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AudioCodes Mediant 600 User Manual

AudioCodes Mediant 600 User Manual

Media gateways
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See also: Troubleshooting Manual , User Manual
User's Manual
Version 5.8
Document #: LTRT-83305
September 2009

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Summary of Contents for AudioCodes Mediant 600

  • Page 1 User's Manual Version 5.8 Document #: LTRT-83305 September 2009...

  • Page 3: Table Of Contents

    SIP User's Manual Contents Table of Contents Overview ......................21     1.1  Mediant 1000 ......................21  1.2  Mediant 600 ......................22  1.3  SIP Overview ......................24  Configuration Concepts ................... 25     Web-Based Management ................. 27     3.1 ...
  • Page 4 Mediant 600 & Mediant 1000 3.3.2 Media Settings ......................74     3.3.2.1 Configuring the Voice Settings ..............74     3.3.2.2 Configuring the Fax/Modem/CID Settings ..........76     3.3.2.3 Configuring the RTP/RTCP Settings ............76     3.3.2.4 Configuring the IPmedia Settings ............
  • Page 5 SIP User's Manual Contents 3.5.2.3 Viewing Registration Status ..............213     3.5.2.4 Viewing SAS/SBC Registered Users ............ 214     3.5.2.5 Viewing IP Connectivity ................. 215     INI File Configuration ..................217     4.1  Secured Encoded ini File ..................217 ...
  • Page 6 Mediant 600 & Mediant 1000 6.3  Debugging and Diagnostics Parameters .............. 261  6.3.1 General Parameters ....................261     6.3.2 CDR and Debug Parameters ................262     6.3.3 Heartbeat Packet Parameters ................264     6.3.4 Remote Alarm Indication Parameters ..............264  ...
  • Page 7 SIP User's Manual Contents 6.17.3 Number Manipulation Parameters ................ 411     6.18  Channel Parameters .................... 421  6.18.1 General Parameters ....................421     6.18.2 Voice Parameters....................423     6.18.3 Coder Parameters ....................426     6.18.4 Fax and Modem Parameters ................427  ...
  • Page 8 Mediant 600 & Mediant 1000 9.7.2.4 Fax / Modem Transparent Mode ............483     9.7.2.5 Fax / Modem Transparent with Events Mode ........484     9.7.2.6 G.711 Fax / Modem Transport Mode ............ 484     9.7.2.7 Fax Fallback ..................485  ...
  • Page 9 SIP User's Manual Contents 9.24  SIP Trunking between Enterprise and ITSPs ............528  9.25  Remote PBX Extension Between FXO and FXS Devices ........532  9.25.1 Dialing from Remote Extension (Phone at FXS) ..........533     9.25.2 Dialing from PBX Line or PSTN ................533  ...
  • Page 10 Mediant 600 & Mediant 1000 Tunneling Applications .................. 571     12.1  TDM Tunneling ..................... 571  12.1.1 DSP Pattern Detector ................... 574     12.2  QSIG Tunneling ....................574  Media Server Capabilities ................577     13.1  Conference Server ....................
  • Page 11     14.3.2 Connecting using Remote Desktop Connection ........... 655     SIP Software Package ..................657     Selected Technical Specifications ..............659     16.1  Mediant 1000 ......................659  Mediant 600 ......................663  16.2  Version 5.8 September 2009...
  • Page 12 Figure 3-26: Log Off Confirmation Box ....................55   Figure 3-27: Web Session Logged Off ....................55   Figure 3-27: Home Page – Mediant 600 ....................56   Figure 3-27: Home Page – Mediant 1000 ..................... 56   Figure 3-28: Shortcut Menu for Assigning Port name ................58  ...
  • Page 13 SIP User's Manual Contents Figure 3-55: WEB User Accounts Page (for Users with 'Security Administrator' Privileges) ....87   Figure 3-56: Web & Telnet Access List Page - Add New Entry ............89   Figure 3-57: Web & Telnet Access List Table ..................89  ...
  • Page 14 Mediant 600 & Mediant 1000 Figure 3-113: Regional Settings Page ....................190   Figure 3-114: Maintenance Actions Page ................... 191   Figure 3-115: Reset Confirmation Message Box ................. 192   Figure 3-116: Device Lock Confirmation Message Box ..............193  ...
  • Page 15 SIP User's Manual Contents Figure 9-20: Configuring IP Groups #1 and #2 in the IP Group Table Page ........530   Figure 9-21: Assign the Trunk to Trunk Group ID #1 in the Trunk Group Table Page ......530   Figure 9-22: Configuring Trunk Group #1 for Registration per Account in Trunk Group Settings Page ................................
  • Page 16 Mediant 600 & Mediant 1000 List of Tables Table 3-1: Description of Toolbar Buttons ..................... 30   Table 3-2: ini File Parameters for Replacing Logo with Text ..............52   Table 3-3: ini File Parameters for Customizing Product Name ............. 52  ...
  • Page 17 SIP User's Manual Contents Table 6-16: Heartbeat Packet Parameters ..................264   Table 6-17: RAI Parameters ........................ 264   Table 6-18: Serial Parameters ......................265   Table 6-19: BootP Parameters ......................266   Table 6-20: General Security Parameters ................... 267  ...
  • Page 18 Table 13-12: VoiceXML Variables and Events ..................637   Table 13-13: ECMAScript Support ...................... 639   Table 15-1: Software Package ......................657   Table 16-1: Mediant 1000 Functional Specifications ................659   Table 16-2: Mediant 600 Functional Specifications ................663   SIP User's Manual Document #: LTRT-83305...

  • Page 19: Weee Eu Directive

    IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, TrunkPack, VoicePacketizer, VoIPerfect, VoIPerfectHD, What’s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are property of their respective owners.

  • Page 20: Related Documentation

    The following naming conventions are used throughout this manual, unless otherwise specified: • The term device refers to the Mediant 1000 and Mediant 600 gateways. • The term Trunk is used synonymously with Hunt. Trunk typically refers to digital modules, while Hunt typically refers to analog modules.

  • 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 AudioCodes 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

    Web browser (such as Microsoft™ Internet Explorer™). Mediant 600 AudioCodes' Mediant 600 (hereafter referred to as device) is a cost-effective, wireline Voice-over-IP (VoIP) Session Initiation Protocol (SIP)-based media gateway. It is designed to interface between Time-Division Multiplexing (TDM) and IP networks in enterprises, small and medium businesses (SMB), and CPE application service providers.
  • Page 23 SIP User's Manual 1. Overview Up to four FXO interfaces (RJ-11 ports) - for connecting analog lines of an enterprise's PBX or the PSTN to the IP network Up to four FXS interfaces (RJ-11 ports) - for connecting legacy telephones, fax machines, and modems 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

    Note: To initialize the device by assigning it an IP address, a firmware file (cmp), and a configuration file (ini file), you can use AudioCodes' BootP/TFTP utility, which accesses the device using its MAC address (refer to the Product Reference Manual).
  • Page 26 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83305...

  • 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: Accessing The Web Interface

    Mediant 600 & Mediant 1000 3.1.2 Accessing the Web Interface The Web interface can be opened using any standard Web browser (refer to ''Computer Requirements'' on page 27). When initially accessing the Web interface, use the default user name ('Admin') and password ('Admin'). For changing the login user name and password, refer to ''Configuring the Web User Accounts'' on page 86).

  • Page 29: Areas Of The Gui

    SIP User's Manual 3. Web-Based Management 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: Title bar: Displays the corporate logo and product name.

  • Page 30: Toolbar

    Mediant 600 & Mediant 1000 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 193).

  • Page 31: Navigation Tree

    SIP User's Manual 3. Web-Based Management 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 32: Displaying Navigation Tree In Basic And Full View

    Mediant 600 & Mediant 1000 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 33: Showing / Hiding The Navigation Pane

    SIP User's Manual 3. Web-Based Management 3.1.5.2 Showing / Hiding the Navigation Pane The Navigation pane can be hidden to provide more space for elements displayed in the Work pane. This is especially useful when the Work pane displays a 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 34: Accessing Pages

    Mediant 600 & Mediant 1000 3.1.6.1 Accessing Pages The configuration pages are accessed by clicking the required page item in the Navigation tree. To open a configuration page in the Work pane: On the Navigation bar, click the required tab: •...

  • Page 35: Figure 3-7: Toggling Between Basic And Advanced Page View

    SIP User's Manual 3. Web-Based Management 3.1.6.2.1 Displaying Basic and Advanced Parameters Some pages provide you with an Advanced Parameter List / Basic Parameter List toggle button that allows you to show or hide advanced parameters (in addition to displaying the basic parameters).

  • Page 36: Figure 3-8: Expanding And Collapsing Parameter Groups

    Mediant 600 & Mediant 1000 3.1.6.2.2 Showing / Hiding Parameter Groups Some pages provide groups of parameters, which can be hidden or shown. To toggle between hiding and showing a group, simply click the group name button that appears above each group. The button appears with a down-pointing or up-pointing arrow, indicating that it can be collapsed or expanded when clicked, respectively.

  • Page 37: Modifying And Saving Parameters

    SIP User's Manual 3. Web-Based Management 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).

  • Page 38: Entering Phone Numbers In Various Tables

    Mediant 600 & Mediant 1000 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...

  • Page 39: Figure 3-11: Adding An Index Entry To A Table

    SIP User's Manual 3. Web-Based Management To add an entry to a table: In the 'Add' field, enter the desired index entry number, and then click Add; 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 40: Figure 3-12: Compacting A Web Interface Table

    Mediant 600 & Mediant 1000 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 41: Searching For Configuration Parameters

    SIP User's Manual 3. Web-Based Management 3.1.7 Searching for Configuration Parameters The Web interface provides a search engine that allows you to search any ini file parameter that is configurable by the Web interface (i.e., has a corresponding Web parameter). You can search for a specific parameter (e.g., "EnableIPSec") or a sub-string of that parameter (e.g., "sec").

  • Page 42: Working With Scenarios

    Mediant 600 & Mediant 1000 3.1.8 Working with Scenarios The Web interface allows you to create your own "menu" with up to 20 pages selected from the menus in the Navigation tree (i.e., pertaining to the Configuration, Management, and Status & Diagnostics tabs). The "menu" is a set of configuration pages grouped into a logical entity referred to as a Scenario.

  • Page 43: Figure 3-15: Creating A Scenario

    SIP User's Manual 3. Web-Based Management 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). When you have added all the required Steps for your Scenario, click the Save &...

  • Page 44: Accessing A Scenario

    Mediant 600 & Mediant 1000 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 45: Editing A Scenario

    SIP User's Manual 3. Web-Based Management In an opened Scenario Step (i.e., page appears in the Work pane), use the following navigation buttons: • Next: opens the next Step listed in the Scenario. • Previous: opens the previous Step listed in the Scenario. Note: If you reset the device while in Scenario mode, after the device resets, you are returned once again to the Scenario mode.

  • Page 46: Saving A Scenario To A Pc

    Mediant 600 & Mediant 1000 • Edit the Step Name: In the Navigation tree, select the required Step. In the 'Step Name' field, modify the Step name. In the page, click Next. • Edit the Scenario Name: In the 'Scenario Name' field, edit the Scenario name.

  • Page 47: Loading A Scenario To The Device

    SIP User's Manual 3. Web-Based Management Click the Get Scenario File button; the 'File Download' window appears. Click Save, and then in the 'Save As' window navigate to the folder to where you want to save the Scenario file. When the file is successfully downloaded to your PC, the 'Download Complete' window appears.

  • Page 48: Deleting A Scenario

    Mediant 600 & Mediant 1000 3.1.8.6 Deleting a Scenario You can delete the Scenario by using the Delete Scenario File button, as described in the procedure below: To delete the Scenario: On the Navigation bar, click the Scenarios tab; a message box appears, requesting...

  • Page 49: Exiting Scenario Mode

    SIP User's Manual 3. Web-Based Management 3.1.8.7 Exiting Scenario Mode When you want to close the Scenario mode after using it for device configuration, follow the procedure below: To close the Scenario mode: Simply click any tab (besides the Scenarios tab) on the Navigation bar, or click the Cancel Scenarios button located at the bottom of the Navigation tree;...

  • Page 50: Customizing The Web Interface

    The figure below shows an example of a customized Title bar. The top image displays the Title bar with AudioCodes logo and product name. The bottom image displays a customized Title bar with a different image logo and product name.

  • Page 51: Figure 3-23: Image Download Screen

    SIP User's Manual 3. Web-Based Management On the left pane, click Image Load to Device; the 'Image Download' page is displayed, as shown in the figure below: Figure 3-23: Image Download Screen Click the Browse button, and then navigate to the folder in which the logo image file is located.

  • Page 52: Customizing The Product Name

    The corporate logo can be replaced with a text string instead of an image. To replace AudioCodes’ default logo with a text string using the ini file, configure the ini file parameters listed in the table below. (For a description on using the ini file, refer to ''Modifying an ini File'' on page 221.)

  • Page 53: Creating A Login Welcome Message

    SIP User's Manual 3. Web-Based Management 3.1.9.3 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 54: 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 55: 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 56: Using The Home Page

    To access the Home page: On the toolbar, click the Home icon; the 'Home' page is displayed. Figure 3-28: Home Page – Mediant 600 Figure 3-29: Home Page – Mediant 1000 Note: The displayed number and type of modules, trunks and channels depends on the device's hardware configuration.
  • Page 57 SIP User's Manual 3. Web-Based Management Item # Description Module slot number (1 to 6). 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 58: Assigning A Port Name

    Mediant 600 & Mediant 1000 Item # Description Fan tray unit status icon: (green): Fan tray operating. (red): Fan tray failure. Power Supply Unit 1 status icon (applicable only to Mediant 1000): (green): Power supply is operating. (red): Power supply failure or no power supply unit installed.

  • Page 59: Viewing Analog Port Information

    SIP User's Manual 3. Web-Based Management To reset a channel: Click the required FXS or FXO port icon, and then from the shortcut menu, choose Reset Channel; the channel is changed to inactive (i.e., the port icon is displayed in grey).

  • Page 60: 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 61: Replacing Modules

    SIP User's Manual 3. Web-Based Management To view information of a specific trunk's channel, in the 'Trunks & Channels Status' page, click the required Channel icon; the 'Basic Channel Information' page appears: Figure 3-36: Basic Channel Information Page Click the buttons located above the 'Basic Channel Information' page to view additional parameters.

  • Page 62: Figure 3-35: Remove Module Button Appears After Clicking Module Name

    Mediant 600 & 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 Remove Module button appears: Figure 3-37: Remove Module Button Appears after Clicking Module Name Click the Remove Module button;...

  • Page 63: Configuration Tab

    SIP User's Manual 3. Web-Based Management Configuration Tab The Configuration tab on the Navigation bar displays menus in the Navigation tree related to device configuration. These menus include the following: Network Settings (refer to ''Network Settings'' on page 63) Media Settings (refer to ''Media Settings'' on page 74) PSTN Settings (refer to “PSTN Settings”...

  • Page 64: Configuring The Multiple Interface Table

    Mediant 600 & Mediant 1000 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...

  • Page 65: Figure 3-39: Ip Settings Page

    SIP User's Manual 3. Web-Based Management To configure the multiple IP interface table: Open the 'IP Settings' page (Configuration tab > Network Settings menu > IP Settings). Figure 3-41: IP Settings Page Under the 'Multiple Interface Settings' group, click the Multiple Interface Table button;...

  • Page 66: Table 3-7: Multiple Interface Table Parameters Description

    Mediant 600 & Mediant 1000 Table 3-7: 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 67 SIP User's Manual 3. Web-Based Management 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. 192.168.0.0/16 is synonymous with 192.168.0.0 and a subnet of 255.255.0.0.

  • Page 68: Configuring The Application Settings

    Mediant 600 & Mediant 1000 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 104).

  • Page 69: Figure 3-41: Application Settings Page

    SIP User's Manual 3. Web-Based Management To configure the Application settings: Open the 'Application Settings' page (Configuration tab > Network Settings menu > Application Settings page item). Figure 3-44: Application Settings Page Configure the parameters as required. For configuring NFS, under the 'NFS Settings' group, click the NFS Table button;...

  • Page 70: Configuring The Nfs Settings

    Mediant 600 & Mediant 1000 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 five different NFS file systems.

  • Page 71: Configuring The Ip Routing Table

    SIP User's Manual 3. Web-Based Management Parameter Description Root Path Path to the root of the remote file system in the format: /[path]. For example, '/audio'. NFS Version NFS version used to access the remote file system. [2] NFS Version 2. [3] NFS Version 3 (default).

  • Page 72: Figure 3-43: Ip Routing Table Page

    Mediant 600 & Mediant 1000 To configure static IP routing: Open the 'IP Routing Table' page (Configuration tab > Network Settings menu > IP Routing Table page item). Figure 3-46: 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 73: Configuring The Qos Settings

    SIP User's Manual 3. Web-Based Management Parameter Description Metric The maximum number of times a packet can be [RoutingTableHopsCountColumn] forwarded (hops) between the device and destination (typically, up to 20). Note: This parameter must be set to a number greater than 0 for the routing rule to be valid.

  • Page 74: Media Settings

    Mediant 600 & Mediant 1000 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 74) Fax/Modem/CID Settings (refer to ''Configuring the Fax/Modem/CID Settings'' on page...

  • Page 75: Figure 3-45: Voice Settings Page

    SIP User's Manual 3. Web-Based Management To configure the Voice parameters: Open the 'Voice Settings' page (Configuration tab > Media Settings menu > Voice Settings page item). Figure 3-48: Voice Settings Page Configure the Voice parameters as required. Click the Submit button to save your changes. To save the changes to flash memory, refer to ''Saving Configuration'' on page 193.

  • Page 76: Configuring The Fax/Modem/Cid Settings

    Mediant 600 & Mediant 1000 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 243.

  • Page 77: Figure 3-47: Rtp/Rtcp Settings Page

    SIP User's Manual 3. Web-Based Management To configure the RTP/RTCP parameters: Open the 'RTP/RTCP Settings' page (Configuration tab > Media Settings menu > RTP / RTCP Settings page item). Figure 3-50: RTP/RTCP Settings 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 193.

  • Page 78: Configuring The Ipmedia Settings

    Mediant 600 & Mediant 1000 3.3.2.4 Configuring the IPmedia Settings The 'IPMedia Settings' page allows you to configure the IP media parameters. This includes Automatic Gain Control (AGC) parameters. AGC equalizes the energy of the output signal to a required level. It estimates the energy of the incoming signal, calculates the essential gain and performs amplification.

  • Page 79: Configuring The General Media Settings

    SIP User's Manual 3. Web-Based Management 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 243. To configure general media parameters: Open the 'General Media Settings' page (Configuration tab >...

  • Page 80: Configuring Media Security

    Mediant 600 & Mediant 1000 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 243. To configure media security: Open the 'Media Security' page (Configuration tab >...

  • Page 81: Figure 3-52: Trunk Settings Page

    SIP User's Manual 3. Web-Based Management For a description of the trunk parameters, refer to ''PSTN Parameters'' on page 352. Notes: • During trunk deactivation, trunk configuration cannot be performed. • A stopped trunk cannot also be activated. To configure the trunks: Open the ‘Trunk Settings’...

  • Page 82: Figure 3-53: Trunk Scroll Bar

    Mediant 600 & Mediant 1000 Select the trunk that you want to configure, by clicking the desired Trunk number icon. The bar initially displays the first eight trunk number icons (i.e., trunks 1 through 8). To scroll through the trunk number icons (i.e., view the next/last or previous/first group of...

  • Page 83: Configuring The Cas State Machines

    SIP User's Manual 3. Web-Based Management Notes: • If the ‘Protocol Type’ field displays 'NONE' (i.e., no protocol type selected) and no other trunks have been configured, after selecting a PRI protocol type, you must reset the device. • The displayed parameters on the page depend on the protocol selected in the ‘Protocol Type’...

  • Page 84: Table 3-10: Cas State Machine Parameters Description

    Mediant 600 & Mediant 1000 Once you have completed the configuration, activate the trunk if required in the 'Trunk Settings' page, by clicking the trunk number in the 'Related Trunks' field, and in the 'Trunk Settings' page, select the required Trunk number icon, and then click Apply Trunk Settings.
  • Page 85 SIP User's Manual 3. Web-Based Management Parameter Description DTMF Min Detection Time Detects digit minimum on time (according to [CasStateMachineDTMFMinOnDetectionTime] DSP detection information event) in msec units. The digit time length must be longer than this value to receive a detection. Any number may be used, but the value must be less than CasStateMachineDTMFMaxOnDetectionTi...

  • Page 86: Security Settings

    Mediant 600 & Mediant 1000 3.3.4 Security Settings The Security Settings menu allows you to configure various security settings. This menu contains the following page items: Web User Accounts (refer to ''Configuring the Web User Accounts'' on page 86) WEB & Telnet Access List (refer to ''Configuring the Web and Telnet Access List'' on page 88) Firewall Settings (refer to “Configuring the Firewall Settings”...

  • Page 87: Table 3-12: Default Attributes For The Web User Accounts

    SIP User's Manual 3. Web-Based Management The default attributes for the two Web user accounts are shown in the following table: Table 3-12: Default Attributes for the Web User Accounts Account / Attribute User Name Password Access Level (Case-Sensitive) (Case-Sensitive) Primary Account Admin Admin...

  • Page 88: Configuring The Web And Telnet Access List

    Mediant 600 & Mediant 1000 To change the user name of an account, perform the following: In the field 'User Name', enter the new user name (maximum of 19 case-sensitive characters). Click Change User Name; if you are currently logged into the Web interface with this account, the 'Enter Network Password' dialog box appears, requesting you to enter the new user name.
  • Page 89 SIP User's Manual 3. Web-Based Management To add authorized IP addresses for Web and Telnet interfaces access: Open the 'Web & Telnet Access List' page (Configuration tab > Security Settings menu > Web & Telnet Access List page item). Figure 3-59: Web & Telnet Access List Page - Add New Entry 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;...

  • Page 90: Configuring The Firewall Settings

    Mediant 600 & Mediant 1000 3.3.4.3 Configuring the Firewall Settings The device provides an internal firewall, allowing you (the security administrator) to define network traffic filtering rules. You can add up to 50 ordered firewall rules. For each packet received on the network interface, the table is scanned from the top down until a matching rule is found.

  • Page 91: Table 3-13: Internal Firewall Parameters

    SIP User's Manual 3. Web-Based Management To activate a de-activated rule: In the 'Edit Rule' column, select the de-activated rule that you want to activate. Click the Activate button; the rule is activated. To de-activate an activated rule: In the 'Edit Rule' column, select the activated rule that you want to de-activate. Click the DeActivate button;...

  • Page 92: Configuring The Certificates

    Mediant 600 & Mediant 1000 Parameter Description Match Count A read-only field providing the number of packets accepted / rejected [AccessList_MatchCount] by the specific rule. 3.3.4.4 Configuring the Certificates The 'Certificates' page is used for the following: Replacing the server certificate (refer to ''Server Certificate Replacement'' on page 92)
  • Page 93 SIP User's Manual 3. Web-Based Management In the 'Subject Name' field, enter the DNS name, and then click Generate CSR. A textual certificate signing request that contains the SSL device identifier is displayed. Copy this text and send it to your security provider. The security provider (also known as Certification Authority or CA) signs this request and then sends you a server certificate for the device.

  • Page 94: Figure 3-63: Ike Table

    Mediant 600 & Mediant 1000 To apply the loaded certificate for IPsec negotiations: Open the ‘IKE Table’ page (refer to ''Configuring the IKE Table'' on page 99); the 'Loaded Certificates Files' group lists the newly uploaded certificates, as shown below: Figure 3-63: IKE Table Listing Loaded Certificate Files Click the Apply button to load the certificates;...
  • Page 95 SIP User's Manual 3. Web-Based Management When operation complete, file parameter HTTPSRequireClientCertificates to 1. Save the configuration (refer to ''Saving Configuration'' on page 193), and then restart the device. When a user connects to the secured Web server: If the user has a client certificate from a CA that is listed in the Trusted Root Certificate file, the connection is accepted and the user is prompted for the system password.

  • Page 96: 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 243.

  • Page 97: Configuring The Ipsec Table

    SIP User's Manual 3. Web-Based Management 3.3.4.6 Configuring the IPSec Table The 'IPSec Table' page allows you to configure the Security Policy Database (SPD) parameters for IP security (IPSec). Note: You can also configure the IPSec table using the ini file table parameter IPSEC_SPD_TABLE (refer to ''Security Parameters'' on page 267).

  • Page 98: Table 3-14: Default Ike Second Phase Proposals

    Mediant 600 & Mediant 1000 If no IPSec methods are defined (Encryption / Authentication), the default settings, shown in the following table are applied. Table 3-14: Default IKE Second Phase Proposals Proposal Encryption Authentication Proposal 0 3DES SHA1 Proposal 1...
  • Page 99 SIP User's Manual 3. Web-Based Management Parameter Name Description Protocol Defines the protocol type to which the [IPSecPolicyProtocol] IPSec mechanism is applied. 0 = Any protocol (default). 17 = UDP. 6 = TCP. Any other protocol type defined by IANA (Internet Assigned Numbers Authority).

  • Page 100: Configuring The Ike Table

    Mediant 600 & Mediant 1000 3.3.4.7 Configuring the IKE Table The 'IKE Table' page is used to configure the Internet Key Exchange (IKE) parameters. Note: You can also configure the IKE table using the ini file table parameter IPSec_IKEDB_Table (refer to ''Security Parameters'' on page 267).

  • Page 101: Table 3-16: Default Ike First Phase Proposals

    SIP User's Manual 3. Web-Based Management If no IKE methods are defined (Encryption / Authentication / DH Group), the default settings (shown in the following table) are applied. Table 3-16: Default IKE First Phase Proposals Proposal Encryption Authentication DH Group Proposal 0 3DES SHA1...
  • Page 102 Mediant 600 & Mediant 1000 Parameter Name Description [IKEPolicyLifeInSec] expires, the SA is re-negotiated. The default value is 28800 (i.e., 8 hours). IKE SA LifeTime (KB) Determines the lifetime (in kilobytes) that the SA negotiated [IKEPolicyLifeInKB] in the first IKE session (main mode) is valid. After this size is reached, the SA is re-negotiated.

  • Page 103: Protocol Configuration

    SIP User's Manual 3. Web-Based Management 3.3.5 Protocol Configuration The Protocol Configuration menu allows you to configure the device's SIP parameters and contains the following submenus: Applications Enabling (refer to “Enabling Applications” on page 103) Media Realms (refer to “Configuring Media Realms” on page 104) Protocol Definition (refer to ''Protocol Definition'' on page 105) Proxies/IpGroups/Registration (refer to ''Proxies, IP Groups, and Registration'' on page 108)

  • Page 104: Configuring Media Realms

    Mediant 600 & Mediant 1000 3.3.5.2 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 'Multiple Interface' table - refer to ''Configuring the Multiple Interface Table'' on page 63) into several realms, where each realm is specified by a UDP port range.
  • Page 105 SIP User's Manual 3. Web-Based Management Parameter Description IPv4 Name Associates the IPv4 interface to the Media Realm. The name [CpMediaRealm_IPv4IF] of this IPv4 interface must be exactly as configured in the 'Multiple Interface' table (InterfaceTable). IPv6 Name Not Applicable. [CpMediaRealm_IPv6IF] Port Range Start Defines the starting port for the range of Media interface...

  • Page 106: Protocol Definition

    Mediant 600 & Mediant 1000 3.3.5.3 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 106) DTMF & Dialing (refer to ''DTMF & Dialing Parameters'' on page 108) 3.3.5.3.1 Configuring SIP General Parameters...
  • Page 107 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 193. Version 5.8 September 2009...

  • Page 108: Proxies, Ip Groups, And Registration

    Mediant 600 & Mediant 1000 3.3.5.3.2 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 243.
  • Page 109 SIP User's Manual 3. Web-Based Management 3.3.5.4.1 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 ''Configuration Parameters Reference'' on page 243.
  • Page 110 Mediant 600 & Mediant 1000 3.3.5.4.2 Configuring the Proxy Sets Table The 'Proxy Sets Table' page allows you to define Proxy Sets. A Proxy Set is a group of Proxy servers defined by IP address or fully qualified domain name (FQDN). You can define up to six Proxy Sets, each having a unique ID number and each containing up to five Proxy server addresses.

  • Page 111: Table 3-19: Proxy Sets Table Parameters

    SIP User's Manual 3. Web-Based Management Click the Submit button to save your changes. To save the changes to flash memory, refer to ''Saving Configuration'' on page 193. Table 3-19: Proxy Sets Table Parameters Parameter Description Web: Proxy Set ID The Proxy Set identification number.
  • Page 112 Mediant 600 & Mediant 1000 Parameter Description Notes: If EnableProxyKeepAlive is set to 1 or 2, the device monitors the connection with the Proxies by using keep-alive messages (OPTIONS or REGISTER). To use Proxy Redundancy, you must specify one or more redundant Proxies.
  • Page 113 SIP User's Manual 3. Web-Based Management Parameter Description Web/EMS: Enable Proxy Keep Determines whether Keep-Alive with the Proxy is enabled or Alive disabled. This parameter is configured per Proxy Set. [EnableProxyKeepAlive] [0] Disable = Disable (default). [1] Using OPTIONS = Enables Keep-Alive with Proxy using OPTIONS.
  • Page 114 Mediant 600 & Mediant 1000 3.3.5.4.3 Configuring the IP Groups The 'IP Group Table' page allows you to create up to nine logical IP entities called IP Groups. These IP Groups are used for call routing. The IP Group can be used as a...

  • Page 115: Table 3-20: Ip Group Parameters

    SIP User's Manual 3. Web-Based Management Configure the IP group parameters according to the table below. Click the Submit button to save your changes. To save the changes to flash memory, refer to ''Saving Configuration'' on page 193. Table 3-20: IP Group Parameters Parameter Description Common Parameters...
  • 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 Routing Mode Defines the routing mode for outgoing SIP INVITE messages. [IPGroup_RoutingMode] [-1] Not Configured = The routing is according to the selected Serving IP Group. If no Serving IP Group is selected, the device routes the call according to the 'Outbound IP Routing' table (refer to “Configuring the Outbound IP Routing Table”...
  • Page 118 Mediant 600 & Mediant 1000 Parameter Description Enable Survivability Determines whether Survivability mode is enabled for USER- [IPGroup_EnableSurvivability] type IP Groups. Disable (default). Enable = Survivability mode is enabled. The device records in its local database the registration messages sent by the clients belonging to the USER-type IP Group.

  • Page 119: Table 3-21: Account Table Parameters Description

    SIP User's Manual 3. Web-Based Management Note: You can also configure the Account table using the ini file table parameter Account (refer to ''SIP Configuration Parameters'' on page 279). To configure Accounts: Open the 'Account Table' page (Configuration tab > Protocol Configuration menu > Proxies/IpGroups/Registration submenu >...
  • 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 114) to where the REGISTER requests (if enabled) are sent or Authentication is performed. The actual...
  • Page 121 'Trunk Group Settings' table for the specific Trunk Group. The Host Name (i.e., host name in SIP From/To headers) and Contact User (user in From/To and Contact headers) are taken from this table upon a successful registration. See the example below: REGISTER sip:audiocodes SIP/2.0 Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac1397582418 From: <sip:ContactUser@HostName>;tag=1c1397576231...

  • Page 122: Coders And Profile Definitions

    Mediant 600 & Mediant 1000 3.3.5.5 Coders and Profile Definitions The Coders And Profile Definitions submenu includes the following page items: Coders (refer to ''Configuring Coders'' on page 122) Coder Group Settings (refer to ''Configuring Coder Groups'' on page 124)
  • Page 123 SIP User's Manual 3. Web-Based Management 3.3.5.5.1 Configuring Coders The 'Coders' page allows you to configure up to five coders (and their attributes) for the device. The first coder in the list has the highest priority and is used by the device whenever possible.

  • Page 124: Configuring Coder Groups

    Mediant 600 & Mediant 1000 Click the Submit button to save your changes. To save the changes to flash memory, refer to ''Saving Configuration'' on page 193. Notes: • Each coder (i.e., 'Coder Name') can appear only once. • If packetization time and/or rate are not specified, the default value is applied.
  • Page 125 SIP User's Manual 3. Web-Based Management To configure coder groups: Open the 'Coder Group Settings' page (Configuration tab > Protocol Configuration menu > Coders And Profile Definition submenu > Coder Group Settings page item). Figure 3-76: Coder Group Settings Page From the 'Coder Group ID' drop-down list, select a coder group ID.
  • Page 126 Mediant 600 & Mediant 1000 3.3.5.5.3 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 127: Configuring Ip Profiles

    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 128 Mediant 600 & Mediant 1000 To configure the IP Profile settings: Open the 'IP Profile Settings' page (Configuration tab > Protocol Configuration menu > Coders And Profile Definition submenu > IP Profile Settings). Figure 3-78: IP Profile Settings Page From the 'Profile ID' drop-down list, select an identification number for the IP Profile.

  • Page 129: 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 130 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 3-79: Advanced Parameters Page Configure the parameters as required.
  • Page 131 SIP User's Manual 3. Web-Based Management 3.3.5.6.2 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 243. For an overview on supplementary services, refer to ''Working with Supplementary Services'' on page 508.
  • Page 132 Mediant 600 & Mediant 1000 Configure the parameters as required. Click the Submit button to save your changes, or click the Subscribe to MWI or Unsubscribe to MWI buttons to save your changes and to subscribe / unsubscribe to the MWI server.
  • Page 133 SIP User's Manual 3. Web-Based Management 3.3.5.6.4 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 'Tel to IP Routing' table.
  • Page 134 Mediant 600 & Mediant 1000 3.3.5.6.5 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 170) Caller ID Restriction (refer to ''Configuring Caller Display Information'' on page 168)

  • Page 135: Sas Parameters

    SIP User's Manual 3. Web-Based Management Configure the keypad features as required. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page 243. Click the Submit button to save your changes. To save the changes to the flash memory, refer to ''Saving Configuration'' on page 193.
  • Page 136 Mediant 600 & Mediant 1000 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 3-84: SAS Configuration Page Configure the parameters as described in ''SIP Configuration Parameters'' on page 279.

  • Page 137: Table 3-23: Sas Routing Table Parameters

    SIP User's Manual 3. Web-Based Management 3.3.5.7.2 Configuring the IP2IP Routing Table (SAS) The 'IP2IP Routing Table' page configures SAS routing when SAS is in Emergency mode. Up to 120 SAS routing rules can be defined. The device routes the SAS call (received SIP INVITE message) once a rule in this table is matched.
  • Page 138 Mediant 600 & Mediant 1000 Parameter Description Destination Username Prefix The prefix of the incoming SIP INVITE's destination URI [IP2IPRouting_DestUsernamePrefix] (usually the Request URI) user part. If this rule is not required, leave the field empty. To denote any prefix, use the asterisk (*) symbol.
  • Page 139 SIP User's Manual 3. Web-Based Management 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]. When using domain names, enter a DNS server IP address or alternatively, define these names in the 'Internal DNS Table' (refer to ''Configuring the Internal...

  • Page 140: Manipulation Tables

    Mediant 600 & Mediant 1000 3.3.5.8 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: Dest Number IP->Tel (refer to ''Configuring the Number Manipulation Tables'' on page 140) Dest Number Tel->IP (refer to ''Configuring the Number Manipulation Tables'' on page...
  • Page 141 SIP User's Manual 3. Web-Based Management The device matches manipulation rules starting at the top of the table. In other words, a rule at the top of the table takes precedence over a rule defined lower down in the table. Therefore, define more specific rules above more generic rules.

  • Page 142: Table 3-24: Number Manipulation Parameters Description

    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 143 SIP User's Manual 3. Web-Based Management Parameter Description Notes: The value -1 indicates that it is ignored in the rule. This parameter is available only in the 'Source Phone Number Manipulation Table for Tel -> IP Calls' and 'Destination Phone Number Manipulation Table for Tel ->...
  • Page 144 Mediant 600 & Mediant 1000 Parameter Description Web: NPI The Numbering Plan Indicator (NPI) assigned to this entry. EMS: Number Plan [0] Unknown (default) [_NumberPlan] [9] Private [1] E.164 Public [-1] Not Configured = value received from PSTN/IP is used...

  • Page 145: Table 3-25: Phone-Context Parameters Description

    SIP User's Manual 3. Web-Based Management For example, for a Tel-to-IP call with NPI/TON set as E164 National (values 1/2), the device sends the outgoing SIP INVITE URI with the following settings: “sip:12365432;phone- context= na.e.164.nt.com“. This is configured for entry 3 in the figure below. In the opposite direction (IP-to-Tel call), if the incoming INVITE contains this Phone-Context (e.g.

  • Page 146: Table 3-26: Npi/Ton Values For Isdn Etsi

    Mediant 600 & Mediant 1000 Parameter Description Select the Number Plan assigned to this entry. [0] Unknown = Unknown (default) [1] E.164 Public = E.164 Public [9] Private = Private For a detailed list of the available NPI/TON values, refer to “Numbering Plans and Type of Number”...

  • Page 147: Routing Tables

    SIP User's Manual 3. Web-Based Management Description Private [9] Unknown [0] A private number, but with no further information about the numbering plan. Level 2 Regional [1] Level 1 Regional [2] A private number with a location, e.g., 3932200. PISN Specific [3] Level 0 Regional (local) [4] A private local extension number, e.g., 2200.
  • Page 148 Mediant 600 & Mediant 1000 3.3.5.9.1 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 243. To configure the general routing parameters: Open the 'Routing General Parameters' page (Configuration tab >...
  • Page 149 SIP User's Manual 3. Web-Based Management This routing table associates called and/or calling telephone number prefixes (originating from a specific Trunk Group), with a destination IP address (or Fully Qualified Domain Name - FQDN) or IP Group. When a call is routed by the device (i.e., a Proxy server isn't used), the called and calling numbers are compared to the list of prefixes in this table.
  • Page 150 Mediant 600 & Mediant 1000 Alternative routing (using this table) is commonly implemented when there is no response to an INVITE message (after INVITE retransmissions). The device then issues an internal 408 'No Response' implicit release reason. If this reason is included in the 'Reasons for Alternative Routing' table, the device immediately initiates a call to the redundant destination using the next matched entry in the 'Tel to IP Routing' table.

  • Page 151: Table 3-27: Tel To Ip Routing Table Parameters

    SIP User's Manual 3. Web-Based Management Table 3-27: Tel to IP Routing Table Parameters Parameter Description Web/EMS: Tel to IP Routing Determines whether to route Tel (or inbound IP) calls to IP before or Mode after manipulation of destination number. [RouteModeTel2IP] [0] Route calls before manipulation = Calls are routed before the number manipulation rules are applied (default).

  • Page 152: Parameter Description

    Mediant 600 & Mediant 1000 Parameter Description [PREFIX_SourcePrefix] Note: The source phone prefix can be a single digit or a range of digits. For available notations representing multiple numbers/digits, refer to ''Dialing Plan Notation for Routing and Manipulation'' on page 459.
  • Page 153 SIP User's Manual 3. Web-Based Management Parameter Description Web: Dest IP Group ID The IP Group (1-9) to where you want to route the call. The SIP EMS: Destination IP Group INVITE messages are sent to the IP address defined for the Proxy Set ID (refer to ''Configuring the Proxy Sets Table'' on page 110) that [PREFIX_DestIPGroupID] is associated with this IP Group.

  • Page 154: Table 3-28: Outbound Ip Routing Table Description

    Mediant 600 & Mediant 1000 This table allows you to configure the device's routing rules for sending inbound IP calls matching some or all of the following criteria to a destination IP address or IP Group: Source IP Group Source host prefix...
  • Page 155 SIP User's Manual 3. Web-Based Management Parameter Description Group. In this scenario, this table is used only if the parameter PreferRouteTable is set to 1. For defining IP Groups, refer to ''Configuring the IP Groups'' on page 114. Src. Host Prefix The prefix of the SIP URI host name in the From header of the [PREFIX_SrcHostPrefix] incoming SIP INVITE message.
  • Page 156 Mediant 600 & Mediant 1000 Parameter Description [2] TLS Note: When 'Not Configured' is selected, the transport type defined by the parameter SIPTransportType (refer to ''SIP General Parameters'' on page 106) is used. Dest. IP Group ID The IP Group (1 to 9) to where you want to route the outbound IP-to- [PREFIX_DestIPGroupID] IP call.
  • Page 157 SIP User's Manual 3. Web-Based Management The IP-to-Tel calls are routed to Trunk Groups according to any one of the following (or a combination thereof) criteria: Destination and source host prefix Destination and source phone prefix Source IP address Once the call is routed to the specific Trunk Group, the call is sent to the device's channels pertaining to that Trunk Group.

  • Page 158: Table 3-29: Ip To Trunk Group Routing Table Description

    Mediant 600 & Mediant 1000 Table 3-29: IP to Trunk Group Routing Table Description Parameter Description IP to Tel Routing Mode Determines whether to route IP calls to the Trunk Group before or [RouteModeIP2Tel] after manipulation of destination number (configured in ''Configuring the Number Manipulation Tables'' on page 140).
  • Page 159 SIP User's Manual 3. Web-Based Management Parameter Description IP Profile ID The IP Profile (configured in ''Configuring P Profiles'' on page 127) [PstnPrefix_ProfileId] that is assigned to the routing rule. Source IP Group ID The source IP Group (1-9) associated with the incoming IP-to-Tel [PstnPrefix_SrcIPGroupID] call.

  • Page 160: Table 3-30: Inbound Ip Routing Table Description

    Mediant 600 & Mediant 1000 Configure the table according to the table below. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to ''Saving Configuration'' on page 193.
  • Page 161 SIP User's Manual 3. Web-Based Management Parameter Description Source IP Group ID The IP Group (1-9) to which you want to assign this inbound IP-to-IP [PstnPrefix_SrcIPGroupID] call. This defines the IP Group (configured in the ''Configuring the IP Groups'' on page 114) from where the SIP INVITE message is received.
  • Page 162 Mediant 600 & Mediant 1000 3.3.5.9.7 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 163 SIP User's Manual 3. Web-Based Management 3.3.5.9.8 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 562).
  • Page 164 Mediant 600 & Mediant 1000 3.3.5.9.9 Configuring Reasons for Alternative Routing The 'Reasons for Alternative Routing' page allows you to define up to four different call release (termination) reasons for IP-to-Tel call releases and for Tel-to-IP call releases. If a call is released as a result of one of these reasons, the device tries to find an alternative route for that call.
  • Page 165 SIP User's Manual 3. Web-Based Management To configure the reasons for alternative routing: Open the 'Reasons for Alternative Routing' page (Configuration tab > Protocol Configuration menu > Routing Tables submenu > Reasons for Alternative Routing page item). Figure 3-96: Reasons for Alternative Routing Page In the 'IP to Tel Reasons' group, select up to four different call failure reasons that invoke an alternative IP-to-Tel routing.

  • Page 166: Endpoint Settings

    Mediant 600 & Mediant 1000 3.3.5.10 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 166) Automatic Dialing (refer to ''Configuring Automatic Dialing'' on page 167)
  • Page 167 SIP User's Manual 3. Web-Based Management To configure the Authentication Table: Set the parameter 'Authentication Mode' to 'Per Endpoint' (refer to ''Configuring Proxy and Registration Parameters'' on page 108). Open the 'Authentication' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu >...
  • Page 168 Mediant 600 & Mediant 1000 To configure Automatic Dialing: Open the 'Automatic Dialing' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu > Automatic Dialing page item). Figure 3-98: Automatic Dialing Page In the 'Destination Phone Number' field corresponding to a port, enter the telephone number that you want automatically dialed.
  • Page 169 SIP User's Manual 3. Web-Based Management To configure the Caller Display Information: Open the 'Caller Display Information' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu > Caller Display Information page item). Figure 3-99: Caller Display Information Page In the' Caller ID/Name' field corresponding to the desired port, enter the Caller ID string (up to 18 characters).

  • Page 170: Table 3-31: Call Forward Table

    Mediant 600 & Mediant 1000 3.3.5.10.4 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 171 SIP User's Manual 3. Web-Based Management 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 'Tel to IP Routing' table (refer to ''Configuring the Tel to IP Routing Table'' on page 148).
  • Page 172 Mediant 600 & Mediant 1000 3.3.5.10.6 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 131).

  • Page 173: Trunk Groups

    SIP User's Manual 3. Web-Based Management 3.3.5.11 Trunk Groups The Trunk Group submenu allows you to configure groups of channels called Trunk Groups. This submenu includes the following page items: Trunk Group (refer to “Configuring the Trunk Group Table” on page 173) Trunk Group Settings (refer to ''Configuring the Trunk Group Settings'' on page 175) 3.3.5.11.1 Configuring the Trunk Group Table The 'Trunk Group Table' page allows you to enable the device's channels by assigning...

  • Page 174: Table 3-32: Trunk Group Table Parameters

    Mediant 600 & Mediant 1000 Table 3-32: Trunk Group Table Parameters Parameter Description Module The module type (i.e., FXS/FXO/BRI/PRI) for which you want [TrunkGroup_Module] to define the Trunk Group. From Trunk Starting physical Trunk number in the Trunk Group. The...
  • Page 175 SIP User's Manual 3. Web-Based Management Parameter Description Notes: Once you have defined a Trunk Group, you must configure the parameter PSTNPrefix (IP to Trunk Group Routing Table) to assign incoming IP calls to the appropriate Trunk Group. If you do not configure this, calls cannot be established.

  • Page 176: Table 3-33: Trunk Group Settings Parameters

    Mediant 600 & Mediant 1000 From the 'Routing Index' drop-down list, select the range of entries that you want to edit (up to 24 entries can be configured). Configure the Trunk Group according to the table below. Click the Submit button to save your changes.
  • Page 177 SIP User's Manual 3. Web-Based Management Parameter Description [4] Don't Register = No registrations are sent by endpoints pertaining to the Trunk Group. For example, if the device is configured globally to register all its endpoints (using the parameter ChannelSelectMode), you can exclude some endpoints from being registered by assigning them to a Trunk Group and configuring the Trunk Group registration mode to 'Don't Register'.

  • Page 178: Configuring Digital Gateway Parameters

    Mediant 600 & Mediant 1000 Parameter Description header contains the source party number. The 'ContactUser' parameter in the 'Account Table' page overrides this parameter. 3.3.5.12 Configuring Digital Gateway Parameters The 'Digital Gateway Parameters' page allows you to configure miscellaneous digital parameters.

  • Page 179: Configuring The Ipmedia Parameters

    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 193. 3.3.5.13 Configuring the IPmedia Parameters The 'IPmedia Parameters' page allows you to configure the IP media parameters. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page 243.

  • Page 180: Configuring Tdm Bus Settings

    Mediant 600 & Mediant 1000 3.3.6 Configuring TDM Bus Settings The 'TDM Bus Settings' page allows you to configure the device's Time-Division Multiplexing (TDM) bus settings. For detailed information on configuring the device's clock settings, refer to ''Clock Settings'' on page 561. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page 243.
  • Page 181 SIP User's Manual 3. Web-Based Management To configure the Voice Mail parameters: Open the 'Voice Mail Settings' page (Configuration tab > Advanced Applications menu > Voice Mail Settings page item). Figure 3-108: Voice Mail Settings Page Configure the parameters as required. Click the Submit button to save your changes.

  • Page 182: Configuring Radius Accounting Parameters

    Mediant 600 & Mediant 1000 3.3.7.2 Configuring RADIUS Accounting Parameters The 'RADIUS Parameters' page is used for configuring the Remote Authentication Dial In User Service (RADIUS) accounting parameters. For a description of these parameters, refer to ''Configuration Parameters Reference'' on page 243.
  • Page 183 SIP User's Manual 3. Web-Based Management To configure the FXO parameters: Open the 'FXO Settings' page (Configuration tab > Advanced Applications menu > FXO Settings page item). Figure 3-110: FXO Settings 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 193.

  • Page 184: Management Tab

    Mediant 600 & Mediant 1000 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 184) Software Update (refer to ''Software Update'' on page 194) 3.4.1...
  • Page 185 SIP User's Manual 3. Web-Based Management To configure the Management parameters: Open the 'Management Settings' page (Management tab > Management Configuration menu > Management Settings page item). Figure 3-111: Management Settings Page Configure the management settings. In addition, you can configure the following SNMP settings: •...

  • Page 186: Table 3-34: Snmp Trap Destinations Parameters Description

    Mediant 600 & Mediant 1000 3.4.1.1.1 Configuring the SNMP Trap Destinations Table The 'SNMP Trap Destinations' page allows you to configure up to five SNMP trap managers. To configure the SNMP Trap Destinations table: Access the 'Management Settings' page, as described in ''Configuring the Management Settings'' on page 184.
  • Page 187 SIP User's Manual 3. Web-Based Management Parameter Description Trap Enable Activates or de-activates the sending of traps to the [SNMPManagerTrapSendingEnable_x] corresponding SNMP Manager. [0] Disable = Sending is disabled. [1] Enable = Sending is enabled (default). 3.4.1.1.2 Configuring the SNMP Community Strings The 'SNMP Community String' page allows you to configure up to five read-only and up to five read-write SNMP community strings, and to configure the community string that is used for sending traps.

  • Page 188: Table 3-35: Snmp Community Strings Parameters Description

    Mediant 600 & Mediant 1000 Table 3-35: SNMP Community Strings Parameters Description Parameter Description Read Only [SNMPReadOnlyCommunityString_x]: Up to five Community String read-only community strings (up to 19 characters each). The default string is 'public'. Read / Write [SNMPReadWriteCommunityString_x]: Up to five read / write community strings (up to 19 characters each).

  • Page 189: Table 3-36: Snmp V3 Users Parameters

    SIP User's Manual 3. Web-Based Management Table 3-36: SNMP V3 Users Parameters Parameter Description Index The table index. [SNMPUsers_Index] The valid range is 0 to 9. User Name Name of the SNMP v3 user. This name must be unique. [SNMPUsers_Username] Authentication Protocol Authentication protocol of the SNMP v3 user.

  • Page 190: Configuring The Regional Settings

    Mediant 600 & Mediant 1000 In the 'SNMP Trusted Managers' field, click the right-pointing arrow button; the 'SNMP Trusted Managers' page appears. Figure 3-115: SNMP Trusted Managers Select the check box corresponding to the SNMP Trusted Manager that you want to enable and for whom you want to define an IP address.

  • Page 191: Maintenance Actions

    SIP User's Manual 3. Web-Based Management 3.4.1.3 Maintenance Actions The 'Maintenance Actions' page allows you to perform the following operations: Reset the device (refer to ''Resetting the Device'' on page 191) Lock and unlock the device (refer to ''Locking and Unlocking the Device'' on page 192) Save the configuration to the device's flash memory (refer to ''Saving Configuration'' on page 193) To access the 'Maintenance Actions' page:...
  • Page 192 Mediant 600 & Mediant 1000 Under the 'Reset Configuration' group, from the 'Graceful Option' drop-down list, select one of the following options: • 'Yes': Reset starts only after the user-defined time in the 'Shutdown Timeout' field (refer to Step 4) expires or after no more active traffic exists (the earliest thereof).
  • Page 193 SIP User's Manual 3. Web-Based Management 3.4.1.3.2 Locking and Unlocking the Device The Lock and Unlock options allow you to lock the device so that it doesn't accept any new incoming calls. This is useful when, for example, you are uploading new software files to the device and you don't want any traffic to interfere with the process.

  • Page 194: Software Update

    Mediant 600 & Mediant 1000 3.4.1.3.3 Saving Configuration The 'Maintenance Actions' page allows you to save (burn) the current parameter configuration (including loaded auxiliary files) to the device's non-volatile memory (i.e., flash). The parameter modifications that you make throughout the Web interface's pages are temporarily saved (to the volatile memory - RAM) when you click the Submit button on these pages.

  • Page 195: Loading Auxiliary Files

    SIP User's Manual 3. Web-Based Management 3.4.2.1 Loading Auxiliary Files The 'Load Auxiliary Files' page allows you to load various auxiliary files to the device. These auxiliary files are briefly described in the table below: Table 3-37: Auxiliary Files Descriptions File Type Description Provisions the device’s parameters.
  • Page 196 Mediant 600 & Mediant 1000 3.4.2.1.1 Loading Auxiliary Files using Web Interface The auxiliary files can be loaded to the device using the Web interface's 'Load Auxiliary Files' page, as described in the procedure below. To load an auxiliary file to the device using the Web interface: Open the 'Load Auxiliary Files' page (Management tab >...

  • Page 197: Loading A Software Upgrade Key

    You can load a Software Upgrade Key using one of the following management tools: Web interface BootP/TFTP configuration utility (refer to Loading via BootP/TFTP on page 200) AudioCodes’ EMS (refer to AudioCodes’ EMS User’s Manual or EMS Product Description) Version 5.8...
  • Page 198 • The Software Upgrade Key is an encrypted key. • The Software Upgrade Key is provided only by AudioCodes. To load a Software Upgrade Key: Open the 'Software Upgrade Key Status' page (Management tab > Software Update menu > Software Upgrade Key page item).
  • Page 199 Open the Software Upgrade Key file and check that the S/N line appears. If it does not appear, contact AudioCodes. Verify that you’ve loaded the correct file. Open the file and ensure that the first line displays [LicenseKeys].

  • Page 200: Software Upgrade Wizard

    3.4.2.2.1 Loading via BootP/TFTP The procedure below describes how to load a Software Upgrade Key to the device using AudioCodes' BootP/TFTP Server utility (for a detailed description on the BootP utility, refer to the Product Reference Manual). To load a Software Upgrade Key file using BootP/TFTP: Place the Software Upgrade Key file (typically, a *.txt file) in the same folder in which...
  • Page 201 SIP User's Manual 3. Web-Based Management Warnings: • To preserve all configuration settings, before upgrading the device to a new major software version (e.g., from version 5.6 to 5.8), save a copy of the device's configuration settings (i.e., ini file) to your PC and ensure that you have all the original auxiliary files (e.g., CPT file) currently used by the device.
  • Page 202 Mediant 600 & Mediant 1000 Click the Start Software Upgrade button; the 'Load a CMP file' Wizard page appears. Figure 3-124: Load a CMP File Wizard Page Note: At this stage, you can quit the Software Update Wizard, by clicking Cancel , without requiring a device reset.
  • Page 203 SIP User's Manual 3. Web-Based Management You can now choose to either: • Click Reset; the device resets, utilizing the new cmp and ini file you loaded up to now as well as utilizing the other auxiliary files. • Click Back; the 'Load a cmp file' page is opened again. •...

  • Page 204: 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 (Voice) 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 205: Status & Diagnostics Tab

    The 'Message Log' page displays Syslog debug messages sent by the device. You can select the Syslog messages in this page, and then copy and paste them into a text editor such as Notepad. This text file (txt) can then be sent to AudioCodes Technical Support for diagnosis and troubleshooting.

  • Page 206: Viewing Ethernet Port Information

    Mediant 600 & Mediant 1000 To activate the Message Log: Set the parameter 'Debug Level' (GwDebugLevel) to 6 (refer ''Configuring Advanced Parameter'' on page 129). This parameter determines the Syslog logging level in the range 0 to 6, where 6 is the highest level.

  • Page 207: 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 3-128: Ethernet Port Information Page Table 3-38: Ethernet Port Information Parameters Parameter Description Displays the active Ethernet port (1 or 2).

  • Page 208: 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. This page also displays any loaded files used by the device (stored in the RAM) and allows you to remove them.

  • Page 209: Viewing Active Alarms

    SIP User's Manual 3. Web-Based Management Diagnostics menu > Performance Statistics page item). Figure 3-131: Performance Statistics Page To reset the performance statistics to zero: Click the Reset Statistics button. 3.5.1.6 Viewing Active Alarms The 'Active Alarms' page displays a list of currently active alarms. You can also access this page from the 'Home' page (refer to ''Using the Home Page'' on page 56).

  • Page 210: 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 211: Table 3-39: Call Counters Description

    SIP User's Manual 3. Web-Based Management Table 3-39: 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 212: Viewing Call Routing Status

    Mediant 600 & Mediant 1000 Counter Description Number of Failed Calls Indicates the number of calls that failed due to mismatched device due to No Matched capabilities. It is incremented as a result of an internal identification of Capabilities capability mismatch. This mismatch is reflected to CDR via the value of...

  • Page 213: Viewing Registration Status

    SIP User's Manual 3. Web-Based Management Parameter Description Not Used = Proxy server isn't defined. Current Proxy IP address and FQDN (if exists) of the Proxy server with which the device currently operates. N/A = Proxy server isn't defined. Current Proxy State OK = Communication with the Proxy server is in order.

  • Page 214: Viewing Sas/Sbc Registered Users

    Mediant 600 & Mediant 1000 3.5.2.4 Viewing SAS/SBC Registered Users The 'SAS Registered Users' page displays a list of Stand Alone Survivability (SAS) and/or IP-to-IP registered users. To view the SAS registered users: Open the 'SAS Registered Users' page (Status & Diagnostics tab > Gateway Statistics menu >...

  • Page 215: 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 'Tel to IP Routing' page (refer to ''Configuring the Tel to IP Routing Table'' on page 148). Notes: •...
  • Page 216 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 217: Ini File Configuration

    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. To overcome this security threat, the AudioCodes' TrunkPack Downloadable Conversion Utility (DConvert) allows you to binary-encode the ini file before loading it to the device (refer to the Product Reference Manual).

  • Page 218: Ini File Format

    Mediant 600 & Mediant 1000 INI File Format The ini file can include any number of parameters and of the following types: Individual parameters, which can be conveniently grouped (optional) by their functionality (refer to ''Structure of Individual ini File Parameters'' on page 218)

  • Page 219: Format Of Ini File Table Parameters

    SIP User's Manual 4. INI File Configuration An example of an ini file containing individual ini file parameters is shown below: [System Parameters] SyslogServerIP = 10.13.2.69 EnableSyslog = 1 ; these are a few of the system-related parameters. [Web Parameters] LogoWidth = '339' WebLogoText = 'My Device' UseWeblogo = 1...
  • Page 220 Mediant 600 & Mediant 1000 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 221: Example Of An Ini File

    SIP User's Manual 4. INI File Configuration The table below displays an example of an ini file table parameter: [ PREFIX ] FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort; PREFIX 0 = 10, 10.13.83.5, *, 0, 255, 0; PREFIX 1 = 20, 10.13.83.7, *, 0, 255, 0;...

  • Page 222: Modifying An Ini File

    Mediant 600 & Mediant 1000 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 223: Element Management System (Ems)

    5. Element Management System (EMS) Element Management System (EMS) This section describes how to configure 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'...

  • Page 224: Securing Ems-Device Communication

    Mediant 600 & Mediant 1000 The MG Tree is a hierarchical tree-like structure that lists all the devices managed by EMS. The tree includes the following icons: Globe : highest level in the tree, from which a Region can be added.

  • Page 225: Changing Ssh Login Password

    SIP User's Manual 5. Element Management System (EMS) The configuration session is now active and all data entered at the terminal is parsed as configuration text (formatted as an ini file). Type the following at the configuration session: [ IPSEC_IKEDB_TABLE ] FORMAT IKE_DB_INDEX = IkePolicySharedKey;...

  • Page 226: Adding The Device In Ems

    Mediant 600 & Mediant 1000 The device responds with the message “Password changed”. Close the SSH client session and reconnect using the new password. Note: The default user name ("Admin") cannot be changed from within an SSH client session. Adding the Device in EMS...

  • Page 227: Figure 5-3: Adding A Region

    SIP User's Manual 5. Element Management System (EMS) Add a Region for your deployed devices, by performing the following: 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).

  • Page 228: Configuring Trunks

    Mediant 600 & Mediant 1000 Note: The Pre-shared Key string defined in the EMS must be identical to the one that you defined for the device. When IPSec is enabled, default IPSec/IKE parameters are loaded to the device. Configuring Trunks This section describes how to provision the E1/T1 PSTN Trunks.

  • Page 229: Configuring Isdn Nfas

    SIP User's Manual 5. Element Management System (EMS) From the 'Clock Master' drop-down list, set the Clock Master to one of the following values: • Clock Master OFF: the Clock Source is recovered from the Trunk line. • Clock Master ON: the Clock Source is provided by the internal TDM bus clock source, according to the parameter TDM Bus Clock Source.

  • Page 230: Figure 5-6: Ems Isdn Settings Screen

    Mediant 600 & Mediant 1000 Select the ISDN Settings tab; the 'ISDN Settings' screen appears. Figure 5-6: EMS ISDN Settings Screen Perform the following configurations: Configure each trunk in the group with the same values for the 'Termination Side' parameter.

  • Page 231: Figure 5-7: General Settings Window

    SIP User's Manual 5. Element Management System (EMS) • Line Code Figure 5-7: General Settings Window Burn and reset the device after all the trunks have been configured. Note: All trunks in the group must be configured with the same values for trunk parameters TerminationSide, ProtocolType, FramingMethod and LineCode.

  • Page 232: 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.

  • Page 233: 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 234: Provisioning Srtp Crypto Offered Suites

    Mediant 600 & Mediant 1000 Select the button to add a new entry, and then click Yes at the confirmation prompt; a row is added to the table. Figure 5-9: IPSec Table Screen Double-click each field to insert required values.

  • Page 235: Provisioning Mlpp Parameters

    SIP User's Manual 5. Element Management System (EMS) Select the Authentication & Security tab; the 'Authentication & Security' screen is displayed. Figure 5-10: Authentication & Security Screen From the 'SRTP Offered Suites' (SRTPofferedSuites) drop-down list, select one of the following: •...

  • Page 236: Configuring The Device To Operate With Snmpv3

    Mediant 600 & Mediant 1000 Figure 5-11: 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 237: 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 238: Configuring Ems To Operate With A Pre-Configured Snmpv3 System

    Mediant 600 & Mediant 1000 To end the PuTTY configuration session, type a full-stop (“.”) on an empty line; the device responds with the following: INI File replaced To save the configuration to the non-volatile memory, type sar; the device reboots with IPSec enabled.

  • Page 239: 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, and then right-click the device.

  • Page 240: 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 241: 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; the 'Software Manager' screen appears.

  • Page 242: Figure 5-17: 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 then navigate to and select the required cmp file; the software version number of the selected file appears in the 'Software Version' field.

  • Page 243: Configuration Parameters Reference

    SIP User's Manual 6. Configuration Parameters Reference Configuration Parameters Reference The device's configuration parameters, default values, and their description are documented in this section. Parameters and values enclosed in square brackets ([...]) represent ini file parameters and their enumeration values; parameters not written in square brackets represent their corresponding Web interface and/or EMS parameters.

  • Page 244: Multiple Ip Interfaces And Vlans Parameters

    Mediant 600 & Mediant 1000 Parameter Description Note: For this parameter to take effect, a device reset is required. Enable LAN Watchdog is relevant only if the Ethernet connection is full duplex. [MIIRedundancyEnable] Enables the Ethernet Interface Redundancy feature. When enabled, the device performs a switchover to the secondary (redundant) Ethernet port upon sensing a link failure in the primary Ethernet port.
  • Page 245 SIP User's Manual 6. Configuration Parameters Reference Parameter Description is required. Each interface index must be unique. Each IP interface must have a unique subnet. Subnets in different interfaces must not be overlapping in any way (e.g., defining two interfaces with 10.0.0.1/8 and 10.50.10.1/24 is invalid).
  • Page 246 Mediant 600 & Mediant 1000 Parameter Description For this parameter to take effect, a device reset is required. VLANs are available only when booting the device from flash. When booting using BootP/DHCP protocols, VLANs are disabled to allow easier maintenance access. In this scenario, multiple network interface capabilities are not available.

  • Page 247: Static Routing Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description [VLANSendNonTaggedOnNative] Specify whether to send non-tagged packets on the native VLAN. [0] = Sends priority tag packets (default). [1] = Sends regular packets (with no VLAN tag). Note: For this parameter to take effect, a device reset is required.

  • Page 248: Quality Of Service Parameters

    Mediant 600 & Mediant 1000 Parameter Description Web: Metric The maximum number of times a packet can be EMS: Primary Routing Metric forwarded (hops) between the device and destination [RoutingTableHopsCountColumn] (typically, up to 20). Notes: For this parameter to take effect, a device reset is required.
  • Page 249 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Media Premium Defines the VLAN priority (IEEE 802.1p) for the EMS: Premium Service Class Media Priority Premium CoS content and media traffic. Priority The valid range is 0 to 7. The default value is 6. [VLANPremiumServiceClassMediaPriority] Web: Control Premium Priority Defines the VLAN priority (IEEE 802.1p) for the...

  • Page 250: 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 251: Nfs Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: NAT IP Address Global (public) IP address of the device to enable static Network EMS: Static NAT IP Address Address Translation (NAT) between the device and the Internet. [StaticNatIP] Note: For this parameter to take effect, a device reset is required. EMS: Disable NAT Enables / disables the Network Address Translation (NAT) [DisableNAT]...

  • Page 252: Dns Parameters

    Mediant 600 & Mediant 1000 Parameter Description Web: NFS Table This ini file table parameter defines up to five Network File Systems EMS: NFS Settings (NFS) file systems so that the device can access a remote server's [NFSServers] shared files and directories for loading cmp, ini, and auxiliary files (using the Automatic Update mechanism).
  • Page 253 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: DNS Secondary The IP address of the second DNS server. Enter the IP address in Server IP dotted-decimal notation, for example, 10.8.2.255. EMS: DNS Secondary Note: For this parameter to take effect, a device reset is required. Server [DNSSecServerIP] Internal DNS Table...

  • Page 254: Dhcp Parameters

    Mediant 600 & Mediant 1000 Parameter Description description of the parameters in this ini file table parameter, refer to ''Configuring the Internal SRV Table'' on page 162. For an explanation on using ini file table parameters, refer to ''Format of ini File Table Parameters'' on page 219.

  • Page 255: Ntp And Daylight Saving Time Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description [1] = Enable This parameter is applicable only if DHCPEnable is set to 0 for cases where booting up the device using DHCP is not desirable, but renewing DHCP leasing is. When the device is powered up, it attempts to communicate with a BootP server.

  • Page 256: Syslog Parameters

    Mediant 600 & Mediant 1000 6.1.10 Syslog Parameters The Syslog parameters are described in the table below. Table 6-10: Syslog Parameters Parameter Description Web/EMS: Syslog Server IP address (in dotted-decimal notation) of the computer you are using to IP Address run the Syslog server.

  • Page 257: Web And Telnet Parameters

    SIP User's Manual 6. Configuration Parameters Reference Web and Telnet Parameters This subsection describes the device's Web and Telnet parameters. 6.2.1 General Parameters The general Web and Telnet parameters are described in the table below. Table 6-11: General Web and Telnet Parameters Parameter Description Web: Web and Telnet...

  • Page 258: Web Parameters

    Mediant 600 & Mediant 1000 6.2.2 Web Parameters The Web parameters are described in the table below. Table 6-12: Web Parameters Parameter Description [DisableWebTask] Disables or enables device management through the Web interface. [0] = Enable Web management (default). [1] = Disable Web management.
  • Page 259 Determines whether the Web interface displays a logo image or text. [0] = Logo image is used (default). [1] = Text string is used instead of a logo image. AudioCodes' default logo (or any other logo defined by the LogoFileName parameter) is replaced with a text string defined by the WebLogoText parameter.

  • Page 260: Telnet Parameters

    Name of the image file that you want to load to the device for being displayed in the Web GUI's title bar (instead of AudioCodes' logo). The file name can be up to 47 characters. The default is AudioCodes’ logo file.

  • Page 261: Debugging And Diagnostics Parameters

    SIP User's Manual 6. Configuration Parameters Reference Debugging and Diagnostics Parameters This subsection describes the device's debugging and diagnostic parameters. 6.3.1 General Parameters The general debugging and diagnostic parameters are described in the table below. Table 6-14: General Debugging and Diagnostic Parameters Parameter Description EMS: Enable Diagnostics...

  • Page 262: Cdr And Debug Parameters

    Mediant 600 & Mediant 1000 Parameter Description Notes: For this parameter to take effect, a device reset is required. This parameter is only applicable to FXS interfaces. To enable Lifeline switching on network failure, LAN watch dog must be activated (EnableLANWatchDog set to 1).
  • Page 263 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Debug Level Syslog debug logging level. [GwDebugLevel] [0] 0 = Debug is disabled (default). [1] 1 = Flow debugging is enabled. [2] 2 = Flow and device interface debugging are enabled. [3] 3 = Flow, device interface, and stack interface debugging are enabled.

  • Page 264: Heartbeat Packet Parameters

    Mediant 600 & Mediant 1000 6.3.3 Heartbeat Packet Parameters The Heartbeat packet parameters are described in the table below. The device sends a heartbeat packet to ensure that the far-end is passing traffic. Table 6-16: Heartbeat Packet Parameters Parameter Description...

  • Page 265: Serial Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.3.5 Serial Parameters The RS-232 serial parameters are described in the table below. (Serial interface is mainly used for debugging.) Table 6-18: Serial Parameters Parameter Description [DisableRS232] Enables or disables the device's RS-232 port. [0] = RS-232 serial port is enabled (default).

  • Page 266: Bootp Parameters

    Mediant 600 & Mediant 1000 6.3.6 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 267: Security Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description [0] = Disable (default). [ExtBootPReqEnable] [1] = 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 268: 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 269 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: HTTPS Port Determines the local Secured HTTPS port of the device. [HTTPSPort] The valid range is 1 to 65535 (other restrictions may apply within this range). The default port is 443. Note: For this parameter to take effect, a device reset is required.

  • Page 270: 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 6-22: SRTP Parameters Parameter Description Web: Media Security Enables Secure Real-Time Transport Protocol (SRTP). EMS: Enable Media Security [0] Disable = SRTP is disabled (default).

  • Page 271: Tls Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.4.4 TLS Parameters The Transport Layer Security (TLS) parameters are described in the table below. Table 6-23: TLS Parameters Parameter Description Web/EMS: TLS Version Defines the supported versions of SSL/TLS (Secure Socket [TLSVersion] Layer/Transport Layer Security.

  • Page 272: Ssh Parameters

    Mediant 600 & Mediant 1000 Parameter Description domain name. If the SubjectAltName is not marked as ‘critical’ and there is no match, the CN value of the SubjectName field is compared with the parameter TLSRemoteSubjectName. If a match is found, the connection is established.

  • Page 273: Ipsec Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.4.6 IPSec Parameters The Internet Protocol security (IPSec) parameters are described in the table below. Table 6-25: IPSec Parameters Parameter Description IPSec Parameters Web: Enable IP Security Enables or disables IPSec on the device. EMS: IPSec Enable [0] Disable = IPSec is disabled (default).

  • Page 274: Ocsp Parameters

    Mediant 600 & Mediant 1000 Parameter Description the Web interface, refer to ''Configuring the IPSec Table'' on page 97. For an explanation on using ini file table parameters, refer to ''Format of ini File Table Parameters'' on page 219. Internet Key Exchange (IKE) Parameters Web/EMS: IKE Table This ini file table parameter configures the IKE table.

  • Page 275: Radius Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: OCSP Server Port Defines the OCSP server's TCP port number. [OCSPServerPort] The default port number is 2560. EMS: OCSP Default Determines the default OCSP behavior when the server cannot be Response contacted.
  • Page 276 Mediant 600 & Mediant 1000 Parameter Description [RadiusTO] Determines the time interval (measured in seconds) the device waits for a response before a RADIUS retransmission is issued. The valid range is 1 to 30. The default value is 10. Web: RADIUS Authentication IP address of the RADIUS authentication server.

  • Page 277: Snmp Parameters

    SIP User's Manual 6. Configuration Parameters Reference SNMP Parameters The SNMP parameters are described in the table below. Table 6-28: SNMP Parameters Parameter Description Web: Enable SNMP Determines whether SNMP is enabled. [DisableSNMP] [0] Enable = SNMP is enabled (default). [1] Disable = SNMP is disabled and no traps are sent.
  • Page 278 Mediant 600 & Mediant 1000 Parameter Description [SNMPSysOid] Defines the base product system OID. The default is eSNMP_AC_PRODUCT_BASE_OID_D. Note: For this parameter to take effect, a device reset is required. [SNMPTrapEnterpriseOid] Defines a Trap Enterprise OID. The default is eSNMP_AC_ENTERPRISE_OID.
  • Page 279 SIP User's Manual 6. Configuration Parameters Reference Parameter Description SNMP Community String Parameters Community String Defines up to five read-only SNMP community strings (up to 19 [SNMPReadOnlyCommuni characters each). The default string is 'public'. tyString_x] Community String Defines up to five read / write SNMP community strings (up to 19 [SNMPReadWriteCommun characters each).

  • Page 280: 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 6-29: General SIP Parameters Parameter Description Web/EMS: PRACK Mode PRACK (Provisional Acknowledgment) mechanism mode for 1xx SIP reliable responses.
  • Page 281 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Minimum Session- Defines the time (in seconds) that is used in the Min-SE header. This Expires header defines the minimum time that the user agent refreshes the EMS: Minimal Session session.
  • Page 282 Mediant 600 & Mediant 1000 Parameter Description [3] Fax Fallback = Initiates T.38 fax relay. If the T.38 negotiation fails, the device re-initiates a fax session using the coder G.711 A-law/μ-law with adaptations (refer to the Note below). Notes: Fax adaptations (for options 2 and 3):...
  • Page 283 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Enable SIPS Enables secured SIP (SIPS URI) connections over multiple hops. [EnableSIPS] [0] Disable (default). [1] Enable. When 'SIP Transport Type' is set to TLS (SIPTransportType = 2) and 'Enable SIPS' is disabled, TLS is used for the next network hop only. When 'SIP Transport Type' is set to TCP or TLS (SIPTransportType = 2 or 1) and 'Enable SIPS' is enabled, TLS is used through the entire connection (over multiple hops).
  • Page 284 Mediant 600 & Mediant 1000 Parameter Description Web: Enable History-Info Enables usage of the History-Info header. Header [0] Disable = Disable (default) EMS: Enable History Info [1] Enable = Enable [EnableHistoryInfo] User Agent Client (UAC) Behavior: Initial request: The History-Info header is equal to the Request URI.
  • Page 285 SIP User's Manual 6. Configuration Parameters Reference Parameter Description call, the 'tgrp' parameter isn't included. If a 'tgrp' value is specified in incoming messages, it is ignored. [2] Send and Receive = The functionality of outgoing SIP messages is identical to the functionality described in option (1). In addition, for incoming SIP messages, if the Request-URI includes a 'tgrp' parameter, the device routes the call according to that value (if possible).
  • Page 286 EMS: User Agent Display Info and SIP response header Server. If not configured, the default string [UserAgentDisplayInfo] 'AudioCodes product-name s/w-version' is used (e.g., User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40.010.006). When configured, the string 'UserAgentDisplayInfo s/w-version' is used (e.g., User-Agent: MyNewOEM/v.5.40.010.006). Note that the version number can't be modified.
  • Page 287 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Redirecting Reason >> SIP Response Code Unknown >> 404 User busy >> 486 No reply >> 408 Deflection >> 487/480 Unconditional >> 302 Others >> 302 If the device receives a Request-URI that includes a 'target' and 'cause' parameter, the 'target' is mapped to the Redirect phone number and the 'cause' is mapped to Redirect number reason.
  • Page 288 Mediant 600 & Mediant 1000 Parameter Description Web: Reliable Connection Determines whether all TCP/TLS connections are set as persistent Persistent Mode and therefore, not released. [ReliableConnectionPersist [0] = Disable (default) - all TCP connections (except those that entMode] are set to a proxy IP) are released if not used by any SIP dialog\transaction.
  • Page 289 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [EnableDelayedOffer] Determines whether the device sends the initial INVITE message with or without an SDP. Sending the first INVITE without SDP is typically done by clients for obtaining the far-end's full list of capabilities before sending their own offer.
  • Page 290 Mediant 600 & Mediant 1000 Parameter Description [EnableSilenceSuppInSDP] Determines the device's behavior upon receipt of SIP Re-INVITE messages that include the silencesupp:off attribute. [0] = Disregard the silecesupp attribute (default). [1] = Handle incoming Re-INVITE messages that include the silencesupp:off attribute in the SDP as a request to switch to the Voice-Band-Data (VBD) mode.
  • Page 291 SIP User's Manual 6. Configuration Parameters Reference Parameter Description that is received in ISDN Proceeding, Progress, and Alert messages is used as described in the options below. (default) [0] No PI = For IP-to-Tel calls, the device sends 180 Ringing SIP response to IP after receiving ISDN Alert or (for CAS) after placing a call to PBX/PSTN.
  • Page 292 Mediant 600 & Mediant 1000 Parameter Description is translated to corresponding SIP 40x or 50x values (e.g., 3 to SIP 404, and 34 to SIP 503). For analog interfaces: For an explanation on mapping PSTN release causes to SIP responses, refer to Mapping PSTN Release Cause to SIP Response on page 503.
  • Page 293 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: Use SIP URI For Sets the URI format in the SIP Diversion header. Diversion Header [0] = 'tel:' (default) [UseSIPURIForDiversionHe [1] = 'sip:' ader] [TimeoutBetween100And18 Defines the timeout (in msec) between receiving a 100 Trying response and a subsequent 18x response.
  • Page 294 Mediant 600 & Mediant 1000 Parameter Description message for T1 PRI trunks supporting these messages (NI-2, 4/5- ESS, DMS-100, and Meridian). These behaviors are performed upon one of the following scenarios: Physically disconnected from the network (i.e., Ethernet cable is disconnected).
  • Page 295 SIP User's Manual 6. Configuration Parameters Reference Parameter Description DTMF Parameters Web: Declare RFC 2833 in Defines the supported Receive DTMF negotiation method. [0] No = Don't declare RFC 2833 telephony-event parameter in EMS: Rx DTMF Option SDP. [RxDTMFOption] [3] Yes = Declare RFC 2833 telephony-event parameter in SDP (default).
  • Page 296 Mediant 600 & Mediant 1000 Parameter Description Tx DTMF Option Table This ini file table parameter determines up to two preferred transmit [TxDTMFOption] DTMF negotiation methods. The format of this parameter is as follows: [TxDTMFOption] FORMAT TxDTMFOption_Index = TxDTMFOption_Type; [\TxDTMFOption] For example: TxDTMFOption 0 = 1;...
  • Page 297 SIP User's Manual 6. Configuration Parameters Reference Parameter Description detection of dial tone before it starts playing DTMF digits. For example, if the called number is '1007766p100', the device places a call with 1007766 as the destination number, then after the call is answered, it waits 1.5 seconds ('p') and plays the rest of the number (100) as DTMF digits.
  • Page 298 Mediant 600 & Mediant 1000 Parameter Description Web: SIP Media Realm Table EMS: Media Realm [CpMediaRealm] This ini file table parameter configures the SIP Media Realm table. The Media Realm table allows you to divide a Media-type interface (defined in the 'Multiple Interface' table) into several realms, where each realm is specified by a UDP port range.
  • Page 299 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: IP Group Table EMS: IP Group This ini file table parameter configures the IP Group table. The [IPGroup] format of this parameter is as follows: [IPGroup] FORMAT IPGroup_Index = IPGroup_Type, IPGroup_Description, IPGroup_ProxySetId, IPGroup_SIPGroupName, IPGroup_ContactUser, IPGroup_EnableSurvivability, IPGroup_ServingIPGroup, IPGroup_SipReRoutingMode,...

  • Page 300: Proxy, Registration And Authentication Parameters

    Mediant 600 & Mediant 1000 6.7.2 Proxy, Registration and Authentication Parameters The proxy server, registration and authentication SIP parameters are described in the table below. Table 6-30: Proxy, Registration and Authentication SIP Parameters Parameter Description Web: Authentication Table EMS: SIP Endpoints > Authentication...
  • Page 301 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Account Table EMS: Account This ini file table parameter configures the Account table for [Account] 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 302 Mediant 600 & Mediant 1000 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] [0] 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 303 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [2] Routing Table = Uses the Routing table to locate the destination and then sends a new INVITE to this destination. Notes: When this parameter is set to [1] and the INVITE sent to the Proxy fails, the device re-routes the call according to the Standard mode [0].
  • Page 304 Mediant 600 & Mediant 1000 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. [0] A-Record = A-Record (default) [1] SRV = SRV...
  • Page 305 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Use Gateway Name for Determines whether the device uses its IP address or gateway OPTIONS name in keep-alive SIP OPTIONS messages. [UseGatewayNameForOptions] [0] No = Use the device's IP address in keep-alive OPTIONS messages (default).
  • Page 306 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Challenge Caching Determines the mode for Challenge Caching, which reduces Mode the number of SIP messages transmitted through the network. [SIPChallengeCachingMode] The first request to the Proxy is sent without authorization. The Proxy sends a 401/407 response with a challenge.
  • Page 307 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Proxy Set Table EMS: Proxy Set This ini file table parameter configures the Proxy Set ID table. [ProxySet] It is used in conjunction with the ini file table parameter ProxyIP, which defines the Proxy Set IDs with their IP addresses.
  • Page 308 Mediant 600 & Mediant 1000 Parameter Description Web: Registrar IP Address The IP address (or FQDN) and optionally, port number of the EMS: Registrar IP SIP Registrar server. The IP address is in dotted-decimal [RegistrarIP] notation, e.g., 201.10.8.1:<5080>. Notes: If not specified, the REGISTER request is sent to the primary Proxy server.
  • Page 309 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Registration Retry Time Defines the time interval (in seconds) after which a [RegistrationRetryTime] Registration request is resent if registration fails with a 4xx response or if there is no response from the Proxy/Registrar server.

  • Page 310: Profile Parameters

    Mediant 600 & Mediant 1000 Parameter Description Web: Set Out-Of-Service On Enables setting an endpoint, trunk, or the entire device (i.e., all Registration Failure endpoints) to out-of-service if registration fails. [OOSOnRegistrationFail] [0] Disable = Disabled (default). [1] Enable = Enabled.
  • Page 311 SIP User's Manual 6. Configuration Parameters Reference Parameter Description The table below lists the supported coders: Coder Name Packetization Rate Payload Silence Time (msec) (kbps) Type Suppression G.711 A-law 10, 20 Always 64 Always 8 Disable [0] [g711Alaw64k] (default), 30, Enable [1] 40, 50, 60, 80, 100, 120...
  • Page 312 Mediant 600 & Mediant 1000 Parameter Description Notes: This parameter can include up to 25 indices (i.e., five coders per five coder groups), where indices 0 through 4 is the default coder group. The coder name is case-sensitive. Each coder type can appear only once per Coder Group.
  • Page 313 SIP User's Manual 6. Configuration Parameters Reference Parameter Description IPProfile_SigIPDiffServ*, IpProfile_SCE, IPProfile_RTPRedundancyDepth, IPProfile_RemoteBaseUDPPort, IPProfile_CNGmode, IPProfile_VxxTransportType, IPProfile_NSEMode, IpProfile_IsDTMFUsed, IPProfile_PlayRBTone2IP, IPProfile_EnableEarlyMedia*, IPProfile_ProgressIndicator2IP*, IPProfile_EnableEchoCanceller*, IPProfile_CopyDest2RedirectNumber, IPProfile_MediaSecurityBehaviour, IPProfile_CallLimit, IPProfile_ DisconnectOnBrokenConnection, IPProfile_FirstTxDtmfOption, IPProfile_SecondTxDtmfOption, IPProfile_RxDTMFOption, IpProfile_EnableHold, IpProfile_InputGain, IpProfile_VoiceVolume, IpProfile_AddIEInSetup, IpProfile_SBCExtensionCodersGroupID, IPProfile_MediaIPVersionPreference, IPProfile_TranscodingMode; [\IPProfile] 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;...
  • Page 314 Mediant 600 & Mediant 1000 Parameter Description DTMF negotiation method: [-1] not configured, use the global parameter; for the remaining options, refer to the global parameter. IP Profiles can also be used when operating with a Proxy server (set the parameter AlwaysUseRouteTable to 1).

  • Page 315: Voice Mail Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description Assign intuitive names (ProfileName) to the Tel Profiles so that you can easily identify them later. To use the settings of the corresponding global parameter, enter the value -1. To apply default settings to a parameter, enter two adjacent dollar signs ($$).
  • Page 316 Mediant 600 & Mediant 1000 Parameter Description completes the call transfer by releasing the line; otherwise, the transfer is cancelled, the device sends a SIP NOTIFY message with a failure reason in the NOTIFY body (such as 486 if busy tone detected), and generates an additional hook- flash towards the FXO line to restore connection to the original call.
  • Page 317 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: MWI Suffix Pattern Determines the digit code used by the device as a suffix for 'MWI EMS: MWI Suffix Code On Digit Pattern' and 'MWI Off Digit Pattern'. This suffix is added [MWISuffixCode] to the generated DTMF string after the extension number.

  • Page 318: Supplementary Services Parameters

    Mediant 600 & Mediant 1000 Parameter Description Web: Forward on No Reason Determines the digit pattern used by the PBX to indicate 'call Digit Pattern (External) forward with no reason' when the original call is received from an EMS: VM Digit Pattern No external line (not an internal extension).
  • Page 319 SIP User's Manual 6. Configuration Parameters Reference Parameter Description value 'id' ('Privacy: id'). Otherwise, for allowed Caller ID, 'Privacy: none' is used. If Caller ID is restricted (received from Tel or configured in the device), the From header is set to <anonymous@anonymous.invalid>.
  • Page 320 Mediant 600 & Mediant 1000 Parameter Description For configuring this table using the Web interface, refer to “Configuring Caller ID Permissions” on page 171. For an explanation on using ini file table parameters, refer to ''Format of ini File Table Parameters'' on page 219.
  • Page 321 SIP User's Manual 6. Configuration Parameters Reference Parameter Description 'Source Number Manipulation' table (table parameter SourceNumberMapIP2Tel). For configuring this table using the Web interface, refer to “Configuring Caller Display Information” on page 168. For an explanation on using ini file table parameters, refer to ''Format of ini File Table Parameters'' on page 219.
  • Page 322 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 323 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [AnalogCallerIDTimingMode] Determines when Caller ID is generated. [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 324: Call Waiting Parameters

    Mediant 600 & Mediant 1000 6.8.2 Call Waiting Parameters The call waiting parameters are described in the table below. Table 6-34: Call Waiting Parameters Parameter Description Web/EMS: Enable Call Waiting Determines whether Call Waiting is enabled. [EnableCallWaiting] [0] Disable = Disable the Call Waiting service.
  • Page 325 SIP User's Manual 6. Configuration Parameters Reference Parameter Description waiting indication signal. When hook-flash is detected, the device switches to the waiting call. The device that initiates the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received.

  • Page 326: Call Forwarding Parameters

    Mediant 600 & Mediant 1000 6.8.3 Call Forwarding Parameters The call forwarding parameters are described in the table below. Table 6-35: Call Forwarding Parameters Parameter Description Web: Enable Call Forward Determines whether Call Forward is enabled. [EnableForward] [0] Disable = Disable the Call Forward service.
  • Page 327 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Notes: The indexing of this parameter starts at 0. Ensure that the Call Forward feature is enabled (default) for the settings of this table parameter to take effect. To enable Call Forwarding, use the parameter EnableForward.

  • Page 328: Message Waiting Indication Parameters

    Mediant 600 & Mediant 1000 6.8.4 Message Waiting Indication Parameters The message waiting indication (MWI) parameters are described in the table below. Table 6-36: MWI Parameters Parameter Description Web: Enable MWI Enables Message Waiting Indication (MWI). EMS: MWI Enable [0] Disable = Disabled (default).
  • Page 329 SIP User's Manual 6. Configuration Parameters Reference 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) [0] UDP [1] TCP [2] TLS Note: When set to ‘Not Configured’, the value of the parameter SIPTransportType is used.

  • Page 330: Call Hold Parameters

    Mediant 600 & Mediant 1000 6.8.5 Call Hold Parameters The call hold parameters are described in the table below. Table 6-37: 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 331: Call Transfer Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.8.6 Call Transfer Parameters The call transfer parameters are described in the table below. Table 6-38: Call Transfer Parameters Parameter Description Web/EMS: Enable Transfer Determines whether call transfer is enabled. [EnableTransfer] [0] Disable = Disable the call transfer service. [1] Enable = The device responds to a REFER message with the Referred-To header to initiate a call transfer(default).

  • Page 332: Three-Way Conferencing Parameters

    Defines the mode of operation when the 3-Way Conference feature is Mode used. EMS: 3 Way Mode [0] AudioCodes Media Server = The Conference-initiating INVITE [3WayConferenceMode] (sent by the device), uses the ConferenceID concatenated with a SIP User's Manual Document #: LTRT-83305...

  • Page 333: Emergency Call Parameters

    Refer-To header value in the REFER messages that are sent to the two remote parties. This conference mode is used when operating with AudioCodes IPMedia conferencing server. (Default) [1] Non-AudioCodes Media Server = The Conference-initiating INVITE (sent by the device) uses only the ConferenceID as the Request-URI.

  • Page 334: Fxs Call Cut-Through Parameter

    Mediant 600 & Mediant 1000 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 message [EmergencyNumbers] 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 335: Automatic Dialing Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.8.10 Automatic Dialing Parameters The automatic dialing upon off-hook parameters are described in the table below. Table 6-42: Automatic Dialing Parameters Parameter Description Web: Automatic Dialing Table EMS: SIP Endpoints > Auto Dial [TargetOfChannel] This ini file table parameter defines telephone numbers that are automatically dialed when a specific FXS or FXO port is used (i.e.,...

  • Page 336: Direct Inward Dialing Parameters

    Mediant 600 & Mediant 1000 6.8.11 Direct Inward Dialing Parameters The Direct Inward Dialing (DID) parameters are described in the table below. Table 6-43: DID Parameters Parameter Description Web/EMS: Enable DID Enables Direct Inward Dialing (DID) using Wink-Start signaling. Wink [0] Disable = Disables DID Wink(default).

  • Page 337: Mlpp Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.8.12 MLPP Parameters The Multilevel Precedence and Preemption (MLPP) parameters are described in the table below. Table 6-44: MLPP Parameters Parameter Description Web/EMS: Call Priority Mode Enables MLPP Priority Call handling. [CallPriorityMode] [0] Disable = Disable (default). [1] MLPP = Priority Calls handling is enabled.
  • Page 338 Mediant 600 & Mediant 1000 Parameter Description Web: MLPP Normalized Service MLPP normalized service domain string. If the device receives Domain an MLPP ISDN incoming call, it uses the parameter (if different EMS: Normalized Service Domain from ‘FFFFFF’) as a Service domain in the SIP Resource- [MLPPNormalizedServiceDomain] Priority header in outgoing INVITE messages.

  • Page 339: 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 Priority precedence call Priority level. [MLPPPriorityRTPDSCP] 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 340 Mediant 600 & Mediant 1000 Parameter Description Web: SAS Local SIP UDP Port Local UDP port for sending and receiving SIP messages for SAS. EMS: Local SIP UDP The SIP entities in the local network need to send the registration [SASLocalSIPUDPPort] requests to this port.
  • Page 341 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: SAS Survivability Mode Determines the Survivability mode used by the SAS application. EMS: Survivability Mode [0] Standard = All incoming INVITE and REGISTER requests [SASSurvivabilityMode] are forwarded to the defined Proxy list in SASProxySet in Normal mode and handled by the SAS application in Emergency mode (default).
  • Page 342 Mediant 600 & Mediant 1000 Parameter Description SASRegistrationManipulation_RemoveFromRight, SASRegistrationManipulation_LeaveFromRight; [\SASRegistrationManipulation] RemoveFromRight = number of digits removed from the right side of the User-Part before saving to the registered user database. LeaveFromRight = number of digits to keep from the right side.

  • Page 343: Media Server Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.10 Media Server Parameters The media processing-related ini file configuration parameters are described in the table below. Table 6-46: Media Server ini File Parameters Parameter Description Web: Number of Media Determines the number of DSP channels that are allocated for IP Channels conferencing, IP streaming, and IP transcoding (other DSP channels EMS: Media Channels...
  • Page 344 Mediant 600 & Mediant 1000 Parameter Description Channel Utilization. Web: Enable Voice Streaming Enables/disables the HTTP Voice Streaming application (play / [EnableVoiceStreaming] record). [0] Disable = Voice Streaming is disabled (default). [1] Enable = Voice Streaming is enabled. Note: For this parameter to take effect, a device reset is required.
  • Page 345 SIP User's Manual 6. Configuration Parameters Reference Parameter Description AMS Parameters [AmsProfile] Must be set to 1 to use advanced audio. [0] = Disable (default) [1] = Enable Note: For this parameter to take effect, a device reset is required. [AASPackagesProfile] Must be set to 3 to use advanced audio.
  • Page 346 Mediant 600 & Mediant 1000 Parameter Description Web: Enable Conference Determines the device logic once a DTMF is received on any DTMF Clamping conference participant. If enabled, the DTMF is not regenerated [EnableConferenceDTMFCl toward the other conference participants. This logic is only relevant amp] for simple conferencing (NetAnn).
  • Page 347 SIP User's Manual 6. Configuration Parameters Reference Parameter Description 6 = 1.75 dB/sec 7 = 2.00 dB/sec 8 = 2.50 dB/sec 9 = 3.00 dB/sec 10 = 3.50 dB/sec 11 = 4.00 dB/sec 12 = 4.50 dB/sec 13 = 5.00 dB/sec 14= 5.50 dB/sec 15 = 6.00 dB/sec 16 = 7.00 dB/sec...
  • Page 348 Mediant 600 & Mediant 1000 Parameter Description EMS: Disable Fast Adaptation Disables the AGC Fast Adaptation mode. [AGCDisableFastAdaptatio [0] = Disable (default) [1] = Enable Note: For this parameter to take effect, a device reset is required. Answer Machine Detector (AMD) Parameters...
  • Page 349 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [PDThreshold] Defines the number of consecutive patterns to trigger the pattern detection event. The valid range is 0 to 31. The default is 5. Note: For this parameter to take effect, a device reset is required. VXML Parameters Web: Enable VXML Enables the VXML stack.
  • Page 350 Mediant 600 & Mediant 1000 Parameter Description [VxmlInterDigitTimeout] Defines the inter-digit timeout value (in msec) used when DTMF is received. The valid range for this parameter is 0 to 7,000 msec. The default value is 3,000. Note: For this parameter to take effect, a device reset is required.
  • Page 351 SIP User's Manual 6. Configuration Parameters Reference 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 352: Pstn Parameters

    Mediant 600 & Mediant 1000 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. Table 6-47: General PSTN Parameters Parameter Description Web/EMS: Protocol Type Defines the PSTN protocol for a the Trunks. To configure the...
  • Page 353 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [53] BRI 5ESS 10 ISDN [54] BRI QSIG [56] BRI NTT = BRI ISDN Japan (Nippon Telegraph) Note: The device simultaneously supports different variants of CAS and PRI protocols on different E1/T1 spans (no more than four simultaneous PRI variants).
  • Page 354 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Clock Master Determines the Tx clock source of the E1/T1 line. [ClockMaster] [0] Recovered = Generate the clock according to the Rx of the E1/T1 line (default). [1] Generated = Generate the clock according to the internal TDM bus.

  • Page 355: Tdm Bus And Clock Timing Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Enable TDM Tunneling Enables TDM tunneling. EMS: TDM Over IP [0] Disable = Disabled (default). [EnableTDMoverIP] [1] Enable = TDM Tunneling is enabled. When TDM Tunneling is enabled, the originating device automatically initiates SIP calls from all enabled B-channels pertaining to E1/T1/J1 spans that are configured with the 'Transparent' protocol.
  • Page 356 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 357 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: TDM Bus PSTN Auto FallBack Enables or disables the PSTN trunk Auto-Fallback Clock Clock feature. EMS: TDM Bus Auto Fall Back [0] Disable (default) = Recovers the clock from the E1/T1 Enable line defined by the parameter TDMBusLocalReference.

  • Page 358: Cas Parameters

    Mediant 600 & Mediant 1000 6.11.3 CAS Parameters The Common Channel Associated (CAS) parameters are described in the table below. Table 6-49: CAS Parameters Parameter Description Web: CAS Table Defines the CAS protocol per trunk (where x denotes the trunk ID)

  • Page 359: 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 6-50: ISDN Parameters Parameter Description Web: ISDN Termination Side Selects the ISDN termination side. EMS: Termination Side [0] User side = ISDN User Termination Equipment (TE) side [TerminationSide] (default) [1] Network side = ISDN Network Termination (NT) side...
  • Page 360 Mediant 600 & Mediant 1000 Parameter Description number. With ISDN Non-Facility Associated Signaling you can use single D- channel to control multiple PRI interfaces. Notes: For this parameter to take effect, a device reset is required. This parameter is applicable only to T1 ISDN protocols.
  • Page 361 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [ISDNRxOverlap] Same as the description for parameter ISDNRxOverlap_x, but for all trunks. Web: PI For Setup Message Determines whether and which Progress Indicator (PI) information [PIForSetupMsg] 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 362 Mediant 600 & Mediant 1000 Parameter Description User provided, user provided: the first one is used Note: When using the ini file to configure the device to support several ISDNInCallsBehavior features, enter a summation of the individual feature values. For example, to support both [2048] and [65536] features, set ISDNInCallsBehavior = 67584 (i.e., 2048 +...
  • Page 363 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [262144] STATUS ERROR CAUSE = Clear call on receipt of STATUS according to cause value. [524288] ACCEPT A LAW =A-Law is also accepted in 5ESS. [2097152] RESTART INDICATION = Upon receipt of a RESTART message, acEV_PSTN_RESTART_CONFIRM is generated.
  • Page 364 Mediant 600 & Mediant 1000 Parameter Description not identified as channel-id #16, but is still carried into the timeslot #16. When this bit is set, the channel ID #16 is considered as a valid B-channel ID, but timeslot values are converted to reflect the range 1 to 15 and 17 to 31.

  • Page 365: Isdn And Cas Interworking Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description calls are set according to the length of the calling number. [2048] = When this bit is set, the device accepts any IA5 character in the called_nb and calling_nb strings and sends any IA5 character in the called_nb, and is not restricted to extended digits only (i.e., 0-9,*,#).
  • Page 366 Mediant 600 & Mediant 1000 Parameter Description When ISDN Tunneling is enabled, the device extracts raw data received in a proprietary SIP header (X- ISDNTunnelingInfo) or a dedicated message body (application/isdn) in the SIP messages and sends the data as ISDN messages to the PSTN side.
  • Page 367 SIP User's Manual 6. Configuration Parameters Reference Parameter Description configured differently. [1] Play on Local = The device configured with CAS protocol type plays a local RBT to PSTN upon receipt of a SIP 180 Ringing response (with or without SDP). Note: Receipt of a 183 response does not cause the device configured with CAS to play an RBT (unless SIP183Behaviour is set to 1).
  • Page 368 Mediant 600 & Mediant 1000 Parameter Description response), the device plays a local RBT if there are no prior received RTP packets. The device stops playing the local RBT as soon as it starts receiving RTP packets. At this stage, if the device receives additional 18x responses, it does not resume playing the local RBT.
  • Page 369 SIP User's Manual 6. Configuration Parameters Reference Parameter Description and use Alarm on non-supporting variants. CAS: Use Alarm. When updating this parameter value at run-time, you must stop the trunk and then restart it for the update to take effect. To determine the method for setting Out-Of-Service state per trunk, use the parameter DigitalOOSBehaviorForTrunk_ID.
  • Page 370 Mediant 600 & Mediant 1000 Parameter Description Web: Release Cause Mapping Table EMS: SIP to ISDN Cause Mapping [CauseMapSIP2ISDN] This ini file table parameter maps SIP Responses to Q.850 Release Causes. The format of this parameter is as follows: [CauseMapSIP2ISDN]...
  • Page 371 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [UserToUserHeaderFormat] Determines the format of the User-to-User SIP header in the INVITE message for interworking the ISDN User to User Information Element (UU IE) data to SIP. [0] = Format: X-UserToUser (default). [1] = Format: User-to-User with Protocol Discriminator (pd) attribute.
  • Page 372 Mediant 600 & Mediant 1000 Parameter Description Web: Set PI in Rx Disconnect Defines the device's behavior when a Disconnect message is Message received from the ISDN before a Connect message is EMS: Set PI For Disconnect Msg received. The ID in the ini file parameter depicts the trunk [PIForDisconnectMsg_ID] number, where 0 is the first trunk.
  • Page 373 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: ISDN Transfer Capabilities Defines the IP-to-ISDN Transfer Capability of the Bearer EMS: Transfer Capability To ISDN Capability IE in ISDN Setup messages. The ID in the ini file [ISDNTransferCapability_ID] parameter depicts the trunk number, where 0 is the first trunk. [-1] Not Configured [0] Audio 3.1 = Audio (default).
  • Page 374 Mediant 600 & Mediant 1000 Parameter Description [DisableFallbackTransferToTDM] Enables or disables "hairpin" TDM transfer upon ISDN (ECT, RLT, or TBCT) call transfer failure. When this feature is enabled and an ISDN call transfer failure occurs, the device sends a SIP NOTIFY message with a SIP 603 Decline response.
  • Page 375 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [0] User Provided = CPN set by user, but not screened (verified). [1] User Passed = CPN set by user, verified and passed. [2] User Failed = CPN set by user, and verification failed. [3] Network Provided = CPN set by network.
  • Page 376 Mediant 600 & Mediant 1000 Parameter Description EMS: DSP Detectors Enable Enables or disables the device's DSP detectors. [EnableDSPIPMDetectors] [0] = Disable (default). [1] = Enable. Notes: For this parameter to take effect, a device reset is required. The device's Feature Key must contain the 'IPMDetector' DSP option.
  • Page 377 SIP User's Manual 6. Configuration Parameters Reference Parameter Description interworking: SIP INVITE to SETUP, SIP 200 OK to CONNECT, SIP INFO to USER INFORMATION, SIP 18x to ALERT, and SIP BYE to DISCONNECT. Notes: The interworking of User-to-User IE to SIP INFO is supported only on the 4ESS PRI variant.
  • Page 378 Mediant 600 & Mediant 1000 Parameter Description [TrunkTransferMode_X] Determines the supported trunk transfer method when a 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...
  • Page 379 SIP User's Manual 6. Configuration Parameters Reference Parameter Description 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 REFER message is sent to the IP side. [5] = IP-to-Tel Blind Transfer mode supported for ISDN PRI/BRI protocols and implemented according to AT&T Toll Free Transfer Connect Service (TR 50075) “Courtesy...
  • Page 380 Mediant 600 & Mediant 1000 Parameter Description [CASAddressingDelimiters] Determines if delimiters are added to the received address or received ANI digits string. [0] = Disable (default) [1] = Enable When this parameter is enabled, delimiters such as '*', '#', and 'ST' are added to the received address or received ANI digits string.
  • Page 381 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Collect ANI In some cases, when the state machine handles the ANI [CasStateMachineCollectANI] collection (not related to MFCR2), you can control the state machine to collect ANI or discard ANI. [0] No = Don't collect ANI.

  • Page 382: Call Disconnect Parameters

    Mediant 600 & Mediant 1000 6.13 Call Disconnect Parameters The call disconnect supervision parameters are described in the table below. Table 6-52: Call Disconnect Parameters Parameter Description Web/EMS: Disconnect on Dial Tone Determines whether the device disconnects a call when a [DisconnectOnDialTone] dial tone is detected from the PBX.
  • Page 383 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Currently, this feature works only if Silence Suppression is disabled. Web: Disconnect Call on Silence Determines whether calls are disconnected after detection Detection of silence. EMS: Disconnect On Detection Of [1] Yes = The device disconnects calls in which silence Silence occurs (in both call directions) for more than a user- [EnableSilenceDisconnect]...
  • Page 384 Mediant 600 & Mediant 1000 Parameter Description Web: Trunk Alarm Call Disconnect Time in seconds to wait (in seconds) after an E1/T1 trunk Timeout "red" alarm (LOS/LOF) is raised before the device [TrunkAlarmCallDisconnectTimeout] disconnects the SIP call. Once this user-defined time elapses, the device sends a SIP BYE message to terminate the call.
  • Page 385 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Enable Current Disconnect Enables call release upon detection of a Current [EnableCurrentDisconnect] Disconnect signal. [0] Disable = Disable the current disconnect service (default). [1] Enable = Enable the current disconnect service. If the current disconnect service is enabled: The FXO releases a call when a current disconnect signal is detected on its port.
  • Page 386 Mediant 600 & Mediant 1000 Parameter Description [CurrentDisconnectDefaultThreshold] Determines the line voltage threshold which, when reached, is considered a current disconnect detection. The valid range is 0 to 20 Volts. The default value is 4 Volts. Notes: This parameter is applicable only to FXO interfaces.

  • Page 387: Tone Parameters

    SIP User's Manual 6. Configuration Parameters Reference 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 6-53: Tone Parameters Parameter Description Web/EMS: Dial Tone Duration Duration (in seconds) that the dial tone is played (for digital [sec] interface: to an ISDN terminal).
  • Page 388 Mediant 600 & Mediant 1000 Parameter Description Web: FXO AutoDial Play Determines whether the device plays a Busy/Reorder tone to the BusyTone PSTN side if a Tel-to-IP call is rejected by a SIP error response [FXOAutoDialPlayBusyTone] (4xx, 5xx or 6xx). If a SIP error response is received, the device...
  • Page 389 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [FirstCallWaitingToneID] Determines the index of the first Call Waiting Tone in the Call Progress Tone (CPT) file. This feature enables the called party to distinguish between four different call origins (e.g., external versus internal calls).
  • Page 390 Mediant 600 & Mediant 1000 Parameter Description 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. [0] Don't Play = Ringback tone isn't played (default).

  • Page 391: Tone Detection Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.14.2 Tone Detection Parameters The signal tone detection parameters are described in the table below. Table 6-54: Tone Detection Parameters Parameter Description Web: Answer Supervision Enables sending of 200 OK upon detection of speech, fax, or EMS: Enable Voice Detection modem.
  • Page 392 Mediant 600 & Mediant 1000 Parameter Description EMS: SIT Enable Enables or disables Special Information Tone (SIT) detection [SITDetectorEnable] according to the ITU-T recommendation E.180/Q.35. [0] = Disable (default). [1] = Enable. To disconnect IP-to-ISDN calls when a SIT tone is detected,...
  • Page 393 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Hook-Flash Option Determines the hook-flash transport type (i.e., method by [HookFlashOption] which hook-flash is sent and received). [0] Not Supported = Hook-Flash indication isn't sent (default). [1] INFO = Send proprietary INFO message with Hook- Flash indication.

  • Page 394: Metering Tone Parameters

    Mediant 600 & Mediant 1000 Parameter Description INFO message containing a hook-flash signal. FXO interfaces: Hook-flash generation period. Notes: For FXO interfaces, a constant of 100 msec must be added to the required hook-flash period. For example, to generate a 450 msec hook-flash, set this parameter to 550.

  • Page 395: Telephone Keypad Sequence Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description ChargeCode_PulsesOnAnswer4; [\ChargeCode] Where, EndTime = Period (1 - 4) end time. PulseInterval = Period (1 - 4) pulse interval. PulsesOnAnswer = Period (1 - 4) pulses on answer. For example: ChargeCode 1 = 7,30,1,14,20,2,20,15,1,0,60,1; ChargeCode 2 = 5,60,1,14,20,1,0,60,1;...
  • Page 396 Mediant 600 & Mediant 1000 Parameter Description [1] = Enable For example, if this parameter is enabled and the prefix string for the external line is defined as "9" (using the parameter Prefix2ExtLine), and the FXS user wants to make a call to destination "123", the device collects and sends all the dialed...
  • Page 397 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Deactivate Keypad sequence that deactivates any of the call forward EMS: Call Forward Deactivation options. After the sequence is pressed, a confirmation tone is [KeyCFDeact] heard. Keypad Feature - Caller ID Restriction Parameters Web: Activate Keypad sequence that activates the restricted Caller ID option.
  • Page 398 Mediant 600 & Mediant 1000 Parameter Description Keypad Feature - Call Waiting Parameters Web: Activate Keypad sequence that activates the Call Waiting option. After EMS: Keypad Features CW the sequence is pressed, a confirmation tone is heard. [KeyCallWaiting] Web: Deactivate Keypad sequence that deactivates the Call Waiting option.

  • Page 399: 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 6-57: General FXO Parameters Parameter Description [CountryCoefficients] Determines the FXO line characteristics (AC and DC) according to USA or TBR21 standard. [66] = TBR21 [70] = United States (default) Note: For this parameter to take effect, a device reset is required.
  • Page 400 Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Waiting For Dial Determines whether the device waits for a dial tone before dialing the Tone phone number for IP-to-Tel (FXO) calls. [IsWaitForDialTone] [0] No = Don't wait for dial tone. [1] Yes = Wait for dial tone (default).
  • Page 401 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Ring Detection Defines the timeout (in seconds) for detecting the second ring after the Timeout [sec] first detected ring. EMS: Timeout Between If automatic dialing is not used and Caller ID is enabled, the device Rings seizes the line after detection of the second ring signal (allowing [FXOBetweenRingTime]...

  • Page 402: Number Manipulation And Routing Parameters

    Mediant 600 & Mediant 1000 6.17 Number Manipulation and Routing Parameters This subsection describes the device's number manipulation and routing parameters. 6.17.1 Routing Parameters The routing parameters are described in the table below. Table 6-58: Routing Parameters Parameter Description Web: Trunk Group Table EMS: SIP Endpoints >...
  • Page 403 Parameter Description [\TrunkGroupSettings] For example: [TrunkGroupSettings] TrunkGroupSettings 0 = 1, 0, 5, audiocodes, user, 1; TrunkGroupSettings 1 = 2, 1, 0, localname, user1, 2; [\TrunkGroupSettings] Notes: This parameter can include up to 240 indices. For configuring Trunk Group Settings using the Web interface, refer to ''Configuring Trunk Group Settings'' on page 175.
  • Page 404 Mediant 600 & Mediant 1000 Parameter Description Notes: The internal numbers of the device's B-channels are defined by the TrunkGroup parameter. For defining the channel select mode per Trunk Group, refer to ''Configuring the Trunk Group Settings'' on page 175.
  • Page 405 SIP User's Manual 6. Configuration Parameters Reference Parameter Description received, the outgoing Source Number and Display Name are set to '100' and the Presentation is set to Allowed (0). When 'from: <sip:100@101.102.103.104>' is received, the outgoing Source Number is set to '100' and the Presentation is set to Restricted (1).
  • Page 406 Mediant 600 & Mediant 1000 Parameter Description Web: IP to Trunk Group Routing Table EMS: SIP Routing > IP to Hunt IP to Trunk Group Routing Table This ini file table parameter configures the routing of IP calls to [PSTNPrefix] Trunk Groups (or inbound IP Group).
  • Page 407 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: IP to Tel Routing Determines whether to route IP calls to the Trunk Group (or IP Mode Group) before or after manipulation of destination number [RouteModeIP2Tel] (configured in ''Configuring the Number Manipulation Tables'' on page 140).

  • Page 408: Alternative Routing Parameters

    Mediant 600 & Mediant 1000 Parameter Description The routing label can be up to 9 (text) characters. Notes: The routing must be configured to be performed before manipulation. For a detailed description of this feature, refer to ''Dial Plan Prefix Tags for IP-to-Tel Routing'' on page 463.
  • Page 409 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [2] QoS = Alternative routing is performed if poor QoS is detected. [3] Both = Alternative routing is performed if either ping to initial destination fails, poor Quality of Service is detected, or DNS host name is not resolved (default).
  • Page 410 Mediant 600 & Mediant 1000 Parameter Description Web: Max Allowed Delay for Alt Transmission delay (in msec) at which the IP connection is Routing [msec] considered a failure and Alternative Routing mechanism is [IPConnQoSMaxAllowedDelay] activated. The range is 100 to 1000. The default value is 250.

  • Page 411: Number Manipulation Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description [\AltRouteCauseIP2Tel] For example: AltRouteCauseIP2Tel 0 = 3 (No Route to Destination) AltRouteCauseIP2Tel 1 = 1 (Unallocated Number) AltRouteCauseIP2Tel 2 = 17 (Busy Here) Notes: This parameter can include up to 5 indices. If the device fails to establish a call to the PSTN because it has no available channels in a specific Trunk Group (e.g., all the channels are occupied, or the spans are...
  • Page 412 Mediant 600 & Mediant 1000 Parameter Description [2] Copy before phone number manipulation = Copies the called number before manipulation. The device first copies the original called number to the SIP Diversion header, and then performs Tel-to-IP destination phone number manipulation. Therefore, this allows you to have different numbers for the called (i.e., SIP To header)
  • Page 413 SIP User's Manual 6. Configuration Parameters Reference Parameter Description the Request-URI as a Phone-Context parameter. Instead, it's added as a prefix to the phone number. The '+' isn't removed from the phone number in the IP-to-Tel direction. To configure the Phone Context table using the Web interface, refer to ''Mapping NPI/TON to Phone-Context'' on page 144.
  • Page 414 Mediant 600 & Mediant 1000 Parameter Description [0] = Leave Source Number empty (default). [CopyDestOnEmptySource] [1] = If the Source Number of a Tel-to-IP call is empty, the Destination Number is copied to the Source Number. Web: Add NPI and TON to Calling...
  • Page 415 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [SwapTel2IPCalled&CallingNumbers] If enabled, the device swaps the calling and called numbers received from the Tel side (for Tel-to-IP calls). The SIP INVITE message contains the swapped numbers. [0] = Disabled (default) [1] = Swap calling and called numbers Web/EMS: Add Prefix to Redirect Defines a string prefix that is added to the Redirect number...
  • Page 416 Mediant 600 & Mediant 1000 Parameter Description NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; Notes: This table parameter can include up to 100 indices. The parameters SourceAddress and IsPresentationRestricted are not applicable. Set these to $$. The parameter RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, LeaveFromRight, NumberType,...
  • Page 417 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Notes: This table parameter can include up to 100 indices. The parameter NumberMapIp2Tel_IsPresentationRestricted is not applicable. Set its value to $$. RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, LeaveFromRight, NumberType, and NumberPlan are applied if the called and calling numbers match the DestinationPrefix, SourcePrefix, and SourceAddress conditions.
  • Page 418 Mediant 600 & Mediant 1000 Parameter Description Web: Source Phone Number This ini file table parameter manipulates the source phone Manipulation Table for Tel to IP Calls number for Tel-to-IP calls. The format of this parameter is EMS: SIP Manipulations > Source...
  • Page 419 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: Source Phone Number This ini file table parameter manipulates the source Manipulation Table for IP to Tel Calls number for IP-to-Tel calls. The format of this parameter is EMS: EMS: SIP Manipulations > as follows: Source IP to Telcom [SourceNumberMapIp2Tel]...
  • Page 420 Mediant 600 & Mediant 1000 Parameter Description For ETSI ISDN variant, the following Number Plan and Type combinations (Plan/Type) are supported in the Destination and Source Manipulation tables: 0,0 = Unknown, Unknown 9,0 = Private, Unknown 9,1 = Private, Level 2 Regional...

  • Page 421: Channel Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.18 Channel Parameters This subsection describes the device's channel parameters. 6.18.1 General Parameters The general channel parameters are described in the table below. Table 6-61: General Channel Parameters Parameter Description Web: Max Number of Defines the maximum number of simultaneous active calls supported Active Calls by the device.
  • Page 422 Mediant 600 & Mediant 1000 Parameter Description Web: Reanswer Time For Analog interfaces: The time interval from when the user hangs up EMS: Regret Time the phone until the call is disconnected (FXS). This allows the user to [RegretTime] hang up and then pick up the phone (before this timeout) to continue the call conversation.

  • Page 423: Voice Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.18.2 Voice Parameters The voice parameters are described in the table below. Table 6-62: Voice Parameters Parameter Description Web/EMS: Input Gain Pulse-code modulation (PCM) input gain control (in decibels). [InputGain] This parameter sets the level for the received (Tel/PSTN-to-IP) signal.
  • Page 424 Mediant 600 & Mediant 1000 Parameter Description Web: Silence Suppression Silence Suppression is a method for conserving bandwidth on EMS: Silence Compression Mode VoIP calls by not sending packets when silence is detected. [EnableSilenceCompression] [0] Disable = Silence Suppression is disabled (default).
  • Page 425 SIP User's Manual 6. Configuration Parameters Reference Parameter Description EMS: Echo Canceller Hybrid Loss Sets the four wire to two wire worst-case Hybrid loss, the ratio [ECHybridLoss] between the signal level sent to the hybrid and the echo level returning from the hybrid. [0] = 6 dB (default) [1] = N/A [2] = 0 dB...

  • Page 426: Coder Parameters

    Mediant 600 & Mediant 1000 6.18.3 Coder Parameters The coder parameters are described in the table below. Table 6-63: Coder Parameters Parameter Description [EnableEVRCVAD] Enables or disables the EVRC voice activity detector. [0] = Disable (default) [1] = Enable Note: Supported for EVRC and EVRC-B coders.

  • Page 427: Fax And Modem Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description DspTemplates 1 = 2, 50; Note: The ini file parameter DSPVersionTemplateNumber is ignored when using the parameters specified in this table. EMS: VBR Coder Header Defines the format of the RTP header for VBR coders. Format [0] = Payload only (no header, no TOC, no m-factor) - similar [VBRCoderHeaderFormat]...
  • Page 428 Mediant 600 & Mediant 1000 Parameter Description Web: Fax Relay Max Rate (bps) Maximum rate (in bps), at which fax relay messages are EMS: Relay Max Rate transmitted (outgoing calls). [FaxRelayMaxRate] [0] 2400 = 2.4 kbps. [1] 4800 = 4.8 kbps.
  • Page 429 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web/EMS: Fax CNG Mode Determines the device's behavior upon detection of a CNG [FaxCNGMode] tone. [0] = Does not send a SIP Re-INVITE upon detection of a fax CNG tone when CNGDetectorMode is set to 1 (default). [1] = Sends a SIP Re-INVITE upon detection of a fax CNG tone when CNGDetectorMode is set to 1.
  • Page 430 Mediant 600 & Mediant 1000 Parameter Description Web: Detect Fax on Answer Tone Determines when the device initiates a T.38 session for fax EMS: Enables Detection of FAX on transmission. Answer Tone [0] Initiate T.38 on Preamble = The device to which the [DetFaxOnAnswerTone] called fax is connected initiates a T.38 session on receiving...
  • Page 431 G.711 coders is a standard one (8 for G.711 A-Law and 0 for G.711 μ-Law). The parameters defining payload type for the 'old' AudioCodes' Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass. The bypass packet interval is selected according to the parameter FaxModemBypassBasicRtpPacketInterval.

  • Page 432: Dtmf Parameters

    Mediant 600 & Mediant 1000 Parameter Description Web: V.32 Modem Transport Type V.32 Modem Transport Type used by the device. EMS: V32 Transport [0] Disable = Disable (Transparent) [V32ModemTransportType] [1] Enable Relay = N/A [2] Enable Bypass = (default) [3] Events Only = Transparent with Events Note: This parameter applies only to V.32 and V.32bis...
  • Page 433 SIP User's Manual 6. Configuration Parameters Reference Parameter Description Web: DTMF Generation Twist Defines the range (in decibels) between the high and low EMS: DTMF Twist Control frequency components in the DTMF signal. Positive [DTMFGenerationTwist] decibel values cause the higher frequency component to be stronger than the lower one.

  • Page 434: Rtp, Rtcp And T.38 Parameters

    Mediant 600 & Mediant 1000 Parameter Description [ReplaceNumberSignWithEscapeChar] Determines whether to replace the number sign (#) with the escape character (%23) in outgoing SIP messages for Tel-to-IP calls. [0] Disable (default) [1] Enable = All number signs #, received in the dialed...
  • Page 435 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [1] 1 = Enable the generation of RFC 2198 redundancy packets (payload type defined by the parameter RFC2198PayloadType). Note: The RTP redundancy dynamic payload type can be included in the SDP, by using the parameter EnableRTPRedundancyNegotiation.
  • Page 436 Mediant 600 & Mediant 1000 Parameter Description Web: RTP Base UDP Port Lower boundary of UDP port used for RTP, RTCP (RTP EMS: Base UDP Port port + 1) and T.38 (RTP port + 2). The upper boundary is [BaseUDPport] the Base UDP Port + 10 * (number of device's channels).
  • Page 437 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 438 Mediant 600 & Mediant 1000 Parameter Description RTCP XR Settings (Note: For a detailed description of RTCP XR reports, refer to the Product Reference Manual.) Web: Enable RTCP XR Enables voice quality monitoring and RTCP Extended EMS: RTCP XR Enable Reports (RTCP XR).

  • Page 439: Digit Collection And Dial Plan Parameters

    SIP User's Manual 6. Configuration Parameters Reference Parameter Description [1] End Call = RTCP XR reports are sent to the ESC at the end of each call. [2] End Call & Periodic = RTCP XR reports are sent to the ESC at the end of each call and periodically according to the parameter RTCPInterval.
  • Page 440 Mediant 600 & Mediant 1000 Parameter Description An example of a digit map is shown below: 11xS|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T In the example above, the last rule can apply to International numbers - 9 for dialing tone, 011 Country Code, and then any number of digits for the local number ('x.').

  • Page 441: Auxiliary And Configuration Files Parameters

    SIP User's Manual 6. Configuration Parameters Reference 6.19 Auxiliary and Configuration Files Parameters This subsection describes the device's auxiliary and configuration files parameters. 6.19.1 Auxiliary/Configuration File Name Parameters The configuration files (i.e., auxiliary files) can be loaded to the device using the Web interface or a TFTP session (refer to ''Loading Auxiliary Files'' on page 194).

  • Page 442: Automatic Update Parameters

    Mediant 600 & Mediant 1000 Parameter Description Web/EMS: Prerecorded Tones File The name (and path) of the file containing the Prerecorded [PrerecordedTonesFileName] Tones. Note: For this parameter to take effect, a device reset is required. Web: CAS File CAS file name (e.g., 'E_M_WinkTable.dat') that defines the EMS: Trunk Cas Table Index CAS protocol, where x denotes the CAS file ID (0-7).
  • Page 443 SIP User's Manual 6. Configuration Parameters Reference Parameter Description [AutoUpdatePredefinedTime] Schedules an automatic update to a predefined time of the day. The range is 'HH:MM' (24-hour format). For example: 20:18 Notes: For this parameter to take effect, a device reset is required. The actual update time is randomized by five minutes to reduce the load on the Web servers.
  • Page 444 Mediant 600 & Mediant 1000 Parameter Description [CptFileURL] Specifies the name of the CPT file and the location of the server (IP address or FQDN) from which it is loaded. For example: http://server_name/file, https://server_name/file. Note: The maximum length of the URL address is 99 characters.

  • Page 445: Default Settings

    SIP User's Manual 7. Default Settings Default Settings You can restore the device's factory default settings or define your own user-defined default settings for the device. Defining Default Settings The device is shipped with factory default configuration values stored on its non-volatile memory (flash).
  • Page 446 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83305...

  • Page 447: 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 Distinctive Ringing in the ini file (refer to “Distinctive Ringing”...
  • Page 448 Mediant 600 & Mediant 1000 The format attribute can be one of the following: Continuous: (e.g., dial tone) a steady non-interrupted sound. 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 449 SIP User's Manual 8. Auxiliary Configuration Files • High Freq Level: generation level. 0 to -31 dBm. The value should be set to 32 in the case of a single tone (not relevant to AM tones). • First Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the first cadence on-off cycle.

  • Page 450: Distinctive Ringing

    Mediant 600 & Mediant 1000 required) First Signal On Time [10msec]=300; the dial tone is detected after 3 sec First Signal Off Time [10msec]=0 Second Signal On Time [10msec]=0 Second Signal Off Time [10msec]=0 8.1.1 Distinctive Ringing Distinctive Ringing is applicable only to FXS interfaces. Using the distinctive ringing section of the Call Progress Tones auxiliary file, you can create up to 16 distinctive ringing patterns.
  • Page 451 SIP User's Manual 8. Auxiliary Configuration Files • Fourth (Burst) Ring On Time [10 msec]: 'Ring Off' period (in 10 msec units) for the fourth cadence on-off cycle. • Fourth (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units) for the fourth cadence on-off cycle.

  • Page 452: Prerecorded Tones File

    Resolution: 8-bit Channels: mono The generated PRT file can then be loaded to the device using AudioCodes' BootP/TFTP utility or the Web interface (refer to ''Loading Auxiliary Files'' on page 194). The prerecorded tones are played repeatedly. This allows you to record only part of the tone and then play the tone for the full duration.

  • Page 453: Cas Files

    Larger files (up to 10 MB) are stored in RAM, and should be loaded again (using BootP/TFTP utility) after the device is reset. AudioCodes provides a professionally recorded English (U.S.) Voice Prompts file. Note: Voice Prompts are applicable only to Mediant 1000.

  • Page 454: Dial Plan File

    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 (for a description, refer to ''Dial Plan Prefix Tags for IP-to-Tel Routing'' on page 463).
  • Page 455 SIP User's Manual 8. Auxiliary Configuration Files An example of a Dial Plan file in ini-file format (i.e., before converted to *.dat) that contains two dial plans is shown below: ; Example of dial-plan configuration. ; This file contains two dial plans: [ PLAN1 ] ;...

  • Page 456: Fxs Coefficient File

    Mediant 600 & Mediant 1000 FXS Coefficient File The FXS Coefficient file (Coeff_FXS.dat) is used to provide best termination and transmission quality adaptation for different line types for FXS interfaces. This adaptation is performed by modifying the telephony interface characteristics (such as DC and AC impedance, feeding current, and ringing voltage).

  • Page 457: User Information File

    SIP User's Manual 8. Auxiliary Configuration Files User Information File The User Information file is a text file that maps PBX extensions connected to the device to global IP numbers. In this context, a global IP phone number (alphanumerical) serves as a routing identifier for calls in the 'IP world'.
  • Page 458 Mediant 600 & Mediant 1000 The User Information file can be loaded to the device by using one of the following methods: ini file, using the parameter UserInfoFileName (described in ''Auxiliary / Configuration Files Parameters'' on page 440) Web interface (refer to ''Loading Auxiliary Files'' on page 194)

  • Page 459: Ip Telephony Capabilities

    SIP User's Manual 9. IP Telephony Capabilities IP Telephony Capabilities This section describes the device's IP telephony capabilities. Dialing Plan Features This section discusses various dialing plan features offered by the device: Dialing plan notations (refer to ''Dialing Plan Notation for Routing and Manipulation'' on page 459) Digit mapping (refer to ''Digit Mapping'' on page 461) External Dial Plan file containing dial plans (refer to ''External Dial Plan File'' on page...

  • Page 460: 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 a number. 54324xx#: represents a 7-digit number at the end of a that starts with 54324. number A single Represents any number.

  • Page 461: Digit Mapping

    SIP User's Manual 9. IP Telephony Capabilities number, in the example, the number is changed to 8888888; 3) the prefix that was previously calculated is then added. 9.1.2 Digit Mapping Digit map pattern rules are used for Tel-to-IP ISDN overlap dialing (for digital interface). The device collects digits until a match is found in the user-defined digit pattern (e.g., for closed numbering schemes) or until a timer expires (e.g., for open numbering schemes).

  • Page 462: External Dial Plan File

    This file is loaded to the device as a *.dat file (binary file), converted from an ini file using AudioCodes TrunkPack Downloadable Conversion utility (DConvert). This file can include up to eight Dial Plans (Dial Plan indices). The required Dial Plan can be selected using the Dial Plan index, using the parameter DialPlanIndex.

  • Page 463: Dial Plan Prefix Tags For Ip-To-Tel Routing

    SIP User's Manual 9. IP Telephony Capabilities ; Emergency number 911 (no additional digits expected). 911,0 [ PLAN2 ] ; Supplementary services such as Call Camping and Last Calls (no additional digits expected), by dialing *41, *42, or *43. *4[1-3],0 Notes: •...

  • Page 464: Figure 9-2: Configuring Dial Plan File Label For Ip-To-Tel Routing

    Mediant 600 & Mediant 1000 To use Dial Plan file routing tags: Load an ini file to the device that selects the Dial Plan index (e.g., 1) for routing tags, as shown below: IP2TelTaggingDialPlanIndex = 1 Define the external Dial Plan file with two routing tags (as shown below): •...

  • Page 465: Ip-To-Ip Routing Application

    9. IP Telephony Capabilities IP-to-IP Routing Application The AudioCodes device supports IP-to-IP VoIP call routing (or SIP trunking). The device enables Enterprises to seamlessly connect their IP-PBX to a SIP trunk provided by an Internet Telephony Service Provider (ITSP). The Enterprise can communicate with the PSTN through the ITSP, which interfaces directly with PSTN.
  • Page 466 Mediant 600 & Mediant 1000 Verify (by using the Web interface or downloaded ini file) that your device has been supplied with the following Feature Keys: • Number of IPmedia Channels: this Feature Key is configured to the maximum number of required DSP resources. (For example, the ini file displays this as "IPMediaDspCh=60".)
  • Page 467 SIP User's Manual 9. IP Telephony Capabilities ♦ TRUNKS module with one Trunk provides 40 channels (2 DSPs * 20 channels), instead of 48 ♦ TRUNKS module with two Trunks provides 60 channels (3 DSPs * 20 channels), instead of 72 ♦...

  • Page 468: Stand-Alone Survivability (Sas) Feature

    Mediant 600 & Mediant 1000 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 469: Configuring Sas

    SIP User's Manual 9. IP Telephony Capabilities The call routing rules for SAS in Emergency mode is configured in the 'IP2IP Routing Table' page (refer to ''SAS Routing Table'' on page 136). This table provides enhanced call routing capabilities (such as built-in ENUM queries and redundant SAS proxy server load balancing) for routing received SIP INVITE messages in Emergency mode.

  • Page 470: Configuring Emergency Calls

    Mediant 600 & Mediant 1000 SASLocalSIPUDPPort = (default 5080) SASRegistrationTime = <expiration time that SAS returns in the 200 OK to REGISTER in Emergency mode> (default 20) SASDefaultGatewayIP = < SAS gateway IP address> SASProxySet = 1 SAS call routing rules in Emergency mode, use the ini file parameter IP2IPRouting (or the Web - refer to ''SAS Routing Table'' on page 136) 9.4.2...

  • Page 471: Emergency Phone Number Services

    SIP User's Manual 9. IP Telephony Capabilities SASDefaultGatewayIP = < SAS gateway IP address> SASProxySet = 1 Emergency Phone Number Services The device supports various emergency phone number services such as 911 (in North America). 9.5.1 Enhanced 911 Support 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 472: Fxo Device Interworking Sip E911 Calls From Service Provider's Ip Network To Psap Did Lines

    Mediant 600 & Mediant 1000 9.5.1.1 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 473: Table 9-3: Dialed Number By Device Depending On Calling Number

    "mmmmmmmmmm" (pANI) None "KPnnSTP" "nn" MF dialed "KPnST" "n" For example: "From: <sip:8>@audiocodes.com>" generates device MF spill of KP 8 ST Table Notes: For all other cases, a SIP 484 response is sent. KP is for *. Version 5.8 September 2009...

  • Page 474: Mediant 600 & Mediant

    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 475: Fxs Device Emulating Psap Using Did Loop-Start Lines

    SIP User's Manual 9. IP Telephony Capabilities 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 Content-Length: 17 event flashhook 9.5.1.2 FXS Device Emulating PSAP using DID Loop-Start Lines...
  • Page 476 Mediant 600 & Mediant 1000 The FXS device collects the MF digits, and then sends a SIP INVITE message to the PSAP with all collected MF digits in the SIP From header as one string. The FXS device generates a mid-call wink signal (two subsequent polarity reversals) toward the E911 tandem switch upon either detection of an RFC 2833 "hookflash"...

  • Page 477: Table 9-4: Dialed Mf Digits Sent To Psap

    SIP User's Manual 9. IP Telephony Capabilities The outgoing SIP INVITE message contains the following headers: INVITE sip:Line@DomainName From: <sip:*81977820#@sipgw>;tag=1c143 To: <sip:Line@DomainName> Where: Line = as configured in the Endpoint Phone Number Table. SipGtw = configured using the SIPGatewayName parameter. From header/user part = calling party number as received from the MF spill.

  • Page 478: Configuring Dtmf Transport Types

    Mediant 600 & Mediant 1000 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 479: Fax And Modem Capabilities

    SIP User's Manual 9. IP Telephony Capabilities 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 (ini file); 'Declare RFC 2833 in SDP' field = 'No' •...

  • Page 480: Fax/Modem Transport Modes

    Mediant 600 & Mediant 1000 9.7.2 Fax/Modem Transport Modes The device supports the following transport modes for fax per modem type (V.22/V.23/Bell/V.32/V.34): T.38 fax relay (refer to ''Fax Relay Mode'' on page 480) Fax and modem bypass: a proprietary method that uses a high bit rate coder (refer to ''Fax/Modem Bypass Mode'' on page 481) NSE Cisco’s Pass-through bypass mode for fax and modem (refer to ''Fax / Modem...

  • Page 481: Fax/Modem Bypass Mode

    SIP User's Manual 9. IP Telephony Capabilities To configure T.38 mode using SIP Re-INVITE messages, set IsFaxUsed to 1. Additional configuration parameters include the following: FaxRelayEnhancedRedundancyDepth FaxRelayRedundancyDepth FaxRelayECMEnable FaxRelayMaxRate Note: The terminating gateway sends T.38 packets immediately after the T.38 capabilities are negotiated in SIP.
  • Page 482 Tip: When the remote (non-AudioCodes’) gateway uses G711 coder for voice and doesn’t change the coder payload type for fax or modem transmission, it is recommended to use the Bypass mode with the following configuration: •...

  • Page 483: Fax / Modem Nse Mode

    The voice channel is optimized for fax/modem transmission (same as for usual bypass mode). The parameters defining payload type for the proprietary AudioCodes’ Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass. When configured for NSE mode, the device includes in its SDP the following line:...

  • Page 484: Fax / Modem Transparent With Events Mode

    Mediant 600 & Mediant 1000 V23ModemTransportType = 0 V32ModemTransportType = 0 V34ModemTransportType = 0 BellModemTransportType = 0 Additional configuration parameters: • CoderName • DJBufOptFactor • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission.

  • Page 485: Fax Fallback

    SIP User's Manual 9. IP Telephony Capabilities Dynamic Jitter Buffer Minimum Delay = 40 Dynamic Jitter Buffer Optimization Factor = 13 After a few seconds upon detection of fax V.21 preamble or super G3 fax signals, the device sends a second Re-INVITE enabling the echo canceller (the echo canceller is disabled only on modem transmission).

  • Page 486: Using Bypass Mechanism For V.34 Fax Transmission

    Mediant 600 & Mediant 1000 Using the ini file parameter V34FaxTransportType, you can determine whether to pass V.34 Fax-over-T.38 fallback to T.30, or use Bypass over the High Bit Rate coder (e.g. PCM A- Law). Note: The CNG detector is disabled (CNGDetectorMode = 0) in all the subsequent examples.

  • Page 487: Support

    SIP User's Manual 9. IP Telephony Capabilities 9.7.4 V.152 Support The device supports the ITU-T recommendation V.152 (Procedures for Supporting Voice- Band Data over IP Networks). Voice-band data (VBD) is the transport of modem, facsimile, and text telephony signals over a voice channel of a packet network with a codec appropriate for such signals.

  • Page 488: Fxo Operating Modes

    Mediant 600 & Mediant 1000 FXO Operating Modes This section provides a description of the device's FXO operating modes: IP-to-Tel calls (refer to ''FXO Operations for IP-to-Tel Calls'' on page 488) Tel-to-IP calls (refer to ''FXO Operations for Tel-to-IP Calls'' on page 491) 9.8.1...

  • Page 489: One-Stage Dialing

    SIP User's Manual 9. IP Telephony Capabilities 9.8.1.1 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. Figure 9-8: Call Flow for One-Stage Dialing One-stage dialing incorporates the following FXO functionality: Waiting for Dial Tone: Enables the device to dial the digits to the Tel side only after...

  • Page 490: Two-Stage Dialing

    Mediant 600 & Mediant 1000 9.8.1.2 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 491: Fxo Operations For Tel-To-Ip Calls

    SIP User's Manual 9. IP Telephony Capabilities 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) DID Wink can be used for connection to EIA/TIA-464B DID Loop Start lines Both FXO (detection) and FXS (generation) are supported...

  • Page 492: Collecting Digits Mode

    Mediant 600 & Mediant 1000 9.8.2.2 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 9-11: Call Flow for Collecting Digits Mode 9.8.2.3...

  • Page 493: Call Termination On Fxo Devices

    SIP User's Manual 9. IP Telephony Capabilities 9.8.3 Call Termination on FXO Devices This section describes the device's call termination capabilities for its FXO interfaces: Calls terminated by a PBX (refer to ''Call Termination by PBX'' on page 493) Calls terminated before call establishment (refer to ''Call Termination before Call Establishment'' on page 494) Ring detection timeout (refer to ''Ring Detection Timeout'' on page 494) 9.8.3.1...

  • Page 494: Call Termination Before Call Establishment

    Mediant 600 & Mediant 1000 Protocol-based termination of the call from the IP side Note: The implemented disconnect method must be supported by the CO or PBX. 9.8.3.2 Call Termination before Call Establishment The device supports the following call termination methods before a call is established:...

  • Page 495: Answer Machine Detector (Amd)

    SIP User's Manual 9. IP Telephony Capabilities Answer Machine Detector (AMD) Answering Machine Detection can be useful in automatic dialing applications. In some of these applications, it is important to detect if a human voice or answering machine is answering the call. Answering Machine Detection can be activated and de-activated only after a channel is already open.
  • Page 496 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 497 CSeq: 1 INFO Contact: <sip:56700@172.22.168.249> Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB SCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-IPmedia 260_UN/v.5.20A.040.004 Content-Type: application/x-detect Content-Length: 34 Type= PTT SubType= SPEECH-START Upon detection of the end of voice (i.e., end of the greeting message of the answering machine), the device sends the Application server the following: INFO sip:sipp@172.22.2.9:5060 SIP/2.0...

  • Page 498: Event Notification Using X-Detect Header

    Mediant 600 & Mediant 1000 9.10 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 499 SIP User's Manual 9. IP Telephony Capabilities For example: INFO sip:5001@10.33.2.36 SIP/2.0 Via: SIP/2.0/UDP 10.33.45.65;branch=z9hG4bKac2042168670 Max-Forwards: 70 From: <sip:5000@10.33.45.65;user=phone>;tag=1c1915542705 To: <sip:5001@10.33.2.36;user=phone>;tag=WQJNIDDPCOKAPIDSCOTG Call-ID: AIFHPETLLMVVFWPDXUHD@10.33.2.36 CSeq: 1 INFO Contact: <sip:2206@10.33.45.65> Supported: em,timer,replaces,path,resource-priority Content-Type: application/x-detect Content-Length: 28 Type= CPT SubType= SIT-IC The X-Detect event notification process is as follows: For IP-to-Tel or Tel-to-IP calls, the device receives a SIP request message (using the X-Detect header) that the remote party wishes to detect events on the media stream.

  • Page 500: Rtp Multiplexing (Throughpacket)

    Mediant 600 & Mediant 1000 Content-Type: Application/X-Detect Content-Length: xxx Type = CPT Subtype = SIT 9.11 RTP Multiplexing (ThroughPacket) The device supports a proprietary method to aggregate RTP streams from several channels to reduce the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP headers, and to reduce the packet/data transmission rate.

  • Page 501: Configuring Alternative Routing (Based On Connectivity And Qos)

    SIP User's Manual 9. IP Telephony Capabilities At the minimum value of 0, the buffer tracks delays only to compensate for clock drift and quickly decays back to the minimum level. This optimizes the delay performance but at the expense of a higher error rate. The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide a good compromise between delay and error rate.

  • Page 502: Alternative Routing Mechanism

    Mediant 600 & Mediant 1000 9.13.1 Alternative Routing Mechanism When a Tel-to-IP call is routed through the device, the destination number is compared to the list of prefixes defined in the 'Tel to IP Routing' table (described in ''Configuring the Tel to IP Routing Table'' on page 148).

  • Page 503: Mapping Pstn Release Cause To Sip Response

    SIP User's Manual 9. IP Telephony Capabilities 9.14 Mapping PSTN Release Cause to SIP Response The device's FXO interface interoperates between the SIP network and the PSTN/PBX. This interoperability includes the mapping of PSTN/PBX Call Progress Tones to SIP 4xx or 5xx responses for IP-to-Tel calls.
  • Page 504 Mediant 600 & Mediant 1000 Attribute Attribute Value Purpose Example Number Name Format Start Up to H323-Conf- H.323/SIP call identifier Stop octets Start H323-Setup- Setup time in NTP format String Time Stop Start The call’s originator: H323-Call- Answer, Answering (IP) or...
  • Page 505 SIP User's Manual 9. IP Telephony Capabilities Attribute Attribute Value Purpose Example Number Name Format Account Request Type Start (start or stop) Acct-Status- Note: ‘start’ isn’t Numeric 1: start, 2: stop Type Stop supported on the Calling Card application. Start No.

  • Page 506: Call Detail Record

    Mediant 600 & Mediant 1000 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 nas-port-type = 0 acct-status-type = 2 acct-input-octets = 4841...
  • Page 507 SIP User's Manual 9. IP Telephony Capabilities Field Name Description SrcPhoneNum Source Phone Number SrcNumBeforeMap Source Number Before Manipulation Destination Phone Number Type Destination Phone Number Plan DstPhoneNum Destination Phone Number DstNumBeforeMap Destination Number Before Manipulation Durat Call Duration Coder Selected Coder Packet Interval Intrv...

  • Page 508: Querying Device Channel Resources Using Sip Options

    Mediant 600 & Mediant 1000 9.17 Querying Device Channel Resources using SIP OPTIONS The device reports its maximum and available channel resources in SIP 200 OK responses upon receipt of SIP OPTIONS messages. The device sends this information in the SIP X-...

  • Page 509: Call Hold And Retrieve

    SIP User's Manual 9. IP Telephony Capabilities 9.18.1 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; the other party is called the held party.

  • Page 510: Figure 9-12: 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 511: 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 512: Call Transfer

    Mediant 600 & Mediant 1000 9.18.3 Call Transfer There are two types of call transfers: Consultation Transfer (REFER and REPLACES): The common way 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 513: Call Forward Reminder Ring

    SIP User's Manual 9. IP Telephony Capabilities No Reply: incoming call is forwarded if it isn't answered for a specified time. On Busy or No Reply: incoming call is forwarded if the port is busy or when calls are not answered after a specified time. Do Not Disturb: immediately reject incoming calls.

  • Page 514: Call Waiting

    Mediant 600 & Mediant 1000 The device generates a Call Forward Reminder ring burst to the FXS endpoint each time it receives a SIP NOTIFY message with “reminder ring” xml body. The NOTIFY request is sent from the Application Server to the device each time the Application Server forwards an incoming call.

  • Page 515: Message Waiting Indication

    SIP User's Manual 9. IP Telephony Capabilities 9.18.7 Message Waiting Indication The device supports Message Waiting Indication (MWI) according to IETF <draft-ietf- sipping-mwi-04.txt>, including SUBSCRIBE (to MWI server). The FXS device can accept an MWI NOTIFY message that indicates waiting messages or that the MWI is cleared. Users are informed of these messages by a stutter dial tone.

  • Page 516: Caller Id

    Mediant 600 & Mediant 1000 9.18.8 Caller ID This section discusses the device's Caller ID support. Note: The Caller ID feature is applicable only to FXS/FXO interfaces. 9.18.8.1 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 517: Debugging A Caller Id Detection On Fxo

    ID. If the above does not solve the problem, you need to record the caller ID signal (and send it to AudioCodes), as described below. To record the caller ID signal using the debug recording mechanism: Access the FAE page (by appending "FAE"...

  • Page 518: Caller Id On The Ip Side

    Mediant 600 & Mediant 1000 9.18.8.3 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 519: Three-Way Conferencing

    The device supports the following conference modes (configured by the parameter 3WayConferenceMode): Conferencing controlled by an external AudioCodes Conference (media) server: The Conference-initiating INVITE sent by the device uses the ConferenceID concatenated with a unique identifier as the Request-URI. This same Request-URI is set as the Refer-To header value in the REFER messages that are sent to the two remote parties.

  • Page 520: Proxy Or Registrar Registration Example

    Mediant 600 & Mediant 1000 9.19 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 521: Sip Call Flow Example

    F1 (10.8.201.108 >> 10.8.201.10 INVITE): INVITE sip:1000@10.8.201.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:8000@10.8.201.108>;tag=1c5354 To: <sip:1000@10.8.201.10> Call-ID: 534366556655skKw-8000--1000@10.8.201.108 CSeq: 18153 INVITE Contact: <sip:8000@10.8.201.108;user=phone> User-Agent: Audiocodes-Sip-Gateway/Mediant 1000/v.5.40.010.006 Supported: 100rel,em Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE, NOTIFY,PRACK,REFER,INFO Content-Type: application/sdp Content-Length: 208 o=AudiocodesGW 18132 74003 IN IP4 10.8.201.108 s=Phone-Call c=IN IP4 10.8.201.108...
  • Page 522 Mediant 600 & Mediant 1000 F2 (10.8.201.10 >> 10.8.201.108 TRYING): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:8000@10.8.201.108>;tag=1c5354 To: <sip:1000@10.8.201.10> Call-ID: 534366556655skKw-8000--1000@10.8.201.108 Server: Audiocodes-Sip-Gateway/Mediant 1000/v.5.40.010.006 CSeq: 18153 INVITE Content-Length: 0 F3 (10.8.201.10 >> 10.8.201.108 180 RINGING): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:8000@10.8.201.108>;tag=1c5354...

  • Page 523: Sip Authentication Example

    F5 (10.8.201.108 >> 10.8.201.10 ACK): ACK sip:1000@10.8.201.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ From: <sip:8000@10.8.201.108>;tag=1c5354 To: <sip:1000@10.8.201.10>;tag=1c7345 Call-ID: 534366556655skKw-8000--1000@10.8.201.108 User-Agent: Audiocodes-Sip-Gateway/Mediant 1000/v.5.40.010.006 CSeq: 18153 ACK Supported: 100rel,em Content-Length: 0 Note: Phone ‘8000’ goes on-hook and device 10.8.201.108 sends a BYE to device 10.8.201.10.
  • Page 524 Since the algorithm is MD5, then: • The username is equal to the endpoint phone number 122. • The realm return by the proxy is audiocodes.com. • The password from the ini file is AudioCodes. • The equation to be evaluated is (according to RFC this part is called A1) ‘122:audiocodes.com:AudioCodes’.
  • Page 525 At this time, a new REGISTER request is issued with the following response: REGISTER sip:10.2.2.222 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200 From: <sip: 122@10.1.1.200>;tag=1c23940 To: <sip: 122@10.1.1.200> Call-ID: 654982194@10.1.1.200 Server: Audiocodes-Sip-Gateway/Mediant 1000/v.5.40.010.006 CSeq: 1 REGISTER Contact: sip:122@10.1.1.200: Expires:3600 Authorization: Digest, username: 122, realm="audiocodes.com”, nonce="11432d6bce58ddf02e3b5e1c77c010d2",...

  • Page 526: Establishing A Call Between Two Devices

    9.22 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.

  • Page 527: Trunk-To-Trunk Routing Example

    SIP User's Manual 9. IP Telephony Capabilities Figure 9-17: Tel-to-IP Routing Rules 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 528: Sip Trunking Between Enterprise And Itsps

    Proxy Sets, IP Groups, and Accounts, you can "design" complex routing schemes. This section provides an example of an elaborate routing scheme for SIP trunking between an Enterprise's PBX and two Internet Telephony Service Providers (ITSP), using AudioCodes' device. Scenario:In this example, the Enterprise wishes to connect its TDM PBX to two different ITSPs, by implementing a device in its network environment.

  • Page 529: Figure 9-19: Configuring Proxy Set Id #1 In The Proxy Sets Table

    SIP User's Manual 9. IP Telephony Capabilities 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'' on page 110), configure two Proxy Sets and for each, enable Proxy Keep-Alive (using SIP OPTIONS) and 'round robin' load-balancing method: •...

  • Page 530: Figure 9-20: Configuring Ip Groups #1 And #2 In The Ip Group Table

    Mediant 600 & Mediant 1000 In the 'IP Group Table' page (refer to ''Configuring the IP Groups'' on page 114), configure the two IP Groups #1 and #2. Assign Proxy Sets #1 and #2 to IP Groups #1 and #2 respectively.

  • Page 531: Figure 9-23: Configuring Accounts For Pbx Registration To Itsps In Account Table

    SIP User's Manual 9. IP Telephony Capabilities In the 'Account Table' page (refer to ''Configuring the Account Table'' on page 118), configure the two Accounts for PBX trunk registration to ITSPs using the same Trunk Group (i.e., ID #1), but different serving IP Groups #1 and #2. For each account, define user name, password, and hostname, and ContactUser.

  • Page 532: Remote Pbx Extension Between Fxo And Fxs Devices

    Mediant 600 & Mediant 1000 9.25 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 533: Dialing From Remote Extension (Phone At Fxs)

    SIP User's Manual 9. IP Telephony Capabilities 9.25.1 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 534: Message Waiting Indication For Remote Extensions

    Mediant 600 & Mediant 1000 9.25.3 Message Waiting Indication for Remote Extensions The device supports the relaying of Message Waiting Indications (MWI) for remote extensions (and voice mail applications). Instead of subscribing to an MWI server to receive notifications of pending messages, the FXO device receives subscriptions from the remote FXS device and notifies the appropriate extension when messages (and the number of messages) are pending.

  • Page 535: Fxs Gateway Configuration

    SIP User's Manual 9. IP Telephony Capabilities 9.25.5 FXS Gateway Configuration The procedure below describes how to configure the FXS interface (at the 'remote PBX extension'). To configure the FXS interface: In the ‘Trunk Group Table’ page (refer to “Configuring the Trunk Group Table” on page 173, assign the phone numbers 100 to 104 to the device's endpoints.

  • Page 536: Fxo Gateway Configuration

    Mediant 600 & Mediant 1000 9.25.6 FXO Gateway Configuration The procedure below describes how to configure the FXO interface (to which the PBX is directly connected). To configure the FXO interface: In the ‘Trunk Group Table’ page (refer to “Configuring the Trunk Group Table” on page 173, assign the phone numbers 200 to 204 to the device’s FXO endpoints.

  • Page 537: Networking Capabilities

    SIP User's Manual 10. Networking Capabilities 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: Manual mode: • 10Base-T Half-Duplex or 10Base-T Full-Duplex • 100Base-TX Half-Duplex or 100Base-TX Full-Duplex Auto-Negotiation: chooses common transmission parameters such as speed and duplex mode...

  • Page 538: Nat (Network Address Translation) Support

    Mediant 600 & Mediant 1000 When Ethernet redundancy is implemented, the two Ethernet ports can be connected to the same switch (segment / hub). In this setup, one Ethernet port is active and the other is redundant. If an Ethernet connection failure is detected, the CPU module switches over to the redundant Ethernet port.

  • Page 539: Stun

    SIP User's Manual 10. Networking Capabilities 10.3.1 STUN Simple Traversal of UDP through NATs (STUN), based on RFC 3489 is a client / server protocol that solves most of the NAT traversal problems. The STUN server operates in the public Internet and the STUN clients are embedded in end-devices (located behind NAT). STUN is used both for the signaling and the media streams.

  • Page 540: First Incoming Packet Mechanism

    No-Op packets are sent. This is performed using the RTPNoOpPayloadType ini parameter (refer to ''Networking Parameters'' on page 243). AudioCodes’ default payload type is 120. T.38 No-Op: T.38 No-Op packets are sent only while a T.38 session is activated. Sent packets are a duplication of the previously sent frame (including duplication of the sequence number).

  • Page 541: 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 542: 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- defined (using the ini file parameters NTPServerIP and NTPUpdateInterval respectively), or an SNMP MIB object (refer to the Product Reference Manual).

  • Page 543: 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 544: 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 545: 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) MEDIA –...

  • Page 546: 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 547: 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 548: 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) Premium Media service class – used for RTP Media traffic Premium Control Service class –...

  • Page 549: 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 550: Multiple Interface Table Configuration Summary And Guidelines

    Mediant 600 & Mediant 1000 10.9.1.3.5 Applications with Assignable Application Type Some applications can be associated with different application types in different setups. These application types are configurable. The applications listed below can be configured to one of two application types:...
  • Page 551 SIP User's Manual 10. Networking Capabilities • One IPv4 interface with "Application Types" OAMP, one other or more IPv4 interfaces with "Application Types" CONTROL, and one or more IPv4 interfaces with "Application Types" MEDIA (with VLANs). • One IPv4 interface with "Application Types" OAMP & MEDIA, one other or more IPv4 interfaces with "Application Types"...

  • Page 552: Troubleshooting The Multiple Interface Table

    Mediant 600 & Mediant 1000 10.9.1.5 Troubleshooting the Multiple Interface Table If any of the Multiple Interface table guidelines are violated, the device falls back to a "safe mode" configuration, consisting of a single IPv4 interface and no VLANs. For more information on validation failures, consult the Syslog messages.

  • Page 553: Routing Table Columns

    SIP User's Manual 10. Networking Capabilities 10.9.2.2 Routing Table Columns Each row of the Routing table defines a routing rule. Traffic destined to the subnet specified in the routing rule is re-directed to a specified gateway, reachable through a specified interface.

  • Page 554: Routing Table Configuration Summary And Guidelines

    Mediant 600 & Mediant 1000 10.9.2.2.4 Interface Column This column defines the interface index (in the Multiple Interface table) from which the gateway address is reached. Figure 10-4: Interface Column 10.9.2.2.5 Metric Column The Metric column must be set to 1 for each routing rule.

  • Page 555: Troubleshooting The Routing Table

    SIP User's Manual 10. Networking Capabilities 10.9.2.4 Troubleshooting the Routing Table When adding or modifying any of the routing rules, the added or modified rule passes a validation test. If errors are found, the route is rejected and is not added to the Routing table.

  • Page 556: Setting Up The Device

    Mediant 600 & Mediant 1000 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 557: Table 10-9: Multiple Interface Table - Example 1

    • The Multiple Interface table configuration using the ini file must have the prefix and suffix to allow AudioCodes INI File parser to correctly recognize the Multiple Interface Table. The following sections show some examples of selected network configurations, and their matching ini file configuration.

  • Page 558: Table 10-11: Multiple Interface Table - Example 2

    Mediant 600 & Mediant 1000 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...

  • Page 559: Table 10-13: Multiple Interface Table - Example 3

    0.0.0.0 CntrlMedia2 Control VLANs are required. The 'Native' VLAN ID is the same VLAN ID as the AudioCodes Management interface (index 0). One routing rule is required to allow remote management from a host in 176.85.49.0/24. Table 10-14: Routing Table - Example 3...
  • Page 560 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83305...

  • Page 561: 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 562: 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'' on page and ''Fixed Mapping of SIP Response to ISDN Release Reason'' on page 564.
  • Page 563 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 564: 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 565 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 566: 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 567: Nfas Interface Id

    SIP User's Manual 11. Advanced PSTN Configuration The NFAS group is identified by an NFAS GroupID number (possible values are 1 to 9). To assign a number of T1 trunks to the same NFAS group, use the ini file parameter NFASGroupNumber_x = groupID (where x is the physical trunk ID (0 to the maximum number of trunks) or the Web interface (refer to ''Configuring the Trunk Settings'' on page 80).

  • Page 568: Working With Dms-100 Switches

    Mediant 600 & Mediant 1000 11.4.2 Working with DMS-100 Switches The DMS-100 switch requires the following NFAS Interface ID definitions: InterfaceID #0 for the Primary trunk InterfaceID #1 for the Backup trunk InterfaceID #2 for a 24 B-channel T1 trunk...

  • Page 569: Creating An Nfas-Related Trunk Configuration

    SIP User's Manual 11. Advanced PSTN Configuration 11.4.3 Creating an NFAS-Related Trunk Configuration The procedures for creating and deleting an NFAS group must be performed in the correct order, as described below. To create an NFAS Group: If there’s a backup (‘secondary’) trunk for this group, it must be configured first. Configure the primary trunk before configuring any NFAS (‘slave’) trunk.

  • Page 570: Automatic Gain Control (Agc)

    Mediant 600 & Mediant 1000 11.6 Automatic Gain Control (AGC) Automatic Gain Control (AGC) adjusts the energy of the output signal to a required level (i.e., volume). This feature compensates for near-far gain differences. AGC estimates the energy of the incoming signal (from the IP or PSTN, determined by the parameter AGCRedirection), calculates the essential gain, and then performs amplification.

  • Page 571: Tunneling Applications

    SIP User's Manual 12. Tunneling Applications Tunneling Applications This section discusses TDM and QISG tunneling, supported by the device. 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 over the IP network (using point-to-point or point-to-multipoint device distributions).
  • Page 572 Mediant 600 & Mediant 1000 For tunneling of E1/T1 CAS trunks, set the protocol type to 'Raw CAS' (ProtocolType = 3 / 9) and enable RFC 2833 CAS relay mode ('CAS Transport Type' parameter is set to 'CAS RFC2833 Relay' - refer to ''Configuring the Voice Settings'' on page 74).
  • Page 573 SIP User's Manual 12. Tunneling Applications TelProfile SigIPDiffServ, TelProfile DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP; TelProfile 1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$; TelProfile 2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$; [\TelProfile] Originating Side: ;E1_TRANSPARENT_31 ProtocolType_0 = 5 ProtocolType_1 = 5 ProtocolType_2 = 5 ProtocolType_3 = 5 ;Channel selection by Phone number.

  • Page 574: Dsp Pattern Detector

    Mediant 600 & Mediant 1000 12.1.1 DSP Pattern Detector For TDM tunneling applications, you can use the DSP pattern detector feature to initiate the echo canceller at call start. The device can be configured to support detection of a specific one-byte idle data pattern transmitted over digital E1/T1 timeslots.
  • Page 575 SIP User's Manual 12. Tunneling Applications 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 576 Mediant 600 & Mediant 1000 Reader’s Notes SIP User's Manual Document #: LTRT-83305...

  • Page 577: Media Server Capabilities

    SIP User's Manual 13. Media Server Capabilities Media Server Capabilities This section provides information on the device's media server capabilities: Multi-party conferencing (refer to ''Conference Server'' on page 577) Playing and recording Announcements (refer to ''Announcement Server'' on page 592) IP-to-IP Transcoding (refer to ''IP-to-IP Transcoding'' on page 616) Voice XML Interpreter (refer to “Voice XML Interpreter”...

  • Page 578: 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: Figure 13-1: Simple Conferencing – SIP Call Flow...

  • Page 579: Creating A Conference

    Media Service is a Conference) and a Unique Conference Identifier (identifying a specific instance of a conference). INVITE sip: conf100@audiocodes.com SIP/2.0 By default, a request to create a conference reserves three resources on the device. It is possible to reserve a larger number of resources in advance by adding the number of required participants to the User Part of the Request-URI.

  • Page 580: 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 581: Joining A Conference

    SIP User's Manual 13. Media Server Capabilities Figure 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 582: 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 583: Figure 13-3: Modifying A Conference - Sip Call Flow

    SIP User's Manual 13. Media Server 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. Figure 13-3: Modifying a Conference - SIP Call Flow Version 5.8 September 2009...

  • Page 584: 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 585: Terminating A Conference

    SIP User's Manual 13. Media Server Capabilities Event notifications are sent in SIP INFO messages, as shown in the example below of XML Response Generated for ASN: <?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>...

  • Page 586: 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). Figure 13-6: Conference Call Flow Example SIP MESSAGE 1: 10.8.29.1:5060 ->...
  • Page 587 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 588 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 589 From: <sip:600@10.8.8.10>;tag=1c201038291 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c1673415884 Call-ID: 1008914574iYgW@10.8.58.6 CSeq: 1 INVITE Contact: <sip:conf100@10.8.58.4> Supported: em,timer,replaces,path 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 v=0 o=AudiocodesGW 886442 597756 IN IP4 10.8.58.4 s=Phone-Call c=IN IP4 10.8.58.4 t=0 0 m=audio 7150 RTP/AVP 4 96...
  • Page 590 Mediant 600 & Mediant 1000 t=0 0 m=audio 6000 RTP/AVP 4 96 a=rtpmap:4 g723/8000 a=fmtp:4 annexa=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:30 a=sendrecv SIP MESSAGE 10: 10.8.58.4:5060 -> 10.8.58.8:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.8.58.8;branch=z9hG4bKaczJpxnnv 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...
  • Page 591 From: <sip:800@10.8.58.8>;tag=1c2419012378 To: <sip:conf100@10.8.58.4;user=phone>;tag=1c3203015250 Call-ID: 150852731NDDC@10.8.58.8 CSeq: 2 BYE Contact: <sip:800@10.8.58.8> 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 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>...

  • Page 592: Announcement Server

    Mediant 600 & Mediant 1000 13.2 Announcement Server The device supports playing and recording of announcements (local Voice Prompts or HTTP streaming) and playing of Call Progress Tones over the IP network. Three different methods are available for playing and recording announcements:...

  • Page 593: Supported Attributes

    SIP User's Manual 13. Media Server Capabilities 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). Delay: defines the delay (in msec) between announcement repetitions.

  • Page 594: Mscml Interface

    Mediant 600 & Mediant 1000 13.2.2 MSCML Interface Media Server Control Markup Language (MSCML), according to IETF draft <draft-vandyke- mscml-06.txt>) 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.

  • Page 595: Operation

    SIP INVITE message with a SIP URI that includes the MSCML Identifier name. For example: INVITE sip:ivr@audiocodes.com SIP/2.0 The left part of the SIP URI includes the MSCML Identifier string ‘ivr’, which can be configured using the ini file (parameter MSCMLID) or Web interface (refer to ''Configuring the IPmedia Parameters'' on page 179).

  • Page 596: Operating With Audio Bundles

    Mediant 600 & Mediant 1000 The device supports basic IVR functions of playing announcements, collecting DTMF digits, and voice stream recording. These services are implemented using the following Request and Response messages: <Play> for playing announcements <PlayCollect> for playing announcements and collecting digits <PlayRecord>...
  • Page 597 SIP User's Manual 13. Media Server Capabilities 13.2.2.2.1 Uploading a Bundle to the Device The audio bundle can be uploaded through FTP, NFS or HTTP. For more information, see the relevant Automatic Update chapter in the Product Reference Manual. 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'...

  • Page 598: Playing Announcements

    Mediant 600 & Mediant 1000 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: id: an optional random number used to synchronize request and response.
  • Page 599 SIP User's Manual 13. Media Server Capabilities extradigittimer: used to enable the following: • Detection of command keys (ReturnKey and EscapeKey). • Not report the shortest match. MGCP Digitmap searches for the shortest possible match. This means that if a digitmap of (123 | 1234) is defined, once the user enters 123, a match is found and a response is sent.

  • Page 600: Playing Announcements And Recording Voice

    Mediant 600 & Mediant 1000 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>...

  • Page 601: Stopping The Playing Of An Announcement

    SIP User's Manual 13. Media Server Capabilities An example is shown below of an MSCML <PlayRecord> Response: <?xml version="1.0" encoding="utf-8"?> <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.

  • Page 602: Signal Events Notifications

    Mediant 600 & Mediant 1000 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 603: Voice Streaming

    SIP User's Manual 13. Media Server Capabilities 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 604 Mediant 600 & Mediant 1000 <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.) The servlet or cgi script can respond by sending a complete audio file or a portion of an audio file.

  • Page 605: Using File Coders With Different Channel Coders

    SIP User's Manual 13. Media Server Capabilities 13.2.3.1.9 Record Files Using LBR You may record a file using low bit rate coders for *.wav and *.raw files. Notes: This feature is relevant for both NFS and HTTP. 13.2.3.1.10 Modifying Streaming Levels Timers Several parameters enable the user to control streaming level timers for NFS and HTTP and also the number of data retransmission when using NFS as the application layer protocol:...

  • Page 606: Maximum Concurrent Playing And Recording

    Mediant 600 & Mediant 1000 WB: Linear PCM 16KHZ Wide Band Coder Note: When recording with an LBR type coder, it is assumed that the same coder is used both as the file coder and the channel coder. Combinations of different LBR coders are currently not supported.

  • Page 607: Lbr Coders Support

    SIP User's Manual 13. Media Server Capabilities 13.2.3.4 LBR Coders Support The following list describes the DSP templates required for using different low bit rate (LBR) coders and their support for *.wav, *.au, and *.raw files. Table 13-5: DSP Templates Coder DSP Templates *.wav file...

  • Page 608: Http Recording Configuration

    Mediant 600 & Mediant 1000 13.2.3.5 HTTP Recording Configuration The HTTP record method (PUT or POST) is configured using the following offline ini parameter: // 0=post (default), 1=put VoiceStreamUploadMethod = 1 The default value is shown below: VoiceStreamUploadPostUri = "/audioupload/servlet/AcAudioUploadServlet"...

  • Page 609: Supported Http Servers

    SIP User's Manual 13. Media Server Capabilities 13.2.3.7 Supported HTTP Servers The following is a list of HTTP servers that are known to be compatible with AudioCodes voice streaming under Linux™: Apache: cgi scripts are used for recording and supporting dynamic URLs.

  • Page 610: Supporting Nfs Servers

    Mediant 600 & Mediant 1000 13.2.3.8 Supporting NFS Servers The table below lists the NFS servers that are known to be compatible with AudioCodes Voice Streaming functionality. Table 13-6: Compatible NFS Servers Operating System Server Versions Solaris™ 5.8 and 5.9...
  • Page 611 If the systems administrator wishes to use a default other than AUTH_SYS in the nfssec.conf file, then you should add "sec=sys" to each line in the dfstab file that is to be shared with an AudioCodes system. For example: > cat /etc/dfs/dfstab...

  • Page 612: Common Troubleshooting

    Mediant 600 & Mediant 1000 13.2.3.8.2 Linux-Based NFS Servers The AudioCodes device uses local UDP ports that are outside of the range of 0..IPPORT_RESERVED(1024). Therefore, when configuring a remote file system to be accessed by an AudioCodes device, use the insecure option in the /etc/exports file. The insecure option allows the nfs daemon to accept mount requests from ports outside of this range.
  • Page 613 13. Media Server Capabilities Problem Probable Cause Corrective Action NFS Voice Streaming Problems Announcement is terminated prematurelyand The AudioCodes media Fix the network problem the Syslog displays the following: 'NFS server has lost or NFS server problem. request aborted … networkError'.

  • Page 614: Announcement Call Flow Example

    The call flow, shown in the following figure, describes SIP messages exchanged between the device (10.33.24.1) and a SIP client (10.33.2.40) requesting to play local announcement #1 (10.8.25.17) using AudioCodes proprietary method. Figure 13-8: Announcement Call Flow 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...
  • Page 615 SIP MESSAGE 4: 10.33.2.40:5060 -> 10.33.24.1:5060 ACK sip:10.33.24.1 SIP/2.0 Via: SIP/2.0/UDP 10.33.2.40;branch=z9hG4bKacnNUEeKP Max-Forwards: 70 From: <sip:103@10.33.2.40>;tag=1c2917829348 To: <sip:annc@10.33.24.1>;tag=1c1528117157 Call-ID: 1414622340oZZq@10.33.2.40 CSeq: 1 ACK Contact: <sip:103@10.33.2.40> Supported: em,timer,replaces,path Allow:REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,IN FO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-4.0 GA/v.4.0 GA Content-Length: 0 Version 5.8 September 2009...

  • Page 616: Ip-To-Ip Transcoding

    Mediant 600 & Mediant 1000 SIP MESSAGE 5: 10.33.24.1:5060 -> 10.33.2.40:5060 BYE sip:103@10.33.2.40 SIP/2.0 Via: SIP/2.0/UDP 10.33.24.1;branch=z9hG4bKacFhtFbFR Max-Forwards: 70 From: <sip:annc@10.33.24.1>;tag=1c1528117157 To: <sip:103@10.33.2.40>;tag=1c2917829348 Call-ID: 1414622340oZZq@10.33.2.40 CSeq: 1 BYE Contact: <sip:10.33.24.1> 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...

  • Page 617: Figure 13-9: Direct Connection (Example)

    SIP User's Manual 13. Media Server Capabilities The device uses two media (DSP) channels for each call, thereby reducing the number of available Transcoding sessions to half of the defined value for MediaChannels. To limit the number of resources available for Transcoding, use the ini file parameter MediaChannels or Web interface (refer to ''Configuring the IPmedia Parameters'' on page 179).

  • Page 618: Voice Xml Interpreter

    Mediant 600 & Mediant 1000 13.4 Voice XML Interpreter The device supports Voice Extensible Markup Language (VoiceXML) version 2.0. VXML is an XML-based scripting language used to prompt and collect information from callers. A VXML-based script may be used to control many types of interactive voice response (IVR) activities, including playing recorded announcements, collecting DTMF digits, recording a caller's voice, recognizing speech (i.e., automatic speech recognition or ASR), and...
  • Page 619 SIP User's Manual 13. Media Server Capabilities service where a prompt is played to the caller, the prescription number is obtained from the caller as speech or DTMF digits, and this data is then saved to an off-board database. There are ramifications in using both these types of scripts. A dynamic script can be customized for each caller, but has to be downloaded and parsed for every call.

  • Page 620: Proprietary Extensions

    To provide the functionality intended by the VXML specification and to extend the functionality of the VXML specification, some proprietary extensions have been included in the AudioCodes VXML Interpreter. These extensions are discussed in the following sections and are intended to enable a VXML script to make use of the advanced audio capabilities provided by the device.

  • Page 621: Audio Extensions

    This reference directs the VXML software to play the audio segment marked with identifier '123'. Using this method of access, the advanced audio structures defined by the AudioCodes Audio Provisioning Server (APS) can be referenced. While these various structures are outside the scope of the current document, they include sets, sequences, and multi- language variables.

  • Page 622: Table 13-8: Say-As Phrase Types

    Mediant 600 & Mediant 1000 The following table lists the supported phrase types, any valid subtypes for the phrases, the expected input format for each phrase type, and any notes for the various phrase types. Table 13-8: Say-as Phrase Types...
  • Page 623 </audio> 13.4.4.2.3 Supplying Values to Provisioned Variables As mentioned previously, the AudioCodes APS provides the capability to provision several types of advanced audio structures, including multi-language variables. A multi-language variable is an instance of one of the supported phrase types such as date and time. The APS assigns a numeric segment identifier to each variable, and the value for the variable can be provided at runtime.

  • Page 624: Language Identifier Support

    Server (APS) User’s Guide. 13.4.4.3 Language Identifier Support The AudioCodes resident VXML engine supports language identifiers as specified by RFC 3066. However, when accessing audio resident on the device using the proprietary extensions described earlier, the country code portion of the identifier is ignored.

  • Page 625: Combining

    The VXML specification supports multiple <audio> elements nested within other elements such as prompts. An example demonstrating this functionality which includes the AudioCodes extensions is useful to show how multiple components can be combined to create a single announcement. The following example shows how an announcement can be constructed that says “Welcome to Acme Corporation.

  • Page 626: Supported Elements And Attributes

    Mediant 600 & Mediant 1000 13.4.7 Supported Elements and Attributes The following status legend should be referenced for all tables in the following subsections: NS: Not Supported PS: Partially Supported S: Supported 13.4.7.1 VoiceXML Supported Elements and Attributes Table 13-10: VoiceXML Supported Elements and Attributes...
  • Page 627 SIP User's Manual 13. Media Server Capabilities Element Parameter Max Size Shadow Variable Status Comments accept next expr event eventexpr message messageexpr fetchaudio fetchtimeout fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <clear> namelist 4 * 32 <disconnect>...
  • Page 628 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments type enum Built-in grammars are supported for recognition against fields, but the match isn't spoken as the built-in type in text-to-speech. slot Default value is the variable name, thus, slot is not needed.
  • Page 629 SIP User's Manual 13. Media Server Capabilities Element Parameter Max Size Shadow Variable Status Comments mode root 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 630 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments <link> next expr event eventexpr message messageexpr dtmf fetchaudio fetchtimeout fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <log> label expr <menu> scope...
  • Page 631 SIP User's Manual 13. Media Server Capabilities Element Parameter Max Size Shadow Variable Status Comments attribute sizes aren't listed. expr cond classid codebase codetype data type archive fetchtimeout fetchhint Default behavior is "safe"; fetch document when it's needed. maxage maxstale <option>...
  • Page 632 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments timeout numeric xml:lang xml:base <property> 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.
  • Page 633 SIP User's Manual 13. Media Server Capabilities Element Parameter Max Size Shadow Variable Status Comments event eventexpr message messageexpr namelist 4 * 32 <script> The script element and all of its attributes are not supported. charset fetchtimeout fetchhint maxage maxstale <subdialog>...
  • Page 634 Mediant 600 & Mediant 1000 Element Parameter Max Size Shadow Variable Status Comments expr namelist 4 * 32 method enum enctype fetchaudio fetchtimeout Fetchhint Default behavior is "safe"; fetch document when it's needed. Maxage maxstale <throw> Event eventexpr message messageexpr <transfer>...
  • Page 635 SIP User's Manual 13. Media Server Capabilities Element Parameter Max Size Shadow Variable Status Comments name$.utterance <value> expr <var> name expr Name <transfer> Expr Cond Dest Only numbers. destexpr Bridge Only Bridge = false type Only type = blind connecttimeout maxtime transferaudio Aaiexpr...

  • Page 636: Srgs And Ssml Support

    Mediant 600 & Mediant 1000 13.4.7.2 SRGS and SSML Support Note that elements associated with either the Speech Recognition Grammar Specification (SRGS) or Speech Synthesis Markup Language (SSML) are used to control the behavior of a remote speech engine for either speech recognition or text-to-speech. These elements would be passed from the VXML interpreter to the remote speech engine and are outside the scope of VXML.

  • Page 637: Voicexml Variables And Events

    SIP User's Manual 13. Media Server Capabilities Platform Properties Status Equivalent ini file parameter or Notes Objectmaxstale 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 because the majority of systems are expected to use DTMF collection and local or streamed announcements as opposed to text- to-speech and speech recognition.
  • Page 638 Mediant 600 & Mediant 1000 Variable/Event Name Status Notes Pre-defined Events 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...

  • Page 639: Ecmascript Support

    13.4.7.5 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 640: Example Of Udt 'Beep' Tone Definition

    Mediant 600 & Mediant 1000 Operand/Operator Examples Status Note Boolean Literals true, false Section 7.8.2, ECMA-262 3rd 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 13.4.8 Example of UDT ‘beep’ Tone Definition The following is an example definition for ‘beep’...

  • Page 641: Open Solution Network (Osn) Server

    SIP User's Manual 14. Open Solution Network (OSN) Server Open Solution Network (OSN) Server This section is intended for customers who wish to install the optional OSN (Open Solution Network) server platform functionality. The device's chassis houses a plug-in OSN Server module for hosting third-party, VoIP applications such as IP-PBX, Pre-Paid, and IP-PBX redundancy.

  • Page 642: Figure 14-1: Connection Module - Cm (Only For Celeron-Based Osn Server)

    Mediant 600 & Mediant 1000 The device's OSN Server package includes the following modules: Connection module (CM) - installed in the front panel: Figure 14-1: Connection Module - CM (Only for Celeron-Based OSN Server) Note: The CM module is applicable only to OSN1.

  • Page 643: Required Working Tools

    SIP User's Manual 14. Open Solution Network (OSN) Server Hard Drive module (HDMX) - housed in the rear panel: Figure 14-4: Hard Drive Module (HDMX) 14.1.1 Required Working Tools The following tools are required for installing the OSN Server module: Phillips screwdriver Flathead screwdriver Wire cutter...

  • Page 644: Installing The Ipmx Module

    Mediant 600 & Mediant 1000 14.1.3 Installing the iPMX Module The iPMX module is installed on the rear panel of the device, as described in the following procedure: To install the iPMX module: On the device's rear panel, remove the black metal cover plates in the first and second slots located on the right side of the power connection, as shown in the figure below.

  • Page 645: Installing The Hdmx Module

    SIP User's Manual 14. Open Solution Network (OSN) Server Insert the iPMX module into the first slot, closest to the power connection, as shown in the figure below. Figure 14-7: Inserting iPMX Module Push the iPMX module into the slot and press on it firmly to ensure it has been fully inserted.

  • Page 646: Replacing The Ipmx Module's Lithium Battery

    Using a flathead screwdriver, tighten the module's mounting pins. 14.1.5 Replacing the iPMX Module's Lithium Battery The iPMX module is equipped with a 3-volt CR-1225 Lithium battery (AudioCodes product number: ACL P/N RBAT00001). Typically, battery life is estimated at two years. However, for various reasons, the battery may last for a shorter duration.

  • Page 647: Figure 14-9: Removing Lithium Battery From Ipmx Module

    SIP User's Manual 14. Open Solution Network (OSN) Server The following procedure describes how to replace the Lithium battery in the iPMX module. To replace the Lithium battery in the iPMX: Remove the iPMX module from the slot in which it's housed in the device rear panel, by performing the following: Using a flathead screwdriver, loosen the module's two lower mounting captive screws.

  • Page 648: Installing Linux™ Operating System

    Mediant 600 & Mediant 1000 14.2 Installing Linux™ Operating System Once the OSN Server modules have been installed in the device, you need to install an operating system (OS) on the OSN server on which the partner application (e.g. IP-PBX) is to run.

  • Page 649: Cabling The Osn1

    SIP User's Manual 14. Open Solution Network (OSN) Server 14.2.2.1 Cabling the OSN1 The OSN1 cabling is performed on the CM module (on the front panel). The serial port is connected to the PC runing Windows™, and the USB port is connected to the external CD- ROM drive.

  • Page 650: Cabling The Osn2

    Mediant 600 & Mediant 1000 14.2.2.2 Cabling the OSN2 The OSN2 cabling is performed on the iPMX module (on the rear panel). The connection to the external CD-ROM drive is done through the USB port; the connection to the PC can be done either through the RS-232 port or through the VGA (USB) port.

  • Page 651: Installing Linux

    SIP User's Manual 14. Open Solution Network (OSN) Server 14.2.3 Installing Linux Once you have cable the device as described in the previous section, you can install the Linux OS. To install Linux: Start a terminal application (e.g. HyperTerminal) on your PC, and create a new connection with the following settings: •...

  • Page 652: Figure 14-13: Enabling System Management Bios

    Mediant 600 & Mediant 1000 Change the System Management BIOS parameter to “Enabled”. Figure 14-13: Enabling System Management BIOS Pess the Esc key to return to the main BIOS window: Figure 14-14: Saving BIOS Settings Choose the Write to CMOS and Exit option, and then press the Y key to save changes and exit.

  • Page 653: Connecting Remotely To Osn Server Running Windows

    SIP User's Manual 14. Open Solution Network (OSN) Server Press the Enter key; the Linux installation begins. Continue installation according to the Linux installation instructions. 14.3 Connecting Remotely to OSN Server Running Windows Typically, for customers requiring Microsoft Windows® operating system (OS), the OSN Server is provided pre-installed with Windows OS.

  • Page 654: Cabling

    Mediant 600 & Mediant 1000 14.3.1 Cabling Before you can connect remotely to the OSN Server, you need to cable a PC (running Remote Desktop Connection) to the OSN Server's iPMX module. Note: The remote PC must be in the same subnet as the OSN server (default IP address 10.10.12).

  • Page 655: Connecting Using Remote Desktop Connection

    SIP User's Manual 14. Open Solution Network (OSN) Server 14.3.2 Connecting using Remote Desktop Connection Once you have cabled the OSN Server's iPMX module, perform the procedure below for connecting the PC remotely to the OSN Server, using the Remote Desktop Connection program.

  • Page 656: Figure 14-17: Entering Ip Address In Remote Desktop Connection

    Mediant 600 & Mediant 1000 Start Microsoft's Remote Desktop Connection program: from the Start menu, point to Programs, to Accessories, to Communications, and then click Remote Desktop Connection. Figure 14-17: Entering IP Address in Remote Desktop Connection In the 'Computer' field, enter the OSN Server's default IP address (i.e., 10.1.10.12).

  • Page 657: Sip Software Package

    M1000_SIP_xxx.cmp Image file containing the software for both FXS and FXO modules. ini Configuration Files SIPgw_M1K.ini Sample ini file for the Mediant 1000 and Mediant 600 devices. M600_Digital_SIP_T1.ini Sample ini file for Mediant 600 E1 devices. M600_Digital_SIP_E1.ini Sample ini file for Mediant 600 T1 devices.
  • Page 658 MIB files, and Utilities) from AudioCodes Web site at www.audiocodes.com/support (customer registration is performed online at this Web site). If you are not a direct customer of AudioCodes, please contact the AudioCodes’ Distributor and Reseller from whom this product was purchased.

  • Page 659: Selected Technical Specifications

    AC impedance matching, hybrid balance, Tx & Rx frequency FXS Capabilities response, Tx & Rx Gains Note: For a specific coefficient file, please contact AudioCodes. Configurable ringing signal: up to four cadences and frequency 10 to 200 Hz. Drive 5 phones per port simultaneously in offhook and Ring states.
  • Page 660 Mediant 600 & Mediant 1000 Function Specification FXO Functionality Short or long haul, up to 7,000 m (24,000 ft.), using 24 AWG line cord 4 ports per FXO module Far-end disconnect detection Lightning and high voltage protection for outdoor operation Programmable Line Characteristics: AC impedance matching, hybrid balance, Tx &...
  • Page 661 SIP User's Manual 16. Selected Technical Specifications Function Specification Simultaneous 3-Way Conferences (Max.) Full-Duplex Parties per Conference Bridge (Max.) Fax/Modem Relay Group 3 fax relay up to 14.4 kbps with auto fallback T.30 (PSTN) and T.38 (IP) compliant, real time fax relay Fax Relay Tolerant network delay (up to 9 seconds round trip) CNG tone detection &...
  • Page 662 Mediant 600 & Mediant 1000 Function Specification Physical Dimensions (W x H x 482.6 mm (19”) x 1U x 350.5 mm (13.8”) Approx. 5 kg (depending on number of installed modules) Weight Supply Voltage and Universal 100 - 240 VAC; 50 - 60 Hz; 1 A max.

  • Page 663: Mediant 600

    Maintenance Web Management via HTTP or HTTPS Telnet 16.2 Mediant 600 The table below lists the main technical specifications of the Mediant 600. Table 16-2: Mediant 600 Functional Specifications Function Specification Interfaces and Capacity 1 or 2 E1/T1/J1 spans using RJ-48c ports...

  • Page 664: Mediant 600 & Mediant

    Mediant 600 & Mediant 1000 Function Specification Lifeline on every FXS factory-preconfigured module FXO Functionality Short or long haul, up to 7,000 m (24,000 ft.), using 24 AWG line cord 4 ports per FXO module Far-end disconnect detection Lightning and high voltage protection for outdoor operation Programmable Line Characteristics: AC impedance matching, hybrid balance, Tx &...
  • Page 665 SIP User's Manual 16. Selected Technical Specifications Function Specification Fax/Modem Relay Group 3 fax relay up to 14.4 kbps with automatic fallback T.30 (PSTN) and T.38 (IP) compliant, real time fax relay Fax Relay Tolerant network delay (up to 9 seconds round trip) CNG tone detection &...
  • Page 666 Mediant 600 & Mediant 1000 Function Specification Physical Dimensions (W x H x 300 mm (11.8”) x 44 mm (1.73") x 272 mm (10.7”) Weight Approx. 2.7 kg (5.95 lbs), depending on number of installed modules Supply Voltage and Universal 100 - 240 VAC; 50 - 60 Hz; 1A max.
  • Page 667 SIP User's Manual 16. Selected Technical Specifications Reader’s Notes Version 5.8 September 2009...
  • Page 668 User's Manual Version 5.8 www.audiocodes.com...

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Mediant 1000

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