User's Manual 6. Configuration Parameters Reference Parameter Description terminating devices.  To enable this function, set the parameter ISDNDuplicateQ931BuffMode to 128 (i.e., duplicate all messages). Web: Enable Hold to ISDN Enables interworking of the Hold/Retrieve supplementary EMS: Enable Hold 2 ISDN service from SIP to PRI.
Page 396 Mediant 600 & Mediant 1000 Parameter Description [IgnoreISDNSubaddress] Determines whether the device ignores the Subaddress from the incoming ISDN Called and Calling numbers when sending to IP.  [0] = If an incoming ISDN Q.931 Setup message contains a Called/Calling Number Subaddress, the Subaddress is interworked to the SIP 'isub' parameter according to RFC (default).
Page 397 SIP User's Manual 6. Configuration Parameters Reference Parameter Description by itself. Note: Receipt of a 183 response does not cause the device with ISDN protocol type to play an RBT; the device issues a Progress message (unless SIP183Behaviour is set to 1). If the parameter SIP183Behaviour is set to 1, the 183 response is handled the same way as a 180 Ringing response.
Page 398 Mediant 600 & Mediant 1000 Parameter Description only).  Alarm = Sends or clears PSTN AIS Alarm (ISDN and CAS).  [4] Block = Blocks trunk (CAS only). Notes:  This parameter is applicable only if the parameter EnableBusyOut is set to 1.
Page 399 SIP User's Manual 6. Configuration Parameters Reference Parameter Description 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 400 Mediant 600 & Mediant 1000 Parameter Description [UserToUserHeaderFormat] Determines the format of the User-to-User SIP header in the INVITE message for interworking the ISDN User to User (UU) IE data to SIP.  = Format: X-UserToUser (default).  = Format: User-to-User with Protocol Discriminator (pd) attribute.
Page 401 SIP User's Manual 6. Configuration Parameters Reference Parameter Description sends a 183 response, enabling the PSTN to play a voice announcement to the IP side. If there isn't a PI in the Disconnect message, the call is released (default).  No PI = Doesn't send a 183 response to IP.
Page 402 Mediant 600 & Mediant 1000 Parameter Description Notes:  This parameter is applicable to ISDN protocols.  The option ‘Any’ is only applicable if TerminationSide is set to 0 (i.e., User side).  The ID in the ini file parameter name represents the trunk number, where 0 is the first trunk.
Page 403 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Notes:  This feature is supported only for SIP-to-ISDN calls.  The parameter AddCicAsPrefix can be used to add the CIC as a prefix to the destination phone number for routing IP-to-Tel calls. EMS: Enable AOC Determines whether ISDN Advice of Charge (AOC) [EnableAOC]...
Page 404 Mediant 600 & Mediant 1000 Parameter Description  When IP Profiles are used for configuring different IE data for Trunk Groups, this parameter is ignored. Web: Enable User-to-User IE for Tel Enables ISDN PRI-to-SIP interworking. to IP  Disable = Disabled (default).
Page 405 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Trunk Transfer Mode Determines the trunk transfer method (for all trunks) when a [TrunkTransferMode] SIP REFER message is received. The transfer method depends on the Trunk's PSTN protocol (configured by the parameter ProtocolType) and is applicable only when one of these protocols are used: PSTN Protocol...
Page 406 Mediant 600 & Mediant 1000 Parameter Description device performs a transfer by sending a Facility message to the PBX, initiating Single Step transfer. Once a success return result is received, the transfer is completed. Tel-to-IP: When a Facility message initiating Single Step transfer is received from the PBX, a SIP REFER message is sent to the IP side.
Page 407 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: ISDN Transfer Capabilities Defines the IP-to-ISDN Transfer Capability of the Bearer Capability IE in ISDN Setup messages. The in the ini file EMS: Transfer Capability To ISDN [ISDNTransferCapability_ID] parameter depicts the trunk number, where 0 is the first trunk. ...
Page 408 Mediant 600 & Mediant 1000 Parameter Description Web: Enable QSIG Transfer Update Determines whether the device interworks QSIG Facility [EnableQSIGTransferUpdate] messages with callTranferComplete invoke application protocol data unit (APDU) to SIP UPDATE messages with P- Asserted-Identity and optional Privacy headers. This feature is supported for IP-to-Tel and Tel-to-IP calls.
Page 409: Answer And Disconnect Supervision Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.13 Answer and Disconnect Supervision Parameters The answer and disconnect supervision parameters are described in the table below. Table 6-54: Answer and Disconnect Parameters Parameter Description Web: Answer Supervision Enables the sending of SIP 200 OK upon detection of EMS: Enable Voice Detection speech, fax, or modem.
Page 410 Mediant 600 & Mediant 1000 Parameter Description Web: Disconnect on Broken Connection Determines whether the device releases the call if RTP EMS: Disconnect Calls on Broken packets are not received within a user-defined timeout. Connection  [DisconnectOnBrokenConnection]  [1] Yes (default) Notes: ...
Page 411 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Silence Detection Method Silence detection method. [FarEndDisconnectSilenceMethod]  None = Silence detection option is disabled.  Packets Count = According to packet count.  [2] Voice/Energy Detectors = N/A  [3] All = N/A.
Page 412 Mediant 600 & Mediant 1000 Parameter Description Web: Disconnect Call on Busy Tone Determines whether a call is disconnected upon detection Detection (CAS) of a busy tone (for CAS) EMS: Disconnect On Detection End  Disable = Do not disconnect call on detection of Tones busy tone.
Page 413 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: Polarity Reversal Type Defines the voltage change slope during polarity reversal [PolarityReversalType] or wink.  [0] = Soft reverse polarity (default).  [1] = Hard reverse polarity. Notes:  This parameter is applicable only to FXS interfaces. ...
Page 414: Tone Parameters
Mediant 600 & Mediant 1000 6.14 Tone Parameters This subsection describes the device's tone parameters. 6.14.1 Telephony Tone Parameters The telephony tone parameters are described in the table below. Table 6-55: Tone Parameters Parameter Description Tone Index Table [ToneIndex] This ini file table parameter configures the Tone Index table, which...
Page 415 SIP User's Manual 6. Configuration Parameters Reference Parameter Description 4" for ports 1 through 4).  You can configure multiple entries with different source prefixes and tones for the same FXS port. Web/EMS: Dial Tone Duration Duration (in seconds) that the dial tone is played (for digital [sec] interfaces, to an ISDN terminal).
Page 416 Mediant 600 & Mediant 1000 Parameter Description Web: Time Before Reorder The delay interval (in seconds) from when the device receives a Tone [sec] SIP BYE message (i.e., remote party terminates call) until the EMS: Time For Reorder Tone device starts playing a reorder tone to the FXS phone.
Page 417 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Note: For ISDN trunks, this option is applicable only if the parameter LocalISDNRBSource is set to 1. Web: Play Ringback Tone to IP Determines whether or not the device plays a ringback tone (RBT) EMS: Play Ring Back Tone To to the IP side for IP-to-Tel calls.
Page 418: Tone Detection Parameters
Mediant 600 & Mediant 1000 6.14.2 Tone Detection Parameters The signal tone detection parameters are described in the table below. Table 6-56: Tone Detection Parameters Parameter Description EMS: DTMF Enable Enables or disables the detection of DTMF signaling. [DTMFDetectorEnable] ...
Page 419: Metering Tone Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description  The IP-to-ISDN call is disconnected on detection of a SIT tone only in call alert state. If the call is in connected state, the SIT does not disconnect the call. Detection of Busy or Reorder tones disconnect these calls also in call connected state.
Page 420 Mediant 600 & Mediant 1000 Parameter Description Web: Analog TTX Voltage Determines the metering signal/pulse voltage level (TTX). Level  = 0 Vrms sinusoidal bursts EMS: TTX Voltage Level  (default) = 0.5 Vrms sinusoidal bursts [AnalogTTXVoltageLevel ]  [2] = 1 Vrms sinusoidal bursts Notes: ...
Page 421: Telephone Keypad Sequence Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.15 Telephone Keypad Sequence Parameters The telephony keypad sequence parameters are described in the table below. Table 6-58: Keypad Sequence Parameters Parameter Description Prefix for External Line [Prefix2ExtLine] Defines a string prefix (e.g., '9' dialed for an external line) that when dialed, the device plays a secondary dial tone (i.e., stutter tone) to the FXS line and then starts collecting the subsequently dialed digits from the FXS line.
Page 422 Mediant 600 & Mediant 1000 Parameter Description Three-Way Conference feature is enabled, i.e., the parameter Enable3WayConference is set to 1 and the parameter 3WayConferenceMode is set to 2). Web: Flash Keys Sequence Flash keys sequence timeout - the time (in msec) that the...
Page 423 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Deactivate Keypad sequence that deactivates the delayed hotline option. EMS: Hot Line Deactivation After the sequence is pressed, a confirmation tone is heard. [KeyHotLineDeact] Keypad Feature - Transfer Parameters Web: Blind Keypad sequence that activates blind transfer for Tel-to-IP EMS: Blind Transfer calls.
Page 424 Mediant 600 & Mediant 1000 Parameter Description [RejectAnonymousCallPerPort] This file table parameter determines whether the device rejects incoming anonymous calls on FXS interfaces. The format of this parameter is as follows: [RejectAnonymousCallPerPort] FORMAT RejectAnonymousCallPerPort_Index = RejectAnonymousCallPerPort_Enable, RejectAnonymousCallPerPort_Port, RejectAnonymousCallPerPort_Module; [\RejectAnonymousCallPerPort] Where, ...
Page 425: General Fxo Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.16 General FXO Parameters The general FXO parameters are described in the table below. Table 6-59: General FXO Parameters Parameter Description Web: FXO Coefficient Type Determines the FXO line characteristics (AC and DC) according to EMS: Country Coefficients USA or TBR21 standard.
Page 426 Mediant 600 & Mediant 1000 Parameter Description  The correct dial tone parameters must be configured in the CPT file.  The device may take 1 to 3 seconds to detect a dial tone (according to the dial tone configuration in the CPT file). If the dial tone is not detected within 6 seconds, the device releases the call and sends a SIP 500 "Server Internal Error”...
Page 427: Fxs Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Guard Time Defines the time interval (in seconds) after a call has ended and a new Between Calls call can be accepted for IP-to-Tel (FXO) calls. [GuardTimeBetweenCalls] The valid range is 0 to 10. The default value is 1. Notes: ...
Page 428: Trunk Groups, Number Manipulation And Routing Parameters
Mediant 600 & Mediant 1000 6.18 Trunk Groups, Number Manipulation and Routing Parameters This subsection describes the device's number manipulation and routing parameters. 6.18.1 Trunk Groups and Routing Parameters The routing parameters are described in the table below. Table 6-61: Routing Parameters...
Page 429 SIP User's Manual 6. Configuration Parameters Reference Parameter Description TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode, TrunkGroupSettings_GatewayName,TrunkGroupSettings_Cont actUser, TrunkGroupSettings_ServingIPGroup, TrunkGroupSettings_MWIInterrogationType; [\TrunkGroupSettings] Where,  MWIInterrogationType = defines QSIG MWI to IP interworking for interrogating MWI supplementary services:  [255] Not Configured  [0] None = disables the feature. ...
Page 430 Mediant 600 & Mediant 1000 Parameter Description channel number in the Trunk Group. When the device reaches the lowest channel number in the Trunk Group, it selects the highest channel number in the Trunk Group and then starts descending again.
Page 431 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Source IP Address Input Determines the IP address that the device uses to determine [SourceIPAddressInput] the source of incoming INVITE messages for IP-to-Tel routing.  [-1] = Auto Decision - if the IP-to-IP feature is enabled, this parameter is automatically set to Layer 3 Source IP.
Page 432 Mediant 600 & Mediant 1000 Parameter Description Notes:  This parameter appears only if the 'Use Default Proxy' parameter is enabled.  The domain name is used instead of a Proxy name or IP address in the INVITE SIP URI.
Page 433 SIP User's Manual 6. Configuration Parameters Reference Parameter Description  For a description of the table's parameters, refer to the corresponding Web parameters in ''Configuring the Inbound IP Routing Table'' on page  To support the In-Call Alternative Routing feature, you can use two entries that support the same call but assigned with a different Trunk Group.
Page 434 Mediant 600 & Mediant 1000 Parameter Description  This feature is supported only for numerical IP addresses in the 'Outbound IP Routing Table'. Web/EMS: Filter Calls to IP Enables filtering of Tel-to-IP calls when a Proxy is used (i.e., [FilterCalls2IP]...
Page 435: Alternative Routing Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description many different networks. The cic parameter is carried in the SIP INVITE and maps to the ISDN Transit Network Selection Information Element (TNS IE) in the outgoing ISDN Setup message (if the EnableCIC parameter is set to 1). The TNS IE identifies the requested transportation networks and allows different providers equal access support, based on customer choice.
Page 436 Mediant 600 & Mediant 1000 Parameter Description Web: Alt Routing Tel to IP Mode Determines the event(s) reason for triggering Alternative EMS: Alternative Routing Mode Routing. [AltRoutingTel2IPMode]  None = Alternative routing is not used.  Connectivity = Alternative routing is performed if a ping to the initial destination fails.
Page 437 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Alternative Routing Determines the duration (in milliseconds) for which the device Tone Duration [ms] plays a tone to the endpoint on each Alternative Routing [AltRoutingToneDuration] attempt. When the device finishes playing the tone, a new SIP INVITE message is sent to the new destination.
Page 438 Mediant 600 & Mediant 1000 Parameter Description  For an explanation on using ini file table parameters, refer to ''Configuring ini File Table Parameters'' on page Web: Reasons for Alternative IP-to-Tel Routing Table EMS: Alt Route Cause IP to Tel...
Page 439 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Forward On Busy Trunk Destination [ForwardOnBusyTrunkDest] This ini file table parameter configures the Forward On Busy Trunk Destination table. This table allows you to define an alternative IP destination (IP address) per Trunk Group for IP- to Tel calls.
Page 440: Number Manipulation Parameters
Mediant 600 & Mediant 1000 6.18.3 Number Manipulation Parameters The number manipulation parameters are described in the table below. Table 6-63: Number Manipulation Parameters Parameter Description Web: Set Redirect number Screening Defines the value of the Redirect Number screening Indicator to TEL indicator in ISDN Setup messages.
Page 441 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Redirect Number IP -> Tel EMS: Redirect Number Map IP to Tel [RedirectNumberMapIp2Tel] This ini file table parameter manipulates the redirect number for IP-to-Tel calls. This manipulates the value of the SIP Diversion, History-Info, or Resource-Priority headers (including the reason the call was redirected).
Page 442 Mediant 600 & Mediant 1000 Parameter Description RedirectNumberMapTel2Ip_RemoveFromLeft, RedirectNumberMapTel2Ip_RemoveFromRight, RedirectNumberMapTel2Ip_LeaveFromRight, RedirectNumberMapTel2Ip_Prefix2Add, RedirectNumberMapTel2Ip_Suffix2Add, RedirectNumberMapTel2Ip_IsPresentationRestricted, RedirectNumberMapTel2Ip_SrcTrunkGroupID, RedirectNumberMapTel2Ip_SrcIPGroupID; [\RedirectNumberMapTel2Ip] For example: RedirectNumberMapTel2Ip 1 = *, 4, 255, 255, 0, 0, 255, , 972, 255, 1, 2; Notes:  This parameter table can include up to 20 indices (1-20).
Page 443 SIP User's Manual 6. Configuration Parameters Reference Parameter Description 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 444 Mediant 600 & Mediant 1000 Parameter Description Web: Replace Empty Destination with Determines whether the internal channel number is used B-channel Phone Number as the destination number if the called number is missing. EMS: Replace Empty Dst With Port ...
Page 445 SIP User's Manual 6. Configuration Parameters Reference 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 [SwapRedirectNumber] number (sometimes referred to as 'original called number') uses the redirect number instead of the called number.
Page 446 Mediant 600 & Mediant 1000 Parameter Description NumberMapTel2Ip_IsPresentationRestricted, NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_ SrcIPGroupID; [\NumberMapTel2Ip] For example: NumberMapTel2Ip 0 = 01,$$,*,0,0,2,$$,$$,971,$$,$$,$$,$$; NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; Notes:  This table parameter can include up to 120 indices (0- 119).  The parameters SourceAddress and IsPresentationRestricted are not applicable.
Page 447 SIP User's Manual 6. Configuration Parameters Reference Parameter Description For example: [NumberMapIp2Tel] NumberMapIp2Tel 0 = 01,034,10.13.77.8,$$,0,$$,2,$$,667,$$; NumberMapIp2Tel 1 = 10,10,1.1.1.1,255,255,3,0,5,100,$$,255; [\NumberMapIp2Tel] Notes:  This table parameter can include up to 100 indices.  The parameter IsPresentationRestricted is not applicable.  RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, LeaveFromRight, NumberType, and NumberPlan...
Page 448 Mediant 600 & Mediant 1000 Parameter Description NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_SrcIPGroupID; [\SourceNumberMapTel2Ip] For example: [SourceNumberMapTel2Ip] SourceNumberMapTel2Ip 0 = 22,03,$$,0,0,$$,2,$$,667,$$,0,$$,$$; SourceNumberMapTel2Ip 0 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; [\SourceNumberMapTel2Ip] Notes:  This table parameter can include up to indices.  RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, LeaveFromRight, NumberType, NumberPlan, and IsPresentationRestricted are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions.
Page 449 SIP User's Manual 6. Configuration Parameters Reference Parameter Description For example: [SourceNumberMapIp2Tel] SourceNumberMapIp2Tel 0 = 22,03,$$,$$,$$,$$,2,667,$$,$$; SourceNumberMapIp2Tel 1 = 034,01,1.1.1.1,$$,0,2,$$,$$,972,$$,10; [\SourceNumberMapIp2Tel] Notes:  RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, LeaveFromRight , NumberType, and NumberPlan are applied if the called and calling numbers match the DestinationPrefix, SourcePrefix, and SourceAddress conditions.
Page 450: Ldap Parameters
Mediant 600 & Mediant 1000 Parameter Description  1/2 - National number in ISDN/Telephony numbering plan  1/4 - Subscriber (local) number in ISDN/Telephony numbering plan  9/4 - Subscriber (local) number in Private numbering plan 6.18.4 LDAP Parameters The Lightweight Directory Access Protocol (LDAP) parameters are described in the table below.
Page 451: Channel Parameters
SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: MS LDAP OCS Number The name of the attribute that represents the user OCS attribute name number in the Microsoft AD database. [MSLDAPOCSNumAttributeName] The valid value is a string of up to 49 characters. The default is "msRTCSIP-PrimaryUserAddress".
Page 452 For this parameter to take effect, a device reset is required.  Using 128 msec reduces the channel capacity to 200 channels.  When housed with an analog/BRI module, the device (Mediant 1000) can use a max. echo canceller length of 64 msec. SIP User's Manual Document #: LTRT-83306...
Page 453 Parameter Description  When housed with PRI TRUNKS module, the device (Mediant 1000) can use a max. echo canceller length of 128 msec.  When set to 128 msec, the number of available channels is reduced by a factor of 5/6 (for Mediant 1000).
Page 454: Coder Parameters
Mediant 600 & Mediant 1000 6.19.2 Coder Parameters The coder parameters are described in the table below. Table 6-66: Coder Parameters Parameter Description Web: Enable RFC 4117 Enables transcoding of calls according to RFC 4117. Transcoding  [0] Disable (default) [EnableRFC4117Transcoding] ...
Page 455: Fax And Modem Parameters
SIP User's Manual 6. Configuration Parameters Reference 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 456 Mediant 600 & Mediant 1000 Parameter Description Web: Fax Relay Redundancy Depth Number of times that each fax relay payload is EMS: Relay Redundancy Depth retransmitted to the network. [FaxRelayRedundancyDepth]  = No redundancy (default).  = One packet redundancy.
Page 457 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: CNG Detector Mode Determines whether the device detects the fax [CNGDetectorMode] Calling tone (CNG).  Disable = The originating device doesn’t detect CNG; the CNG signal passes transparently to the remote side (default). ...
Page 458 Mediant 600 & Mediant 1000 Parameter Description EMS: Relay Volume (dBm) Determines the fax gain control. [FaxModemRelayVolume] The range is -18 to -3, corresponding to -18 dBm to -3 dBm in 1-dB steps. The default is -6 dBm fax gain control.
Page 459 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: NSE Mode Cisco compatible fax and modem bypass mode. [NSEMode]  = NSE disabled (default)  [1] = NSE enabled Notes:  This feature can be used only if VxxModemTransportType = 2 (Bypass). ...
Page 460 Mediant 600 & Mediant 1000 Parameter Description  Enable Relay = N/A  Enable Bypass = (default)  Events Only = Transparent with Events Web: V.32 Modem Transport Type V.32 Modem Transport Type used by the device. EMS: V32 Transport ...
Page 461: Dtmf Parameters
SIP User's Manual 6. Configuration Parameters Reference 6.19.4 DTMF Parameters The dual-tone multi-frequency (DTMF) parameters are described in the table below. Table 6-68: DTMF Parameters Parameter Description Web/EMS: DTMF Transport Determines the DTMF transport type. Type  DTMF Mute = Erases digits from voice stream and doesn't [DTMFTransportType] relay to remote.
Page 462: Rtp, Rtcp And T.38 Parameters
Mediant 600 & Mediant 1000 Parameter Description Web: Enable Special Digits Determines whether the asterisk (*) and pound (#) digits can be EMS: Use '#' For Dial used in DTMF. Termination  [0] Disable = Use '*' or '#' to terminate number collection (refer [IsSpecialDigits] to the parameter UseDigitForSpecialDTMF).
Page 463 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Enable RTP Redundancy Determines whether the device includes the RTP Negotiation redundancy dynamic payload type in the SDP, according to [EnableRTPRedundancyNegotiation] RFC 2198.  Disable (default)  Enable When enabled, the device includes in the SDP message the RTP payload type "RED"...
Page 464 Mediant 600 & Mediant 1000 Parameter Description  = The device uses the received GARP packets to change the MAC address of the transmitted RTP packets (default).  [3] = Options 1 and 2 are used. Note: For this parameter to take effect, a device reset is required.
Page 465 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: RTP Multiplexing Local UDP Port Determines the local UDP port used for outgoing [L1L1ComplexTxUDPPort] multiplexed RTP packets (applies to RTP multiplexing). The valid range is the range of possible UDP ports: 6,000 to 64,000.
Page 466 Mediant 600 & Mediant 1000 Parameter Description Web: Minimum Gap Size Voice quality monitoring - minimum gap size (number of EMS: GMin frames). The default is 16. [VQMonGMin] Web/EMS: Burst Threshold Voice quality monitoring - excessive burst alert threshold. if [VQMonBurstHR] set to -1 (default), no alerts are issued.
Page 467: Auxiliary And Configuration Files Parameters
Notes:  For this parameter to take effect, a device reset is required.  This parameter is applicable only to Mediant 1000. Web/EMS: Prerecorded Tones The name (and path) of the file containing the Prerecorded Tones. File Note: For this parameter to take effect, a device reset is required.
Page 468: Automatic Update Parameters
Mediant 600 & Mediant 1000 Parameter Description Web: CAS File CAS file name (e.g., 'E_M_WinkTable.dat') that defines the CAS EMS: Trunk Cas Table Index protocol (where x denotes the CAS file ID 0 to 7). It is possible to [CASFileName_x] define up to eight different CAS files by repeating this parameter.
Page 469 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [ResetNow] Invokes an immediate device reset. This option can be used to activate offline (i.e., not on-the-fly) parameters that are loaded using the parameter IniFileUrl.  = The immediate restart mechanism is disabled (default). ...
Page 470  The maximum length of the URL address is 99 characters.  This parameter is applicable only to Mediant 1000. [CasFileURL] Specifies the name of the CAS file and the path to the server (IP address or FQDN) on which it is located.
Page 471: Restoring Factory Default Settings
SIP User's Manual 7. Restoring Factory Default Settings Restoring Factory Default Settings The device provides you with the following methods for restoring the device's configuration to factory default settings:  Using the CLI (refer to ''Restoring Defaults using CLI'' on page ...
Page 472 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83306...
Page 473: Auxiliary Configuration Files
SIP User's Manual 8. Auxiliary Configuration Files Auxiliary Configuration Files This section describes the auxiliary files that can be loaded (in addition to the ini file) to the device:  Call Progress Tones (refer to '' Call Progress Tones File'' on page ...
Page 474 Mediant 600 & Mediant 1000 The format attribute can be one of the following:  Continuous: A steady non-interrupted sound (e.g., a dial tone). Only the 'First Signal On time' should be specified. All other on and off periods must be set to zero. In this case, the parameter specifies the detection period.
Page 475 SIP User's Manual 8. Auxiliary Configuration Files • High Freq [Hz: Frequency (in Hz) of the higher tone component in case of dual frequency tone, or zero (0) in case of single tone (not relevant to AM tones). • Low Freq Level [-dBm]: Generation level 0 dBm to -31 dBm in dBm (not relevant to AM tones).
Page 476: Distinctive Ringing
Mediant 600 & Mediant 1000 For example, to configure the dial tone to 440 Hz only, enter the following text: [NUMBER OF CALL PROGRESS TONES] Number of Call Progress Tones=1 #Dial Tone [CALL PROGRESS TONE #0] Tone Type=1 Tone Form =1 (continuous)
Page 477 SIP User's Manual 8. Auxiliary Configuration Files • First (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units) for the first cadence on-off cycle. • Second (Burst) Ring On Time [10 msec]: 'Ring On' period (in 10 msec units) for the second cadence on-off cycle.
Page 478: Fxs Distinctive Ringing And Call Waiting Tones Per Source Number
Mediant 600 & Mediant 1000 [Ringing Pattern #1] Ring Type=1 Freq [Hz]=20 First Ring On Time [10msec]=200 First Ring Off Time [10msec]=400 #GR-506-CORE Ringing Pattern 2 [Ringing Pattern #2] Ring Type=2 Freq [Hz]=20 First Ring On Time [10msec]=80 First Ring Off Time [10msec]=40...
Page 479: Prerecorded Tones File
SIP User's Manual 8. Auxiliary Configuration Files Prerecorded Tones File The CPT file mechanism has several limitations such as a limited number of predefined tones and a limited number of frequency integrations in one tone. To overcome these limitations and provide tone generation capability that is more flexible, the Prerecorded Tones (PRT) file can be used.
Page 480: Cas Files
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 481 SIP User's Manual 8. Auxiliary Configuration Files Note: To use this Dial Plan, you must also use a special CAS *.dat file that supports this feature (contact your AudioCodes sales representative).  Prefix tags (for IP-to-Tel routing): Provides enhanced routing rules based on Dial Plan prefix tags.
Page 482: User Information File
Mediant 600 & Mediant 1000 012,7-14 014,7-14 ; Defines emergency number 911. ; No additional digits are expected. 911,0 [ PLAN2 ] ; Defines area codes 02, 03, 04. ; In these area codes, phone numbers have 7 digits. 0[2-4],7 ;...
Page 483: Figure 8-1: Example Of A User Information File
SIP User's Manual 8. Auxiliary Configuration Files An example of a User Information file is shown in the figure below: Figure 8-1: 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 484 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83306...
Page 485: Ip Telephony Capabilities
SIP User's Manual 9. IP Telephony Capabilities IP Telephony Capabilities This section describes the device's main IP telephony capabilities. Dialing Plan Features This section discusses various dialing plan features supported by the device:  Dialing plan notations (refer to ''Dialing Plan Notation for Routing and Manipulation'' page ...
Page 486: Figure 9-1: Prefix To Add Field With Notation
Mediant 600 & Mediant 1000 Notation Description Example Represents any single digit. Pound sign (#) Represents the end of 54324xx#: represents a 7-digit number that starts with at the end of a a number. 54324. number A single *: represents any number (i.e., all numbers).
Page 487: Digit Mapping
SIP User's Manual 9. IP Telephony Capabilities In this configuration, the following manipulation process occurs: 1) the prefix is calculated, 020215 in the example; 2) the first seven digits from the left are removed from the original number, in the example, the number is changed to 8888888; 3) the prefix that was previously calculated is then added.
Page 488: External Dial Plan File
Mediant 600 & Mediant 1000 Below is an example of a digit map pattern containing eight rules: DigitMapping = 11xS|00[1- 7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x|x.T In the example above, 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 489: Modifying Isdn-To-Ip Calling Party Number
SIP User's Manual 9. IP Telephony Capabilities An example of a Dial Plan file with indices (in ini- file format before conversion to binary *.dat) is shown below: [ PLAN1 ] ; Area codes 02, 03, - phone numbers include 7 digits. 02,7 03,7 ;...
Page 490: Dial Plan Prefix Tags For Ip-To-Tel Routing
Mediant 600 & Mediant 1000 The device adds the newly manipulated calling number to the URI user part in the From header, and to the Contact header of the SIP INVITE sent to the IP side. For example, a received Calling Number Party of 0567811181 that is changed to 04343434181 (see Dial Plan file example above) is sent to the IP with a SIP INVITE as follows: Via: SIP/2.0/UDP 211.192.160.214:5060;branch=z9hG4bK3157667347...
Page 491: Figure 9-2: Configuring Dial Plan File Label For Ip-To-Tel Routing
SIP User's Manual 9. IP Telephony Capabilities IP2TelTaggingDestDialPlanIndex = 1 Define the external Dial Plan file with two routing tags (as shown below): • "LOCL" - for local calls • "LONG" - for long distance calls [ PLAN1 ] 42520[3-5],0,LOCL 425207,0,LOCL 42529,0,LOCL 425200,0,LONG...
Page 492: Routing Applications
Mediant 600 & Mediant 1000 Routing Applications 9.2.1 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 493: Theory Of Operation
SIP User's Manual 9. IP Telephony Capabilities 9.2.1.1 Theory of Operation The device's IP-to-IP SIP session is performed by implementing Back-to-Back User Agent (B2BUA). The device acts as a user agent for both ends (legs ) of the SIP call (from call establishment to termination).
Page 494 Mediant 600 & Mediant 1000 9.2.1.1.1 Proxy Sets A Proxy Set is a group containing up to five Proxy servers (for Proxy load balancing and redundancy), defined by IP address or fully qualified domain name (FQDN). The Proxy Set is assigned to IP Groups (of type SERVER only), representing the address of the IP Group to where the device sends the INVITE message (destination of the call).
Page 495: Figure 9-5: Ip-To-Ip Routing/Registration/Authentication Of Remote Ip-Pbx Users (Example)
SIP User's Manual 9. IP Telephony Capabilities For registrations of USER IP Groups, the device updates its internal database with the AOR and Contacts of the users (refer to the figure below) Digest authentication using SIP 401/407 responses (if needed) is performed by the Serving IP Group (e.g., IP-PBX). The device forwards these responses directly to the remote SIP users.
Page 496: Figure 9-6: Ip-To-Ip Routing For Ip-Pbx Remote Users In Survivability Mode (Example)
Mediant 600 & Mediant 1000 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...
Page 497: Configuring Ip-To-Ip Routing
SIP User's Manual 9. IP Telephony Capabilities 9.2.1.1.4 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 498 Mediant 600 & Mediant 1000 • The enterprise also includes remote IP-PBX users that communicate with the IP- PBX via the device. All dialed calls from the IP-PBX consisting of four digits starting with digit "4" are routed to the remote IP-PBX users.
Page 499: Figure 9-8: Sip Trunking Setup Scenario Example
SIP User's Manual 9. IP Telephony Capabilities The figure below provides an illustration of this example scenario: Figure 9-8: SIP Trunking Setup Scenario Example The steps for configuring the device according to the scenario above can be summarized as follows: ...
Page 500: Figure 9-9: Defining Required Media Channels
Mediant 600 & Mediant 1000 page 508).  Configure IP Profiles (refer to ''Step 7: Configure IP Profiles for Voice Coders'' on page  Configure inbound IP routing rules (refer to ''Step 8: Configure Inbound IP Routing'' page  Configure outbound IP routing rules (refer to ''Step 9: Configure Outbound IP Routing'' on page ...
Page 501 SIP User's Manual 9. IP Telephony Capabilities 9.2.1.2.3 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. This Trunk Group is also used for alternative routing to the legacy PSTN network in case of a loss of connection with the ITSP's.
Page 502: Figure 9-10: Proxy Set Id #1 For Itsp-A
Mediant 600 & Mediant 1000 9.2.1.2.4 Step 4: Configure the Proxy Sets This step describes how to configure the following Proxy Sets:  Proxy Set ID #1 defined with two FQDN's for ITSP-A.  Proxy Set ID #2 defined with two IP addresses for ITSP-B.
Page 503: Figure 9-11: Proxy Set Id #2 For Itsp-B
SIP User's Manual 9. IP Telephony Capabilities Configure Proxy Set ID #2 for ITSP-B: From the 'Proxy Set ID' drop-down list, select "2". In the 'Proxy Address' column, enter the IP addresses of the ITSP-B SIP trunk (e.g., 216.182.224.202 and 216.182.225.202). From the 'Transport Type' drop-down list corresponding to the IP address entered above, select "UDP".
Page 504: Figure 9-12: Proxy Set Id #3 For The Ip-Pbx
Mediant 600 & Mediant 1000 In the 'Enable Proxy Keep Alive' drop-down list, select "Using Options" – this is used in Survivability mode for remote IP-PBX users. Figure 9-12: Proxy Set ID #3 for the IP-PBX 9.2.1.2.5 Step 5: Configure the IP Groups This step describes how to create the IP Groups for the following entities in the network: ...
Page 505: Figure 9-13: Defining Ip Group 1
SIP User's Manual 9. IP Telephony Capabilities Contact User = name that is sent in the SIP Request's Contact header for this IP Group (e.g., ITSP-A). Figure 9-13: Defining IP Group 1 Define IP Group #2 for ITSP-B: From the 'Type' drop-down list, select 'SERVER'. In the 'Description' field, type an arbitrary name for the IP Group (e.g., ITSP B).
Page 506: Figure 9-14: Defining Ip Group 2
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 9-14: Defining IP Group 2 Define IP Group #3 for the IP-PBX: From the 'Type' drop-down list, select 'SERVER'.
Page 507: Figure 9-15: Defining Ip Group 3
SIP User's Manual 9. IP Telephony Capabilities Contact User = name that is sent in the SIP Request Contact header for this IP Group (e.g., PBXUSER). Figure 9-15: Defining IP Group 3 Define IP Group #4 for the remote IP-PBX users: From the 'Type' drop-down list, select 'USER'.
Page 508: Figure 9-16: Defining Ip Group 4
Mediant 600 & Mediant 1000 From the 'Serving IP Group ID' drop-down list, select "3" (i.e. the IP Group for the IP-PBX). Figure 9-16: Defining IP Group 4 Note: No Serving IP Groups are defined for ITSP-A and ITSP-B. Instead, the...
Page 509: Figure 9-17: Defining Accounts For Registration
SIP User's Manual 9. IP Telephony Capabilities  To configure the Account table: Open the 'Account Table' page (Protocol Configuration menu > Proxies, Registration, IP Groups submenu > Account Table Figure 9-17: Defining Accounts for Registration Configure Account ID #1 for IP-PBX authentication and registration with ITSP-A: •...
Page 510: Figure 9-18: Defining Coder Group Id 1
Mediant 600 & Mediant 1000 9.2.1.2.7 Step 7: Configure IP Profiles for Voice Coders Since different voice coders are used by the IP-PBX (G.711) and the ITSP's (G.723), you need to define two IP Profiles:  Profile ID #1 - configured with G.711 for the IP-PBX ...
Page 511: Figure 9-20: Defining Ip Profile Id 1
SIP User's Manual 9. IP Telephony Capabilities Configure Profile ID #1 for the IP-PBX (as shown below): From the 'Profile ID' drop-down list, select '1'. From the 'Coder Group' drop-down list, select 'Coder Group 1'. Click Submit Figure 9-20: Defining IP Profile ID 1 Configure Profile ID #2 for the ITSP's: From the 'Profile ID' drop-down list, select '2'.
Page 512: Figure 9-21: Defining Inbound Ip Routing Rules
Mediant 600 & Mediant 1000 9.2.1.2.8 Step 8: Configure Inbound IP Routing This step defines how to configure the device for routing inbound (i.e., received) IP-to-IP calls. The table in which this is configured uses the IP Groups that you defined in ''Step 5:...
Page 513: Figure 9-22: Defining Outbound Ip Routing Rules
SIP User's Manual 9. IP Telephony Capabilities Index #4: identifies IP calls received from ITSP-B as IP-to-IP calls and assigns them to the IP Group ID configured for ITSP-B: • 'Dest Phone Prefix': ITSP-B assigns the Enterprise a range of numbers that start with 0200.
Page 514 Mediant 600 & Mediant 1000 Index #1: routes IP calls received from ITSP-A to the IP-PBX: • 'Source IP Group ID': select "1" to indicate received (inbound) calls identified as belonging to the IP Group configured for ITSP-A. • 'Dest Phone Prefix' and 'Source Phone Prefix' : enter the asterisk (*) symbol to indicate all destinations and callers respectively.
Page 515: Figure 9-23: Defining Destination Phone Number Manipulation Rules
SIP User's Manual 9. IP Telephony Capabilities Index #6: routes dialed calls (four digits starting with digit 4) from IP-PBX to remote IP- PBX users. The device searches its database for the remote users registered number, and then sends an INVITE to the remote user's IP address (listed in the database): •...
Page 516: Stand-Alone Survivability (Sas) Feature
Mediant 600 & Mediant 1000 9.2.2 Stand-Alone Survivability (SAS) Feature 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. In case of failure of the IP Centrex, IP-PBX servers...
Page 517: Configuring Sas
SIP User's Manual 9. IP Telephony Capabilities The received INVITE message is routed as depicted in the flow chart below: Figure 9-24: SAS Routing in Emergency Mode 9.2.2.1 Configuring SAS For configuring the device to operate with SAS, perform the following configurations: ...
Page 518: Configuring Sas Emergency Calls
Mediant 600 & Mediant 1000  SASDefaultGatewayIP = < SAS gateway IP address>  SASProxySet = 1  IP2IPRouting (SAS call routing rules) 9.2.2.2 Configuring SAS Emergency Calls The device's SAS agent can be configured to detect a user-defined emergency number (e.g.
Page 519: Dsp Channel Resources For Sbc/Ip-To-Ip/Ip Media Functionality
SIP User's Manual 9. IP Telephony Capabilities 9.2.3 DSP Channel Resources for SBC/IP-to-IP/IP Media Functionality The device supports the IP-to-IP call routing application as well as IP media capabilities. The device provides the required DSP resources (channels) for these applications (in addition to the DSP resources needed for the PRI Trunk interfaces).
Page 520 Mediant 600 & Mediant 1000 ♦ With Conferencing: when the MPM modules are housed in chassis slots 4, 5, and 6, up to 100 DSP resources are supported with call conferencing (up to 60 conference participants). These channels are allocated as follows: ...
Page 521 SIP User's Manual 9. IP Telephony Capabilities • IPmediaChannels: defines the number of DSP channels that are “borrowed” (used) from each TRUNKS module for IP-to-IP routing, and/or IP media, as shown in the example below: [IPMediaChannels] FORMAT IPMediaChannels_Index = IPMediaChannels_ModuleID, IPMediaChannels_DSPChannelsReserved;...
Page 522: Multiple Sip Signaling/Media Interfaces Environment
Mediant 600 & Mediant 1000 9.2.4 Multiple SIP Signaling/Media Interfaces Environment The device supports multiple logical SIP signaling interfaces and RTP (media) traffic interfaces. This allows you to separate SIP signaling messages and media traffic between different applications (i.e., SAS, Gateway\IP-to-IP), and/or between different networks (e.g., when working with multiple ITSP's).
Page 523: Sip Interfaces
SIP User's Manual 9. IP Telephony Capabilities 9.2.4.3 SIP Interfaces A SIP Interface represents one SIP signaling entity, which is a combination of UDP, TCP, and TLS ports relating to one specific IP address (network interface, configured in the Multiple Interface table). The SIP Interface is configured with a corresponding SRD. This allows User Agents on the network to communicate with a specific SRD, using the SIP Interface (signaling interface) associated with it.
Page 524: Configuration Example
Mediant 600 & Mediant 1000 9.2.4.4 Configuration Example Below is an example configuration for implementing multiple SIP signaling and RTP interfaces. In this example, the device serves as the communication interface between the enterprise's PBX (connected using an E1/T1 trunk) and two ITSP', as shown in the figure...
Page 525: Figure 9-28: Defining Trunk Group
SIP User's Manual 9. IP Telephony Capabilities  To configure the scenario example: Configure Trunk Group ID #1 in the 'Trunk Group Table' page (Configuration tab > Protocol Configuration menu > Trunk Group submenu > Trunk Group ), as shown in the figure below: Figure 9-28: Defining Trunk Group Configure the Trunk in the 'Trunk Settings' page (PSTN Settings menu >...
Page 526: Figure 9-32: Defining Sip Interfaces
Mediant 600 & Mediant 1000 Configure the SIP Interfaces in the SIP Interface table (Configuration tab > Protocol Configuration menu > Application Network Settings submenu > SIP Interface Table Figure 9-32: Defining SIP Interfaces Configure Proxy Sets in the Proxy Set table (Configuration tab > Protocol Configuration menu >...
Page 527: Figure 9-34: Defining Ip Groups
SIP User's Manual 9. IP Telephony Capabilities Configure IP Groups in the IP Groups table (Configuration tab > Protocol Configuration menu > Proxies, Registration, IP Groups 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 528: Transcoding Using Third-Party Call Control
Mediant 600 & Mediant 1000 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 411C (refer to ''Using RFC 4117'' on page ...
Page 529: Figure 9-37: Direct Connection (Example)
SIP User's Manual 9. IP Telephony Capabilities It is assumed that the device is controlled by a third-party, Application server (or any SIP user agent) that instructs the device to start an IP Transcoding call by sending two SIP INVITE messages with SIP URI that includes the Transcoding Identifier name. For example: Invite sip:trans123@audiocodes.com SIP/2.0 The left part of the SIP URI includes the Transcoding ID (the default string is ‘trans’) and is...
Page 530: Emergency Phone Number Services - E911
Mediant 600 & Mediant 1000 The figure below illustrates an example of implementing an Application server: Emergency Phone Number Services - E911 The device supports various emergency phone number services. The device supports the North American emergency telephone number system known as Enhanced 911 (E911), according to the TR-TSY-000350 and Bellcore's GR-350-Jun2003 standards.
Page 531: Fxs Device Emulating Psap Using Did Loop-Start Lines
SIP User's Manual 9. IP Telephony Capabilities 9.4.1 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 9-38: 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 532: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 When the call is answered by the PSAP operator, the PSAP sends a SIP 200 OK to the FXS device, and the FXS device then generates a polarity reversal signal to the E911 switch. 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.
Page 533: Table 9-3: Dialed Mf Digits Sent To Psap
SIP User's Manual 9. IP Telephony Capabilities Typically, the MF spills are sent from the E911 tandem switch to the PSAP, as shown in the table below: Table 9-3: Dialed MF Digits Sent to PSAP Digits of Calling Number Dialed MF Digits 8 digits "nnnnnnnn"...
Page 534: Fxo Device Interworking Sip E911 Calls From Service Provider's Ip Network To Psap Did Lines
Mediant 600 & Mediant 1000 9.4.2 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 535: Table 9-4: Dialed Number By Device Depending On Calling Number
SIP User's Manual 9. IP Telephony Capabilities Following the "hookflash" Wink signal, the PSAP sends DTMF digits. These digits are detected by the device and forwarded to the IP, using RFC 2833 telephony events (or inband, depending on the device's configuration). Typically, this Wink signal followed by the DTMF digits initiates a call transfer.
Page 536 Mediant 600 & Mediant 1000  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. The gap duration is 60 msec between any two MF digits.
Page 537 SIP User's Manual 9. IP Telephony Capabilities 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 Content-Type: application/broadsoft...
Page 538: Routing Based On Ldap Active Directory Queries
Mediant 600 & Mediant 1000 Routing Based on LDAP Active Directory Queries The device supports Lightweight Directory Access Protocol (LDAP), allowing the device to make call routing decisions based on information stored on a third-party LDAP server (or Microsoft’s Active Directory-based enterprise directory server). This feature enables the usage of one common, popular database to manage and maintain information regarding user’s availability, presence, and location.
Page 539: Ad-Based Tel-To-Ip Routing In Microsoft Ocs 2007 Environment
SIP User's Manual 9. IP Telephony Capabilities 9.5.2 AD-Based Tel-to-IP Routing in Microsoft OCS 2007 Environment Typically, enterprises wishing to deploy Microsoft’s Office Communication Server 2007 (OCS 2007) are faced with a complex, call routing dial plan when migrating users from their existing PBX/IP-PBX to the OCS 2007 platform.
Page 540: Figure 9-40: Active Directory-Based Routing Rules In Outbound Ip Routing Table
Mediant 600 & Mediant 1000 For enabling alternative routing, you need to enable the alternative routing mechanism and configure corresponding SIP reasons for alternative routing. For this feature, alternative routing always starts again from the top of the table (first routing rule entry) and not from the next row.
Page 541: Configuring Dtmf Transport Types
SIP User's Manual 9. IP Telephony Capabilities Configuring DTMF Transport Types You can control the way DTMF digits are transported over the IP network to the remote endpoint, by using one of the following modes:  Using INFO message according to Nortel IETF draft: DTMF digits are carried to the remote side in INFO messages.
Page 542: Fxs And Fxo Capabilities
Mediant 600 & Mediant 1000  Using INFO message according to Korea mode: DTMF digits are carried to the remote side in INFO messages. To enable this mode, define the following: • RxDTMFOption = 0 (i.e., disabled) • TxDTMFOption = 3 Note that in this mode, DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0).
Page 543: Fxo Operating Modes
SIP User's Manual 9. IP Telephony Capabilities  Frequency response in transmit and receive direction  Hook thresholds  Ringing generation and detection parameters 9.7.2 FXO Operating Modes This section provides a description of the device's FXO operating modes:  For IP-to-Tel calls (refer to ''FXO Operations for IP-to-Tel Calls'' on page ...
Page 544: Figure 9-41: Call Flow For One-Stage Dialing
Mediant 600 & Mediant 1000 9.7.2.1.1 One-Stage Dialing One-stage dialing is when the FXO device receives an IP-to-Tel call, off-hooks the PBX line connected to the telephone, and then immediately dials the destination telephone number. In other words, the IP caller doesn't dial the PSTN number upon hearing a dial tone.
Page 545: Figure 9-42: Call Flow For Two-Stage Dialing
SIP User's Manual 9. IP Telephony Capabilities 9.7.2.1.2 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 546: Fxo Operations For Tel-To-Ip Calls
Mediant 600 & Mediant 1000 The "start dial" signal is a wink from the PBX to the FXO device. The FXO then sends the last four to five DTMF digits of the called number. The PBX uses these digits to complete the routing directly to an internal station (telephone or equivalent) ...
Page 547: Figure 9-44: Call Flow For Collecting Digits Mode
SIP User's Manual 9. IP Telephony Capabilities 9.7.2.2.2 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 9-44: Call Flow for Collecting Digits Mode 9.7.2.2.3 FXO Supplementary Services The FXO supplementary services include the following: ...
Page 548: Call Termination On Fxo Devices
Mediant 600 & Mediant 1000 • PBX performs the transfer internally  Hold / Transfer toward the IP side: The FXO device doesn't initiate hold / transfer as a response to input from the Tel side. If the FXO receives a REFER request (with or without replaces), it generates a new INVITE according to the Refer-To header.
Page 549 SIP User's Manual 9. IP Telephony Capabilities  Interruption of RTP stream: Relevant parameters: BrokenConnectionEventTimeout and DisconnectOnBrokenConnection. Note: This method operates correctly only if silence suppression is not used.  Protocol-based termination of the call from the IP side Note: The implemented disconnect method must be supported by the CO or PBX.
Page 550: Remote Pbx Extension Between Fxo And Fxs Devices
Mediant 600 & Mediant 1000 9.7.3 Remote PBX Extension Between FXO and FXS Devices Remote PBX extension offers a company the capability of extending the "power" of its local PBX by allowing remote phones (remote offices) to connect to the company's PBX over the IP network (instead of via PSTN).
Page 551: Dialing From Remote Extension (Phone At Fxs)
SIP User's Manual 9. IP Telephony Capabilities 9.7.3.1 Dialing from Remote Extension (Phone at FXS) The procedure below describes how to dial from the 'remote PBX extension' (i.e., phone connected to the FXS interface).  To make a call from the FXS interface: Off-hook the phone and wait for the dial tone from the PBX.
Page 552: Call Waiting For Remote Extensions
Mediant 600 & Mediant 1000 Figure 9-46: MWI for Remote Extensions 9.7.3.4 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 553: Fxo Gateway Configuration
SIP User's Manual 9. IP Telephony Capabilities In the ‘Automatic Dialing’ page (refer to ''Automatic Dialing'' on page ), enter the phone numbers of the FXO device in the ‘Destination Phone Number’ fields. When a phone connected to Port #1 off-hooks, the FXS device automatically dials the number ‘200’.
Page 554: Configuring Alternative Routing (Based On Connectivity And Qos)
Mediant 600 & Mediant 1000 In the ‘Automatic Dialing’ page, enter the phone numbers of the FXS device in the ‘Destination Phone Number’ fields. When a ringing signal is detected at Port #1, the FXO device automatically dials the number ‘100’.
Page 555: Alternative Routing Mechanism
SIP User's Manual 9. IP Telephony Capabilities 9.8.1 Alternative Routing Mechanism When the device routes a Tel-to-IP call, the destination number is compared to the list of prefixes defined in the 'Outbound IP Routing Table' (described in ''Configuring the Outbound IP Routing Table'' on page ).
Page 556: Fax And Modem Capabilities
Mediant 600 & Mediant 1000 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 (refer to ''Fax/Modem Operating Modes'' on page  Fax and modem transport modes (refer to ''Fax/Modem Transport Modes'' on page ...
Page 557: Fax Relay Mode
SIP User's Manual 9. IP Telephony Capabilities 9.9.2.1 T.38 Fax Relay Mode In Fax Relay mode, fax signals are transferred using the T.38 protocol. T.38 is an ITU standard for sending fax across IP networks in real-time mode. The device currently supports only the T.38 UDP syntax.
Page 558: Fax / Modem Transport Mode
Mediant 600 & Mediant 1000 9.9.2.1.2 Automatically Switching to T.38 Mode without SIP Re-INVITE In the Automatically Switching to T.38 Mode without SIP Re-INVITE mode, when a fax signal is detected, the channel automatically switches from the current voice coder to answer tone mode, and then to T.38-compliant fax relay mode.
Page 559: Fax/Modem Bypass Mode
SIP User's Manual 9. IP Telephony Capabilities  Dynamic Jitter Buffer Minimum Delay = 40  Dynamic Jitter Buffer Optimization Factor = 13 When the device initiates a fax session using G.711, a ‘gpmd’ attribute is added to the SDP according to the following format: ...
Page 560: Fax / Modem Nse Mode
Mediant 600 & Mediant 1000 • FaxModemBypassBasicRTPPacketInterval • FaxModemBypassDJBufMinDelay Note: When the device is configured for modem bypass and T.38 fax, V.21 low- speed modems are not supported and fail as a result. 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...
Page 561: Fax / Modem Transparent With Events Mode
SIP User's Manual 9. IP Telephony Capabilities  V34ModemTransportType = 2  BellModemTransportType = 2 9.9.2.6 Fax / Modem Transparent with Events Mode In this mode, fax and modem signals are transferred using the current voice coder with the following automatic adaptations: ...
Page 562: Rfc 2833 Ans Report Upon Fax/Modem Detection
Mediant 600 & Mediant 1000 • DJBufOptFactor • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission. Instead, use the modes Bypass (refer to ''Fax/Modem Bypass Mode'' on page ) or Transparent with Events (refer to ''Fax / Modem...
Page 563: Using Relay Mode For Both T.30 And V.34 Faxes
SIP User's Manual 9. IP Telephony Capabilities  V32ModemTransportType = 2  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) ...
Page 564: Working With Supplementary Services
Mediant 600 & Mediant 1000 Below is an example of media descriptions of an SDP indicating support for V.152. 0 0 IN IPV4 <IPAdressA> t=0 0 p=+1 c=IN IP4 <IPAddressA m=audio <udpPort A> RTP/AVP 18 0 a=ptime:10 a=rtpmap:96 PCMU/8000 a=gpmd: 96 vbd=yes In the example above, V.152 implementation is supported (using the dynamic payload type...
Page 565: Call Hold And Retrieve
SIP User's Manual 9. IP Telephony Capabilities 9.10.1 Call Hold and Retrieve Initiating Call Hold and Retrieve:  Active calls can be put on-hold by pressing the phone's hook-flash button.  The party that initiates the hold is called the holding party;...
Page 566: Figure 9-54: Double Hold Sip Call Flow
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 567: Consultation Feature
SIP User's Manual 9. IP Telephony Capabilities The flowchart above describes the following "double" call-hold scenario: A calls B and establishes a voice path. A places B on hold; A hears a Dial tone and B hears a Held tone. A calls C and establishes a voice path.
Page 568: Call Transfer
Mediant 600 & Mediant 1000 9.10.3 Call Transfer There are two types of call transfers:  Consultation Transfer (REFER and REPLACES): The common method to perform a consultation transfer is as follows: In the transfer scenario there are three parties - Party A = transferring, Party B = transferred, Party C = transferred to.
Page 569 SIP User's Manual 9. IP Telephony Capabilities Call forward performed by the PSTN side: When the device sends the INVITE message to the remote SIP entity and receives a SIP 302 response, the device sends a Facility message with the same IE mentioned above to the PSTN, and waits for the PSTN side to disconnect the call.
Page 570: Call Forward Reminder Ring
Mediant 600 & Mediant 1000 9.10.4.1 Call Forward Reminder Ring The device supports the Call Forward Reminder Ring feature for FXS interfaces, whereby the device's FXS endpoint emits a short ring burst (only if in onhook state) when a third- party Application Server (e.g., softswitch) forwards an incoming call to another destination.
Page 571: Call Waiting
SIP User's Manual 9. IP Telephony Capabilities For playing the special dial tone, the received SIP NOTIFY message must contain the following headers:  From and To: contain the same information, indicating the specific endpoint  Event: ua-profile  Content-Type: "application/simservs+xml"...
Page 572: Message Waiting Indication
Mediant 600 & Mediant 1000 To configure a delay interval before a Call Waiting Indication is played to the currently busy port, use the parameter TimeBeforeWaitingIndication. This enables the caller to hang up before disturbing the called party with Call Waiting Indications. Applicable only to FXS modules.
Page 573 SIP User's Manual 9. IP Telephony Capabilities The process for sending the MWI status upon request from a softswitch is as follows: The softswitch sends a SIP SUBSCRIBE message to the device. The device responds by sending an empty SIP NOTIFY to the softswitch, and then sending an ISDN Setup message with Facility IE containing an MWI Interrogation request to the PBX.
Page 574: Caller Id
Mediant 600 & Mediant 1000 9.10.7 Caller ID This section discusses the device's Caller ID support. Note: The Caller ID feature is applicable only to FXS/FXO interfaces. 9.10.7.1 Caller ID Detection / Generation on the Tel Side By default, generation and detection of Caller ID to the Tel side is disabled. To enable Caller ID, set the parameter EnableCallerID to 1.
Page 575: Debugging A Caller Id Detection On Fxo
SIP User's Manual 9. IP Telephony Capabilities 9.10.7.2 Debugging a Caller ID Detection on FXO The procedure below describes debugging caller ID detection in FXO interfaces.  To debug a Caller ID detection on an FXO interface: Verify that the parameter EnableCallerID is set to 1. Verify that the caller ID standard (and substandard) of the device matches the standard (using...
Page 576: Caller Id On The Ip Side
Mediant 600 & Mediant 1000 9.10.7.3 Caller ID on the IP Side Caller ID is provided by the SIP From header containing the caller's name and "number", for example: From: “David” <SIP:101@10.33.2.2>;tag=35dfsgasd45dg If Caller ID is restricted (received from Tel or configured in the device), the From header is set to: From: “anonymous”...
Page 577: Three-Way Conferencing
SIP User's Manual 9. IP Telephony Capabilities 9.10.8 Three-Way Conferencing The device supports three-way conference calls. These conference calls can also occur simultaneously. The following example demonstrates three-way conferencing. This example assumes that a telephone "A" connected to the device wants to establish a three-way conference call with two remote IP phones "B"...
Page 578: Routing Examples
Mediant 600 & Mediant 1000 9.11 Routing Examples 9.11.1 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 579  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.6.00.010.006 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 580 Mediant 600 & Mediant 1000 SIP/2.0 200 OK 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 Call-ID: 534366556655skKw-6000--2000@10.8.201.108 CSeq: 18153 INVITE Contact: <sip:2000@10.8.201.161;user=phone> Server: Audiocodes-Sip-Gateway/Mediant 1000/v.6.00.010.006 Supported: 100rel,em Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE, NOTIFY,PRACK,REFER,INFO Content-Type: application/sdp Content-Length: 206 o=AudiocodesGW 30221 87035 IN IP4 10.8.201.161 s=Phone-Call c=IN IP4 10.8.201.10...
Page 581: Sip Authentication Example
9. IP Telephony Capabilities SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud 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 Server: Audiocodes-Sip-Gateway/Mediant 1000/v.6.00.010.006 CSeq: 18154 BYE Supported: 100rel,em Content-Length: 0 9.11.2 SIP Authentication Example The device supports basic and digest (MD5) authentication types, according to SIP RFC 3261 standard.
Page 582 Mediant 600 & Mediant 1000 According to the sub-header present in the WWW-Authenticate header, the correct REGISTER request is created. Since the algorithm is MD5: • The username is equal to the endpoint phone number 122. • The realm return by the proxy is audiocodes.com.
Page 583 SIP User's Manual 9. IP Telephony Capabilities SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.200 From: <sip: 122@10.1.1.200>;tag=1c23940 To: <sip: 122@10.1.1.200> Call-ID: 654982194@10.1.1.200 Cseq: 1 REGISTER Date: Thu, 26 Jul 2001 09:34:42 GMT Server: Columbia-SIP-Server/1.17 Content-Length: 0 Contact: <sip:122@10.1.1.200>; expires="Thu, 26 Jul 2001 10:34:42 GMT";...
Page 584: Proxy Or Registrar Registration Example
Mediant 600 & Mediant 1000 9.11.3 Proxy or Registrar Registration Example Below is an example of Proxy and Registrar registration: REGISTER sip:servername SIP/2.0 VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234 From: <sip:GWRegistrationName@sipgatewayname>;tag=1c29347 To: <sip:GWRegistrationName@sipgatewayname> Call-ID: 10453@212.179.22.229 Seq: 1 REGISTER Expires: 3600 Contact: sip:GWRegistrationName@212.179.22.229 Content-Length: 0 The ‘servername’...
Page 585: Establishing A Call Between Two Devices
SIP User's Manual 9. IP Telephony Capabilities 9.11.4 Establishing a Call between Two Devices This section provides an example on configuring two AudioCodes' devices with FXS interfaces for establishing call communication. After configuration, you can make calls between telephones connected to the same device and between the two devices. This example assumes the following: ...
Page 586: Trunk-To-Trunk Routing Example
Mediant 600 & Mediant 1000 Figure 9-59: Routing Calls Between Devices Make a call. Pick up the phone connected to port #1 of the first device and dial 102 (to the phone connected to port #2 of the same device). Listen for progress tones at the calling phone and for the ringing tone at the called phone.
Page 587: Sip Trunking Between Enterprise And Itsps
SIP User's Manual 9. IP Telephony Capabilities • PSTNPrefix = 4,4 At Device B , remove the first digit from each IP-to-PSTN number before it is used in an outgoing call: NumberMapIP2Tel = *,1. 9.11.6 SIP Trunking between Enterprise and ITSPs By implementing the device's enhanced and flexible routing capabilities, you can "design"...
Page 588: Figure 9-61: Configuring Proxy Set Id #1 In The Proxy Sets Table
Mediant 600 & Mediant 1000  To configure call routing between an Enterprise and two ITSPs: Enable the device to register to a Proxy/Registrar server using the parameter IsRegisterNeeded. In the 'Proxy Sets Table' page (refer to ''Configuring the Proxy Sets Table''...
Page 589: Figure 9-63: Assigning Trunks To Trunk Group Id #1
SIP User's Manual 9. IP Telephony Capabilities 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 9-63: 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 590: Mapping Pstn Release Cause To Sip Response
Mediant 600 & Mediant 1000 9.12 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. The converse is also true - for Tel-to-IP calls, the SIP 4xx or 5xx responses are mapped to tones played to the PSTN/PBX.
Page 591: Answer Machine Detector (Amd)
SIP User's Manual 9. IP Telephony Capabilities 9.14 Answer Machine Detector (AMD) Answering Machine Detection (AMD) can be useful in automatic dialing applications. In some of these applications, it is important to detect if a human voice or an answering machine is answering the call.
Page 592: Table 9-6: Approximate Amd Detection High Sensitivity (Based On North American English)
Mediant 600 & Mediant 1000 Performance AMD Detection Sensitivity Success Rate for Live Calls Success Rate for Answering Machine 3 (Default) 88.57% 94.76% 88.94% 94.31% 90.42% 91.64% 90.66% 91.30% 7 (Best for Live 94.72% 76.14% Calls) Table 9-6: Approximate AMD Detection High Sensitivity (Based on North American English)
Page 593 SIP User's Manual 9. IP Telephony Capabilities Note: The device's AMD feature is based on voice detection for North American English. If you want to implement AMD in a different language or region, you must provide AudioCodes with a database of recorded voices in the language on which the device's AMD mechanism can base its voice detector algorithms for detecting these voices.
Page 594 Mediant 600 & Mediant 1000 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 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 595: Event Notification Using X-Detect Header
SIP User's Manual 9. IP Telephony Capabilities 9.15 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 596: Table 9-8: Special Information Tones (Sits) Reported By The Device
Mediant 600 & Mediant 1000 The device can map these SIT tones to a Q.850 cause and then map them to SIP 5xx/4xx responses, using parameters SITQ850Cause, SITQ850CauseForNC, SITQ850CauseForIC, SITQ850CauseForVC, and SITQ850CauseForRO. Table 9-8: Special Information Tones (SITs) Reported by the device...
Page 597: Supported Radius Attributes
SIP User's Manual 9. IP Telephony Capabilities Each time the device detects a supported event, the event is notified to the remote party by sending an INFO message with the following message body: • Content-Type: application/X-DETECT • Type = [AMD | CPT | FAX | PTT…] •...
Page 598 Mediant 600 & Mediant 1000 Attribute Attribute Value Purpose Example Number Name Format Start NAS-IP- IP address of the Numeric 192.168.14.43 Address requesting device Stop Start Service- Type of service requested Numeric 1: login Type Stop Start H323- Up to...
Page 599 SIP User's Manual 9. IP Telephony Capabilities Attribute Attribute Value Purpose Example Number Name Format The call's terminator: Call- PSTN-terminated call Stop String Yes, No Terminator (Yes); IP-terminated call (No). Start String 8004567145 Destination phone Stop String 2427456425 number Start Calling Party Number String 5135672127...
Page 600: Call Detail Record
Mediant 600 & Mediant 1000 Attribute Attribute Value Purpose Example Number Name Format Session-ID identifier – match start & stop Below is an example of RADIUS Accounting, where the non-standard parameters are preceded with brackets. Accounting-Request (361) user-name = 111 acct-session-id = 1 nas-ip-address = 212.179.22.213...
Page 601 SIP User's Manual 9. IP Telephony Capabilities Field Name Description EPTyp Endpoint Type Orig Call Originator (IP, Tel) SourceIp Source IP Address DestIp Destination IP Address Source Phone Number Type Source Phone Number Plan SrcPhoneNum Source Phone Number SrcNumBeforeMap Source Number Before Manipulation Destination Phone Number Type Destination Phone Number Plan DstPhoneNum...
Page 602: Rtp Multiplexing (Throughpacket)
Mediant 600 & Mediant 1000 Field Name Description Redirection Phone Number Plan RedirectPhonNum Redirection Phone Number 9.18 RTP Multiplexing (ThroughPacket) The device supports a proprietary method to aggregate RTP streams from several channels. This reduces the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP headers and reduces the packet/data transmission rate.
Page 603: Dynamic Jitter Buffer Operation
SIP User's Manual 9. IP Telephony Capabilities 9.19 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 605: Networking Capabilities
SIP User's Manual 10. Networking Capabilities Networking Capabilities This section provides an overview of the device's networking capabilities. 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: ...
Page 606: 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 607: Stun
SIP User's Manual 10. Networking Capabilities To resolve these issues, the following mechanisms are available:  STUN (refer to STUN on page 607)  First Incoming Packet Mechanism (refer to '' First Incoming Packet Mechanism'' page  RTP No-Op packets according to the avt-rtp-noop draft (refer to ''No-Op Packets'' page For information on SNMP NAT traversal, refer to the Product Reference Manual .
Page 608: First Incoming Packet Mechanism
Mediant 600 & Mediant 1000 10.3.2 First Incoming Packet Mechanism If the remote device resides behind a NAT device, it’s possible that the device can activate the RTP/RTCP/T.38 streams to an invalid IP address / UDP port. To avoid such cases, the device automatically compares the source address of the incoming RTP/RTCP/T.38 stream...
Page 609: Robust Receipt Of Rtp Streams
SIP User's Manual 10. Networking Capabilities 10.5 Robust Receipt of RTP 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 610: Ip Qos Via Differentiated Services (Diffserv)
Mediant 600 & Mediant 1000 The client requests a time update from a specified NTP server at a specified update interval. In most situations, this update interval is every 24 hours based on when the system was restarted. The NTP server identity (as an IP address) and the update interval are user-...
Page 611: Multiple Network Interfaces And Vlans
SIP User's Manual 10. Networking Capabilities 10.9.1 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 10-2: Multiple Network Interfaces This figure above depicts a typical configuration featuring in which the device is configured with three network interfaces for:...
Page 612: Overview Of Multiple Interface Table
Mediant 600 & Mediant 1000 10.9.1.1 Overview of Multiple Interface Table The Multiple Interfaces scheme allows you to define up to 16 different IP addresses and VLANs in a table format, as shown below: Table 10-1: Multiple Interface Table Index...
Page 613: Table 10-2: Application Types
SIP User's Manual 10. Networking Capabilities 10.9.1.2.2 Application Types Column This column defines the types of applications that are allowed on this interface:  OAMP – Operations, Administration, Maintenance and Provisioning applications such as Web, Telnet, SSH, SNMP  CONTROL – Call Control Protocols (i.e., SIP) ...
Page 614: Table 10-3: Configured Default Gateway Example
Mediant 600 & Mediant 1000 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 31.
Page 615: Other Related Parameters
SIP User's Manual 10. Networking Capabilities 10.9.1.2.7 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.
Page 616: Table 10-5: Quality Of Service Parameters
Mediant 600 & Mediant 1000 10.9.1.3.4 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 617: Table 10-6: Traffic / Network Types And Priority
SIP User's Manual 10. Networking Capabilities The mapping of an application to its CoS and traffic type is shown in the table below: Table 10-6: Traffic / Network Types and Priority Application Traffic / Network Types Class-of-Service (Priority) Debugging interface Management Bronze Telnet...
Page 618: Multiple Interface Table Configuration Summary And Guidelines
Mediant 600 & Mediant 1000 Table 10-7: Application Type Parameters Parameter Description EnableDNSasOAM This parameter applies to both Multiple IPs and VLAN mechanisms. Multiple IPs: Determines the network type for DNS services. VLAN: Determines the traffic type for DNS services.
Page 619: Troubleshooting The Multiple Interface Table
SIP User's Manual 10. Networking Capabilities  Apart from the interface having the default gateway defined, the Gateway column for all other interfaces must be set to "0.0.0.0" for IPv4.  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.
Page 620: Routing Table
Mediant 600 & Mediant 1000  An IPv4 interface was defined with "Interface Type" different than "IPv4 Manual" (10).  Gateway column is filled in more than one row of the same address family.  Gateway is defined in an interface not having MEDIA as one of its "Application Types".
Page 621: Figure 10-3: Prefix Length And Subnet Masks Columns
SIP User's Manual 10. Networking Capabilities 10.9.2.2.2 Prefix Length and Subnet Mask Columns These two columns offer two notations for the mask. You can either enable the Subnet Mask in dotted-decimal notation, or the CIDR-style representation. Please note that only one of these is needed.
Page 622: Routing Table Configuration Summary And Guidelines
Mediant 600 & Mediant 1000 10.9.2.2.5 Metric Column The Metric column must be set to 1 for each routing rule. 10.9.2.3 Routing Table Configuration Summary and Guidelines The Routing table configurations must adhere to the following rules:  Up to 25 different routing rules may be defined.
Page 623: Setting Up The Device
SIP User's Manual 10. Networking Capabilities 10.9.3 Setting up the Device 10.9.3.1 Using the Web Interface The Web interface is a convenient user interface for configuring the device's network configuration. 10.9.3.2 Using the ini File When configuring the network configuration using the ini File, use a textual presentation of the Interface and Routing Tables, as well as some other parameters.
Page 624: Table 10-9: Multiple Interface Table - Example1
Mediant 600 & Mediant 1000  VLANs are disabled, 'Native' VLAN ID is set to 1.  Values for the Class Of Service parameters are assigned.  The DNS application is configured to act as an OAMP application and the NTP application is configured to act as an OAMP application.
Page 625: Table 10-11: Multiple Interface Table - Example 2
SIP User's Manual 10. Networking Capabilities RoutingTableHopsCountColumn = 1, 1 Example 2: Three Interfaces, one for each application exclusively - the Multiple Interface table is configured with three interfaces, one exclusively for each application type: one interface for OAMP applications, one for Call Control applications, and one for RTP Media applications: Table 10-11: Multiple Interface Table - Example 2 Prefix...
Page 626: Table 10-13: Multiple Interface Table - Example 3
Mediant 600 & Mediant 1000 Example 3 - One interface exclusively for management (OAMP applications) and two others for Call Control and RTP (CONTROL and MEDIA applications): The Multiple Interface table is configured with four interfaces. One is exclusively for Management and the are for Call Control and RTP Media applications.
Page 627: Advanced Pstn Configuration
SIP User's Manual 11. Advanced PSTN Configuration Advanced PSTN Configuration This section discusses advanced PSTN configurations. 11.1 Clock Settings In a traditional TDM service network such as PSTN, both ends of the TDM connection must be synchronized. If synchronization is not achieved, voice frames are either dropped (to prevent a buffer overflow condition) or inserted (to prevent an underflow condition).
Page 628: Release Reason Mapping
Mediant 600 & Mediant 1000 11.2 Release Reason Mapping This section describes the available mapping mechanisms of SIP responses to Q.850 Release Causes and vice versa. The existing mapping of ISDN Release Causes to SIP Responses is described in ''Fixed Mapping of ISDN Release Reason to SIP Response''...
Page 629 SIP User's Manual 11. Advanced PSTN Configuration ISDN Release Description Description Response Reason Channel unacceptable Not acceptable Call awarded and being delivered in an Server internal error established channel Normal call clearing User busy Busy here No user responding Request timeout No answer from the user Temporarily unavailable Call rejected...
Page 630: Fixed Mapping Of Sip Response To Isdn Release Reason
Mediant 600 & Mediant 1000 ISDN Release Description Description Response Reason Invalid call reference value 502* Bad gateway Identified channel does not exist 502* Bad gateway Suspended call exists, but this call 503* Service unavailable identity does not Call identity in use...
Page 631 SIP User's Manual 11. Advanced PSTN Configuration ISDN Release Description Description Response Reason Payment required Call rejected Forbidden Call rejected Not found Unallocated number Method not allowed Service/option unavailable Not acceptable Service/option not implemented Proxy authentication Call rejected required Request timeout Recovery on timer expiry Conflict Temporary failure...
Page 632: Isdn Overlap Dialing
Mediant 600 & Mediant 1000 11.3 ISDN Overlap Dialing Overlap dialing is a dialing scheme used by several ISDN variants to send and/or receive called number digits one after the other (or several at a time). This is in contrast to en-bloc dialing in which a complete number is sent.
Page 633: Isdn Non-Facility Associated Signaling (Nfas)
SIP User's Manual 11. Advanced PSTN Configuration  DigitMapping  MinOverlapDigitsForRouting For configuring ISDN overlap dialing using the Web interface, refer to ''Configuring the Trunk Settings'' on page 11.4 ISDN Non-Facility Associated Signaling (NFAS) In regular T1 ISDN trunks, a single 64 kbps channel carries signaling for the other 23 B- channels of that particular T1 trunk.
Page 634: Nfas Interface Id
Mediant 600 & Mediant 1000 11.4.1 NFAS Interface ID Several ISDN switches require an additional configuration parameter per T1 trunk that is called ‘Interface Identifier’. In NFAS T1 trunks, the Interface Identifier is sent explicitly in Q.931 Setup / Channel Identification IE for all NFAS trunks, except for the B-channels of the Primary trunk (refer to note below).
Page 635: Creating An Nfas-Related Trunk Configuration
SIP User's Manual 11. Advanced PSTN Configuration NFASGroupNumber_0 = 1 NFASGroupNumber_1 = 1 NFASGroupNumber_2 = 1 NFASGroupNumber_3 = 1 DchConfig_0 = 0 ;Primary T1 trunk DchConfig_1 = 1 ;Backup T1 trunk DchConfig_2 = 2 ;B-Channel NFAS trunk DchConfig_3 = 2 ;B-channel NFAS trunk If there is no NFAS Backup trunk, the following configuration should be used: ISDNNFASInterfaceID_0 = 0...
Page 636: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 Notes: • All trunks in the group must be configured with the same values for trunk parameters TerminationSide, ProtocolType, FramingMethod, and LineCode. • After stopping or deleting the backup trunk, delete the group and then reconfigure it.
Page 637: Redirect Number And Calling Name (Display)
SIP User's Manual 11. Advanced PSTN Configuration 11.5 Redirect Number and Calling Name (Display) The following tables define the device's redirect number and calling name (Display) support for various ISDN variants according to NT (Network Termination) / TE (Termination Equipment) interface direction: Table 11-3: Calling Name (Display) NT/TE Interface DMS-100...
Page 639: Tunneling Applications
SIP User's Manual 12. Tunneling Applications Tunneling Applications This section discusses the device's support for tunneling applications. 12.1 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 routing (without Proxy control) capabilities to receive voice and data streams from TDM (E1/T1/J1/ ) spans or individual timeslots, convert them into packets, and then transmit them...
Page 640 Mediant 600 & Mediant 1000 By utilizing the ‘Profiles’ mechanism (refer to ''Coders and Profile Definitions'' on page you can configure the TDM Tunneling feature to choose different settings based on a timeslot or groups of timeslots. For example, you can use low-bit-rate vocoders to transport voice and ‘Transparent’...
Page 641 SIP User's Manual 12. Tunneling Applications CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = g7231; CodersGroup0 1 = Transparent; [ \CodersGroup0 ] [TelProfile] FORMAT TelProfile_Index = TelProfile_ProfileName, TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP;...
Page 642: Dsp Pattern Detector
Mediant 600 & Mediant 1000 ;E1_TRANSPARENT_31 ProtocolType_0 = 5 ProtocolType_1 = 5 ProtocolType_2 = 5 ProtocolType_3 = 5 ;Channel selection by Phone number. ChannelSelectMode = 0 [TrunkGroup] FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId, TrunkGroup_LastTrunkId, TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel, TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId, TrunkGroup_Module; TrunkGroup 0 = 0,0,0,1,31,1000,1;...
Page 643: Qsig Tunneling
SIP User's Manual 12. Tunneling Applications  EnablePatternDetector = 1  Pattern Detector Threshold: PDThreshold - defines the number of consecutive patterns to trigger the pattern detection event. For example: PDThreshold = 5 Detection Pattern: PDPattern - defines the patterns that can be detected by the Pattern Detector.
Page 644 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 645 SIP User's Manual 12. Tunneling Applications Reader’s Notes Version 6.0 March 2010...
Page 647: Ip Media Capabilities
Voice XML Interpreter (refer to Voice XML Interpreter on page 688) Note: This section is applicable only to 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 648: Simple Conferencing (Netann)
Mediant 600 & Mediant 1000 Note: The conference application is a special order option. 13.1.1 Simple Conferencing (NetAnn) 13.1.1.1 SIP Call Flow A SIP call flow for simple conferencing is shown below: Table 13-1: Simple Conferencing SIP Call Flow SIP User's Manual...
Page 649: Creating A Conference
SIP User's Manual 13. IP Media Capabilities 13.1.1.2 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. The User Part of that Request-URI includes both the Conference Service Identifier (indicating that the requested Media Service is a Conference) and a Unique Conference Identifier (identifying a specific instance of a conference).
Page 650: Pstn Participants
Mediant 600 & Mediant 1000 A disconnects. A joins (not guaranteed). Sending a BYE request to the device terminates the participant's SIP session and removes it from the conference. The final BYE from the last participant ends the conference and releases all conference resources.
Page 651: Joining A Conference
SIP User's Manual 13. IP Media Capabilities Table 13-2: Advanced Conferencing SIP Call Flow 13.1.2.2 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>...
Page 652: Modifying A Conference
Mediant 600 & Mediant 1000 The <configure_team> element enables clients to create personalized mixes for scenarios where the standard mixmode settings do not provide sufficient control. <configure_team> element is a child of <configure_leg>. The <configure_team> element, containing one or more <teammate> elements, specifies those participants that should be present in this participant’s personalized mix.
Page 653: Table 13-4: Modifying A Conference - Sip Call Flow
SIP User's Manual 13. IP Media Capabilities To modify a certain Participant Leg, a <configure_leg> MSCML request body is sent in an INFO message on that leg SIP dialog. Using this request, the Application Server can modify any of the attributes defined for the <configure_leg> request. Table 13-4: Modifying a Conference - SIP Call Flow Version 6.0 March 2010...
Page 654: Applying Media Services On A Conference
Mediant 600 & Mediant 1000 13.1.2.4 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 655: Terminating A Conference
SIP User's Manual 13. IP Media Capabilities <?xml version="1.0" encoding="utf-8"?> <MediaServerControl version="1.0"> <notification> <conference uniqueID="3331" numtalkers="1"> <activetalkers> <talker callID="9814266171512000193619@10.8.27.118"/> </activetalkers> </conference> </notification> </MediaServerControl> 13.1.2.6 Terminating a Conference To remove a leg from a conference, the Application Server issues a SIP BYE request on the selected dialog representing the conference leg.
Page 656: Conference Call Flow Example
Mediant 600 & Mediant 1000 13.1.3 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). Table 13-7: Conference Call Flow Example SIP MESSAGE 1: 10.8.29.1:5060 ->...
Page 657 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 658 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 659 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 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 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>...
Page 660 Mediant 600 & Mediant 1000 INVITE sip:conf100@10.8.58.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.58.8;branch=z9hG4bKaczJpxnnv Max-Forwards: 70 From: <sip:800@10.8.58.8>;tag=1c2419012378 To: <sip:conf100@10.8.58.4;user=phone> Call-ID: 150852731NDDC@10.8.58.8 CSeq: 1 INVITE 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...
Page 661 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 INVITE Contact: <sip:conf100@10.8.58.4> 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-Type: application/sdp Content-Length: 236 o=AudiocodesGW 385533 708665 IN IP4 10.8.58.4 s=Phone-Call c=IN IP4 10.8.58.4 t=0 0 m=audio 7140 RTP/AVP 4 96 a=rtpmap:4 g723/8000...
Page 662: Announcement Server
Mediant 600 & Mediant 1000 SIP MESSAGE 14: 10.8.58.4:5060 -> 10.8.58.8:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.58.8;branch=z9hG4bKackSIyGww 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:conf100@10.8.58.4> 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 15: 10.8.58.6:5060 -> 10.8.58.4:5060 BYE sip:conf100@10.8.58.4 SIP/2.0...
Page 663: Netann Interface
SIP User's Manual 13. IP Media Capabilities 13.2.1 NetAnn Interface The device supports playing announcements using NetAnn format (according to RFC 4240). 13.2.1.1 Playing a Local Voice Prompt To play a single local Voice Prompt, 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 664: Supported Attributes
Mediant 600 & Mediant 1000 13.2.1.3 Supported Attributes When playing announcements, the following attributes are available:  Repeat: defines the number of times the announcement is repeated. The default value is 1. The valid range is 1 to 1000, or -1 (i.e., repeats the message forever).
Page 665: Mscml Interface
SIP User's Manual 13. IP Media Capabilities 13.2.2 MSCML Interface Media Server Control Markup Language (MSCML), according to IETF RFC 5022 is a protocol used in conjunction with SIP to provide advanced announcements handling. MSCML is implemented by adding an XML body to existing SIP INFO messages. Only a single message body (containing a single request or response) is allowed per message.
Page 666: Operation
Mediant 600 & Mediant 1000 13.2.2.1 Operation The APS server can be used to generate two files - the audio package as a VP.dat file, and an XML file (segments.xml) that contains indices to the announcements stored on the VP.dat file for playing announcements. These two files can be loaded to the device using the Web interface.
Page 667: Operating With Audio Bundles
SIP User's Manual 13. IP Media Capabilities  <PlayCollect> for playing announcements and collecting digits  <PlayRecord> for playing announcements and recording voice  <Stop> for stopping the playing of an announcement The device sends a Response to each Request that is issued by the Application server. The <Play>, <PlayCollect>, and <PlayRecord>...
Page 668 Mediant 600 & Mediant 1000 To upload a voice bundle to the device, the following ini file parameters should be set: APSEnabled = 1 AMSProfile = 1 VpFileUrl = 'url-dat-file/dat-file' APSSegmentsFileUrl = ‘url-xml-file/xml-file’' Where url-dat-file and url-xml-file relate to the location of the relevant .dat and .xml files,...
Page 669: Playing Announcements
SIP User's Manual 13. IP Media Capabilities 13.2.2.3 Playing Announcements A <Play> request is used to play an announcement to the caller. Each <Play> request contains a single Prompt block and the following request-specific parameters:  : an optional random number used to synchronize request and response. ...
Page 670 Mediant 600 & Mediant 1000  interdigittimer: defines the amount of time (in milliseconds) the user does not enter any digits after the first DTMF digit is received, after which a response is sent indicating timeout.  extradigittimer: used to enable the following: •...
Page 671: Playing Announcements And Recording Voice
SIP User's Manual 13. IP Media Capabilities 13.2.2.5 Playing Announcements and Recording Voice The <PlayRecord> request is used to play an announcement to the caller and to then record the voice stream associated with that caller. The play part of the <PlayRecord> request is identical to the <Play>...
Page 672: Stopping The Playing Of An Announcement
Mediant 600 & Mediant 1000 <MediaServerControl version="1.0"> <response request=“playrecord” id=”75899” code=”200” text=”OK” reclength=”15005”> </response> </MediaServerControl> 13.2.2.6 Stopping the Playing of an Announcement The Application server issues a <stop> request when it requires that the device stops a request in progress and not initiate another operation. The only (optional) request-specific parameter is id.
Page 673: Signal Events Notifications
SIP User's Manual 13. IP Media Capabilities 13.2.2.8 Signal Events Notifications The device supports Signal Events Notifications as defined in RFC 4722/5022 - MSCML. MSCML defines event notifications that are scoped to a specific SIP dialog or call leg. These events allow a client to be notified of various call progress signals. Subscriptions for call leg events are performed by sending an MSCML <configure_leg>...
Page 674: Voice Streaming
Mediant 600 & Mediant 1000 13.2.3 Voice Streaming The voice streaming layer provides you with the ability to play and record different types of files while using an NFSor HTTP server. 13.2.3.1 Voice Streaming Features The following subsections summarizes the Voice Streaming features supported on HTTP and NFS servers, unless stated otherwise.
Page 675 SIP User's Manual 13. IP Media Capabilities where,  :<port> is optional.  <path> is a path to a server-side script.  <searchpart> is of the form: key=value[&key=value]* Note: At least one key=value pair is required. Another example of a dynamic URL is shown below: http://MyServer:8080/prompts/servlet?action=play&language=eng&file =welcome.raw&format=1 (See also RFC 2396 URI: Generic Syntax.)
Page 676 Mediant 600 & Mediant 1000 13.2.3.1.9 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. Note: This feature is relevant for both NFS and HTTP.
Page 677: Using File Coders With Different Channel Coders
SIP User's Manual 13. IP Media Capabilities 13.2.3.2 Using File Coders with Different Channel Coders The tables in the following subsections describe the support for different combinations of file coders (used for recording or playing a file) and channel coders (used when opening a voice channel).
Page 678: Maximum Concurrent Playing And Recording
Mediant 600 & Mediant 1000 13.2.3.2.2 Recording a File The table below lists the device's support of channel coders and file coders for recording a file. Table 13-11: Coder Combinations - Recording a File File File Type Coder Channel Coder...
Page 679: Http Recording Configuration
SIP User's Manual 13. IP Media Capabilities Coder DSP Templates *.wav file *.raw file *.au file AMR (Rate 4.75) 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) 13.2.3.5 HTTP Recording Configuration...
Page 680: Supported Http Servers
Mediant 600 & Mediant 1000 Notes: • The combination of Host/IP and Root Path should be unique for each row in the table. For example, there should be only one row in the table with a Host/IP of 192.168.1.1 and Root Path of /audio.
Page 681: Supporting Nfs Servers
SIP User's Manual 13. IP Media Capabilities  Using mode perl, fix the mod_perl to the following: <IfModule mod_perl.c> <Location /cgi-bin> SetHandler perl-script PerlResponseHandler ModPerl::Registry Options +ExecCGI PerlOptions +ParseHeaders Order allow,deny Allow from all </Location> </IfModule>  Apache MPM worker: it is recommended to use the Multi-Processing Module implementing a hybrid multi-threaded multi-process Web server.
Page 682 Mediant 600 & Mediant 1000 13.2.3.8.1 Solaris-Based NFS Servers If you are using a Solaris™-based NFS server, then the following nfsd configuration modification is recommended, especially if you are planning to support voice recording:  Edit the file /etc/default/nfs and set the value of NFSD_SERVERS to N*2, where N the maximum number of record and play sessions that you expect to have in progress at any one time.
Page 683 SIP User's Manual 13. IP Media Capabilities  To restart the nfs daemon on Solaris, invoke the following two commands: > /etc/init.d/nfs.server stop > /etc/init.d/nfs.server start  To view a log of directories which were shared on the previous restart of the nfs daemon, type the sharetab file.
Page 684: Common Troubleshooting
Mediant 600 & Mediant 1000 13.2.3.9 Common Troubleshooting Always inspect the Syslog for any problem you may encounter; in many cases, the cause appears there. Table 13-14: Troubleshooting Problem Probable Cause Corrective Action General Voice Streaming Problems Attempts to perform voice streaming...
Page 685: Announcement Call Flow Example
SIP User's Manual 13. IP Media Capabilities Problem Probable Cause Corrective Action Remote file system is not being mounted and On a Linux NFS server, The NFS server is not the Syslog displays the following: configured to accept use the insecure option 'NFS mount failed, reason=permission requests on ports outside in the /etc/exports file...
Page 686: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 SIP MESSAGE 1: 10.33.2.40:5060 -> 10.33.24.1:5060 INVITE sip:annc@10.33.24.1;play=http://10.3.0.2/hello.wav;repeat=2 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...
Page 687 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> 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.5.40.010.006D Content-Length: 0 SIP MESSAGE 6: 10.33.2.40:5060 -> 10.33.24.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.24.1;branch=z9hG4bKacFhtFbFR 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:103@10.33.2.40>...
Page 688: 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, automatic speech recognition (ASR) and text-to-speech (TTS) are not supported.
Page 689: Vxml Scripts
SIP User's Manual 13. IP Media Capabilities 13.3.3 VXML Scripts Conceptually, there are two different types of VXML scripts that can be used (simultaneously or only one) by the device:  Dynamic scripts: This script is downloaded as needed for an individual call and usually contains customized content for that particular call.
Page 690: Record
Mediant 600 & Mediant 1000 13.3.4.1 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 691: Audio Extensions
SIP User's Manual 13. IP Media Capabilities <?xml version="1.0"?> <vxml version="2.0" xmlns=\"http://www.w3.org/2001/vxml\"> <var name=”recordpath” expr = “’http://192.168.1.2/recordings/greetings/’”/> <form id="form1"> <record name="msg" finalsilence="3000ms" maxtime="60s" dtmfterm="true" destexpr="recordpath + ‘callersspeech.wav’"> <audio src= "http://192.168.1.2/prompts/recordprompt.wav"/> <filled> <audio src = “http://192.168.1.2/prompts/confirm.wav”/> <audio expr= " recordpath + ‘callersspeech.wav’"/> <exit/>...
Page 692: Table 13-15: Say-As Phrase Types
Mediant 600 & Mediant 1000 13.3.4.2.2 Say-as Tag for the Audio Element While the VXML <say-as> tag is typically used as a directive to a text-to-speech engine in association with a VXML <prompt> element, the AudioCodes resident VXML Interpreter allows the <say-as> tag to also be used with the <audio> element. In this context, the <say- as>...
Page 693 SIP User's Manual 13. IP Media Capabilities Say-as Token Variable Variable Subtype Variable Input Note Type Format 64 digits including 0-9, * and #. telephone:ndn digits North American DN Must be 10 digits 0-9. telephone:gen digits generic directory A string of up to number 64 digits including 0-9, *,...
Page 694 Mediant 600 & Mediant 1000 In this example, the device outputs the date “January 20th, 2008”. 13.3.4.2.4 Supplying Selector Values to Provisioned Variables and to Say-as Phrases Another concept supported by the device is “selectors”. A selector is a keyword and value pair that is used by the device software to build announcements.
Page 695: Language Identifier Support
SIP User's Manual 13. IP Media Capabilities Selectors can also be useful in combination with <say-as> elements. For example, the following illustrates making the same announcement from the previous example using a <say-as> element: <audio src=”?sel=lang=fr&gender=F”> <say-as interpret-as=”date”> 20070620 </say-as> </audio> For more information regarding available, selectors, please refer to the Audio Provisioning Server (APS) User’s Guide .
Page 696: Combining
Mediant 600 & Mediant 1000 Language Code Language Turkish Vietnamese Cantonese Mandarin 13.3.5 Combining <audio> Elements 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.
Page 697: Voicexml Supported Elements And Attributes
SIP User's Manual 13. IP Media Capabilities  S: Supported 13.3.7.1 VoiceXML Supported Elements and Attributes Table 13-17: 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 698 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments event eventexpr message 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 699 SIP User's Manual 13. IP Media Capabilities Element Parameter Max Size Shadow Variable Status Comments modal true/false name$.utterance enum name$.inputmode name$.interpretation numeric name$.confidence <filled> mode namelist 4 * 32 <form> scope enum <goto> next expr nextitem expritem fetchaudio fetchtimeout fetchhint maxage maxstale <grammar>...
Page 700 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments xml:base scope enum In this release, a document scope grammar isn't active in a dialog scope form. type enum Built-in grammars are supported for recognition against fields, but the match is not spoken as the built-in type in text-to-speech.
Page 701 SIP User's Manual 13. IP Media Capabilities Element Parameter Max Size Shadow Variable Status Comments eventexpr message messageexpr dtmf fetchaudio 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...
Page 702 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments classid codebase codetype data type archive fetchtimeout fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <option> dtmf accept It's not obvious how to instruct the...
Page 703 SIP User's Manual 13. IP Media Capabilities Element Parameter Max Size Shadow Variable Status Comments name value <record> name expr cond modal Grammars are not supported, thus, modal doesn't apply. beep true/false Requires that a user-defined tone be added to the system. Please Example of UDT refer to '' ‘beep’...
Page 704 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments messageexpr namelist 4 * 32 <script> The script element and all of its attributes are not supported. charset fetchtimeout fetchhint maxage maxstale <subdialog> Playing a prompt from a sub-dialog element is not supported in this release.
Page 705 SIP User's Manual 13. IP Media Capabilities Element Parameter Max Size Shadow Variable Status Comments fetchaudio fetchtimeout Fetchhint Default behavior is "safe"; fetch document when it's needed. Maxage maxstale <throw> Event eventexpr message messageexpr <transfer> Name Expr Cond Dest Only numbers. destexpr Bridge Only blind transfer supported...
Page 706: Srgs And Ssml Support
Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments name expr Name <transfer> Expr Cond Dest Only numbers. destexpr Bridge Only Bridge = false type Only type = blind connecttimeout maxtime transferaudio Aaiexpr name$.duration name$.inputmode name$.utterance <value>...
Page 707: Voicexml Supported Properties
SIP User's Manual 13. IP Media Capabilities 13.3.7.3 VoiceXML Supported Properties Table 13-18: VoiceXML Supported Properties Platform Properties Status Equivalent ini file parameter or Notes Recognizer confidencelevel VxmlConfidenceLevel Sensitivity VxmlSensitivityLevel speedvsaccuracy VxmlSpeedVsAccuracy Completetimeout VxmlCompleteTimeout incompletetimeout VxmlInCompleteTimeout maxspeechtimeout VxmlMaxSpeechTimeout DTMF Recognizer Interdigittimeout VxmlInterDigitTimeout Termtimeout...
Page 708: Voicexml Variables And Events
Mediant 600 & Mediant 1000 Platform Properties Status Equivalent ini file parameter or Notes Scriptfetchhint Scriptmaxage Scriptmaxstale Fetchaudio Fetchaudiodelay fetchaudiominimum Fetchtimeout Miscellaneous Inputmodes VxmlSystemInputModes. Note that the system default is 0 (DTMF) vs 2 (Voice and DTMF) as specified in the specification. This is...
Page 709 SIP User's Manual 13. IP Media Capabilities Variable/Event Name Status Notes Note: while throwing and catching events from scripts are supported, throwing events asynchronously from within the interpreter (e.g., an event.badfetch) is currently not supported. catch connection.disconnect.hangup connection.disconnect.transfer exit help noinput nomatch maxspeechtimeout...
Page 710: Ecmascript Support
Mediant 600 & Mediant 1000 Variable/Event Name Status Notes error.unsupported.transfer.bridge error.unsupported.uri 13.3.7.5 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 711: Example Of Udt 'Beep' Tone Definition
SIP User's Manual 13. IP Media Capabilities Operand/Operator Examples Status Note Edition December, 1999 Numeric Literals Section 7.8.3, ECMA-262 3rd Edition December, 1999 String Literals Section 7.8.4, ECMA-262 3rd Edition December, 1999 13.3.8 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...
Page 712 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83306...
Page 713: Sip Software Package
Sample ini file for Mediant 600 E1 devices. M600_Digital_SIP_E1.ini Sample ini file for Mediant 600 T1 devices. SIPgw_M1K.ini Sample ini file for the Mediant 1000 and Mediant 600 devices. M1000_Digital_SIP_T1.ini Sample ini file for Mediant 1000 E1 devices. M1000_Digital_SIP_E1.ini Sample ini file for Mediant 1000 T1 devices.
Page 714 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83306...
Page 715: Selected Technical Specifications
SIP User's Manual 15. Selected Technical Specifications Selected Technical Specifications 15.1 Mediant 1000 The table below lists the main technical specifications of the Mediant 1000. Table 15-1: Mediant 1000 Functional Specifications Function Specification Interfaces Modularity and Capacity Voice interface: Equipped with 6 Slots that can host voice modules.
Page 716 Mediant 600 & Mediant 1000 Function Specification  OSN1: One SODIMM slot 512M or 1G RAM Memory  OSN2: 1 or 2 GRAM  OSN1: Single/Dual hard disk drives Storage  OSN2: Single SATA HDD  OSN1: 10/100Base-TX, USB, RS-232, NB relay, MOH Interfaces ...
Page 717: Mediant 600
SIP User's Manual 15. Selected Technical Specifications 15.2 Mediant 600 The table below lists the main technical specifications of the Mediant 600. Table 15-2: Mediant 600 Functional Specifications Function Specification Interfaces E1/T1/J1 1, 2 or fractional (15 DS0) span spans using RJ-48c connectors BRI S/T 4 or 8 ports (8/16 calls) using RJ-45 connectors Analog...
Page 718: Mediant 600 & Mediant
Mediant 600 & Mediant 1000 Function Specification Security Security Protocols IPSec, HTTPS, TLS (SIPS), SSL, Web access list, RADIUS login and SRTP Hardware Specifications Power Supply Single universal 90-260 V AC power supply Physical 1U high, 19-inch wide Dimensions 306 x 273 x 44 mm...
Page 719 SIP User's Manual 15. Selected Technical Specifications Reader’s Notes Version 6.0 March 2010...
Page 720 User's Manual Version 6.0 www.audiocodes.com...

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AudioCodes Mediant 1000 User Manual

1 Overview

2 Configuration Concepts

3 Web-Based Management

4 INI File Configuration

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Summary of Contents for AudioCodes Mediant 1000