Page 1 Mediant™ 800 MSBG Multi‐Service Business Gateway SIP Protocol User's Manual Version 6.2 February 2011 Document # LTRT‐12804 ...
Page 3: Table Of Contents
SIP User's Manual Contents Table of Contents Overview ......................25 Configuration Concepts ................... 27 2.1 Configuration Tools ....................27 2.2 Main Operating Modes ................... 27 2.2.1 Operating in VoIP and Data-Routing Mode ............. 27 2.2.1.1 Configuring Data-Routing LAN Interface ..........28 ...
Page 4 Mediant 800 MSBG 3.3.2.5 Media ..................... 103 3.3.2.6 Services ....................112 3.3.2.7 Applications Enabling ................113 3.3.2.8 Control Network ..................113 3.3.2.9 SIP Definitions ..................130 3.3.2.10 Coders and Profiles ................138 ...
Page 5 SIP User's Manual Contents EMS-Based Management ................373 5.1 Familiarizing yourself with EMS GUI ..............373 5.2 Adding the Device in EMS ................... 374 5.3 Configuring Trunks ....................376 5.3.1 General Trunk Configuration .................376 5.3.2 Configuring ISDN NFAS ..................377 ...
Page 6 Mediant 800 MSBG 8.3.3.2 FXO Device Interworking SIP E911 Calls from Service Provider's IP Network to PSAP DID Lines ................. 424 8.3.3.3 Pre-empting Existing Calls for E911 IP-to-Tel Calls ......427 8.3.4 Configuring DTMF Transport Types ..............428 ...
Page 7 SIP User's Manual Contents 8.4.5.4 Interworking DTMF Methods ..............498 8.4.5.5 Transcoding Modes ................499 8.4.5.6 Coder Restrictions Control ..............499 8.4.5.7 SRTP-RTP Transcoding ................ 501 8.4.5.8 Multiple RTP Media Streams per Call Session ........502 ...
Page 8 Mediant 800 MSBG 9.4 Simple Network Time Protocol Support ............... 619 9.5 Network Configuration ..................620 9.5.1 Multiple Network Interfaces and VLANs ..............620 9.5.1.1 Overview of Multiple Interface Table ............. 621 9.5.1.2 Columns of the Multiple Interface Table ..........622 ...
Page 9 SIP User's Manual Contents 12.3 Debugging and Diagnostics Parameters .............. 665 12.3.1 General Parameters ....................665 12.3.2 Syslog, CDR and Debug Parameters ..............666 12.3.3 Remote Alarm Indication Parameters ..............669 12.3.4 Serial Parameters ....................670 12.4 ...
Page 10 Mediant 800 MSBG 12.12.9.2 T one Detection Parameters ..............826 12.12.9.3 M etering Tone Parameters ..............828 12.12.10 Telephone Keypad Sequence Parameters ............829 12.12.11 General FXO Parameters .................833 12.12.12 FXS Parameters ....................835 ...
Page 11 SIP User's Manual Contents List of Figures Figure 1-1: Typical Application ....................... 26 Figure 2-1: Connections Page ........................ 28 Figure 2-2: Defining LAN Data Routing IP Address ................28 Figure 2-3: Configuring the DHCP Server ....................29 ...
Page 12 Mediant 800 MSBG Figure 3-30: WEB User Accounts Page (for Users with 'Security Administrator' Privileges) ....74 Figure 3-31: Web Security Page ......................76 Figure 3-32: Telnet/SSH Settings Page ....................76 Figure 3-33: Web & Telnet Access List Page - Add New Entry ............. 77 ...
Page 13 SIP User's Manual Contents Figure 3-89: Supplementary Services Page ..................178 Figure 3-90: Keypad Features Page ....................180 Figure 3-91: Metering Tones Page .......................181 Figure 3-92: Charge Codes Table Page ....................182 Figure 3-93: FXO Settings Page ......................183 Figure 3-94: Authentication Page ......................184 ...
Page 14 Mediant 800 MSBG Figure 3-147: Defining a DMZ Host ......................244 Figure 3-148: Configuring Port Triggering ....................245 Figure 3-149: Editing Port Triggering Rule ...................245 Figure 3-150: Defining Trigger Ports ....................245 Figure 3-151: Configuring Website Restrictions ...................246 ...
Page 15 SIP User's Manual Contents Figure 3-206: Defining Network Object Type ..................285 Figure 3-207: Configuring Scheduler Rules ..................285 Figure 3-208: Defining Scheduler Rule Name ..................286 Figure 3-209: Defining Time Segment ....................286 Figure 3-210: Defining Hour Range .....................286 ...
Page 16 Mediant 800 MSBG Figure 3-264: Defining L2TP Properties ....................315 Figure 3-265: L2TP Server Added Successfully ..................315 Figure 3-266: Defining Advanced L2TP Properties ................315 Figure 3-267: Selecting VPN Type for IPSec ..................316 Figure 3-268: Selecting IPSec ......................316 ...
Page 17 SIP User's Manual Contents Figure 5-3: Adding a Region.........................375 Figure 5-4: Defining the IP Address .....................375 Figure 5-5: DS1 Trunks List Table .......................376 Figure 5-6: Trunks Channels Table ......................376 Figure 5-7: General Settings Screen ....................377 Figure 5-8: EMS ISDN Settings Screen ....................378 ...
Page 18 Mediant 800 MSBG Figure 8-43: Configuring Username and Password for Authenticating Channels 5-8 ......477 Figure 8-44: Configuring Account for Registration to ITSP 1 ...............478 Figure 8-45: Configuring ITSP-to-Hunt Group Routin ................478 Figure 8-46: Configuring Hunt Group to ITSP Routing ................478 ...
Page 19 SIP User's Manual Contents Figure 8-102: SAS Redundant Mode in Emergency State (Example) ..........586 Figure 8-103: Flowchart of INVITE from UA's in SAS Normal State ............587 Figure 8-104: Flowchart of INVITE from Primary Proxy in SAS Normal State ........588 ...
Page 20 Mediant 800 MSBG Table of Tables Table 2-1: Default LAN Data-Routing IP Address .................. 28 Table 3-1: Description of Toolbar Buttons ....................44 Table 3-2: ini File Parameter for Welcome Login Message ..............56 Table 3-3: Areas of the Home Page ....................... 60 ...
Page 21 SIP User's Manual Contents Table 8-10: MLPP Structure .........................529 Table 8-11: Reason Structure ......................530 Table 8-12: URL Structure........................530 Table 8-13: Enum Agent Role ......................531 Table 8-14: Enum Event Package ......................531 Table 8-15: Enum MLPP Reason Type ....................532 ...
Page 22 Mediant 800 MSBG Table 12-22: RADIUS Parameters .......................677 Table 12-23: SNMP Parameters ......................679 Table 12-24: SIP Media Realm Parameters ..................682 Table 12-25: Proxy, Registration and Authentication SIP Parameters ..........683 Table 12-26: SIP Network Application Parameters ................696 ...
Page 23: Weee Eu Directive
This document is subject to change without notice. Date Published: February-21-2011 Trademarks AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP Sounds Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect, VoIPerfectHD, What’s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or...
Page 24: Related Documentation
Related Documentation Manual Name SIP CPE Release Notes Product Reference Manual for SIP CPE Devices Mediant 800 MSBG Installation Manual MSBG CLI Reference Guide Note: Throughout this manual, unless otherwise specified, the term device refers to the Mediant 800 MSBG.
Page 25: Overview
SIP User's Manual 1. Overview Overview The Mediant 800 Multi-Service Business Gateway (MSBG) is a networking device that combines multiple service functions such as a Media Gateway, Session Border Controller (SBC), Data Router and Firewall, LAN switch, WAN access, Stand Alone Survivability (SAS) and an integrated general-purpose server.
Page 26: Figure 1-1: Typical Application
Mediant 800 MSBG The device also provides an integrated Open Solution Network (OSN) Server module. The OSN can host a variety of third-party applications such as IP-PBX, Call Center, and Conferencing. Figure 1-1: Typical Application The device provides Foreign Exchange Station (FXS) and/or Foreign Exchange Office (FXO) telephony module interfaces, depending on ordered hardware configuration.
Page 27: Configuration Concepts
Web browser (described in ''Web-based Management'' on page 41). A configuration ini file loaded to the device (see ''ini File Configuration'' on page 367). AudioCodes’ Element Management System (see ''Element Management System (EMS)'' on page 373). Simple Network Management Protocol (SNMP) browser software (refer to the Product Reference Manual).
Page 28: Configuring Data-Routing Lan Interface
Mediant 800 MSBG 2.2.1.1 Configuring Data-Routing LAN Interface The default IP addresses of the LAN data-routing interface is listed in the table below. Table 2-1: Default LAN Data-Routing IP Address Parameter Default Value IP Address 192.168.0.1 Subnet Mask 255.255.255.0 Default Gateway 0.0.0.0...
Page 29: Configuring Device's Dhcp Server
SIP User's Manual 2. Configuration Concepts 2.2.1.2 Configuring Device's DHCP Server The device's embedded DHCP server for the LAN is enabled, and with default IP pool addresses relating to the default subnet LAN. After reconfiguring the LAN IP addresses, the IP pool addresses should be changed accordingly. You can either disable the DHCP server or modify the IP address pool.
Page 30: Figure 2-4: Selecting Wan Connection
Mediant 800 MSBG To assign a WAN IP address: Cable the device to the WAN network (i.e., ADSL or Cable modem), using the WAN port. Access the device's Web interface with the Voice and Management IP address. Access the 'Settings' page (Configuration tab > Data menu > WAN Access >...
Page 31: Assign Wan Interface To Voip Traffic
SIP User's Manual 2. Configuration Concepts 2.2.1.4 Assign WAN Interface to VoIP Traffic Once you have defined the WAN IP address for the data-routing interface, you then need to associate it with VoIP traffic (i.e., SIP signaling and media / RTP interfaces). The available WAN interfaces depend on the hardware configuration (i.e., Ethernet, T1, or SHDSL) and/or whether VLANs are defined for the WAN interface.
Page 32: Configuring Quality Of Service
Mediant 800 MSBG Open the 'SIP Media Realm Table' page (Configuration tab > VoIP menu > Media submenu > Media Realm Configuration) and define media interface(s) on the WAN interface. Figure 2-7: Assigning WAN Interface to Media Realm Define SRDs and associate them with these SIP signaling and media interfaces.
Page 33: Figure 2-9: Selecting Device For Traffic Shaping
SIP User's Manual 2. Configuration Concepts Click the New button; the following page appears. Figure 2-9: Selecting Device for Traffic Shaping From the 'Device' drop-down list, select 'Default WAN device', and then click OK; the following page appears: Figure 2-10: Defining Traffic Shaping In the ‘Tx Bandwidth’...
Page 34: Figure 2-12: Defining Shaping Class (For Voip Tx Traffic)
Mediant 800 MSBG Click the newly added class name; the following page appears: Figure 2-12: Defining Shaping Class (for VoIP Tx Traffic) Configure the following: From the 'Queue Priority' drop-down list, select '0 (Highest)', i.e., the highest priority. In the 'Bandwidth - Reserved' field, enter "1000" (i.e., 1 Mbps).
Page 35: Figure 2-14: Match Rules Page
SIP User's Manual 2. Configuration Concepts 2.2.1.5.3 Defining VoIP Traffic Matching Rules Once you have defined the VoIP Tx traffic shaping class (e.g., "VOIP Tx") in ''Defining VoIP Tx Shaping Class'' on page 33, you need to define traffic matching rules (QoS outbound rules) for VoIP RTP media traffic as well as for SIP signaling traffic, and then assign the shaping class to these traffic rules.
Page 36: Figure 2-15: Adding A Traffic Priority Rule
Mediant 800 MSBG Under the 'QoS Output Rules' table, click the New button corresponding to the 'WAN Ethernet Rules' rule ID; the following page appears: Figure 2-15: Adding a Traffic Priority Rule Add a new output traffic rule for VoIP SIP signaling to the WAN: From the 'Protocol' drop-down list, select ‘Show All Services’...
Page 37: Figure 2-18: Configured Ports For Incoming Sip
SIP User's Manual 2. Configuration Concepts Define the TCP and UDP ports, and then click OK; the following page appears displaying the configured ports. Figure 2-18: Configured Ports for Incoming SIP Click OK again. Perform steps 2b through 2e to configure ports for outgoing SIP packets. Under the 'Operation' group, select the 'Set Tx Class Name' check box, and then from the corresponding drop-down list, select the traffic shaping class 'VOIP Tx', which you defined for Rx packets (in ''Defining VoIP Tx Shaping...
Page 38: Configuring Virtual Routing And Forwarding
Mediant 800 MSBG Add a new traffic matching rule for transmitted (Tx) VoIP RTP packets to the WAN. Perform steps 2 through 3, except for the 'Protocol' group, select the protocol 'RTP' and only port 'UDP', as shown below. Figure 2-20: Matching Rule for Received RTP Traffic The final traffic matching rule configuration for WAN Tx RTP and SIP signaling is shown below.
Page 39: Enabling Remote Http/S Web Management
SIP User's Manual 2. Configuration Concepts 2.2.1.7 Enabling Remote HTTP/S Web Management If you want to access the device’s Web interface remotely through HTTP or HTTPS, you need to define the WAN HTTP/S port. To define WAN HTTP/S port for remote Web management: Open the 'WEB Security Settings' page (Configuration tab >...
Page 40: Figure 2-24: Multiple Interface Table
Mediant 800 MSBG Configure VoIP IP network interfaces in the ‘Multiple Interface’ table (Configuration tab > VoIP menu > Network Settings > IP Settings) - see ''Configuring IP Interface Settings'' on page 83. The configuration depends on whether or not you want to...
Page 41: 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 42: Getting Acquainted With The Web Interface
Mediant 800 MSBG Getting Acquainted with the Web Interface This section describes the Web interface with regards to its graphical user interface (GUI) and basic functionality. 3.1.1 Computer Requirements To use the device's Web interface, the following is required: A connection to the Internet network (World Wide Web).
Page 43: Figure 3-1: Login Screen
SIP User's Manual 3. Web-Based Management To access the Web interface: Open a standard Web browser application. In the Web browser's Uniform Resource Locator (URL) field, specify the device's IP address (e.g., http://10.1.10.10); the Web interface's Login screen appears, as shown in the figure below: Figure 3-1: Login Screen In the 'User Name' and 'Password' fields, enter the case-sensitive, user name and...
Page 44: Areas Of The Gui
Mediant 800 MSBG 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: 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 45: Navigation Tree
SIP User's Manual 3. Web-Based Management Icon Button Description Name Device Opens a drop-down menu list with frequently needed commands: Actions Load Configuration File: opens the 'Configuration File' page for loading an ini file (see ''Backing Up and Loading Configuration File'' on page 344).
Page 46: Displaying Navigation Tree In Basic And Full View
Mediant 800 MSBG Figure 3-4: Navigation Tree To view menus in the Navigation tree: On the Navigation bar, select the required tab: • Configuration (see ''Configuration Tab'' on page 65) • Maintenance (see ''Maintenance Tab'' on page 333) • Status & Diagnostics (see ''Status & Diagnostics Tab'' on page 346)
Page 47: Showing / Hiding The Navigation Pane
SIP User's Manual 3. Web-Based Management To toggle between Full and Basic view: Select the Basic option (located below the Navigation bar) to display a reduced menu tree; select the Full option to display all the menus. By default, the Basic option is selected.
Page 48: Working With Configuration
Mediant 800 MSBG Figure 3-6: Show / Hide Navigation Tree 3.1.6 Working with Configuration Pages The configuration pages contain the parameters for configuring the device. The configuration pages are displayed in the Work pane, which is located to the right of the Navigation pane.
Page 49: Viewing Parameters
SIP User's Manual 3. Web-Based Management Notes: • You can also access certain pages from the Device Actions button located on the toolbar (see ''Toolbar'' on page 44). • To view all the menus in the Navigation tree, ensure that the Navigation tree is in 'Full' view (see ''Displaying Navigation Tree in Basic and Full View'' on page 46).
Page 50: Figure 3-7: Toggling Between Basic And Advanced View
Mediant 800 MSBG 3.1.6.2.1 Displaying Basic and Advanced Parameters Some pages provide you with an Advanced Parameter List / Basic Parameter List toggle button that allows you to show or hide advanced parameters (in addition to displaying the basic parameters). This button is located on the top-right corner of the page...
Page 51: Modifying And Saving Parameters
SIP User's Manual 3. Web-Based Management 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.
Page 52: Entering Phone Numbers
Mediant 800 MSBG To save configuration changes on a page to the device's volatile memory (RAM): Click the Submit button, which is located near the bottom of the page in which you are working; modifications to parameters with on-the-fly capabilities are immediately applied to the device and take effect;...
Page 53: Working With Tables
SIP User's Manual 3. Web-Based Management 3.1.6.5 Working with Tables The Web interface includes many configuration pages that provide tables for configuring the device. Some of these tables provide the following command buttons: Add Index: adds an index entry to the table. Duplicate: duplicates a selected, existing index entry.
Page 54: Searching For Configuration Parameters
Mediant 800 MSBG To edit an existing index table entry: In the 'Index' column, select the index corresponding to the table row that you want to edit. Click Edit; the fields in the corresponding index row become available. Modify the values as required, and then click Apply; the new settings are applied.
Page 55: Figure 3-13: Searched Result Screen
SIP User's Manual 3. Web-Based Management To search for ini file parameters configurable in the Web interface: On the Navigation bar, click the Search tab; the Search engine appears in the Navigation pane. In the 'Search' field, enter the parameter name or sub-string of the parameter name that you want to search.
Page 56: Creating A Login Welcome Message
Mediant 800 MSBG 3.1.8 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 57: Getting Help
SIP User's Manual 3. Web-Based Management 3.1.9 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 58: 3.1.10 Logging Off The Web Interface
Mediant 800 MSBG 3.1.10 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, see User Accounts. To log off the Web interface: On the toolbar, click the Log Off button;...
Page 59: Using The Home Page
SIP User's Manual 3. Web-Based Management Using the Home Page The 'Home' page provides you with a graphical display of the device's front panel, displaying color-coded status icons for monitoring the functioning of the device. The 'Home' page also displays general device information (in the 'General Information' pane) such as the device's IP address and firmware version.
Page 60: Table 3-3: Areas Of The Home Page
Mediant 800 MSBG • LOCKED - device is locked (i.e. no new calls are accepted) • UNLOCKED - device is not locked • SHUTTING DOWN - device is currently shutting down To perform these operations, see ''Maintenance Actions'' on page 333.
Page 61 SIP User's Manual 3. Web-Based Management Item # Description For trunk ports, you can view the status of trunk channels by clicking the trunk port icon (see Viewing Trunks' Channels on page 64). If you right-click a port, a shortcut menu appears allowing you to perform the following: (Analog ports only) Reset the channel port (see Resetting an Analog Channel on page 62) View the channel's port settings (see ''Viewing Analog Port Information'' on page...
Page 62: Assigning A Port Name
Mediant 800 MSBG 3.2.1 Assigning a Port Name The 'Home' page allows you to assign an arbitrary name or a brief description to each port. This description appears as a tooltip when you move your mouse over the port. To add a port description: Click the required port icon;...
Page 63: Viewing Analog Port Information
SIP User's Manual 3. Web-Based Management 3.2.3 Viewing Analog Port Information The 'Home' page allows you to view detailed information on a specific FXS or FXO analog port such as RTP/RTCP and voice settings. To view detailed port information: Click the port for which you want to view port settings; the shortcut menu appears. Figure 3-22: Shortcut Menu for Viewing Port Information From the shortcut menu, click Port Settings;...
Page 64: Viewing Trunk Channels
Mediant 800 MSBG 3.2.4 Viewing Trunk 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 trunk port icon of whose status you want to view;...
Page 65: 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. This tab provides the following main menus: System (see ''System Settings'' on page 65) VoIP (see VoIP Settings on page 83) Data (see Data Settings on page 222) 3.3.1 System Settings...
Page 66: Configuring Nfs Settings
Mediant 800 MSBG Configure the parameters as required. For configuring NFS, under the 'NFS Settings' group, click the NFS Table button; the 'NFS Settings' page appears. For a description of configuring this page, see ''Configuring NFS Settings'' on page 66.
Page 67: Table 3-5: Nfs Settings Parameters
SIP User's Manual 3. Web-Based Management Table 3-5: NFS Settings Parameters Parameter Description Index The row index of the remote file system. The valid range is 1 to 16. Host Or IP The domain name or IP address of the NFS server. If a domain name is provided, a DNS server must be configured.
Page 68: Configuring Syslog Settings
Mediant 800 MSBG 3.3.1.3 Configuring Syslog Settings The 'Syslog Settings' page allows you to configure the device's embedded Syslog client. For a detailed description on the Syslog parameters, see ''Syslog, CDR and Debug Parameters'' on page 666. For a detailed description on Syslog messages and using third- party Syslog servers, refer to the Product Reference Manual.
Page 69: Configuring Regional Settings
SIP User's Manual 3. Web-Based Management 3.3.1.4 Configuring Regional Settings The 'Regional Settings' page allows you to define and view the device's internal date and time. To configure the device's date and time: Open the 'Regional Settings' page (Configuration tab > System menu > Regional Settings).
Page 70: Figure 3-29: Certificates Signing Request Page
Mediant 800 MSBG 3.3.1.5.1 Server Certificate Replacement The device is supplied with a working Secure Socket Layer (SSL) configuration consisting of a unique self-signed server certificate. If an organizational Public Key Infrastructure (PKI) is used, you may wish to replace this certificate with one provided by your security administrator.
Page 71: Client Certificates
SIP User's Manual 3. Web-Based Management -----BEGIN CERTIFICATE----- MIIDkzCCAnugAwIBAgIEAgAAADANBgkqhkiG9w0BAQQFADA/MQswCQYDVQQGEwJGUj ETMBEGA1UEChMKQ2VydGlwb3N0ZTEbMBkGA1UEAxMSQ2VydGlwb3N0ZSBTZXJ2ZXVy MB4XDTk4MDYyNDA4MDAwMFoXDTE4MDYyNDA4MDAwMFowPzELMAkGA1UEBhMCRlIxEz ARBgNVBAoTCkNlcnRpcG9zdGUxGzAZBgNVBAMTEkNlcnRpcG9zdGUgU2VydmV1cjCC ASEwDQYJKoZIhvcNAQEBBQADggEOADCCAQkCggEAPqd4MziR4spWldGRx8bQrhZkon WnNm`+Yhb7+4Q67ecf1janH7GcN/SXsfx7jJpreWULf7v7Cvpr4R7qIJcmdHIntmf7 JPM5n6cDBv17uSW63er7NkVnMFHwK1QaGFLMybFkzaeGrvFm4k3lRefiXDmuOe+FhJ gHYezYHf44LvPRPwhSrzi9+Aq3o8pWDguJuZDIUP1F1jMa+LPwvREXfFcUW+w== -----END CERTIFICATE----- In the 'Certificates Files' group, click the Browse button corresponding to 'Send Server Certificate...', navigate to the cert.txt file, and then click Send File. After the certificate successfully loads to the device, save the configuration (see ''Saving Configuration'' on page 336) and restart the device;...
Page 72 Mediant 800 MSBG In the 'Certificates Files' group, click the Browse button corresponding to 'Send "Trusted Root Certificate Store" file ...', navigate to the file, and then click Send File. When the operation is complete, set the HTTPSRequireClientCertificate ini file parameter to 1.
Page 73: Management Settings
SIP User's Manual 3. Web-Based Management 3.3.1.6 Management Settings The Management submenu includes the following: WEB User Accounts item (see ''Configuring Web User Accounts'' on page 73) Web Security Settings item (see ''Configuring Web Security Settings'' on page 76) Telnet/SSH Settings item (see ''Configuring Telnet and SSH Settings'' on page 76) WEB &...
Page 74: Figure 3-30: Web User Accounts Page (For Users With 'Security Administrator' Privileges)
Mediant 800 MSBG The default attributes for the two Web user accounts are shown in the following table: Table 3-7: Default Attributes for the Web User Accounts Account / Attribute User Name Password Access Level (Case-Sensitive) (Case-Sensitive) Primary Account Admin...
Page 75 SIP User's Manual 3. Web-Based Management Notes: • The access level of the primary Web user account is 'Security Administrator', which cannot be modified. • The access level of the secondary account can only be modified by the primary account user or a secondary account user with 'Security Administrator' access level.
Page 76: Figure 3-31: Web Security Page
Mediant 800 MSBG 3.3.1.6.2 Configuring Web Security Settings The 'WEB Security Settings' page is used to define a secure Web access communication method. For a description of these parameters, see ''Web and Telnet Parameters'' on page 662. To define Web access security: Open the 'WEB Security Settings' page (Configuration tab >...
Page 77: Figure 3-33: Web & Telnet Access List Page - Add New Entry
SIP User's Manual 3. Web-Based Management 3.3.1.6.4 Configuring Web and Telnet Access List The 'Web & Telnet Access List' page is used to define IP addresses (up to ten) that are permitted to access the device's Web, Telnet, and SSH interfaces. Access from an undefined IP address is denied.
Page 78: Figure 3-35: Radius Parameters Page
Mediant 800 MSBG 3.3.1.6.5 Configuring RADIUS Settings The 'RADIUS Settings' page is used for configuring the Remote Authentication Dial In User Service (RADIUS) accounting parameters. For a description of these parameters, see ''Configuration Parameters Reference'' on page 653. To configure RADIUS: Open the ‘RADIUS Settings' page (Configuration tab >...
Page 79: Figure 3-36: Snmp Community String Page
SIP User's Manual 3. Web-Based Management To configure the SNMP community strings: Open the 'SNMP Community String' page (Maintenance tab > System menu > Management submenu > SNMP submenu > SNMP Community String). Figure 3-36: SNMP Community String Page Configure the SNMP community strings parameters according to the table below. Click the Submit button to save your changes.
Page 80: Figure 3-37: Snmp Trap Destinations Page
Mediant 800 MSBG 3.3.1.6.6.2 Configuring SNMP Trap Destinations The 'SNMP Trap Destinations' page allows you to configure up to five SNMP trap managers. To configure SNMP trap destinations: Open the 'SNMP Trap Destinations' page (Maintenance tab > System menu >...
Page 81: Figure 3-38: Snmp Trusted Managers
SIP User's Manual 3. Web-Based Management 3.3.1.6.6.3 Configuring SNMP Trusted Managers The 'SNMP Trusted Managers' page allows you to configure up to five SNMP Trusted Managers, based on IP addresses. By default, the SNMP agent accepts SNMP Get and Set requests from any IP address, as long as the correct community string is used in the request.
Page 82: Table 3-10: Snmp V3 Users Parameters
Mediant 800 MSBG Click the Apply button to save your changes. To save the changes, see ''Saving Configuration'' on page 336. Notes: • For a description of the web interface's table command buttons (e.g., Duplicate and Delete), see ''Working with Tables'' on page 53.
Page 83: Voip Settings
SIP User's Manual 3. Web-Based Management 3.3.2 VoIP Settings The VoIP menu includes the following main submenus: Network (see ''Network'' on page 83) TDM (see TDM on page 94) Security (see ''Security'' on page 94) PSTN (see PSTN on page 98) Media (see ''Media'' on page 103) Services (see Configuring LDAP Settings on page 112) Applications Enabling (see Enabling Applications on page 113)
Page 84: Figure 3-40: Multiple Interface Table Page
Mediant 800 MSBG Application type: OAMP + Media + Control IP address: 192.168.0.2 with prefix length 24 (i.e., subnet mask 255.255.255.0) Default gateway: 192.168.0.1 Name: "Voice" VLAN ID: 1 When using data-routing functionality, the network interfaces for the data-router are configured using the Data Settings menu (see Data Settings on page 222).
Page 85: Table 3-11: Multiple Interface Table Parameters Description
SIP User's Manual 3. Web-Based Management Click Done to validate the interface. If the interface is not valid (e.g., if it overlaps with another interface in the table or if it does not adhere to the other rules as summarized in ''Multiple Interface Table Configuration Summary and Guidelines'' on page 627), a warning message is displayed.
Page 86 Mediant 800 MSBG Parameter Description Web: Interface Mode Determines the method that this interface uses to calculate its [InterfaceTable_InterfaceMode] IP address. [3] IPv6 Manual Prefix = IPv6 manual prefix IP address assignment. [4] IPv6 Manual = IPv6 manual IP address assignment.
Page 87 SIP User's Manual 3. Web-Based Management Parameter Description Web/EMS: Interface Name Defines a string (up to 16 characters) to name this interface. [InterfaceTable_InterfaceName] This name is displayed in management interfaces (Web, CLI and SNMP) for clarity (and has no functional use), as well as in the 'SIP Media Realm' and 'SIP Interface' tables.
Page 88: Figure 3-41: Ip Routing Table Page
Mediant 800 MSBG 3.3.2.1.2 Configuring the IP Routing Table The 'IP Routing Table' page allows you to define up to 30 static IP routing rules for the device. These rules can be associated with a network interface (defined in the Multiple Interface table) and therefore, the routing decision is based on the source subnet/VLAN.
Page 89 SIP User's Manual 3. Web-Based Management Parameter Description The address of the host/network you want to reach is determined by an AND operation that is applied to the fields 'Destination IP Address' and 'Destination Mask'. For example, to reach the network 10.8.x.x, enter 10.8.0.0 in the field 'Destination IP Address' and 255.255.0.0 in the field 'Destination Mask'.
Page 90: Figure 3-42: Diffserv Table Page
Mediant 800 MSBG To configure QoS: Open the 'Diff Serv Table' page (Configuration tab > VoIP menu > Network submenu > QoS Settings). Figure 3-42: DiffServ Table Page Configure DiffServ to VLAN priority mapping (Layer-2 QoS): Enter an index entry, and then click Add.
Page 91: Figure 3-43: Dns Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.1.4 DNS The DNS submenu includes the following items: DNS Settings (refer to ''Configuring DNS Settings'' on page 91) Internal DNS Table (refer to ''Configuring the Internal DNS Table'' on page 91) Internal SRV Table (refer to ''Configuring the Internal SRV Table'' on page 92) 3.3.2.1.4.1 Configuring DNS Settings The 'DNS Settings' page defines the VoIP Domain Name System (DNS) server IP addresses.
Page 92: Figure 3-44: Internal Dns Table Page
Mediant 800 MSBG To configure the internal DNS table: Open the 'Internal DNS Table' page (Configuration tab > VoIP menu > Network submenu > DNS submenu > Internal DNS Table). Figure 3-44: Internal DNS Table Page In the 'Domain Name' field, enter the host name to be translated. You can enter a string of up to 31 characters.
Page 93: Figure 3-45: Internal Srv Table Page
SIP User's Manual 3. Web-Based Management To configure the Internal SRV table: Open the 'Internal SRV Table' page (Configuration tab > VoIP menu > Network submenu > DNS submenu > Internal SRV Table). Figure 3-45: Internal SRV Table Page In the 'Domain Name' field, enter the host name to be translated. You can enter a string of up to 31 characters.
Page 94: Tdm
Mediant 800 MSBG 3.3.2.2 The TDM submenu contains the following item: TDM (see Configuring TDM Bus Settings on page 94) 3.3.2.2.1 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, see ''Clock Settings'' on page 637.
Page 95: Figure 3-47: Firewall Settings Page
SIP User's Manual 3. Web-Based Management Limit traffic to a pre-defined rate (blocking the excess) Limit traffic to specific protocols, and specific port ranges on the device For each packet received on the network interface, the table is scanned from the top down until a matching rule is found.
Page 96: Table 3-13: Internal Firewall Parameters
Mediant 800 MSBG The previous figure shows the following access list settings: Rule #1: traffic from the host 'mgmt.customer.com' destined to TCP ports 0 to 80, is always allowed. Rule #2: traffic from the 192.xxx.yyy.zzz subnet, is limited to a rate of 40 Kbytes per second (with an allowed burst of 50 Kbytes).
Page 97 SIP User's Manual 3. Web-Based Management Parameter Description (network mask of 255.0.0.0). A value of 16 corresponds to IPv4 subnet class B (network mask of 255.255.0.0). A value of 24 corresponds to IPv4 subnet class C (network mask of 255.255.255.0). The IP address of the sender of the incoming packet is trimmed in accordance with the prefix length (in bits) and then compared to the parameter ‘Source IP’.
Page 98: Pstn
Mediant 800 MSBG Parameter Description bytes/sec, then this allowance would be consumed within 10 seconds, after which all traffic exceeding the allocated 40000 bytes/sec is dropped. If the actual traffic rate then slowed to 30000 bytes/sec, then the allowance would be replenished within 5 seconds.
Page 99: Figure 3-49: Cas State Machine Page
SIP User's Manual 3. Web-Based Management 3.3.2.4.1 Configuring CAS State Machines The 'CAS State Machine' page allows you to modify various timers and other basic parameters to define the initialization of the CAS state machine without changing the state machine itself (no compilation is required). The change doesn't affect the state machine itself, but rather the configuration.
Page 100: Table 3-14: Cas State Machine Parameters Description
Mediant 800 MSBG 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 101: Figure 3-50: Trunk Scroll Bar (Used Only As An Example)
SIP User's Manual 3. Web-Based Management 3.3.2.4.2 Configuring Trunk Settings The 'Trunk Settings' page allows you to configure the device's trunks. This includes selecting the PSTN protocol and configuring related parameters. Some parameters can be configured when the trunk is in service, while others require you to take the trunk out of service (by clicking the Stop button).
Page 102: Figure 3-51: Trunk Scroll Bar (Used Only As An Example)
Mediant 800 MSBG On the top of the page, a bar with Trunk number icons displays the status of each trunk, according to the following color codes: • Grey: Disabled • Green: Active • Yellow: RAI alarm (also appears when you deactivate a Trunk by clicking the Deactivate button) •...
Page 103: Media
SIP User's Manual 3. Web-Based Management Configure the trunk parameters as required. Click the Apply Trunk Settings button to apply the changes to the selected trunk (or click Apply to All Trunks to apply the changes to all trunks); the Stop Trunk button replaces Apply Trunk Settings and the ‘Trunk Configuration State’...
Page 104: Figure 3-52: Voice Settings Page
Mediant 800 MSBG 3.3.2.5.1 Configuring Voice Settings The 'Voice Settings' page configures various voice parameters such as voice volume, silence suppression, and DTMF transport type. For a detailed description of these parameters, see ''Configuration Parameters Reference'' on page 653. To configure the voice parameters: Open the 'Voice Settings' page (Configuration tab >...
Page 105: Figure 3-53: Fax/Modem/Cid Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.5.2 Configuring 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, see ''Configuration Parameters Reference'' on page 653. To configure the fax, modem, and CID parameters: Open the 'Fax/Modem/CID Settings' page (Configuration tab >...
Page 106: Figure 3-54: Rtp/Rtcp Settings Page
Mediant 800 MSBG 3.3.2.5.3 Configuring RTP/RTCP Settings The 'RTP/RTCP Settings' page configures the Real-Time Transport Protocol (RTP) and Real-Time Transport (RTP) Control Protocol (RTCP) parameters. For a detailed description of the parameters appearing on this page, refer to ''Configuration Parameters Reference'' on page 653.
Page 107: Figure 3-55: Ipmedia Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.5.4 Configuring IP Media Settings The 'IPMedia Settings' page allows you to configure the IP media parameters. For a detailed description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 653. To configure the IP media parameters: Open the 'IPMedia Settings' page (Configuration tab >...
Page 108: Figure 3-56: General Media Settings Page
Mediant 800 MSBG 3.3.2.5.5 Configuring General Media Settings The 'General Media Settings' page allows you to configure various media parameters. For a detailed description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 653. To configure general media parameters: Open the 'General Media Settings' page (Configuration tab >...
Page 109: Figure 3-58: Sip Media Realm Table Page
SIP User's Manual 3. Web-Based Management 3.3.2.5.7 Configuring Media Realms The 'SIP Media Realm Table' page allows you to define a pool of up to 64 SIP media interfaces, termed Media Realms. This table allows you to divide a Media-type interface (defined in the 'Multiple Interface' table - see ''Configuring IP Interface Settings'' on page 83) into several realms, where each realm is specified by a UDP port range.
Page 110 Mediant 800 MSBG Parameter Description IPv4 Interface Name Associates the IPv4 interface to the Media Realm. [CpMediaRealm_IPv4IF] Note: The name of this interface must be exactly (i.e., case- sensitive etc.) as configured in the 'Multiple Interface' table (InterfaceTable parameter). For the VoIP WAN IP address, you must enter the string "WAN"...
Page 111: Figure 3-59: Media Security Page
SIP User's Manual 3. Web-Based Management 3.3.2.5.8 Configuring Media Security The 'Media Security' page allows you to configure media security. For a detailed description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 653. To configure media security: Open the 'Media Security' page (Configuration tab >...
Page 112: Services
Mediant 800 MSBG 3.3.2.6 Services The Services submenu contains the following page item: LDAP Settings (see ''Configuring LDAP Settings'' on page 112) 3.3.2.6.1 Configuring LDAP Settings The 'LDAP Settings' page is used for configuring the Lightweight Directory Access Protocol (LDAP) parameters. For a description of these parameters, see ''Configuration Parameters Reference'' on page 653.
Page 113: Applications Enabling
SIP User's Manual 3. Web-Based Management 3.3.2.7 Applications Enabling 3.3.2.7.1 Enabling Applications The 'Applications Enabling' page allows you to enable the following applications: Stand-Alone Survivability (SAS) application Session Border Control (SBC) application Notes: • This page displays the application only if the device is installed with the relevant Software Upgrade Key supporting the application (see ''Loading Software Upgrade Key'' on page 339).
Page 114 Mediant 800 MSBG 3.3.2.8.1 Configuring SRD Table The 'SRD Settings' page allows you to configure up to 32 signaling routing domains (SRD). An SRD is configured with a unique name and assigned a Media Realm (defined in the 'SIP Media Realm' table - see ''Configuring Media Realms'' on page 109). In addition, other attributes such as media anchoring and user registration can also be configured.
Page 115: Figure 3-62: Srd Settings Page
SIP User's Manual 3. Web-Based Management To configure SRDs: Open the 'SRD Settings' page (Configuration tab > VoIP menu > Control Network submenu > SRD Table). Figure 3-62: SRD Settings Page From the 'SRD Index' drop-down list, select an index for the SRD, and then configure it according to the table below.
Page 116 Mediant 800 MSBG Parameter Description Media Realm Determines the media ports associated with the specific SRD. This is [SRD_MediaRealm] the name as defined in the 'SIP Media Realm' table (CpMediaRealm). The valid value is a string of up to 40 characters.
Page 117 SIP User's Manual 3. Web-Based Management Parameter Description Enable Un-Authenticated Determines whether the device blocks REGISTER requests from new Registrations users (i.e., users not registered in the device's registration database) [SRD_EnableUnAuthenticat when the destination IP Group is of type USER. edRegistrations] [0] No = The device sends REGISTER requests to the SIP proxy server and only if authenticated by the server does the device add...
Page 118: Figure 3-63: Sip Interface Table Page
Mediant 800 MSBG To configure the SIP Interface table: Open the 'SIP Interface Table' page (Configuration tab > VoIP menu > Control Network submenu > SIP Interface Table). Figure 3-63: SIP Interface Table Page Add an entry and then configure it according to the table below.
Page 119 SIP User's Manual 3. Web-Based Management Parameter Description TCP Port Determines the listening TCP port. [SIPInterface_TCPPort] The valid range is 1 to 65534. The default is 5060. Note: This port must be outside of the RTP port range. TLS Port Determines the listening TLS port.
Page 120: Figure 3-64: Ip Group Table
Mediant 800 MSBG Notes: • When operating with multiple IP Groups, the default Proxy server must not be used (i.e., the parameter IsProxyUsed must be set to 0). • If different SRDs are configured in the ‘IP Group’ and ‘Proxy Set’ tables, the SRD defined for the Proxy Set takes precedence.
Page 121: Table 3-18: Ip Group Parameters
SIP User's Manual 3. Web-Based Management Table 3-18: IP Group Parameters Parameter Description Common Parameters Type The IP Group can be defined as one of the following types: [IPGroup_Type] [0] SERVER = used when the destination address (configured by the Proxy Set) of the IP Group (e.g., ITSP, Proxy, IP-PBX, or Application server) is known.
Page 122 Mediant 800 MSBG Parameter Description routing to the IP Group is done. Notes: This field is available only if the SBC or IP-to-IP application is enabled. (The IP-to-IP application will be supported in the next applicable release.) Currently, the GATEWAY IP Group type can only be configured using the IPGroup ini file parameter.
Page 123 SIP User's Manual 3. Web-Based Management Parameter Description Media Realm The Media Realm name (defined in Configuring Media Realms [IPGroup_MediaRealm] on page 109) associated with this IP Group. This value must be identical (including case-sensitive) to that defined in the Media Realm table Note: For this parameter to take effect, a device reset is required.
Page 124 Mediant 800 MSBG Parameter Description [0] Standard = INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI, according to the Refer-To header in the REFER message or Contact header in the 3xx response (default).
Page 125 SIP User's Manual 3. Web-Based Management Parameter Description Serving IP Group ID If configured, INVITE messages initiated from the IP Group are [IPGroup_ServingIPGroup] sent to this Serving IP Group (range 1 to 9). In other words, the INVITEs are sent to the address defined for the Proxy Set associated with this Serving IP Group.
Page 126: Figure 3-65: Proxy Sets Table Page
Mediant 800 MSBG 3.3.2.8.4 Configuring Proxy Sets Table The 'Proxy Sets Table' page allows you to define Proxy Sets. A Proxy Set is a group of Proxy servers defined by IP address or fully qualified domain name (FQDN). You can define up to 32 Proxy Sets, each with a unique ID number and up to five Proxy server addresses.
Page 127: Table 3-19: Proxy Sets Table Parameters
SIP User's Manual 3. Web-Based Management From the 'Proxy Set ID' drop-down list, select an ID for the desired group. Configure the Proxy parameters according to the following table. Click the Submit button to save your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 336. Table 3-19: Proxy Sets Table Parameters Parameter Description...
Page 128 Mediant 800 MSBG Parameter Description sent to the next Proxy in the list. The same logic applies to REGISTER messages (if RegistrarIP is not defined). 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).
Page 129 SIP User's Manual 3. Web-Based Management Parameter Description using the UsePingPongKeepAlive parameter. Web: Proxy Keep Alive Time Defines the Proxy keep-alive time interval (in seconds) between EMS: Keep Alive Time Keep-Alive messages. This parameter is configured per Proxy [ProxySet_ProxyKeepAliveTim Set. The valid range is 5 to 2,000,000.
Page 130: Sip Definitions
Mediant 800 MSBG Parameter Description HotSwapRtx), the message is resent to the next redundant Proxy/Registrar server. Web/EMS: Redundancy Mode Determines whether the device switches back to the primary [ProxySet_ProxyRedundancyM Proxy after using a redundant Proxy (per this Proxy Set). ode] [-1] = Not configured –...
Page 131: Figure 3-66: Sip General Parameters Page
SIP User's Manual 3. Web-Based Management To configure general SIP parameters: Open the 'SIP General Parameters' page (Configuration tab > VoIP menu > SIP Definitions submenu > General Parameters). Figure 3-66: SIP General Parameters Page Configure the parameters as required. Version 6.2 February 2011...
Page 132: Figure 3-67: Sip General Parameters Page
Mediant 800 MSBG Click the Submit button to save your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 336. 3.3.2.9.2 Configuring Advanced Parameters The 'Advanced Parameters' page allows you to configure advanced SIP control parameters. For a description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 653.
Page 133: Figure 3-68: Account Table Page
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, see ''Saving Configuration'' on page 336. 3.3.2.9.3 Configuring Account Table The 'Account Table' page allows you to define up to 32 Accounts per Hunt Group (Served Hunt Group) or source IP Group (Served IP Group) for registration and/or digest authentication (user name and password) to a destination IP address (Serving IP Group).
Page 134: Table 3-20: Account Table Parameters Description
Mediant 800 MSBG Note: For a description of the Web interface's table command buttons (e.g., Duplicate and Delete), see ''Working with Tables'' on page 53. Table 3-20: Account Table Parameters Description Parameter Description Served Trunk Group The Hunt Group ID for which you want to register and/or authenticate to a destination IP Group (i.e., Serving IP Group).
Page 135 SIP User's Manual 3. Web-Based Management Parameter Description Username Digest MD5 Authentication user name (up to 50 characters). [Account_Username] Password Digest MD5 Authentication password (up to 50 characters). [Account_Password] Note: After you click the Apply button, this password is displayed as an asterisk (*).
Page 136 Mediant 800 MSBG Parameter Description Contact User Defines the AOR user name. It appears in REGISTER From/To [Account_ContactUser] headers as ContactUser@HostName, and in INVITE/200 OK Contact headers as ContactUser@<device's IP address>. If not configured, the 'Contact User' parameter from the 'IP Group Table' page is used instead.
Page 137: Figure 3-69: Proxy & Registration Page
SIP User's Manual 3. Web-Based Management To configure the Proxy and registration parameters: Open the 'Proxy & Registration' page (Configuration tab > VoIP menu > SIP Definitions submenu > Proxy & Registration). Figure 3-69: Proxy & Registration Page Configure the parameters as required. Click the Submit button to save your changes.
Page 138: 3.3.2.10 Coders And Profiles
Mediant 800 MSBG Click the Register or Un-Register buttons to save your changes and register/unregister to a Proxy/Registrar. To save the changes to flash memory, see ''Saving Configuration'' on page 336. Click the Proxy Set Table button to open the 'Proxy Sets Table' page to configure groups of proxy addresses.
Page 139: Figure 3-70: Coders Page
SIP User's Manual 3. Web-Based Management 3.3.2.10.1 Configuring Coders The 'Coders' page allows you to configure up to 10 coders for the device. The first coder in the list has the highest priority and is used by the device whenever possible. If the far-end device cannot use the first coder, the device attempts to use the next coder in the list, and so on.
Page 140: Figure 3-71: Coder Group Settings Page
Mediant 800 MSBG From the 'Packetization Time' drop-down list, select the packetization time (in msec) for the selected coder. The packetization time determines how many coder payloads are combined into a single RTP packet. From the 'Rate' drop-down list, select the bit rate (in kbps) for the selected coder.
Page 141 SIP User's Manual 3. Web-Based Management From the 'Coder Group ID' drop-down list, select a Coder Group ID. From the 'Coder Name' drop-down list, select the first coder for the Coder Group. From the 'Packetization Time' drop-down list, select the packetization time (in msec) for the coder.
Page 142: Figure 3-72: Tel Profile Settings Page
Mediant 800 MSBG To configure Tel Profiles: Open the 'Tel Profile Settings' page (Configuration tab > VoIP menu > Coders And Profiles submenu > Tel Profile Settings). Figure 3-72: Tel Profile Settings Page From the 'Profile ID' drop-down list, select the Tel Profile identification number you want to configure.
Page 143 SIP User's Manual 3. Web-Based Management 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. If both IP and Tel profiles apply to the same call, the coders and other common parameters (noted by an asterisk in the description of the parameter TelProfile) of the preferred Profile are applied to that call.
Page 144: Figure 3-73: Ip Profile Settings Page
Mediant 800 MSBG To configure IP Profiles: Open the 'IP Profile Settings' page (Configuration tab > VoIP menu > Coders And Profiles submenu > IP Profile Settings). Figure 3-73: IP Profile Settings Page From the 'Profile ID' drop-down list, select an identification number for the IP Profile.
Page 145: 3.3.2.11 Gw And Ip To Ip
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 146: Figure 3-74: Hunt Group Table Page
Mediant 800 MSBG 3.3.2.11.1.1 Configuring Hunt Group Table The 'Hunt Group Table' page allows you to define up to 120 Hunt Groups. This table enables the device's channels by assigning them telephone numbers, Hunt Group IDs and Tel Profiles. Channels that are not defined in this table are disabled. Hunt Groups are used for routing calls (Tel-to-IP and IP-to-Tel) on the channels associated with the Hunt Group.
Page 147 SIP User's Manual 3. Web-Based Management Parameter Description [TrunkGroup_LastBChannel] channel numbers. You can enter a range of channels by using the format [n-m], where n represents the lower channel number and m the higher channel number. For example, [1-4] specifies channels 1 through 4.
Page 148: Figure 3-75: Hunt Group Settings Page
Mediant 800 MSBG 3.3.2.11.1.2 Configuring Hunt Group Settings The 'Hunt Group Settings' page allows you to configure the settings of up to 24 Hunt Groups. These Hunt Groups are configured in the 'Hunt Group Table' page (see Configuring Hunt Group Table on page 146).
Page 149: Table 3-22: Hunt Group Settings Parameters
SIP User's Manual 3. Web-Based Management Table 3-22: Hunt Group Settings Parameters Parameter Description Trunk Group ID The Hunt Group ID that you want to configure. [TrunkGroupSettings_TrunkGroup Channel Select Mode The method for which IP-to-Tel calls are assigned to channels [TrunkGroupSettings_ChannelSel pertaining to a Hunt Group.
Page 150 Mediant 800 MSBG Parameter Description Notes: To enable Hunt Group registrations, configure the global parameter IsRegisterNeeded to 1. This is unnecessary for 'Per Account' registration mode. If no mode is selected, the registration is performed according to the global registration parameter ChannelSelectMode.
Page 151: Figure 3-76: General Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.11.2 Manipulation The Manipulation Tables submenu allows you to configure number manipulation and mapping of NPI/TON to SIP messages. This submenu includes the following items: General Settings (see ''Configuring General Settings'' on page 151) Manipulation tables (see ''Configuring Number Manipulation Tables'' on page 152): •...
Page 152 Mediant 800 MSBG 3.3.2.11.2.2 Configuring Number Manipulation Tables The device provides number manipulation tables for incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. These tables are used to modify the destination and/or source telephone numbers so that the calls can be routed correctly. For example, telephone number manipulation can be implemented by the following: Stripping or adding dialing plan digits from or to the number, respectively.
Page 153: Figure 3-77: Source Phone Number Manipulation Table For Tel-To-Ip Calls
SIP User's Manual 3. Web-Based Management Notes: • Number manipulation can occur before or after a routing decision is made. For example, you can route a call to a specific Hunt Group according to its original number, and then you can remove or add a prefix to that number before it is routed.
Page 154: Table 3-23: Number Manipulation Parameters Description
Mediant 800 MSBG The previous figure shows an example of the use of manipulation rules for Tel-to-IP source phone number manipulation: • Index 1: When the destination number has the prefix 03 (e.g., 035000), source number prefix 201 (e.g., 20155), and from source IP Group ID 2, the source number is changed to, for example, 97120155.
Page 155 SIP User's Manual 3. Web-Based Management Parameter Description Web/EMS: Source IP Source IP address of the caller (obtained from the Contact header in Address the INVITE message). Notes: This parameter is applicable only to the Number Manipulation tables for IP-to-Tel calls. The source IP address can include the 'x' wildcard to represent single digits.
Page 156 Mediant 800 MSBG Parameter Description Web: TON The Type of Number (TON) assigned to this entry. EMS: Number Type If you selected 'Unknown' for the NPI, you can select Unknown [0]. If you selected 'Private' for the NPI, you can select Unknown [0], Level 2 Regional [1], Level 1 Regional [2], PISN Specific [3] or Level 0 Regional (Local) [4].
Page 157: Figure 3-78: Redirect Number Ip To Tel Page
SIP User's Manual 3. Web-Based Management To configure Redirect Number IP-to-Tel manipulation rules: Open the 'Redirect Number IP > Tel' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations submenu > Redirect Number IP > Tel). Figure 3-78: Redirect Number IP to Tel Page Configure the rules according to the table below.
Page 158 Mediant 800 MSBG Parameter Description parameter is set to 'P-Asserted', the From header in the INVITE message includes the following: From: 'anonymous' <sip: anonymous@anonymous.invalid> and 'privacy: id' header. Web/EMS: Source IP Source IP address of the caller (obtained from the Contact header in Address the INVITE message).
Page 159: Figure 3-79: Redirect Number Tel To Ip Page
SIP User's Manual 3. Web-Based Management To configure redirect Tel-to-IP manipulation rules: Open the 'Redirect Number Tel > IP' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations submenu > Redirect Number Tel > IP). Figure 3-79: Redirect Number Tel to IP Page The figure below shows an example configuration in which the redirect prefix "555"...
Page 160: Figure 3-80: Phone Context Table Page
Mediant 800 MSBG Parameter Description Web/EMS: Prefix to Add The number or string that you want added to the front of the telephone number. For example, if you enter '9' and the phone number is 1234, the new number is 91234.
Page 161: Table 3-26: Phone-Context Parameters Description
SIP User's Manual 3. Web-Based Management Configure the Phone Context table according to the table below. Click the Submit button to save your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 336. Notes: • Several rows with the same NPI-TON or Phone-Context are allowed. In such a scenario, a Tel-to-IP call uses the first match.
Page 162: Table 3-27: Npi/Ton Values For Isdn Etsi
Mediant 800 MSBG 3.3.2.11.2.6 Numbering Plans and Type of Number The IP-to-Tel destination or source number manipulation tables allow you to classify numbers by their Numbering Plan Indication (NPI) and Type of Number (TON). The device supports all NPI/TON classifications used in the standard. The list of ISDN ETSI NPI/TON...
Page 163: Figure 3-81: Release Cause Mapping Page
SIP User's Manual 3. Web-Based Management To configure Release Cause Mapping: Open the 'Release Cause Mapping' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations submenu > Release Cause Mapping). Figure 3-81: Release Cause Mapping Page In the 'Release Cause Mapping from ISDN to SIP' group, map different Q.850 Release Causes to SIP Responses.
Page 164: Figure 3-82: Routing General Parameters Page
Mediant 800 MSBG 3.3.2.11.3 Routing The Routing submenu allows you to configure call routing rules. This submenu includes the following page items: General Parameters (see ''Configuring General Routing Parameters'' on page 164) Tel to IP Routing (see ''Configuring Outbound IP Routing Table'' on page 165)
Page 165 SIP User's Manual 3. Web-Based Management 3.3.2.11.3.2 Configuring Outbound IP Routing Table The 'Outbound IP Routing Table' page allows you to configure up to 180 Tel-to- IP/outbound IP call routing rules. The device uses these rules to route calls (from the Tel or IP) to IP destinations.
Page 166: Figure 3-83: Locating Srd
Mediant 800 MSBG Since each call must have a destination IP Group (even in cases when the destination type is not to an IP Group), in cases when the IP Group is not specified, the SRD's default IP Group is used (the first defined IP Group that belongs to the SRD).
Page 167 SIP User's Manual 3. Web-Based Management Assign IP Profiles: IP Profiles can be assigned to destination addresses (also when a proxy is used). Alternative Routing (when a proxy isn't used): An alternative IP destination can be configured for specific call. To associate an alternative IP address to a called telephone number prefix, assign it with an additional entry (with a different IP address), or use an FQDN that resolves into two IP addresses.
Page 168: Figure 3-84: Outbound Ip Routing Table Page
Mediant 800 MSBG To configure outbound IP routing rules: Open the 'Outbound IP Routing Table' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Routing submenu > Tel to IP Routing). Figure 3-84: Outbound IP Routing Table Page The figure above displays the following outbound IP routing rules: •...
Page 169: Table 3-28: Outbound Ip Routing Table Parameters
SIP User's Manual 3. Web-Based Management Table 3-28: Outbound IP Routing Table Parameters Parameter Description Web/EMS: Tel to IP Determines whether to route received calls to an IP destination before or after Routing Mode manipulation of the destination number. [RouteModeTel2IP] [0] Route calls before manipulation = Calls are routed before the number manipulation rules are applied (default).
Page 170 Mediant 800 MSBG Parameter Description Web/EMS: Source Prefix of the calling telephone number. Phone Prefix The prefix can include up to 50 digits. Note: To denote any prefix, enter an asterisk (*) symbol. The prefix can be a single digit or a range of digits. For available notations, see ''Dialing Plan Notation for Routing and Manipulation'' on page 413.
Page 171 SIP User's Manual 3. Web-Based Management Parameter Description Web: Dest IP Group The IP Group to where you want to route the call. The SIP INVITE message is sent to the IP address defined for the Proxy Set ID associated with the IP EMS: Destination IP Group.
Page 172: Figure 3-85: Inbound Ip Routing Table
Mediant 800 MSBG 3.3.2.11.3.3 Configuring Inbound IP Routing Table The 'Inbound IP Routing Table' page allows you to configure up to 24 inbound call routing rules: For IP-to-IP routing: identifying IP-to-IP calls and assigning them to IP Groups (referred to as Source IP Groups). These IP-to-IP calls, now pertaining to an IP Group, can later be routed to an outbound destination IP Group (see Configuring the Outbound IP Routing Table).
Page 173: Table 3-29: Inbound Ip Routing Table Description
SIP User's Manual 3. Web-Based Management The previous figure displays the following configured routing rules: • Rule 1: If the incoming IP call destination phone prefix is between 10 and 19, the call is assigned settings configured for IP Profile ID 2 and routed to Hunt Group ID 1.
Page 174: Configuring Alternative Routing Reasons
Mediant 800 MSBG Parameter Description Note: The prefix can be a single digit or a range of digits. For available notations, see ''Dialing Plan Notation for Routing and Manipulation'' on page 413. Source IP Address The source IP address of the incoming IP call (obtained from the Contact header in the INVITE message) that can be used for routing decisions.
Page 175: Figure 3-86: Reasons For Alternative Routing Page
SIP User's Manual 3. Web-Based Management Release reason for Tel-to-IP calls: Reason for call release on the IP side, provided in SIP 4xx, 5xx, and 6xx response codes. As a result of a release reason, an alternative IP address is provided. For defining an alternative IP address, see ''Configuring Outbound IP Routing Table'' on page 165.
Page 176: Figure 3-87: Forward On Busy Trunk Destination Page
Mediant 800 MSBG 3.3.2.11.3.5 Configuring Call Forward upon Busy Trunk The 'Forward on Busy Trunk Destination' page allows you to configure forwarding of IP-to- Tel calls (call redirection) to a different (alternative) IP destination, using SIP 3xx responses, upon the following scenarios: For digital interfaces: If a Trunk Group has no free channels (i.e., “busy”...
Page 177: Figure 3-88: Dtmf & Dialing Page
SIP User's Manual 3. Web-Based Management 3.3.2.11.4 DTMF and Supplementary The DTMF and Supplementary submenu allows you to configure DTMF and supplementary parameters. This submenu includes the following page items: DTMF & Dialing (see ''Configuring DTMF and Dialing'' on page 177) Supplementary Services (see ''Configuring Supplementary Services'' on page 177) 3.3.2.11.4.1 Configuring DTMF and Dialing...
Page 178: Figure 3-89: Supplementary Services
Mediant 800 MSBG To configure supplementary services parameters: Open the 'Supplementary Services' page (Configuration tab > VoIP menu > GW and IP to IP submenu > DTMF & Supplementary submenu > Supplementary Services). Figure 3-89: Supplementary Services Page Configure the parameters as required.
Page 179 SIP User's Manual 3. Web-Based Management 3.3.2.11.5 Analog Gateway The Analog Gateway submenu allows you to configure analog settings. This submenu includes the following page items: Keypad Features (see ''Configuring Keypad Features'' on page 179) Metering Tones (see ''Configuring Metering Tones'' on page 180) Charge Codes (see ''Configuring Charge Codes'' on page 181) FXO Settings (see ''Configuring FXO Settings'' on page 182) Authentication (see ''Configuring Authentication'' on page 183)
Page 180: Figure 3-90: Keypad Features
Mediant 800 MSBG To configure the keypad features Open the 'Keypad Features' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Keypad Features). Figure 3-90: Keypad Features Page Configure the keypad features as required. For a description of these parameters, see ''Configuration Parameters Reference'' on page 653.
Page 181: Figure 3-91: Metering Tones
SIP User's Manual 3. Web-Based Management Notes: • The 'Metering Tones' page is available only for FXS interfaces. • Charge Code rules can be assigned to routing rules in the 'Outbound IP Routing Table' (see ''Configuring Outbound IP Routing Table'' on page 165).
Page 182: Figure 3-92: Charge Codes Table
Mediant 800 MSBG To configure the Charge Codes: Access the 'Charge Codes Table' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Charge Codes). Alternatively, you can access this page from the 'Metering Tones' page (see ''Configuring Metering Tones'' on page 180).
Page 183: Figure 3-93: Fxo Settings
SIP User's Manual 3. Web-Based Management To configure the FXO parameters: Open the 'FXO Settings' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > FXO Settings). Figure 3-93: FXO Settings Page Configure the parameters as required.
Page 184: Figure 3-94: Authentication
Mediant 800 MSBG To configure the Authentication Table: Set the parameter 'Authentication Mode' (AuthenticationMode ) to 'Per Endpoint'. Open the 'Authentication' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Authentication).
Page 185: Figure 3-95: Automatic Dialing
SIP User's Manual 3. Web-Based Management To configure Automatic Dialing: Open the 'Automatic Dialing' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Automatic Dialing). Figure 3-95: Automatic Dialing Page In the 'Destination Phone Number' field corresponding to a port, enter the telephone number that you want automatically dialed.
Page 186: Figure 3-96: Caller Display Information
Mediant 800 MSBG To configure the Caller Display Information: Open the 'Caller Display Information' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Analog Gateway submenu > Caller Display Information). Figure 3-96: 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 187: Table 3-30: Call Forward Table
SIP User's Manual 3. Web-Based Management Notes: • Ensure that the Call Forward feature is enabled (default) for the settings on this page to take effect. To enable Call Forward, use the parameter EnableForward (''Configuring Supplementary Services'' on page 177). •...
Page 188: Figure 3-98: Caller Id Permissions
Mediant 800 MSBG Parameter Description Time for No Reply If you have set the 'Forward Type' for this port to 'No Answer', enter the Forward number of seconds the device waits before forwarding the call to the phone number specified.
Page 189: Figure 3-99: Caller Waiting
SIP User's Manual 3. Web-Based Management 3.3.2.11.5.10 Configuring Call Waiting The 'Call Waiting' page allows you to enable or disable call waiting per device FXS port. Notes: • This page is applicable only to FXS interfaces. • Instead of using this page, you can enable or disable call waiting for all the device's ports, using the global call waiting parameter 'Enable Call Waiting' (see ''Configuring Supplementary Services'' on page 177).
Page 190: Figure 3-100: Digital Gateway Parameters
Mediant 800 MSBG 3.3.2.11.6 Digital Gateway The Digital Gateway submenu allows you to configure digital PSTN settings. This submenu includes the following page items: Digital Gateway Parameters (see ''Configuring Digital Gateway Parameters'' on page 190) ISDN Supp Services (see "Configuring ISDN Supplementary Services" on page 191) 3.3.2.11.6.1...
Page 191 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, see ''Saving Configuration'' on page 336. 3.3.2.11.6.2 Configuring ISDN Supplementary Services The 'ISDN Supp Services Table' page allows you to configure supplementary services for Integrated Services Digital Network (ISDN) Basic Rate Interface (BRI) phones connected to the device.
Page 192: Table 3-31: Isdn Supp Services Table Parameters
Mediant 800 MSBG To configure BRI supplementary services: Open the 'ISDN Supp Services Table' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Digital Gateway submenu > ISDN Supp Services). Figure 3-101: ISDN Supp Services Table Page Configure the parameters as described in the table below.
Page 193: Figure 3-102: Isdn Supp Services Table
SIP User's Manual 3. Web-Based Management 3.3.2.11.7 Advanced Applications The Advanced Applications menu allows you to configure advanced SIP-based applications. This menu includes the following page item: Voice Mail Settings (see Configuring Voice Mail Parameters on page 193) 3.3.2.11.7.1 Configuring Voice Mail Parameters The 'Voice Mail Settings' page allows you to configure the voice mail parameters.
Page 194: 3.3.2.12 Sbc
Mediant 800 MSBG 3.3.2.12 SBC The SBC submenu allows you to configure the SBC application. This submenu includes the following items: General Settings (see ''Configuring General Settings'' on page 194) Admission Control (see ''Configuring Admission Control'' on page 195) Allowed Coders Group (see ''Configuring Allowed Coder Groups'' on page 197) Routing SBC: •...
Page 195: Figure 3-104: Admission Control
SIP User's Manual 3. Web-Based Management Click the Submit button to save your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 336. 3.3.2.12.2 Configuring Admission Control The 'Admission Control' page allows you to define up to 100 rules for limiting the number of concurrent calls (SIP dialogs).
Page 196: Table 3-32: Admission Control Parameters
Mediant 800 MSBG Table 3-32: Admission Control Parameters Parameter Description Limit Type Limitation rule defined per IP group or SRD. [0] IP Group (default) [1] SRD IP Group ID IP Group to which you want to apply the SIP dialog limit. To apply the rule to all IP Groups, set this parameter to -1 (default).
Page 197 SIP User's Manual 3. Web-Based Management Parameter Description MaxBurst The maximum number of tokens (SIP dialogs) that the bucket can hold, where 0 is unlimited (default). The device only accepts a SIP dialog if a token exists in the bucket. Once the SIP dialog is accepted, a token is removed from the bucket.
Page 198: Figure 3-105: Allowed Coders Group
Mediant 800 MSBG To configure Allowed Coder Groups: Open the 'Allowed Coders Group' page (Configuration tab > VoIP menu > SBC submenu > Allowed Coders Group). Figure 3-105: Allowed Coders Group Page From the 'Allowed Coders Group ID' drop-down list, select an ID for the Allowed Coder Group.
Page 199: Figure 3-106: Classification Table
SIP User's Manual 3. Web-Based Management Compares P-Asserted/From URL to the registered AOR If the database search is unsuccessful, then the classification process proceeds with locating a Proxy Set (associated with the SIP dialog request’s IP address) and then finding a match with a corresponding IP Group in the 'IP Group' table.
Page 200: Table 3-33: Classification Table Parameters
Mediant 800 MSBG Table 3-33: Classification Table Parameters Parameter Description Matching Characteristics Source SRD ID The SRD ID (configured in the SRD table) from where the [Classification_SrcSRDID] SIP dialog request is received. The default is -1. Note: The source SRD is defined according to the UDP/TCP/TLS port at which the incoming SIP dialog request is received.
Page 201 SIP User's Manual 3. Web-Based Management Parameter Description This IP Group is used for SBC routing and manipulations To define IP Groups, see ''Configuring IP Groups'' on page 119. 3.3.2.12.4.2 Configuring SBC IP-to-IP Routing The 'IP2IP Routing Table' page configures up to 120 SBC IP-to-IP routing rules. This table provides enhanced IP-to-IP call routing capabilities for routing received SIP dialog messages (e.g., INVITE) to a destination IP address.
Page 202: Table 3-34: Ip2Ip Routing Table Parameters
Mediant 800 MSBG Notes: • For a specific IP-to-IP routing rule to be effective, the incoming SIP dialog message must match the characteristics configured for that rule. • The 'IP2IP Routing' table can also be configured using the ini file table parameter IP2IPRouting (see ''SBC Parameters'' on page 858).
Page 203 SIP User's Manual 3. Web-Based Management Parameter Description For available notations, see ''Dialing Plan Notation for Routing and Manipulation'' on page 413. Destination Host The host part of the incoming SIP SIP dialog’s destination [IP2IPRouting_DestHost] URI (usually the Request-URI). If this rule is not required, leave the field empty.
Page 204 Mediant 800 MSBG Parameter Description Notes: This parameter is only relevant if the parameter 'Destination Type' is set to 'IP Group'. However, regardless of the settings of the parameter 'Destination Type', the IP Group is still used - only for determining the IP Profile or outgoing SRD.
Page 205 SIP User's Manual 3. Web-Based Management Parameter Description Alternative Route Options Determines whether this routing rule is the main routing rule [IP2IPRouting_AltRouteOptions] or an alternative routing rule (to the rule defined directly above it in the table). [0] Route Row (default) = Main routing rule - the device first attempts to route the call to this route if the incoming SIP dialog's input characteristics matches this rule.
Page 206: Figure 3-108: Alternative Routing Reasons
Mediant 800 MSBG 3.3.2.12.4.3 Configuring Alternative Routing Reasons The 'SBC Alternative Routing Reasons' page allows you to define up to five different call release (termination) reasons for call releases. If a call is released as a result of one of these reasons provided in SIP 4xx, 5xx, and 6xx response codes, the device tries to find an alternative route for the call.
Page 207: Figure 3-109: Message Manipulations
SIP User's Manual 3. Web-Based Management Notes: • For a detailed description on the syntax for configuring SIP message manipulation rules in the Message Manipulation table, see ''SIP Message Manipulation Description'' on page 506. • SIP message manipulation is done on the inbound and outbound legs (i.e., only after classification, inbound/outbound manipulations, and routing).
Page 208: Table 3-35: Message Manipulations Parameters
Mediant 800 MSBG The previous figure shows the following message manipulation rules: • Index 1: adds the suffix ".com" to the host part of the To header. • Index 2: changes the user part of the SIP From header to 200.
Page 209 SIP User's Manual 3. Web-Based Management Parameter Description Operation Action Subject SIP header upon which the manipulation is performed. [ActionSubject] Action Type The type of manipulation to perform. [ActionType] [0] Add (default) = adds new header/param/body (header or parameter elements). [1] Remove = removes header/param/body (header or parameter elements).
Page 210: Figure 3-110: Ip To Ip Inbound Manipulation
Mediant 800 MSBG 3.3.2.12.5.2 Configuring IP-to-IP Inbound Manipulations The 'IP to IP Inbound Manipulation' page allows you to configure up to 100 manipulation rules for manipulating the SIP URI user part (source and destination) of inbound SIP dialog requests. You can apply these manipulations to different SIP dialog message types (e.g., INVITE or REGISTER).
Page 211: Table 3-36: Ip To Ip Inbound Manipulation Parameters
SIP User's Manual 3. Web-Based Management Table 3-36: IP to IP Inbound Manipulation Parameters Parameter Description Matching Characteristics Is Additional Manipulation Determines whether additional SIP URI user part manipulation is done [IsAdditionalManipulation] for the table entry rule listed directly above it. [0] 0 = Regular manipulation rule (not done in addition to the rule above it).
Page 212 Mediant 800 MSBG Parameter Description [3] SUBSCRIBE = only SIP SUBSCRIBE messages [4] INVITE and REGISTER = all SIP messages except SUBSCRIBE [5] INVITE and SUBSCRIBE = all SIP messages except REGISTER Operation Manipulation Rule (when match occurs in characteristics) Remove From Left The number of digits to remove from the left of the user name prefix.
Page 213: Table 3-37: Ip To Ip Outbound Manipulation Table Parameters
SIP User's Manual 3. Web-Based Management To configure IP-to-IP outbound manipulation rules: Open the 'IP to IP Outbound Manipulation' page (Configuration tab > VoIP menu > SBC submenu > Manipulations SBC submenu > IP to IP Outbound). Figure 3-111: IP to IP Outbound Manipulation Page Add an entry and then configure it according to the table below.
Page 214 Mediant 800 MSBG Parameter Description Source Username Prefix The prefix of the source SIP URI user name (usually in the From [SrcUsernamePrefix] header). For any prefix, enter an asterisk (*), which is the default. Note: The prefix can be a single digit or a range of digits. For available notations, see ''Dialing Plan Notation for Routing and Manipulation'' on page 413.
Page 215 SIP User's Manual 3. Web-Based Management Parameter Description Suffix to Add The number or string that you want added to the end of the user name. [Suffix2Add] For example, if you enter '01' and the user name is "bobby", the new user name is "bobby01".
Page 216: 3.3.2.13 Sas
Mediant 800 MSBG 3.3.2.13 SAS The SAS submenu allows you to configure the SAS application. This submenu includes the Stand Alone Survivability item page (see ''Configuring Stand-Alone Survivability'' on page 216), from which you can also access the 'IP2IP Routing Table' page for configuring SAS routing rules (see ''Configuring IP2IP Routing Table (SAS)'' on page 218).
Page 217: Figure 3-112: Sas Configuration
SIP User's Manual 3. Web-Based Management To configure SAS: Open the 'SAS Configuration' page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability). Figure 3-112: SAS Configuration Page Configure the individual parameters as described in SIP Configuration Parameters. Configure the SAS Registration Manipulation table to manipulate the SIP Request-URI user part of incoming INVITE messages and of incoming REGISTER request AoR (in the To header), before it is saved to the registered users database.
Page 218: Figure 3-113: Ip2Ip Routing
Mediant 800 MSBG Click the Submit button to apply your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 336. To configure the SAS Routing table, under the SAS Routing group, click the SAS Routing Table button to open the 'IP2IP Routing Table' page.
Page 219: Table 3-38: Sas Ip2Ip Routing Table Parameters
SIP User's Manual 3. Web-Based Management Table 3-38: SAS IP2IP Routing Table Parameters Parameter Description Matching Characteristics Source IP Group ID This parameter is not applicable. [IP2IPRouting_SrcIPGroupID] Source Username Prefix The prefix of the user part of the incoming INVITE’s source [IP2IPRouting_SrcUsernamePrefix] URI (usually the From URI).
Page 220 Mediant 800 MSBG Parameter Description overridden and these fields take precedence. [3] ENUM = An ENUM query is sent to include the destination address. If the fields 'Destination Port' and 'Destination Transport Type' are configured, the incoming Request URI parameters are overridden and these fields take precedence.
Page 221 SIP User's Manual 3. Web-Based Management Parameter Description Destination Transport Type The transport layer type for sending the call: [IP2IPRouting_DestTransportType] [-1] Not Configured (default) [0] UDP [1] TCP [2] TLS Note: When this parameter is set to -1, the transport type is determined by the parameter SIPTransportType.
Page 222: Data Settings
Management interfaces (Web, CLI and SNMP), and SIP (when using the device’s VoIP component). For a complete list of features supported only on the default VRF, please contact AudioCodes. For a detailed description of CLI configuration, refer to the MSBG Series CLI Reference Guide.
Page 223: Getting Acquainted With Data Configuration
SIP User's Manual 3. Web-Based Management 3.3.3.1 Getting Acquainted with Data Configuration Pages This section provides you with an overview on working with the data-routing configuration pages. 3.3.3.1.1 Working with Tables Throughout the data section of the Web interface, various configuration icons are provided in the configuration tables.
Page 224: Wan Access
Mediant 800 MSBG 3.3.3.1.2 Using the Configuration Buttons Throughout the Data section of the Web interface, various buttons appear in the configuration pages, as described in the table below: Table 3-40: Description of the Main Configuration Buttons Button Name Description Applies and saves the settings.
Page 225: Figure 3-116: Wan Access
SIP User's Manual 3. Web-Based Management device. For more information, refer to ''Configuring SHDSL WAN Interface'' on page 234. Note: The supported WAN connection methods depend on the installed Software Upgrade Key. For installing a Software Upgrade Key, refer to ''Loading Software Upgrade Key'' on page 339.
Page 226: Figure 3-117: Manual Wan Connection
Mediant 800 MSBG From the 'Connection Type' drop-down list, select the required WAN connection type: • 'Automatic IP Address Ethernet Connection' (see figure above) • 'Manual IP Address Ethernet Connection': Figure 3-117: Manual WAN Connection Specify the following: ♦ IP address ♦...
Page 227: Figure 3-120: L2Tp Wan Connection Type
SIP User's Manual 3. Web-Based Management Specify the following: ♦ PPTP Server Host Name or IP Address ♦ Login User Name ♦ Login Password ♦ Internet Protocol - select the method used by your ISP for assigning an IP address. •...
Page 228: Figure 3-121: Wan Access Page For T1 Wan Interface
Mediant 800 MSBG Notes: • Each physical T1 link is configured separately. • Local-line loopback on the T1 WAN interface can only be configured using the device's CLI. In this loopback, packets from the Tx line interface connect to the Rx line interface. The maximum time of this loopback is enabled and configured using the loopback CLI command.
Page 229: Figure 3-123: Settings Tab For Ppp Over T1
SIP User's Manual 3. Web-Based Management Click the Click here for Advanced Settings link; the General tab page is displayed. The General tab allows you to define an arbitrary name for the connection (in the ‘Name’ field), view various statistical information of the connection, and disable the connection (by clicking Disable).
Page 230: Figure 3-125: T1 Tab
Mediant 800 MSBG • Line Build Out - pulse shape of the T1 analog interface: ♦ Line Loss (pulse shape of the T1 analog interface): the Line CI code, as defined by ANSI T1.403 Annex H. ♦ Max Cable Loss: the maximum customer cable loss, as defined by ANSI T1.403 Annex H...
Page 231: Figure 3-127: Hdlc Over T1
SIP User's Manual 3. Web-Based Management From the ‘Connection Type’ drop-down list, select "HDLC Over T1"; the page refreshes, displaying the relevant parameters: Figure 3-127: HDLC Over T1 In the ‘IP Address’ and ‘Subnet Mask’ fields, enter the IP address supplied by your ISP for this connection, and then click OK.
Page 232: Figure 3-129: Wan Access Page For T1 Wan Interface
Mediant 800 MSBG 3.3.3.2.2.3 ML-PPP over T1 WAN The procedure below describes how to configure ML-PPP over T1 WAN interface. To configure ML-PPP over T1 WAN interface: Open the 'WAN Access' page (Configuration tab > Data menu > WAN Access >...
Page 233: Figure 3-131: Settings Tab For Mlp Over T1
SIP User's Manual 3. Web-Based Management Click the Click here for Advanced Settings link; the General tab page appears. The General tab allows you to define an arbitrary name for the connection (in the ‘Name’ field), view various statistical information of the connection, and disable the connection (by clicking Disable).
Page 234: Figure 3-133: T1 Tab
Mediant 800 MSBG ♦ Use Fragmented Mode: whether to use a fragmented mode: Yes: each packet is fragmented per the bandwidth ratio of the physical links in the bundle or non fragmented mode No: each packet is sent as a whole on a single link while alternating...
Page 235: Figure 3-134: Shdsl Line Mode
SIP User's Manual 3. Web-Based Management Multiple wire-pairs are bonded into a single broadband access link using G.991.2 multiple- pair (also known as "m-pair") technology when the transmission control is ATM, or by 802.3ah PMD Aggregation Function (PAF) when using Ethernet in the First Mile (EFM). To configure the SHDSL WAN interface: Obtain the connection information from your ISP, including the following data: •...
Page 236: Figure 3-135: Adding A New Group
Mediant 800 MSBG Click New Group to add a new SHDSL wire. Figure 3-135: Adding a New Group Note: When using EFM, only one wire-pair group can be defined. Configure the annex, line rate, and pair numbers as provided by your ISP, and then click OK.
Page 237: Figure 3-137: Protocol Interface Settings
SIP User's Manual 3. Web-Based Management If EFM mode was selected, skip the following steps and proceed to ''Configuring Ethernet WAN Interface'' on page 225. Open the 'Protocol Interface Settings' page (Configuration tab > Data menu > WAN Access > Protocol Interface Settings); the current ATM interface list is displayed. Figure 3-137: Protocol Interface Settings Page Click New Connection, select the 'Internet Connection' option, and then click Next.
Page 238: Firewall And Acl
Mediant 800 MSBG Configure the VPI, VCI, encapsulation variant and class-of-service parameters as provided by your ISP. Note that the VPI/VCI combination must be unique in an SHDSL group. If required by your ISP, configure the IP addressing parameters (IP address, network mask, DNS server information);...
Page 239 SIP User's Manual 3. Web-Based Management DMZ Host: allows you to configure a LAN host to receive all traffic arriving at your device, which does not belong to a known session (see ''Configuring DMZ Host'' on page 244). Port Triggering: allows you to define port triggering entries to dynamically open the firewall for specific protocols or ports (see ''Configuring Port Triggering'' on page 244).
Page 240: Figure 3-141: Configuring General Security
Mediant 800 MSBG To configure basic security: Click the General Security item (Configuration tab > Data menu > Firewall and ACL submenu > General Security); the following page appears: Figure 3-141: Configuring General Security Select one of the pre-defined security levels.
Page 241: Figure 3-142: Configuring Lan Restriction Rules
SIP User's Manual 3. Web-Based Management Note: When Web Filtering is enabled, HTTP services cannot be blocked by Access Control. To configure LAN restrictions rule: Click the LAN Restrictions item (Configuration tab > Data menu > Firewall and ACL submenu > LAN Restrictions); the following page appears: Figure 3-142: Configuring LAN Restriction Rules Click the New icon;...
Page 242: Figure 3-144: Disabled Lan Restrictions - Cleared Check Box
Mediant 800 MSBG You can disable a LAN restriction rule to make a service available without having to delete the rule. This may be useful if you wish to make the service temporarily available and expect to reinstate the restriction in the future.
Page 243: Figure 3-145: Configuring Port Forwarding
SIP User's Manual 3. Web-Based Management To configure a port forwarding service: Click the Port Forwarding item (Configuration tab > Data menu > Firewall and ACL submenu > Port Forwarding); the following page appears: Figure 3-145: Configuring Port Forwarding Click the New Entry link; the following page appears: Figure 3-146: Adding Port Forwarding Rule Select the 'Specify Public IP Address' check box if you want to apply this rule on the device's non-default IP address, defined in the 'NAT' page (see ''Configuring NAT'' on...
Page 244: Figure 3-147: Defining A Dmz Host
Mediant 800 MSBG 3.3.3.3.4 Configuring DMZ Host The DMZ (Demilitarized) Host feature allows a single local computer to be exposed to the Internet. You can designate a DMZ host for the following scenario examples: You wish to use a special-purpose Internet service, such as an on-line game or video conferencing program that is not present in the Port Forwarding list and for which no port range information is available.
Page 245: Figure 3-148: Configuring Port Triggering
SIP User's Manual 3. Web-Based Management To configure port triggering: Click the Port Triggering item (Configuration tab > Data menu > Firewall and ACL submenu > Port Triggering); the following page appears: Figure 3-148: Configuring Port Triggering From the drop-down list, you can select a pre-configured service by selecting 'Show All Services', and then from the refreshed drop-down list, selecting a service.
Page 246: Figure 3-151: Configuring Website Restrictions
Mediant 800 MSBG 3.3.3.3.6 Configuring Website Restrictions You can configure the device to block specific Internet Web sites so that they cannot be accessed from computers in the home network. Moreover, restrictions can be applied to a comprehensive and automatically-updated table of sites to which access is not recommended.
Page 247: Figure 3-153: Configuring Nat
SIP User's Manual 3. Web-Based Management 3.3.3.3.7 Configuring NAT The device features a configurable Network Address Translation (NAT) and Network Address Port Translation (NAPT) mechanism, allowing you to control the network addresses and ports of packets routed through the device. When enabling multiple computers on your network to access the Internet using a fixed number of public IP addresses, you can define static NAT/NATP rules which map (translate) LAN IP addresses (LAN computers) to NAT IP addresses and/or ports.
Page 248: Figure 3-154: Defining Public Ip Address
Mediant 800 MSBG Under the 'NAT IP Addresses Pool' group, click the New IP Address link; the following page appears: Figure 3-154: Defining Public IP Address From the 'Network Object Type' drop-down list, select between 'IP Address', 'IP Subnet' or 'IP Range', and then enter the information respectively.
Page 249: Figure 3-156: Access Lists Table
SIP User's Manual 3. Web-Based Management Configure the 'Operation' group parameters to define the operation that will be applied to the IP addresses matching the criteria defined above. The operations available are NAT or NAPT: • NAT: The NAT address into which the original IP address is translated. The drop- down list displays all of your available NAT addresses/ranges, from which you can select an entry.
Page 250: Figure 3-157: Defining Access List Name
Mediant 800 MSBG Add a new Access List group name: Click the New ACL link; the 'Access List Name' page appears. Figure 3-157: Defining Access List Name In the 'Access List Name' field, enter a name for the Access List rule group, and then click OK;...
Page 251: Figure 3-159: Added Access List Rules
SIP User's Manual 3. Web-Based Management ♦ DSCP: Select this check box to display two DSCP fields, which enable you to specify a hexadecimal DSCP value and its mask assigned to the packets matching the priority rule. ♦ Priority: Select this check box to display a drop-down list in which you can select a priority level assigned to the packets matching the priority rule.
Page 252: Qos
Mediant 800 MSBG 3.3.3.3.9 Configuring Advanced Filtering The Advanced Filtering allows you to assign Access List rules (defined in ''Configuring the Access List'' on page 249) to the device's LAN and/or WAN interfaces. To assign Access List rules to the device's LAN/WAN interfaces: Click the Advanced Filtering item (Configuration tab >...
Page 253: Figure 3-161: Configuring General Wan Bandwidth
SIP User's Manual 3. Web-Based Management 3.3.3.4.1 Configuring General QoS Settings The General QoS item allows you to configure your WAN bandwidth. To configure the device's WAN bandwidth: Click the General QoS item (Configuration tab > Data menu > QoS submenu > General QoS);...
Page 254: Figure 3-162: Configuring Traffic Priority
Mediant 800 MSBG The matching of packets by rules is connection-based, known as Stateful Packet Inspection (SPI), using the same connection-tracking mechanism used by the device's firewall. Once a packet matches a rule, all subsequent packets with the same attributes receive the same QoS parameters, both inbound and outbound.
Page 255: Figure 3-163: Adding A Traffic Priority Rule
SIP User's Manual 3. Web-Based Management Click the New Entry link corresponding to the traffic direction (i.e., 'QoS Input Rules' or 'QoS Output Rules') and the device on which to set the rule; the following page appears: Figure 3-163: Adding a Traffic Priority Rule Under the 'Matching' group, define characteristics of the packets matching the QoS rule: •...
Page 256 Mediant 800 MSBG • Priority: Select this check box to display a drop-down list from which you can select a priority level assigned to the packets matching the priority rule. • Device: Select this check box to display a drop-down list from which you can select a network device on which the packet-rule matching is performed.
Page 257: Figure 3-164: Configuring Traffic Shaping
SIP User's Manual 3. Web-Based Management The router sends traffic as fast as it is received, while its well-designed QoS algorithms are left unused. Traffic shaping limits the bandwidth of the router, artificially forcing the router to be the bottleneck. A traffic shaper is essentially a regulated queue that accepts uneven and/or bursty flows of packets and transmits them in a steady, predictable stream so that the network is not overwhelmed with traffic.
Page 258: Figure 3-166: Defining Device Traffic Shaping
Mediant 800 MSBG Click OK; the following page appears: Figure 3-166: Defining Device Traffic Shaping From the 'Tx Bandwidth' drop-down list, select the device's bandwidth transmission rate limit. If you want to specify a TX bandwidth, see Step 8. From the 'TCP Serialization' drop-down list, select whether TCP Serialization is enabled or disabled.
Page 259: Figure 3-168: Class Name Added To Table
SIP User's Manual 3. Web-Based Management In the 'Name' field, enter a new Tx traffic shaping class name (e.g., Class A), and then click OK to save the settings; the class is added to the table. Figure 3-168: Class Name Added to Table Click the newly added class name;...
Page 260 Mediant 800 MSBG ♦ Policy: Class policy determines the policy of routing packets inside the class: Priority: Priority queuing utilizes multiple queues, so that traffic is distributed among queues based on priority. This priority is defined according to packet's priority, which can be defined explicitly by a DSCP value or by a 802.1p value.
Page 261: Figure 3-170: Configuring Dscp Settings
SIP User's Manual 3. Web-Based Management To add, edit or delete DSCP settings: Click the DSCP Settings item (Configuration tab > Data menu > QoS submenu > DSCP Settings); the following page appears: Figure 3-170: Configuring DSCP Settings To edit an existing entry, click the corresponding Edit icon.
Page 262: Vpn
Mediant 800 MSBG 3.3.3.4.5 Configuring 802.1p Settings The IEEE 802.1p priority marking method is a standard for prioritizing network traffic at the data link/Mac sub-layer. 802.1p traffic is simply classified and sent to the destination, with no bandwidth reservations established. The 802.1p header includes a 3-bit prioritization field, which allows packets to be grouped into eight levels of priority (0-7), where level 7 is the highest.
Page 263: Figure 3-173: Configuring Vpn Ipsec
SIP User's Manual 3. Web-Based Management Services supported by the IPSec protocols (AH, ESP) include confidentiality (encryption), authenticity (proof of sender), integrity (detection of data tampering), and replay protection (defense against unauthorized resending of data). IPSec also specifies methodologies for key management.
Page 264: Figure 3-175: Ipsec Log Settings
Mediant 800 MSBG Click the Recreate Key button to recreate the public key, or the Refresh button to refresh the displayed key. Click Close; you are returned to the previous page. Configure the IPSec log display for identifying and analyzing the history of the IPSec package commands, attempts to create connections, etc: Click the Log Settings button;...
Page 265: Figure 3-176: Configuring Vpn Pptp Server
SIP User's Manual 3. Web-Based Management 3.3.3.5.2 Configuring PPTP Server The device can act as a Point-to-Point Tunneling Protocol Server (PPTP Server), accepting PPTP client connection requests. To configure PPTP: Click the PPTP item (Configuration tab > Data menu > VPN submenu > PPTP); the following page appears: Figure 3-176: Configuring VPN PPTP Server Under the 'Server' group, perform the following:...
Page 266: Figure 3-177: Configuring Vpn L2Tp Server
Mediant 800 MSBG 3.3.3.5.3 Configuring L2TP Server The device can act as a Layer 2 Tunneling Protocol Server (L2TP Server), accepting L2TP client connection requests. To configure L2PT: Click the L2TP item (Configuration tab > Data menu > VPN submenu > L2TP); the...
Page 267: Figure 3-178: Adding Users
SIP User's Manual 3. Web-Based Management ♦ From the 'MPPE Encryption Mode' drop-down list, select the Microsoft Point- to-Point Encryption mode: Stateless or Stateful. Under the 'Remote Address Range' group, in the 'Start IP Address' and 'End IP Address' fields, specify the range of IP addresses that are granted by the L2TP server to the L2TP client.
Page 268: Figure 3-180: Defining Outgoing Mail Server
Mediant 800 MSBG Under the 'General' group, configure the following parameters: Full Name: remote user's full name. User Name: name that a user uses to access your network. New Password: user's password. Retype New Password: if a new password is assigned, type it again to verify its correctness.
Page 269: Figure 3-181: Adding Users
SIP User's Manual 3. Web-Based Management Under the 'Security Logging' group, configure the following parameters: • Security Log Buffer Size: size of the security log buffer in Kilobytes. • Remote Security Notify Level: remote security notification level - None, Error, Warning, and Information.
Page 270: Data Services
Mediant 800 MSBG 3.3.3.6 Data Services The Data Services submenu allows you to configure various services (applications), and includes the following menus: DDNS (see ''Configuring DDNS'' on page 270) DNS Server (see ''Configuring DNS Server'' on page 271) DHCP Server (see ''Configuring DHCP Server'' on page 272) 3.3.3.6.1 Configuring DDNS...
Page 271 SIP User's Manual 3. Web-Based Management In the 'Host Name' field, enter your full DDNS domain name. From the 'Connection' field, select the connection to which you want to couple the DDNS service. The DDNS service only uses the selected device, unless failover is enabled.
Page 272: Figure 3-185: Configuring A Dns Server
Mediant 800 MSBG Permits a computer to have multiple host names Permits a host name to have multiple IPs (needed if a host has multiple network cards) The DNS server does not require configuration. However, you may wish to view the list of computers known by the DNS, edit the host name or IP address of a computer on the list, or manually add a new computer to the list.
Page 273: Figure 3-187: Configuring Dhcp Server
SIP User's Manual 3. Web-Based Management Note: By default, the device’s DHCP server is enabled. Therefore, when connecting the device to your enterprise’s LAN, the device responds to DHCP requests and consequently distributes IP addresses (instead of your Enterprise’s DHCP server, if exists). Your device's DHCP server: Displays a list of all DHCP host devices connected to the device Defines the range of IP addresses that can be allocated in the LAN...
Page 274: Figure 3-188: Defining Ip Distribution Type
Mediant 800 MSBG Click the Edit icon corresponding to the required interface; the following page appears: Figure 3-188: Defining IP Distribution Type From the 'IP Address Distribution' drop-down list, choose either 'DHCP Server', 'DHCP Relay' (or 'Disabled' if you want to disable DHCP).
Page 275: Figure 3-190: Defining Dhcp Relay (Dhcp For Lan Bridge)
SIP User's Manual 3. Web-Based Management Option 66 - TFTP Server Name: This option is used to identify a TFTP server. Option 67 - Boot File Name: This option is used to identify the boot file name. Option 2 - Time offset: Specifies the offset of the client's subnet in seconds from Coordinated Universal Time (UTC).
Page 276: Data Routing
Mediant 800 MSBG To view a list of computers currently recognized by the DHCP server and to add a new computer with a static IP address: Click the DHCP Server item (Configuration tab > Data Settings menu > Services >...
Page 277 SIP User's Manual 3. Web-Based Management 3.3.3.7.1 Configuring General Routing You can choose to setup your device to use static or dynamic routing. Dynamic routing automatically adjusts how packets travel on the network, whereas static routing specifies a fixed routing path to destinations. The Data Routing item allows you to add, edit and delete routing rules from the routing table.
Page 278: Figure 3-194: Configuring General Routing
Mediant 800 MSBG Notes: • Only default route devices can participate in load balancing. • DSCP-based policy routing takes precedence over load balancing. If most of the traffic falls under the DSCP-based policy routing rules, it is then forwarded accordingly, regardless of the load balancing. Load balancing, in this case, is by best-effort load balancing, and balances the remaining traffic not directed by the DSCP-based policy routing rules.
Page 279: Figure 3-196: Editing The Default Route
SIP User's Manual 3. Web-Based Management ♦ Netmask: network mask is used in conjunction with the destination to determine when a route is used. ♦ Gateway: enter the device's IP address. ♦ Metric: measurement of a route's preference. Typically, the lowest metric is the most preferred route.
Page 280: Figure 3-198: Adding Dscp-Based Route
Mediant 800 MSBG To add a DSCP-based policy route: Under the 'DSCP-Based Policy Routing' group, click the New Route link; the following page appears: Figure 3-198: Adding DSCP-Based Route From the 'Device' drop-down list, select the network device. In the 'DSCP' field, specify the DSCP value. All traffic matching this DSCP value is routed to the selected device.
Page 281 SIP User's Manual 3. Web-Based Management To enable Internet Group Management Protocol (IGMP) multicasting: Under the 'Internet Group Management Protocol (IGMP)' group, select the 'Enabled' check box. When a host sends a request to join a multicast group, the device listens and intercept the group's traffic, forwarding it to the subscribed host.
Page 282: Figure 3-200: Page Displaying Area For Configuration File
Mediant 800 MSBG To enable BGP and OSPF: Click the BGP & OSPF item (Configuration tab > Data menu > Data Routing submenu > BGP & OSPF); the following page appears: Figure 3-200: Page Displaying Area for Configuration File Create a configuration file for the protocol daemon and also for Zebra. Zebra is Quagga's IP routing management daemon which provides kernel routing table updates, interface lookups, and redistribution of routes between the routing protocols.
Page 283: Objects And Rules
SIP User's Manual 3. Web-Based Management • Zebra: interface eth1 ; instructs the daemon to query and update routing information via a specific WAN device log syslog Click OK to save the settings. 3.3.3.8 Objects and Rules The Objects and Rules submenu allows you to configure objects and rules. Once defined, they can later be used in other configurations (e.g., in Access List rules).
Page 284: Figure 3-202: Adding A Service Protocol
Mediant 800 MSBG Click the New Entry link; the following page appears: Figure 3-202: Adding a Service Protocol In the 'Service Name' field, enter a name for the service protocol. In the 'Service Description' field, enter a brief description of this service.
Page 285: Figure 3-205: Defining Name For Network Object
SIP User's Manual 3. Web-Based Management Click the New Entry link; the following page appears: Figure 3-205: Defining Name for Network Object In the 'Description' field, enter a name for the network object. Click the New Entry link; the following page appears: Figure 3-206: Defining Network Object Type From the 'Network Object Type' drop-down list, select a network object type;...
Page 286: Figure 3-208: Defining Scheduler Rule Name
Mediant 800 MSBG Click the New Entry link; the following page appears: Figure 3-208: Defining Scheduler Rule Name In the 'Name' field, enter a name for the Scheduler rule. Under the 'Rule Activity Settings' group, specify whether the rule is active or inactive during the designated time period, by selecting the appropriate option.
Page 287: Configuring Network Connections
SIP User's Manual 3. Web-Based Management Enter the desired start and end time values for the rule. Note: The defined start and end time is applied to all days of the week that you selected previously. Click OK to return to the previous page, and then click OK again to return to the main page.
Page 288: Figure 3-211: Configuring Network Connections
Mediant 800 MSBG • Internet Protocol Security Server LAN Ethernet switch Advanced connections: • LAN Bridging • VLAN Interface • Internet Protocol over Internet Protocol • General Routing Encapsulation To access the Network Connection list table: Click the Connections item (Configuration tab > Data menu > Data System submenu >...
Page 289: Figure 3-213: Defining Internet Connection Type
SIP User's Manual 3. Web-Based Management Select whether you want to configure an Internet connection, a VPN connection, or advanced connections: • For configuring an Internet connection: Select the 'Internet Connection' option, and then click Next; the following wizard page appears: Figure 3-213: Defining Internet Connection Type Select the required Internet connection type, click Next, and then follow the instructions provided by the wizard.
Page 290: Figure 3-215: Defining Virtual Private Network Over Internet
Mediant 800 MSBG • For configuring a VPN-over-Internet connection: Select the 'Connect to a Virtual Private Network over the Internet' option, and then click Next; the following wizard page appears: Figure 3-215: Defining Virtual Private Network over Internet Select the VPN connection type, click Next, and then follow the instructions provided by the wizard.
Page 291 SIP User's Manual 3. Web-Based Management • For manually configuring a new connection: Select the 'Advanced Connection' option, and then click Next; the following wizard page appears: Select the required connection type, click Next, and then follow the instructions provided by the wizard. Version 6.2 February 2011...
Page 292: Figure 3-217: Advanced Connection Wizard Tree
Mediant 800 MSBG The Advanced Connection wizard tree is illustrated below: Figure 3-217: Advanced Connection Wizard Tree When the wizard completes the initial configuration (by clicking Finish), the new connection type appears listed in the Network Connections page. 3.3.3.9.2 LAN Switch The LAN Switch interface represents all the device's ports.
Page 293: Figure 3-219: Switch Tab
SIP User's Manual 3. Web-Based Management Select the Switch tab; the displayed table lists all available ports, their status, and the VLANs of which they are members. Untagged packets (packets with no VLAN tag) that arrive at a port are tagged with the VLAN number that appears under the PVID (Port VLAN Identifier) column.
Page 294: Figure 3-222: Stp Tab
Mediant 800 MSBG Select the STP tab. Figure 3-222: STP Tab Select the 'STP' check box to enable the Spanning Tree Protocol on the device. You should use this to ensure that there are no loops in your network configuration, and apply these settings in case your network consists of multiple switches, or other bridges apart from those created by the device.
Page 295: Figure 3-223: Ethernet Connection Option
SIP User's Manual 3. Web-Based Management ♦ Point-to-Point: Specifies if a point-to-point links is established, or permits the device to establish a point-to-point link. The possible field values are Enable, Disable, or Auto. ♦ Edge: Specifies if a edge links is established, or permits the device to establish a point-to-point link.
Page 296: Figure 3-225: Internet Connection For External Cable Modem Added
Mediant 800 MSBG Select the 'Dynamic Negotiation (DHCP)' option, and then click Next; a summary of the new connection is shown. Figure 3-225: Internet Connection for External Cable Modem Added Select the 'Edit the Connection' check box if you want to edit the new connection after clicking Finish.
Page 297: Figure 3-228: Manual Ip Address Configuration
SIP User's Manual 3. Web-Based Management Select the 'Manual IP Address Configuration' option, and then click Next; a summary of the new connection is shown. Figure 3-228: Manual IP Address Configuration Enter the IP address, subnet mask, default gateway, and DNS server addresses in their respective fields.
Page 298: Figure 3-230: Defining Internet Connection Type
Mediant 800 MSBG To create a PPPoE connection: In the 'Connections' page, click the New icon; the Connection Wizard opens. Select the 'Internet Connection' option, and then click Next. Figure 3-230: Defining Internet Connection Type Select the 'Point-to-Point Protocol over Ethernet (PPPoE)' option, and then click Next.
Page 299 SIP User's Manual 3. Web-Based Management To edit the PPPoE connection: In the 'Connections' page, click the WAN PPPoE link; the General tab appears displaying general properties. Select the Settings tab to edit various settings (see ''Editing Existing Connections'' on page 328).
Page 300: Figure 3-234: Selecting Lan Interfaces For Bridge Connection
Mediant 800 MSBG ♦ Support Encryption (40 Bit Keys): Select this check box if your peer supports 40 bit encryption keys. ♦ Support Maximum Strength Encryption (128 Bit Keys): Select this check box if your peer supports 128 bit encryption keys.
Page 301: Figure 3-235: Lan Bridge Successfully Added
SIP User's Manual 3. Web-Based Management Click Next; the LAN bridge is successfully added. Figure 3-235: LAN Bridge Successfully Added Select the 'Edit the Connection' check box if you want to edit the new connection after clicking Finish. Click Finish to save the settings; the new bridge is added to the network connections list.
Page 302: Figure 3-237: Assigning Vlan To Lan Ports
Mediant 800 MSBG In the 'VLAN ID' field, enter a value for the VLAN ID, and then click Next. If you chose to create the VLAN over the WAN, skip to Step 9. If you chose to create the VLAN over the LAN bridge, the following page appears.
Page 303: Figure 3-239: Vlan Interface Advanced Tab
SIP User's Manual 3. Web-Based Management To edit the VLAN interface connection: In the 'Connections' page, click the VLAN link (e.g., "LAN Switch VLAN 401"); the General tab appears displaying general properties. Select the Settings tab to edit various settings (see ''Editing Existing Connections'' on page 328).
Page 304: Figure 3-241: Defining Pptp Properties
Mediant 800 MSBG 3.3.3.9.6 Point-to-Point Tunneling Protocol (PPTP) Point-to-Point Tunneling Protocol (PPTP) is a protocol developed by Microsoft targeted at creating VPN connections over the Internet. This enables remote users to access the device via any ISP that supports PPTP on its servers. PPTP encapsulates network traffic, encrypts content using Microsoft's Point-to-Point Encryption (MPPE) protocol that is based on RC4, and routes using the generic routing encapsulation (GRE) protocol.
Page 305: Figure 3-243: Selecting Vpn Type For Ipsec
SIP User's Manual 3. Web-Based Management The following procedure describes how to create a PPTP VPN connection. To create a PPTP VPN connection: In the 'Connections' page, click the New icon; the Connection Wizard opens. Select the 'Connect to a Virtual Private Network over the Internet' option, and then click Next.
Page 306: Figure 3-246: Pptp Vpn Successfully Added
Mediant 800 MSBG Click Next; the following is displayed if successfully configured: Figure 3-246: PPTP VPN Successfully Added Select the 'Edit the Newly Created Connection' check box if you want to edit the new connection after clicking Finish. Click Finish to save the settings; the new PPTP VPN connection is added to the network connections list.
Page 307: Figure 3-248: Pptp Tab
SIP User's Manual 3. Web-Based Management • PPP Authentication: PPP currently supports four authentication protocols: Password Authentication Protocol (PAP), Challenge Handshake Authentication Protocol (CHAP), and Microsoft CHAP version 1 and 2. Select the authentication protocols that the device may use when negotiating with a PPTP server. Select all the protocols if no information is available about the server's authentication protocols.
Page 308: Figure 3-249: Vpn Connection Type
Mediant 800 MSBG 3.3.3.9.7 Point-to-Point Tunneling Protocol Server (PPTP Server) The device can act as a Point-to-Point Tunneling Protocol Server (PPTP Server), accepting PPTP client connection requests. To create a PPTP server: In the 'Connections' page, click the New icon; the Connection Wizard opens.
Page 309: Figure 3-252: Pptp Server Added Successfully
SIP User's Manual 3. Web-Based Management Specify the IP address range that the device reserves for remote users, and then click Next; the following is displayed if successfully configured: Figure 3-252: PPTP Server Added Successfully Note that the attention message alerting that there are no users with VPN permissions.
Page 310: Figure 3-254: Defining L2Tp Properties
Mediant 800 MSBG Connecting the device to a remote network using a Virtual Private Network (VPN) tunnel over the Internet. This enables secure transfer of data to another location over the Internet, using private and public keys for encryption and digital certificates, and user name and password for authentication.
Page 311: Figure 3-256: Selecting Vpn Type For Ipsec
SIP User's Manual 3. Web-Based Management To create a L2TP VPN connection: In the 'Connections' page, click the New icon; the Connection Wizard opens. Select the 'Connect to a Virtual Private Network over the Internet' option, and then click Next. Figure 3-256: Selecting VPN Type for IPSec Select the 'VPN Client or Point-To-Point' option, and then click Next.
Page 312: Figure 3-259: L2Tp Successfully Added
Mediant 800 MSBG In the 'IPSec Shared Secret' field, enter the IPSec shared secret, which is the encryption key jointly decided upon with the network you are trying to access. Click Next; the following is displayed if successfully configured: Figure 3-259: L2TP Successfully Added Select the 'Edit the Newly Created Connection' check box if you want to edit the new connection after clicking Finish.
Page 313 SIP User's Manual 3. Web-Based Management • On Demand: Select this check box to initiate the PPP session only when packets are sent over the Internet. • Time Between Reconnect Attempts: Specify the duration between PPP reconnected attempts, as provided by your ISP. •...
Page 314: Figure 3-261: L2Tp Tab
Mediant 800 MSBG Select the L2TP tab. Figure 3-261: L2TP Tab • In the 'L2TP Server Host Name or IP Address' field, enter the connection's host name or IP address obtained from your ISP. • In the 'Shared Secret' field, enter the shared secret value obtained from your ISP.
Page 315: Figure 3-264: Defining L2Tp Properties
SIP User's Manual 3. Web-Based Management Select the 'Layer 2 Tunneling Protocol Server (L2TP Server)' option, and then click Next. Figure 3-264: Defining L2TP Properties In the 'Start IP Address' and 'End IP Address' fields, specify the address range that the device reserves for remote users.
Page 316: Figure 3-267: Selecting Vpn Type For Ipsec
Mediant 800 MSBG Click the Click here to create VPN users link to define remote users that will be granted access to your home network. Click OK to save settings; the new L2TP server connection is added to the Network Connection list.
Page 317: Figure 3-269: Defining Ipsec Properties
SIP User's Manual 3. Web-Based Management Select the 'Internet Protocol Security Server (IPSec)' option, and then click Next. Figure 3-269: Defining IPSec Properties In the 'Host Name or IP Address of Destination Gateway' field, enter the host or IP address of the destination gateway. From the 'Remote IP' drop-down list, select the method for specifying the remote IP address, which serves as the tunnel's endpoint.
Page 318: Figure 3-271: Ipsec Tab
Mediant 800 MSBG To define (edit) additional properties, click the Edit icon corresponding to the VPN IPSec connection in the connection list; the General, Settings, Routing, and IPSec tabs appear. For descriptions of the parameters in the General, Settings, and Routing tabs, see ''Editing Existing Connections'' on page 328.
Page 319 SIP User's Manual 3. Web-Based Management ♦ Remote Subnet: This section is identical to the 'Local Subnet' section above, but is for defining the remote endpoint. Compress (Support IPComp protocol): Select this check box to compress packets during encapsulation with the IP Payload Compression protocol.
Page 320: Figure 3-272: Ipsec Tab - Ipsec Automatic Phase 1
Mediant 800 MSBG IPSec Automatic Phase 1 – Peer Authentication: Figure 3-272: IPSec Tab - IPSec Automatic Phase 1 Mode: Select the IPSec mode – either 'Main Mode' or 'Aggressive Mode'. Main mode is a secured but slower mode, which presents negotiable propositions according to the authentication algorithms that you select in the check boxes.
Page 321: Figure 3-273: Ipsec Tab - Ipsec Automatic Phase 2
SIP User's Manual 3. Web-Based Management Hash Algorithm: Select the hash algorithms that the device attempts to use when negotiating with the IPSec peer. Group Description Attribute: Select the Diffie-Hellman (DH) group description(s). Diffie-Hellman is a public-key cryptography scheme that allows two parties to establish a shared secret over an insecure communications channel.
Page 322: Figure 3-274: Ipsec Tab - Ipsec Manual
Mediant 800 MSBG ♦ Manual key definition: Figure 3-274: IPSec Tab - IPSec Manual Security Parameter Index (SPI): A 32 bit value that together with an IP address and a security protocol, uniquely identifies a particular security association. The local and remote values must be coordinated with their respective values on the IPSec peer.
Page 323: Figure 3-276: Vpn Protocols
SIP User's Manual 3. Web-Based Management Select the 'VPN Server' option, and then click Next. Figure 3-276: VPN Protocols Select the 'Internet Protocol Security Server (IPSec Server)' option, and then click Next. Figure 3-277: IPSec Shared Secret Key Enter the IPSec shared secret, which is the encryption key jointly decided upon with the network you are trying to access, and then click Next;...
Page 324: Figure 3-279: Configuring General Ipip Parameters
Mediant 800 MSBG 3.3.3.9.12 Internet Protocol over Internet Protocol (IPIP) The device allows you to create an IPIP tunnel to another router, by encapsulating IP packets in IP. This tunnel can be managed as any other network connection. Supported by many routers, this protocol enables using multiple network schemes.
Page 325: Figure 3-281: Ipip Tab
SIP User's Manual 3. Web-Based Management Select the Routing tab to edit the routing parameters (see ''Editing Existing Connections'' on page 328). Select the IPIP tab to define the tunnels's remote endpoint IP address. Figure 3-281: IPIP Tab Select the Advanced tab to enable the firewall for this network connection (see ''Editing Existing Connections'' on page 328).
Page 326: Figure 3-284: Editing Gre Remote Endpoint Ip Address
Mediant 800 MSBG Select the 'Edit the Connection' check box if you want to edit the new connection after clicking Finish. Click Finish to save the settings; the new GRE tunnel is added to the network connections list. To edit the GRE tunnel connection: In the 'Connections' page, click the "WAN GRE"...
Page 327: Figure 3-285: Example Scenario Setup
SIP User's Manual 3. Web-Based Management The devices' WAN ports are connected to the WAN (where the DHCP server is available) Figure 3-285: Example Scenario Setup To create a tunnel, each MSBG device must be made aware of the other's WAN IP address (the information must be exchanged).
Page 328: Figure 3-287: Defining Gre Tunnel For Device B
Mediant 800 MSBG Create a GRE tunnel for device "B": In the 'Connections' page, click the New icon. Select the 'Advanced Connection' option, and then click Next. Select the 'General Routing Encapsulation (GRE)' option, and then click Next. Figure 3-287: Defining GRE Tunnel for Device B Enter 10.71.81.191 as the tunnel's remote endpoint IP address.
Page 329: Figure 3-288: Editing Network Connection - General Tab
SIP User's Manual 3. Web-Based Management To edit connections: Access the configuration tabs: • In the 'Connections' page, click the Edit icon corresponding to the network connection that you want to edit; the General tab is displayed, showing general properties of the connection type (e.g., WAN Ethernet connection). •...
Page 330 Mediant 800 MSBG • Schedule: by default, the connection is always active. However, if you have defined scheduler rules (see ''Configuring Scheduler Rules'' on page 285), you can select one of these (time segments during which the connection is active).
Page 331: Figure 3-290: Editing Network Connection - Routing Tab
SIP User's Manual 3. Web-Based Management Select the Routing tab: Figure 3-290: Editing Network Connection - Routing Tab You can choose to setup your device to use static or dynamic routing. Dynamic routing automatically adjusts how packets travel on the network, whereas static routing specifies a fixed routing path to destinations.
Page 332: Figure 3-291: Editing Network Connection - Advanced Tab
Mediant 800 MSBG Click OK to save the settings. Select the Advanced tab: Figure 3-291: Editing Network Connection - Advanced Tab • Internet Connection Firewall: Your device's firewall helps protect your computer by preventing unauthorized users from gaining access to it through a network such as the Internet.
Page 333: Maintenance Tab
SIP User's Manual 3. Web-Based Management Maintenance Tab The Maintenance tab on the Navigation bar displays menus in the Navigation tree related to device maintenance procedures. These menus include the following: Maintenance (see ''Maintenance'' on page 333) Software Update (see ''Software Update'' on page 337) 3.4.1 Maintenance The Maintenance menu allows you to perform various maintenance procedures.
Page 334: Resetting The Device
Mediant 800 MSBG 3.4.1.1.1 Resetting the Device The 'Maintenance Actions' page allows you to remotely reset the device. In addition, before resetting the device, you can choose the following options: Save the device's current configuration to the device's flash memory (non-volatile).
Page 335: Figure 3-293: Reset Confirmation Message Box
SIP User's Manual 3. Web-Based Management Click the Reset button; a confirmation message box appears, requesting you to confirm. Figure 3-293: Reset Confirmation Message Box Click OK to confirm device reset; if the parameter 'Graceful Option' is set to 'Yes' (in Step 3), the reset is delayed and a screen displaying the number of remaining calls and time is displayed.
Page 336 Mediant 800 MSBG Click OK to confirm device Lock; if 'Graceful Option' is set to 'Yes', the lock is delayed and a screen displaying the number of remaining calls and time is displayed. Otherwise, the lock process begins immediately. The 'Current Admin State' field displays the current state: LOCKED or UNLOCKED.
Page 337: Software Update
SIP User's Manual 3. Web-Based Management 3.4.2 Software Update The Software Update menu allows you to upgrade the device's software, install Software Upgrade Key, and load/save configuration file. This menu includes the following page items: Load Auxiliary Files (see ''Loading Auxiliary Files'' on page 337) Software Upgrade Key (see ''Loading Software Upgrade Key'' on page 339) Software Upgrade Wizard (see ''Software Upgrade Wizard'' on page 341) Configuration File (see ''Backing Up and Loading Configuration File'' on page 344)
Page 338: Figure 3-295: Load Auxiliary Files
Mediant 800 MSBG Notes: • You can schedule automatic loading of updated auxiliary files using HTTP/HTTPS (for more details, refer to the Product Reference Manual). • For a detailed description on auxiliary files, see ''Auxiliary Configuration Files'' on page 393.
Page 339: Loading Software Upgrade Key
You can load a Software Upgrade Key using one of the following management tools: Web interface AudioCodes’ EMS (refer to EMS User’s Manual or EMS Product Description) Warning: Do not modify the contents of the Software Upgrade Key file.
Page 340: Figure 3-296: Software Upgrade Key Status
Mediant 800 MSBG To load a Software Upgrade Key: Open the 'Software Upgrade Key Status' page (Maintenance tab > Software Update menu > Software Upgrade Key). Figure 3-296: Software Upgrade Key Status Page Backup your current Software Upgrade Key as a precaution so that you can re-load this backup key to restore the device's original capabilities if the new key doesn’t...
Page 341: Software Upgrade Wizard
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 342: Figure 3-298: Start Software Upgrade Wizard Screen
• If you upgraded your cmp and the "SW version mismatch" message appears in the Syslog or Web interface, then your Software Upgrade Key does not support the new cmp version. Contact AudioCodes support for assistance. • If you use the wizard to load an ini file, parameters excluded from the ini...
Page 343 SIP User's Manual 3. Web-Based Management Click the Start Software Upgrade button; the 'Load a CMP file' Wizard page appears. Note: At this stage, you can quit the Software Update Wizard, by clicking Cancel , without requiring a device reset. However, once you start uploading a cmp file, the process must be completed with a device reset.
Page 344: Backing Up And Loading Configuration File
Mediant 800 MSBG After the device resets, the 'End Process' screen appears displaying the burned configuration files: Figure 3-299: End Process Wizard Page Click End Process to close the wizard; the Web Login dialog box appears. Enter your login user name and password, and then click OK; a message box appears informing you of the new cmp file.
Page 345: Figure 3-300: Configuration File
SIP User's Manual 3. Web-Based Management To save the ini / data file: Open the 'Configuration File' page (Maintenance tab > Software Update menu > Configuration File). You can also access this page from the toolbar, by clicking Device Actions, and then choosing Load Configuration File or Save Configuration File.
Page 346: Status & Diagnostics Tab
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 347: Figure 3-301: Device Information
SIP User's Manual 3. Web-Based Management To access the 'Device Information' page: Open the 'Device Information' page (Status & Diagnostics tab > System Status menu > Device Information). Figure 3-301: Device Information Page To delete a loaded file: Click the Delete button corresponding to the file that you want to delete. Deleting a file takes effect only after device reset (see ''Resetting the Device'' on page 334).
Page 348: Viewing Ethernet Port Information
Mediant 800 MSBG 3.5.1.2 Viewing Ethernet Port Information The 'Ethernet Port Information' page displays read-only information on the device's Ethernet connection. This includes indicating the active port, duplex mode, and speed. You can also access this page from the 'Home' page (see ''Using the Home Page'' on page 59).
Page 349: Carrier-Grade Alarms
SIP User's Manual 3. Web-Based Management To view WAN port information: Open the ‘WAN Port Information’ page (Status & Diagnostics tab > System Status menu > WAN Port Information). Figure 3-303: WAN Port Information Page 3.5.1.4 Carrier-Grade Alarms The Carrier-Grade Alarms submenu contains the following item: Active Alarms (see ''Viewing Active Alarms'' on page 349) 3.5.1.4.1 Viewing Active Alarms The 'Active Alarms' page displays a list of currently active alarms.
Page 350: Voip Status
Mediant 800 MSBG 3.5.2 VoIP Status The VoIP Status menu allows you to monitor real-time activity of VoIP entities such as IP connectivity, call details, and call statistics. This menu includes the following page items: IP Interface Status (see ''Viewing Active IP Interfaces'' on page 350)
Page 351: Viewing Call Counters
SIP User's Manual 3. Web-Based Management To view performance statistics: Open the 'Basic Statistics’ page (Status & Diagnostics tab > VoIP Status menu > Performance Statistics). Figure 3-306: Basic Statistics Page To reset the performance statistics to zero: Click the Reset Statistics button. 3.5.2.3 Viewing Call Counters The 'IP to Tel Calls Count' and 'Tel to IP Calls Count' pages provide you with statistical...
Page 352: Table 3-43: Call Counters Description
Mediant 800 MSBG Table 3-43: 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. The number of failed calls is represented by the failed-call counters.
Page 353: Viewing Sas/Sbc Registered Users
SIP User's Manual 3. Web-Based Management Counter Description Number of Failed Calls Indicates the number of calls that failed due to unavailable resources or due to No Resources a device lock. The counter is incremented as a result of one of the following release reasons: GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED RELEASE_BECAUSE_GW_LOCKED...
Page 354: Viewing Call Routing Status
Mediant 800 MSBG 3.5.2.5 Viewing Call Routing Status The 'Call Routing Status' page provides you with information on the current routing method used by the device. This information includes the IP address and FQDN (if used) of the Proxy server with which the device currently operates.
Page 355: Figure 3-310: Registration Status
SIP User's Manual 3. Web-Based Management To view Registration status: Open the 'Registration Status' page (Status & Diagnostics tab > VoIP Status menu > Registration Status). Figure 3-310: Registration Status Page Registered Per Gateway: • 'YES' = registration is per device •...
Page 356: Viewing Ip Connectivity
Mediant 800 MSBG 3.5.2.7 Viewing IP Connectivity The 'IP Connectivity' page displays online, read-only network diagnostic connectivity information on all destination IP addresses configured in the 'Outbound IP Routing Table' page (see ''Configuring Outbound IP Routing Table'' on page 165).
Page 357 SIP User's Manual 3. Web-Based Management Column Name Description Fail = Remote side doesn't respond. Init = Connectivity queries not started (e.g., IP address not resolved). Disable = The connectivity option is disabled, i.e., parameter 'Alt Routing Tel to IP Mode' (AltRoutingTel2IPMode ini) is set to 'None' or 'QoS'. Quality Status Determines the QoS (according to packet loss and delay) of the IP address.
Page 358: Data Status
Mediant 800 MSBG 3.5.3 Data Status The Data Status menu is used to view and monitor the device's data routing functionality. This menu includes the following page items: WAN Status (see ''Viewing WAN Status'' on page 358) Connection Statistics (see ''Viewing Network Connection Statistics'' on page 359)
Page 359: Viewing Network Connection Statistics
SIP User's Manual 3. Web-Based Management To run Internet connectivity tests: Click the WAN Status item (Status & Diagnostics tab > Data Status menu > WAN Status), and then click the Have Internet Connection problems? Click here link; the following page appears: Figure 3-313: Running Internet Connectivity Diagnostics Tests Click the Run button.
Page 360: Viewing Logged Security Events
Mediant 800 MSBG To view data on network connections: Click the Connection Statistics item (Status & Diagnostics tab > Data Status menu > Connection Statistics); the following page appears: Figure 3-314: Connection Statistics Page To update the display, click the Refresh button, or click the Automatic Refresh On button to constantly update the displayed parameters.
Page 361: Figure 3-316: Log Settings
SIP User's Manual 3. Web-Based Management The log table displays the following details: • Time: time the event occurred. • Event: there are five kinds of events: ♦ Inbound Traffic: event is a result of an incoming packet. ♦ Outbound Traffic: event is a result of outgoing packet. ♦...
Page 362: Viewing Qos Queues Statistics
Mediant 800 MSBG • Blocked Events group: ♦ All Blocked Connection Attempts: generates a log message for each blocked attempt to establish an inbound connection to the home network or vice versa. You can enable logging of blocked packets of specific types by disabling this option, and enabling some of the more specific options listed below it.
Page 363: Viewing Logged Data Events
SIP User's Manual 3. Web-Based Management 3.5.3.5 Viewing Logged Data Events The Data Log item displays a list of recent events occurred on the device. To view logged messages: Click the Log item (Status & Diagnostics tab > Data Status menu > Data Log); the following page appears: Figure 3-318: System Log Page By default, all log messages are displayed one after another, sorted by their order of...
Page 364: Figure 3-319: Adding A New Filter
Mediant 800 MSBG To add a new log display filter: In the 'Filters' group, click the New Filter link; the 'Filters' group displays a new Component entry. Figure 3-319: Adding a New Filter Using the drop-down lists, select the component and severity level by which to sort the log messages.
Page 365: Running Diagnostic Tests
SIP User's Manual 3. Web-Based Management 3.5.3.6 Running Diagnostic Tests The Diagnostics item can assist you in testing network connectivity and viewing statistics such as the number of packets transmitted and received, round-trip time and success status. This page allows you to run network connectivity tests (ping), query the physical address (MAC) of a host, and run a trace route test.
Page 366 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 367: Ini File-Based Management
SIP User's Manual 4. INI File-Based Management INI File-Based Management The device can also be configured by loading an ini file, which contains user-defined parameters. The ini file can be loaded to the device using the following method: Web interface (see ''Backing Up and Loading Configuration File'' on page 344) The ini file configuration parameters are saved in the device's non-volatile memory when the file is loaded to the device.
Page 368: Configuring Ini File Table Parameters
Mediant 800 MSBG 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 369 SIP User's Manual 4. INI File-Based Management The following displays an example of the structure of an ini file table parameter. [Table_Title] ; This is the title of the table. FORMAT Index = Column_Name1, Column_Name2, Column_Name3; ; This is the Format line. Index 0 = value1, value2, value3;...
Page 370: General Ini File Formatting Rules
Mediant 800 MSBG 4.1.3 General ini File Formatting Rules The ini file must adhere to the following formatting rules: The ini file name must not include hyphens (-) or spaces; if necessary, use an underscore (_) instead. Lines beginning with a semi-colon (;) are ignored. These can be used for adding remarks in the ini file.
Page 371: Secured Encoded Ini File
Typically, it is loaded to or retrieved from the device using HTTP. These protocols are not secure and are vulnerable to potential hackers. To overcome this security threat, the AudioCodes' TrunkPack Downloadable Conversion Utility (DConvert) utility allows you to binary-encode (encrypt) the ini file before loading it to the device (refer to the Product Reference Manual).
Page 372 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 373: Ems-Based Management
EMS-Based Management This section provides a brief description on configuring various device configurations using AudioCodes Element Management System (EMS). The EMS is an advanced solution for standards-based management of MSBGs within VoP networks, covering all areas vital for the efficient operation, administration, management and provisioning (OAM&P) of AudioCodes' families of MSBGs.
Page 374: Adding The Device In Ems
Mediant 800 MSBG 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. Region : defines a group (e.g., geographical location) to which devices can be...
Page 375: Figure 5-3: Adding A Region
SIP User's Manual 5. EMS-Based Management Add a Region for your deployed device, 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), and then click OK;...
Page 376: Configuring Trunks
Mediant 800 MSBG Configuring Trunks This section describes the provisioning of trunks: E1/T1Trunk configuration (see ''General Trunk Configuration'' on page 376) ISDN NFAS (see ''Configuring ISDN NFAS'' on page 377) 5.3.1 General Trunk Configuration This section describes how to provision a PSTN trunk.
Page 377: Configuring Isdn Nfas
SIP User's Manual 5. EMS-Based Management Select a trunk, and then in the Configuration pane, click Trunk SIP Frame; the Trunk SIP Provisioning screen is displayed with the General Settings tab selected. Figure 5-7: General Settings Screen From the 'Protocol Type' drop-down list, select the required protocol. From the 'Framing Method' drop-down list, select the required framing method.
Page 378: Figure 5-8: Ems Isdn Settings Screen
Mediant 800 MSBG With NFAS it is possible to define a group of T1 trunks, called an NFAS group, in which a single D-channel carries ISDN signaling messages for the entire group. The NFAS group’s B-channels are used to carry traffic such as voice or data. The NFAS mechanism also enables definition of a backup D-channel on a different T1 trunk, to be used if the primary D-channel fails.
Page 379 SIP User's Manual 5. EMS-Based Management Select the General Settings tab, and then configure each trunk in the group with the same values for the following parameters: • Protocol Type • Framing Method • Line Code 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 380: Configuring Basic Sip Parameters
Mediant 800 MSBG 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 Navigation pane, select VoIP > SIP, and then in the Configuration pane, select SIP Protocol Definitions;...
Page 381 SIP User's Manual 5. EMS-Based Management 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 382: Provisioning Sip Srtp Crypto Offered Suites
Mediant 800 MSBG Provisioning SIP SRTP Crypto Offered Suites This section describes how to configure offered SRTP crypto suites in the SDP. To configure SRTP crypto offered suites: In the Navigation pane, select VoIP > SIP, and then in the Configuration pane, select SIP Protocol Definitions;...
Page 383: Configuring The Device To Operate With Snmpv3
SIP User's Manual 5. EMS-Based Management Select the MLPP tab; the 'MLPP' screen appears. 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), DiffServ value with PremiumServiceClassMediaDiffserv global gateway parameter, or by using IP...
Page 384: Configuring Snmpv3 Using Ssh
Mediant 800 MSBG 5.7.1 Configuring SNMPv3 using SSH The procedure below describes how to configure SNMPv3 using SSH. This is a more secure way of configuring the SNMPv3 connection between the EMS and the device, i.e., before you have a secure SNMP connection, there could be eavesdropping.
Page 385: Configuring Ems To Operate With A Pre-Configured Snmpv3 System
SIP User's Manual 5. EMS-Based Management 5.7.2 Configuring EMS to Operate with a Pre-configured SNMPv3 System The procedure below describes how to configure the device with a pre-configured SNMPv3. To configure EMS to operate with a pre-configured SNMPv3 system: In the MG Tree, select the required Region to which the device belongs, and then right-click the device.
Page 386: Configuring Snmpv3 To Operate With Non-Configured Snmpv3 System
Mediant 800 MSBG 5.7.3 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; the device is displayed in the Main pane.
Page 387: Cloning Snmpv3 Users
SIP User's Manual 5. EMS-Based Management 5.7.4 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 388: Upgrading The Device's Software
Mediant 800 MSBG 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 389: Figure 5-17: Files Manager Screen
SIP User's Manual 5. EMS-Based Management Select the cmp file, by performing the following: Ensure that the CMP File Only option is selected. In the 'CMP' field, click the browse button and navigate to the required cmp file; the software version number of the selected file appears in the 'Software Version' field.
Page 390 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 391: Restoring Factory Default Settings
SIP User's Manual 6. Restoring Factory Default Settings Restoring Factory Default Settings You can restore the device's configuration to factory defaults using one of the following methods: Using the CLI (see ''Restoring Defaults using CLI'' on page 391) Loading an empty ini file (see ''Restoring Defaults using an ini File'' on page 392) Using the hardware Reset button (see Restoring Defaults using Hardware Reset Button on page 392) Restoring Defaults using CLI...
Page 392: Restoring Defaults Using An Ini File
Mediant 800 MSBG The CLI commands are shown in the terminal emulation program (e.g., HyperTerminal) below: Restoring Defaults using an ini File You can restore the device to factory default settings by loading an empty ini file to the device, using the Web interface's 'Configuration File' page (see ''Backing Up and Loading Configuration File'' on page 344).
Page 393: Auxiliary Configuration Files
SIP User's Manual 7. Auxiliary Configuration Files Auxiliary Configuration Files This section describes the auxiliary files that can be loaded to the device: Call Progress Tones (see ''Call Progress Tones File'' on page Distinctive Ringing in the ini file (see Distinctive Ringing on page 396) Prerecorded Tones (see ''Prerecorded Tones File'' on page CAS (see CAS Files on page 399) Dial Plan (see Dial Plan File on page 400)
Page 394 Mediant 800 MSBG The format attribute can be one of the following: Continuous: A steady non-interrupted sound (e.g., a dial tone). Only the 'First Signal On time' should be specified. All other on and off periods must be set to zero. In this case, the parameter specifies the detection period.
Page 395 SIP User's Manual 7. Auxiliary Configuration Files • High Freq [Hz: Frequency (in Hz) of the higher tone component in case of dual frequency tone, or zero (0) in case of single tone (not relevant to AM tones). • Low Freq Level [-dBm]: Generation level 0 dBm to -31 dBm in dBm (not relevant to AM tones).
Page 396: Distinctive Ringing
Mediant 800 MSBG For example, to configure the dial tone to 440 Hz only, enter the following text: [NUMBER OF CALL PROGRESS TONES] Number of Call Progress Tones=1 #Dial Tone [CALL PROGRESS TONE #0] Tone Type=1 Tone Form =1 (continuous)
Page 397 SIP User's Manual 7. Auxiliary Configuration Files • First (Burst) Ring Off Time [10 msec]: 'Ring Off' period (in 10 msec units) for the first cadence on-off cycle. • Second (Burst) Ring On Time [10 msec]: 'Ring On' period (in 10 msec units) for the second cadence on-off cycle.
Page 398 Mediant 800 MSBG First Ring On Time [10msec]=200 First Ring Off Time [10msec]=400 #GR-506-CORE Ringing Pattern 2 [Ringing Pattern #2] Ring Type=2 Freq [Hz]=20 First Ring On Time [10msec]=80 First Ring Off Time [10msec]=40 Second Ring On Time [10msec]=80 Second Ring Off Time [10msec]=400 7.1.2...
Page 399: Prerecorded Tones File
SIP User's Manual 7. Auxiliary Configuration Files Prerecorded Tones File The CPT file mechanism has several limitations such as a limited number of predefined tones and a limited number of frequency integrations in one tone. To overcome these limitations and provide tone generation capability that is more flexible, the Prerecorded Tones (PRT) file can be used.
Page 400: 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 detailed description, see Dial Plan Prefix Tags for IP-to-Tel Routing on page 418.
Page 401 SIP User's Manual 7. 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 402: Table 7-1: User Information Items
Mediant 800 MSBG 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'. The PBX extension uses this mapping to emulate the behavior of an IP phone.
Page 403 SIP User's Manual 7. Auxiliary Configuration Files Note: The last line in the User Information file must end with a carriage return (i.e., by pressing the <Enter> key). 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 and Configuration Files Parameters'' on page 881)
Page 404 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 405: Ip Telephony Capabilities
SIP User's Manual 8. IP Telephony Capabilities IP Telephony Capabilities This section describes the device's main IP telephony capabilities. Multiple SIP Signaling and Media Interfaces The device supports multiple, logical SIP signaling interfaces and RTP (media) traffic interfaces. This allows you to separate SIP signaling messages and media traffic between different applications (i.e., SAS, Gateway\IP-to-IP, and SBC), and/or between different networks (e.g., when operating with multiple ITSP's).
Page 406: Media Realms
Mediant 800 MSBG traffic between different customers. In such a scenario, the device is configured with multiple SRD's. Typically, one SRD is defined per group of SIP User Agents/UA (e.g. proxies, IP phones, application servers, gateways, softswitches) that communicate with each other. This...
Page 407: Figure 8-2: Back-To-Back Sbc Call Flow (Rtp And Signaling)
SIP User's Manual 8. IP Telephony Capabilities Each SRD may be associated with up to three SIP Interfaces (one per application type - SAS, Gateway\IP-to-IP, and SBC). Each SIP Interface must have a unique signaling port (i.e., no two SIP Interfaces can share the same port - no overlapping). SIP Interfaces are used for the following: Defining different SIP signaling ports (listening UDP, TCP, and TLS, and the UDP source ports) for single or multiple interfaces.
Page 408: Multiple Sip Signaling And Media Configuration Example
Mediant 800 MSBG 8.1.2 Multiple SIP Signaling and Media Configuration Example This section provides an example for configuring multiple SIP signaling and RTP interfaces. In this example, the device serves as the interface between the enterprise's PBX (connected using an E1/T1 trunk) and two ITSP's, as shown in the figure below:...
Page 409: Figure 8-5: Defining A Trunk Group For Pstn
SIP User's Manual 8. IP Telephony Capabilities To configure multiple SIP signaling and RTP interfaces: Configure Trunk Group ID #1 in the 'Trunk Group Table' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Hunt Group > Hunt Group), as shown in the figure below: Figure 8-5: Defining a Trunk Group for PSTN Configure the Trunk in the 'Trunk Settings' page ((Configuration tab >...
Page 410: Figure 8-8: Defining Srds
Mediant 800 MSBG Configure SRDs in the 'SRD Table' page (Configuration tab > VoIP menu > Control Network submenu > SRD Table): • SRD1 associated with media realm "Realm1". • SRD2 associated with media realm "Realm2". Figure 8-8: Defining SRDs Configure the SIP Interfaces in the 'SIP Interface Table' page (Configuration tab >...
Page 411: Figure 8-11: Defining Ip Groups
SIP User's Manual 8. IP Telephony Capabilities Configure IP Groups in the 'IP Group Table' page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). The figure below configures IP Group for ITSP A. Do the same for ITSP B but for Index 2 with SRD 1 and Media Realm to "Realm2".
Page 412: Dynamic Jitter Buffer Operation
Mediant 800 MSBG Dynamic Jitter Buffer Operation Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames.
Page 413: Table 8-1: Dialing Plan Notations
SIP User's Manual 8. IP Telephony Capabilities Gateway and IP-to-IP This section describes various Gateway and IP-to-IP application features. 8.3.1 Dialing Plan Features This section discusses various dialing plan features supported by the device: Dialing plan notations (see ''Dialing Plan Notation for Routing and Manipulation'' on page 413) Digit mapping (see ''Digit Mapping'' on page 414) External Dial Plan file containing dial plans (see ''External Dial Plan File'' on page 415)
Page 414: Table 8-2: Digit Map Pattern Notations
Mediant 800 MSBG Notation Description Example Pound sign (#) Represents the end of a 54324xx#: represents a 7-digit number that starts with at the end of a number. 54324. number A single asterisk Represents any *: represents any number (i.e., all numbers).
Page 415: External Dial Plan File
SIP User's Manual 8. IP Telephony Capabilities Below is an example of a digit map pattern containing eight rules: DigitMapping = 11xS|00[1- 7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x|xx.T In the example, the rule "00[1-7]xxx" denotes dialed numbers that begin with 00, and then any digit from 1 through 7, followed by three digits (of any number). Once the device receives these digits, it does not wait for additional digits, but starts sending the collected digits (dialed number) immediately.
Page 416 Mediant 800 MSBG The prefix can include asterisks ("*") and number signs ("#"). The number of additional digits can include a numerical range in the format x-y. Empty lines and lines beginning with a semicolon (";") are ignored. An example of a Dial Plan file with indices (in ini-file format before conversion to binary *.dat) is shown below:...
Page 417 SIP User's Manual 8. IP Telephony Capabilities 8.3.1.3.1 Modifying ISDN-to-IP Calling Party Number The device can use the Dial Plan file to change the Calling Party Number value (source number) of the incoming ISDN call when sending to IP. For this feature, the Dial Plan file supports the following syntax: <ISDN Calling Party Number>,0,<new calling number>...
Page 418: Dial Plan Prefix Tags For Ip-To-Tel Routing
Mediant 800 MSBG 8.3.1.4 Dial Plan Prefix Tags for IP-to-Tel Routing The device supports the use of string labels (or "tags") in the external Dial Plan file for tagging incoming IP-to-Tel calls. The special “tag” is added as a prefix to the called party number, and then the 'Inbound IP Routing Table' uses this “tag”...
Page 419: Manipulating Number Prefix
SIP User's Manual 8. IP Telephony Capabilities • The 'Dest. Phone Prefix' field is set to the value "LONG" and this rule is assigned to a long distance Hunt Group (e.g. Hunt Group ID 2). Figure 8-14: Configuring Dial Plan File Label for IP-to-Tel Routing The above routing rules are configured to be performed before manipulation (described in the step below).
Page 420: Emergency Phone Number Services - E911
Mediant 800 MSBG For example, assume that you want to manipulate an incoming IP call with destination number +5492028888888 (area code 202 and phone number 8888888) to the number 0202158888888. To perform such a manipulation, the following configuration is required in...
Page 421: Fxs Device Emulating Psap Using Did Loop-Start Lines
SIP User's Manual 8. IP Telephony Capabilities 8.3.3.1 FXS Device Emulating PSAP using DID Loop-Start Lines The FXS device can be configured to emulate PSAP (using DID loop start lines), according to the Telcordia GR-350-CORE specification. Figure 8-17: FXS Device Emulating PSAP using DID Loop-Start Lines The call flow of an E911 call to the PSAP is as follows: The E911 tandem switch seizes the line.
Page 422 Mediant 800 MSBG When the call is answered by the PSAP operator, the PSAP sends a SIP 200 OK to the FXS device, and the FXS device then generates a polarity reversal signal to the E911 switch. After the call is disconnected by the PSAP, the PSAP sends a SIP BYE to the FXS device, and the FXS device reverses the polarity of the line toward the tandem switch.
Page 423: Table 8-3: Dialed Mf Digits Sent To Psap
SIP User's Manual 8. IP Telephony Capabilities Typically, the MF spills are sent from the E911 tandem switch to the PSAP, as shown in the table below: Table 8-3: Dialed MF Digits Sent to PSAP Digits of Calling Number Dialed MF Digits 8 digits "nnnnnnnn"...
Page 424: Network To Psap Did Lines
Mediant 800 MSBG 8.3.3.2 FXO Device Interworking SIP E911 Calls from Service Provider's IP Network to PSAP DID Lines The FXO device can interwork SIP emergency E911 calls from the Service Provider's IP network to the analog PSAP DID lines. The standards that define this interface include TR- TSY-000350 or Bellcore’s GR-350-Jun2003.
Page 425: Table 8-4: Dialed Number By Device Depending On Calling Number
SIP User's Manual 8. IP Telephony Capabilities Following the "hookflash" Wink signal, the PSAP sends DTMF digits. These digits are detected by the device and forwarded to the IP, using RFC 2833 telephony events (or inband, depending on the device's configuration). Typically, this Wink signal followed by the DTMF digits initiates a call transfer.
Page 426 Mediant 800 MSBG 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 427: Pre-Empting Existing Calls For E911 Ip-To-Tel Calls
SIP User's Manual 8. 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 8.3.3.3 Pre-empting Existing Calls for E911 IP-to-Tel Calls...
Page 428: Configuring Dtmf Transport Types
Mediant 800 MSBG 8.3.4 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 429 SIP User's Manual 8. 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 (i.e., disabled) • TxDTMFOption = 3 Note that in this mode, DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0).
Page 430: Fxs And Fxo Capabilities
Mediant 800 MSBG 8.3.5 FXS and FXO Capabilities 8.3.5.1 FXS/FXO Coefficient Types The FXS Coefficient and FXO Coefficient types used by the device can be one of the following: US line type of 600 ohm AC impedance and 40 V RMS ringing voltage for REN = 2...
Page 431: Figure 8-19: Call Flow For One-Stage Dialing
SIP User's Manual 8. IP Telephony Capabilities • Time to wait before dialing • Answer supervision Two-stage dialing (see ''Two-Stage Dialing'' on page 432) Dialing time: DID wink (see ''DID Wink'' on page 433) 8.3.5.2.1.1 One-Stage Dialing One-stage dialing is when the FXO device receives an IP-to-Tel call, off-hooks the PBX line connected to the telephone, and then immediately dials the destination telephone number.
Page 432: Figure 8-20: Call Flow For Two-Stage Dialing
Mediant 800 MSBG Answer Supervision: The Answer Supervision feature enables the FXO device to determine when a call is connected, by using one of the following methods: • Polarity Reversal: device sends a 200 OK in response to an INVITE only when it detects a polarity reversal.
Page 433 SIP User's Manual 8. IP Telephony Capabilities 8.3.5.2.1.3 DID Wink The device's FXO ports support Direct Inward Dialing (DID). DID is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant.
Page 434: Figure 8-21: Call Flow For Automatic Dialing
Mediant 800 MSBG 8.3.5.2.2.1 Automatic Dialing Automatic dialing is defined using the ini file parameter table TargetOfChannel (see Analog Telephony Parameters) or the embedded Web server's 'Automatic Dialing' screen (see ''Automatic Dialing'' on page 184). The SIP call flow diagram below illustrates Automatic Dialing.
Page 435: Figure 8-22: Call Flow For Collecting Digits Mode
SIP User's Manual 8. IP Telephony Capabilities The SIP call flow diagram below illustrates the Collecting Digits Mode. Figure 8-22: Call Flow for Collecting Digits Mode 8.3.5.2.2.3 FXO Supplementary Services The FXO supplementary services include the following: Hold / Transfer toward the Tel side: The ini file parameter LineTransferMode must be set to 0 (default).
Page 436 Mediant 800 MSBG 8.3.5.2.3 Call Termination on FXO Devices This section describes the device's call termination capabilities for its FXO interfaces: Calls terminated by a PBX (see ''Call Termination by PBX'' on page 436) Calls terminated before call establishment (see ''Call Termination before Call...
Page 437: Remote Pbx Extension Between Fxo And Fxs Devices
SIP User's Manual 8. IP Telephony Capabilities 8.3.5.2.3.2 Call Termination before Call Establishment The device supports the following call termination methods before a call is established: Call termination upon receipt of SIP error response (in Automatic Dialing mode): By default, when the FXO device operates in Automatic Dialing mode, there is no method to inform the PBX if a Tel-to-IP call has failed (SIP error response - 4xx, 5xx or 6xx - is received).
Page 438: Figure 8-23: Fxo-Fxs Remote Pbx Extension (Example)
Mediant 800 MSBG The following is required: FXO interfaces with ports connected directly to the PBX lines (shown in the figure below) FXS interfaces for the 'remote PBX extension' Analog phones (POTS) PBX (one or more PBX loop start lines)
Page 439: Figure 8-24: Mwi For Remote Extensions
SIP User's Manual 8. IP Telephony Capabilities 8.3.5.3.2 Dialing from PBX Line or PSTN The procedure below describes how to dial from a PBX line (i.e., from a telephone directly connected to the PBX) or from the PSTN to the 'remote PBX extension' (i.e., telephone connected to the FXS interface).
Page 440: Figure 8-25: Call Waiting For Remote Extensions
Mediant 800 MSBG 8.3.5.3.4 Call Waiting for Remote Extensions When the FXO device detects a Call Waiting indication (FSK data of the Caller Id - CallerIDType2) from the PBX, it sends a proprietary INFO message, which includes the caller identification to the FXS device. Once the FXS device receives this INFO message, it plays a call waiting tone and sends the caller ID to the relevant port for display.
Page 441: Figure 8-28: Fxs Tel-To-Ip Routing Configuration
SIP User's Manual 8. IP Telephony Capabilities In the ‘Outbound IP Routing Table’ page (see ''Configuring Outbound IP Routing Table'' on page 165), enter 20 for the destination phone prefix, and 10.1.10.2 for the IP address of the FXO device. Figure 8-28: FXS Tel-to-IP Routing Configuration Note: For the transfer to function in remote PBX extensions, Hold must be disabled...
Page 442: Configuring Alternative Routing (Based On Connectivity And Qos)
Mediant 800 MSBG 8.3.6 Configuring Alternative Routing (Based on Connectivity and QoS) The Alternative Routing feature enables reliable routing of Tel-to-IP calls when a Proxy isn’t used. The device periodically checks the availability of connectivity and suitable Quality of Service (QoS) before routing. If the expected quality cannot be achieved, an alternative IP route for the prefix (phone number) is selected.
Page 443: Fax And Modem Capabilities
SIP User's Manual 8. IP Telephony Capabilities 8.3.7 Fax and Modem Capabilities This section describes the device's fax and modem capabilities, and includes the following main subsections: Fax and modem operating modes (see ''Fax/Modem Operating Modes'' on page 443) Fax and modem transport modes (see ''Fax/Modem Transport Modes'' on page 443) V.34 fax support (see V.34 Fax Support on page 449) V.152 support (see ''V.152 Support'' on page 452) 8.3.7.1...
Page 444 Mediant 800 MSBG 8.3.7.2.1 T.38 Fax Relay Mode In Fax Relay mode, fax signals are transferred using the T.38 protocol. T.38 is an ITU standard for sending fax across IP networks in real-time mode. The device currently supports only the T.38 UDP syntax.
Page 445 SIP User's Manual 8. IP Telephony Capabilities To configure automatic T.38 mode, perform the following configurations: IsFaxUsed = 0 FaxTransportMode = 1 Additional configuration parameters: • FaxRelayEnhancedRedundancyDepth • FaxRelayRedundancyDepth • FaxRelayECMEnable • FaxRelayMaxRate 8.3.7.2.2 G.711 Fax / Modem Transport Mode In this mode, when the terminating device detects fax or modem signals (CED or AnsAM), it sends a Re-INVITE message to the originating device requesting it to re-open the channel in G.711 VBD with the following adaptations:...
Page 446 Mediant 800 MSBG For G.711A-law: a=gpmd:0 vbd=yes;ecan=on For G.711 µ-law: a=gpmd:8 vbd=yes;ecan=on In this mode, the parameter FaxTransportMode is ignored and automatically set to ‘transparent’. To configure fax fallback mode, set IsFaxUsed to 3. 8.3.7.2.4 Fax/Modem Bypass Mode In this proprietary mode, when fax or modem signals are detected, the channel automatically switches from the current voice coder to a high bit-rate coder (according to the parameter FaxModemBypassCoderType).
Page 447 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 448 Mediant 800 MSBG V34ModemTransportType = 2 BellModemTransportType = 2 8.3.7.2.6 Fax / Modem Transparent with Events Mode In this mode, fax and modem signals are transferred using the current voice coder with the following automatic adaptations: Echo Canceller = on (or off, for modems)
Page 449: V.34 Fax Support
SIP User's Manual 8. IP Telephony Capabilities • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission. Instead, use the modes Bypass (see ''Fax/Modem Bypass Mode'' on page 446) or Transparent with Events (see ''Fax / Modem Transparent with Events Mode'' on page 448) for modem.
Page 450 Mediant 800 MSBG V34ModemTransportType = 2 (Modem bypass) V32ModemTransportType = 2 V23ModemTransportType = 2 V22ModemTransportType = 2 Configure the following parameters to use bypass mode for V.34 faxes and T.38 for T.30 faxes: FaxTransportMode = 1 (Relay) V34FaxTransportType = 2 (Bypass)
Page 451 Contact: <sip:318@10.8.6.55:5060> Supported: em,100rel,timer,replaces,path,resource-priority,sdp- anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB SCRIBE,UPDATE Remote-Party-ID: <sip:318@10.8.211.250>;party=calling;privacy=off;screen=no;screen- ind=0;npi=1;ton=0 Remote-Party-ID: <sip:2001@10.8.211.250>;party=called;npi=1;ton=0 User-Agent: Audiocodes-Sip-Gateway-/v.6.00A.013.007 Content-Type: application/sdp Content-Length: 433 o=AudiocodesGW 1938931006 1938930708 IN IP4 10.8.6.55 s=Phone-Call c=IN IP4 10.8.6.55 t=0 0 m=audio 6010 RTP/AVP 18 97 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no...
Page 452: V.152 Support
Mediant 800 MSBG a=sendrecv m=image 6012 udptl t38 a=T38FaxVersion:3 a=T38MaxBitRate:33600 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy 8.3.7.4 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 453: Working With Supplementary Services
SIP User's Manual 8. IP Telephony Capabilities 8.3.8 Working with Supplementary Services The device supports the following supplementary services: Call Hold and Retrieve (see ''Call Hold and Retrieve'' on page 453) BRI suspend-resume (see BRI Suspend and Resume on page 455) Consultation (see Consultation Feature on page 455) Call Transfer (see ''Call Transfer'' on page 456) Call Forward (see ''Call Forward'' on page 457)
Page 454: Figure 8-32: Double Hold Sip Call Flow
Mediant 800 MSBG IP address 0.0.0.0 as the connection IP address or the string ‘a=inactive’ in the received Re-INVITE SDP cause the device to enter Hold state and to play the Held tone (configured in the device) to the PBX/PSTN. If the string ‘a=sendonly’ is received in the SDP message, the device stops sending RTP packets, but continues to listen to the incoming RTP packets.
Page 455: Bri Suspend And Resume
SIP User's Manual 8. 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 456: Call Transfer
Mediant 800 MSBG complete), the user can retrieve the held call by pressing hook-flash. The held call can’t be retrieved while Ringback tone is heard. After the Consultation call is connected, the user can toggle between the held and active call by pressing the hook-flash key.
Page 457: Call Forward
SIP User's Manual 8. IP Telephony Capabilities • While hearing Ringback – transfer from alert. • While speaking to C - transfer from active. The device also supports attended (consultation) call transfer for BRI phones (user side) connected to the device and using the Euro ISDN protocol. BRI call transfer is according to ETSI TS 183 036, Section G.2 (Explicit Communication Transfer –...
Page 458: Figure 8-33: Call Forward Reminder With Application Server
Mediant 800 MSBG Disturb, the 603 Decline SIP response code is sent. Three forms of forwarding parties are available: Served party: party configured to forward the call (FXS device). Originating party: party that initiates the first call (FXS or FXO device).
Page 459 SIP User's Manual 8. IP Telephony Capabilities The following parameters are used to configure this feature: EnableNRTSubscription ASSubscribeIPGroupID NRTRetrySubscriptionTime CallForwardRingToneID 8.3.8.5.2 Call Forward Reminder (Off-Hook) Special Dial Tone The device plays a special dial tone (Stutter Dial tone - Tone Type #15) to a specific FXS endpoint when the phone is off-hooked and when a third-party Application server (AS), e.g., a softswitch is used to forward calls intended for the endpoint, to another destination.
Page 460: Call Waiting
Mediant 800 MSBG 8.3.8.5.3 BRI Call Forwarding The device supports call forwarding (CF) services initiated by ISDN Basic BRI phones connected to it. Upon receipt of an ISDN Facility message for call forward from the BRI phone, the device sends a SIP INVITE to the softswitch with a user-defined code in the SIP To header, representing the reason for the call forward.
Page 461: Message Waiting Indication
SIP User's Manual 8. IP Telephony Capabilities Define the Call Waiting indication and Call Waiting Ringback tones in the Call Progress Tones file. You can define up to four Call Waiting indication tones (refer to the parameter FirstCallWaitingToneID in SIP Configuration Parameters). To configure the Call Waiting indication tone cadence, modify the following parameters: NumberOfWaitingIndications,...
Page 462 Mediant 800 MSBG CallerIDType (determines the standard for detection of MWI signals) ETSIVMWITypeOneStandard BellcoreVMWITypeOneStandard VoiceMailInterface EnableVMURI The device supports the following MWI features for its digital PSTN interfaces: For BRI interfaces: This feature provides support for MWI on BRI phones connected to the device and using the Euro ISDN BRI variant.
Page 463: Caller Id
SIP User's Manual 8. IP Telephony Capabilities 8.3.8.8 Caller ID This section discusses the device's Caller ID support. Note: The Caller ID feature is applicable only to FXS/FXO interfaces. 8.3.8.8.1 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 464 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 465: Three-Way Conferencing
SIP User's Manual 8. IP Telephony Capabilities The P-Asserted (or P-Preferred) headers are used to present the originating party’s caller ID even when the caller ID is restricted. These headers are used together with the Privacy header. If Caller ID is restricted: •...
Page 466 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 467: Table 8-5: Mlpp Call Priority Levels (Precedence) And Dscp Configuration Parameters
SIP User's Manual 8. IP Telephony Capabilities 8.3.8.10 Multilevel Precedence and Preemption The device's Multilevel Precedence and Preemption (MLPP) service can be enabled using the CallPriorityMode parameter. MLPP is a call priority scheme, which does the following: Assigns a precedence level (priority level of call) to specific phone calls or messages. Allows higher priority calls (precedence call) and messages to preempt lower priority calls and messages (i.e., terminates existing lower priority calls) that are recognized within a user-defined domain (MLPP domain ID).
Page 468 Mediant 800 MSBG • Reason: preemption ;cause=2 ;text=”Reserved Resources Preempted” • Reason: preemption ;cause=3 ;text=”Generic Preemption” • Reason: preemption ;cause=4 ;text=”Non-IP Preemption” • Reason: preemption; cause=5; text=”Network Preemption” Cause=4: The Reason cause code "Non-IP Preemption" indicates that the session preemption has occurred in a non-IP portion of the infrastructure. The device sends this code in the following scenarios: •...
Page 469: Sip Call Routing Examples
F1 INVITE (10.8.201.108 >> 10.8.201.161): INVITE sip:2000@10.8.201.161;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161> Call-ID: 534366556655skKw-6000--2000@10.8.201.108 CSeq: 18153 INVITE Contact: <sip:8000@10.8.201.108;user=phone> User-Agent: Audiocodes-Sip-Gateway/Mediant 800 MSBG/v.6.00.010.006 Supported: 100rel,em Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE, NOTIFY,PRACK,REFER,INFO Content-Type: application/sdp Content-Length: 208 o=AudiocodesGW 18132 74003 IN IP4 10.8.201.108 s=Phone-Call c=IN IP4 10.8.201.108...
Page 470 F2 TRYING (10.8.201.161 >> 10.8.201.108): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161> Call-ID: 534366556655skKw-6000--2000@10.8.201.108 Server: Audiocodes-Sip-Gateway/Mediant 800 MSBG/v.6.00.010.006 CSeq: 18153 INVITE Content-Length: 0 F3 RINGING 180 (10.8.201.161 >> 10.8.201.108): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161>;tag=1c7345...
Page 471: Sip Authentication Example
F5 ACK (10.8.201.108 >> 10.8.201.10): ACK sip:2000@10.8.201.161;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161>;tag=1c7345 Call-ID: 534366556655skKw-6000--2000@10.8.201.108 User-Agent: Audiocodes-Sip-Gateway/Mediant 800 MSBG/v.6.00.010.006 CSeq: 18153 ACK Supported: 100rel,em Content-Length: 0 Note: Phone ‘6000’ goes on-hook and device 10.8.201.108 sends a BYE to device 10.8.201.161.
Page 472 Since the algorithm is MD5: • The username is equal to the endpoint phone number 122. • The realm return by the proxy is audiocodes.com. • The password from the ini file is AudioCodes. • The equation to be evaluated is (according to RFC this part is called A1) ‘122:audiocodes.com:AudioCodes’.
Page 473 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 800 MSBG/v.6.00.010.006 CSeq: 1 REGISTER Contact: sip:122@10.1.1.200: Expires:3600 Authorization: Digest, username: 122, realm="audiocodes.com”, nonce="11432d6bce58ddf02e3b5e1c77c010d2",...
Page 474: Establishing A Call Between Two Devices
8.3.9.3 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 475: Sip Trunking Between Enterprise And Itsps
SIP User's Manual 8. IP Telephony Capabilities 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 476: Figure 8-39: Configuring Proxy Set Id #1 In The Proxy Sets Table
Mediant 800 MSBG To configure call routing between an Enterprise and two ITSPs: Enable the device to register to a Proxy/Registrar server using the parameter IsRegisterNeeded. In the 'Proxy Sets Table' page (see ''Configuring Proxy Sets Table'' on page 126),...
Page 477: Figure 8-40: Configuring Ip Groups #1 And #2 In The Ip Group Table
SIP User's Manual 8. IP Telephony Capabilities In the 'IP Group Table' page (see ''Configuring IP Groups'' on page 119), configure the two IP Groups #1 and #2. Assign Proxy Sets #1 and #2 to IP Groups #1 and #2 respectively.
Page 478: Mapping Pstn Release Cause To Sip Response
Mediant 800 MSBG In the 'Account Table' page, configure a single Account for Hunt Group ID #1, including an authentication user name and password, and enable registration for this Account to ITSP 1 (i.e., Serving IP Group is 1). Figure 8-44: Configuring Account for Registration to ITSP 1 In the 'Inbound IP Routing Table' page, configure that INVITEs with "ITSP1"...
Page 479: Querying Device Channel Resources Using Sip Options
SIP User's Manual 8. IP Telephony Capabilities 8.3.11 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- Resources header with the following parameters: telchs: specifies the total telephone channels as well as the number of free (available) telephone channels...
Page 480: Sbc Application
Mediant 800 MSBG SBC Application This section provides a detailed description of the device's SBC application. This section includes the following subsections: Overview of the SBC application (see ''Overview'' on page 480) SIP networking definitions (see ''SIP Network Definitions'' on page 482)
Page 481: Nat Traversal
SIP User's Manual 8. IP Telephony Capabilities 8.4.1.1 NAT Traversal The device supports NAT traversal, allowing, for example, communication with ITSPs with globally unique IP addresses, for LAN-to-WAN VoIP signaling (and bearer), using two independent legs. In addition, it also enables communication for "far-end" users located behind a NAT on the WAN.
Page 482: Sip Normalization
Mediant 800 MSBG 8.4.1.4 SIP Normalization The device supports SIP normalization, whereby the SBC application can overcome interoperability problems between SIP user agents. This is achieved by the following: Manipulation of SIP URI user and host parts Connection to ITSP SIP trunks on behalf of an IP-PBX - the device can register and utilize user and password to authenticate for the IP-PBX 8.4.1.5...
Page 483: Determining Source And Destination Url
SIP User's Manual 8. IP Telephony Capabilities The flowchart below illustrates this process: Figure 8-47: Routing Process 8.4.3.1 Determining Source and Destination URL The SIP protocol has more than one URL in a dialog establishing request that might represent the source and destination URL. When handling an incoming request, the device determines which SIP headers are used for source and destination URLs.
Page 484: Source Ip Group Classification
Mediant 800 MSBG • Source URL: if exists, obtained from the P-Asserted\Preferred-Identity header; otherwise, from the From header • Destination URL: obtained from the Request-URI REGISTER dialogs: • Source URL: obtained from the To header • Destination URL: obtained from the Request-URI 8.4.3.2...
Page 485: Sbc Ip-To-Ip Routing
SIP User's Manual 8. IP Telephony Capabilities The flowchart below illustrates the classification process: Figure 8-48: Classification Process (Identifying IP Group or Rejecting Call) 8.4.3.3 SBC IP-to-IP Routing The device's SBC application employs a comprehensive and flexible routing scheme: Routing rules according to Layer-3/4 and SIP characteristics Routing to different destination types: •...
Page 486: Ip-To-Ip Inbound And Outbound Manipulation
Mediant 800 MSBG • Destination IP Group (address defined by Proxy Set associated with the IP Group) with the ability of load balancing and redundancy • ENUM query Alternative Routing Routing between two different Layer-3 networks Transport protocol translator (UDP to TCP to TLS) Source and destination user name manipulation (pre/post routing) The device's IP-to-IP routing rules are configured in the IP-to-IP Routing table.
Page 487: Figure 8-50: Sip Uri Manipulation In Ip-To-Ip Routing
SIP User's Manual 8. IP Telephony Capabilities Manipulated destination user and host are performed on the following SIP headers: Request-URI, To, and Remote-Party-ID (if exists). Manipulated source user and host are performed on the following SIP headers: From, P-Asserted (if exists), P-Preferred (if exists), and Remote-Party-ID (if exists).
Page 488: Sip Header Manipulation
Mediant 800 MSBG Below is an example of a call flow and consequent SIP URI manipulations: Figure 8-51: SIP INVITE (Manipulations) from LAN to WAN The SIP message manipulations in the example above (contributing to typical topology hiding) are as follows:...
Page 489: Figure 8-52: Sip Header Manipulation Example
SIP User's Manual 8. IP Telephony Capabilities Translating one SIP response code to another. Topology hiding (generally present in SIP headers such as Via, Record Route, Route and Service-Route). Configurable identity hiding (information related to identity of subscribers for example, P-Asserted-Identity, Referred-By, Identity and Identity-Info).
Page 490: User Registration And Internal Database
Mediant 800 MSBG Notes: • Unknown SIP parts can only be added or removed. • SIP manipulations do not allow you to remove or add mandatory SIP headers. Only the modify option is available for mandatory headers and is performed only on requests that initiate new dialogs. Mandatory SIP headers include To, From, Via, CSeq, Call-Id, and Max-Forwards.
Page 491: Internal Database
SIP User's Manual 8. IP Telephony Capabilities Internal registration database: If the source IP Group is of type User and registration succeeds (replied with 200 OK by the IP-PBX), then the device adds a record to its database that identified the specific contact of this specific user (AOR). This record is used later to route requests to this specific user (either in normal or in survivability modes).
Page 492: Routing Using Internal Database
Mediant 800 MSBG 8.4.4.3 Routing using Internal Database Typically, routing using the database is applicable to all method types other than registrations. To route to a registered user (using the internal dynamic database), the following steps must be taken: An IP2IP Routing rule with the desired input parameters (matching characteristics) and the destination type as IP Group (operation rule).
Page 493: Sbc Media Handling
SIP User's Manual 8. IP Telephony Capabilities Blocking Incoming Calls from Unregistered Users: You can block incoming calls (INVITE requests) from unregistered users (pertaining to USER-type IP Groups). By default, calls from unregistered users are not blocked. This is configured using the parameter SRD.
Page 494: Media Anchoring Without Transcoding (Transparent)
Mediant 800 MSBG Even though the device usually does not change the negotiated media capabilities (mainly performed by the remote user agents), it does examine the media exchange to control negotiated media types (if necessary) and to know how to open the RTP media channels (IP addresses, coder type, payload type etc.).
Page 495: Media Anchoring With Transcoding
SIP User's Manual 8. IP Telephony Capabilities Media port number RTCP media attribute IP address and port (if the parameter EnableRTCPAttribute is set to 1) Each SBC leg allocates and uses the device's local ports (e.g., for RTP\RTCP\fax). The local ports are allocated from a Media Realm associated with each leg. The legs are associated with a Media Realm as follows: If the leg's IP Group is configured with a Media Realm, then this is the associated Media Realm;...
Page 496: Figure 8-55: Transcoding Using Extended Coders (Example)
Mediant 800 MSBG In the scenario depicted in the figure below, the IP phone on the LAN side initiates a call to the IP phone on the WAN. The initial SDP offer (from the LAN leg) includes codec G.711 as its supported codec. Since this is sent to a Destination IP Group that is configured with an extended coder list, on the WAN leg the device adds another supported codec G.729 to...
Page 497: No Media Anchoring
SIP User's Manual 8. IP Telephony Capabilities standard) INVITE without SDP, offer in 180, and answer in PRACK PRACK and UPDATE transactions can also be used for initiating subsequent offer\answer transactions before the INVITE 200 OK response. In a SIP dialog life time, media characteristics after originally determined by the first offer\answer transaction can be changed by using subsequent offer\answer transactions.
Page 498: Interworking Dtmf Methods
Mediant 800 MSBG The No Media Anchoring process is as follows: Identifying a No Media Anchoring call - according to configuration and the call’s properties (such as source, destination, IP Group, and SRD). Handing the identified No Media Anchoring call.
Page 499: Transcoding Modes
SIP User's Manual 8. IP Telephony Capabilities SBCAlternativeDTMFMethod – the device's first priority for DTMF method at each leg is RFC 2833. Therefore, if a specific leg negotiates RFC 2833 successfully, then the chosen DTMF method for this leg is RFC 2833. For legs where RFC 2833 is not negotiated successfully, the device uses this parameter to determine the DTMF method for the leg.
Page 500 Mediant 800 MSBG This feature also allows the definition of a coders preference policy for the SDP offered coders. Coders Preference is done on both legs on the original SDP offer (without the extended coders), and the offered side selects its chosen coders from the suggested coders list.
Page 501: Srtp-Rtp Transcoding
SIP User's Manual 8. IP Telephony Capabilities If the Allowed Coders policy on SDP returns an empty coders list, the device (source leg) rejects the call (SIP 488 or ACK and BYE). If both Coders Extension and Allowed Coders policies on SDP (in this order) returns an empty coders list, the second leg rejects the call (SIP 488, or ACK and BYE).
Page 502: Multiple Rtp Media Streams Per Call Session
Mediant 800 MSBG If two SBC legs (after offer\answer negotiation) use different security types (i.e., one RTP and the other SRTP), then the device performs RTP-SRTP transcoding. To transcode between RTP and SRTP, the following prerequisites must be met: At least one supported SDP "crypto" attribute and parameters...
Page 503: Handling Sip 3Xx Redirect Responses
SIP User's Manual 8. IP Telephony Capabilities If there are no tokens, the request is dropped. New tokens are added to the bucket at a user-defined rate (token rate). If the bucket contains the maximum number of tokens, tokens to be added at that moment are dropped.
Page 504: Figure 8-57: Sip 3Xx Response Handling
Mediant 800 MSBG The prefix is removed before the resultant INVITE is sent to the destination. Figure 8-57: SIP 3xx Response Handling The process of this feature is described using an example: The device receives the Redirect server's SIP 3xx response (e.g., Contact: <sip:User@IPPBX:5060;transport=tcp;param=a>;q=0.5).
Page 505: Table 8-6: Handling Of Sip Diversion And History-Info Headers
SIP User's Manual 8. IP Telephony Capabilities 8.4.8 Interworking SIP Diversion and History-Info Headers This device can be configured to interwork between the SIP Diversion and History-Info headers. This is important, for example, to networks that support the Diversion header but not the History-Info header, or vice versa.
Page 506: Table 8-7: Message Manipulation Actions
Mediant 800 MSBG 8.4.9 SIP Message Manipulation Syntax This section provides a detailed description on the support and syntax for configuring SIP message manipulation rules. For configuring message manipulation rules, see ''Configuring Message Manipulations'' on page 206. 8.4.9.1 Actions The table below lists the actions that can be performed on SIP message manipulation in the Message Manipulations table.
Page 507 SIP User's Manual 8. IP Telephony Capabilities 8.4.9.2.2 Accept-Language An example of the header is shown below: Accept-Language: da, en-gb;q=0.8, en;q=0.7 The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Below is a header manipulation example:...
Page 508 Mediant 800 MSBG 8.4.9.2.4 Call-Id An example of the header is shown below: Call-ID: JNIYXOLCAIWTRHWOINNR@10.132.10.128 The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes String Read Only...
Page 509 SIP User's Manual 8. IP Telephony Capabilities 8.4.9.2.6 Cseq An example of the header is shown below: CSeq: 1 INVITE The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Integer Read Only...
Page 510 Mediant 800 MSBG Below are header manipulation examples: Example 1 Rule: Add a Diversion header to all INVITE messages: MessageManipulations 0 = 1, invite, , header.Diversion, 0," '<tel:+101>;reason=unknown; counter=1;screen=no; privacy=off'", 0; Result: Diversion: <tel:+101>;reason=user- busy;screen=no;privacy=off;counter=1 Example 2 Rule: Modify the Reason parameter in the header to 1, see ''Reason (Diversion)''...
Page 511 SIP User's Manual 8. IP Telephony Capabilities header.event.EVENTKEY.EVENTPACKAGE, 2, "'2'", 0; Result: Event: refer;id=5678 8.4.9.2.9 From An example of the header is shown below: From: <sip:555@10.132.10.128;user=phone>;tag=YQLQHCAAYBWKKRVIMWEQ The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword...
Page 512 Mediant 800 MSBG 8.4.9.2.10 History-Info An example of the header is shown below: History-Info: <sip:UserA@ims.example.com;index=1> History-Info: <sip:UserA@audc.example.com;index=2> The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes HistoryInfo...
Page 513 SIP User's Manual 8. IP Telephony Capabilities Below are header manipulation examples: Example 1 Rule: Add a Min-Se header to the message using a value of 50: MessageManipulations 1 = 1, any, , header.min-se, 0, '50', 0; Result: Min-SE: 50 Example 2 Rule: Modify a Min-Expires header with the min-expires value and add an...
Page 514 Mediant 800 MSBG 8.4.9.2.13 P-Associated-Uri An example of the header is shown below: P-Associated-URI: <sip:12345678@itsp.com> The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Name String Read/Write...
Page 515 SIP User's Manual 8. IP Telephony Capabilities Result: P-Called-Party-ID: <sip:2000@gw.itsp.com> Append a parameter (p1) to all P-Called-Party-Id headers: Example 2 Rule: MessageManipulations 9 = 1, invite, , header.p-called- party-id.param.p1, 0, 'red', 0; Result: P-Called-Party-ID: <sip:2000@gw.itsp.com>;p1=red Example 3 Rule: Add a display name to the P-Called-Party-Id header: MessageManipulations 3 = 1, any, , header.p-called- party-id.name, 2, 'Secretary', 0;...
Page 516 Mediant 800 MSBG Below are header manipulation examples: Example 1 Rule: Add a P-Preferred-Identity header to all messages: MessageManipulations 1 = 1, any, , header.P-Preferred- Identity, 0, "'Cullen Jennings <sip:fluffy@abc.com>'", Result: P-Preferred-Identity: "Cullen Jennings" <sip:fluffy@abc.com> Modify the display name in the P-Preferred-Identity header:...
Page 517 SIP User's Manual 8. IP Telephony Capabilities Below are header manipulation examples: Example 1 Rule: Add a Proxy-Require header to the message: MessageManipulations 1 = 1, any, , header.Proxy- Require, 0, "'sec-agree'", 0; Result: Proxy-Require: sec-agree Example 2 Rule: Modify the Proxy-Require header to itsp.com: MessageManipulations 2 = 1, any, , header.Proxy- Require, 2, "itsp.com'", 0;...
Page 518 Mediant 800 MSBG 8.4.9.2.20 Referred-By An example of the header is shown below: Referred-By: <sip:referrer@referrer.example>; The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes param param Read/Write...
Page 519 SIP User's Manual 8. IP Telephony Capabilities Below are header manipulation examples: Example 1 Rule: Add a basic header: MessageManipulations 0 = 1, any, ,header.Refer-to, 0, "'<sip:referto@referto.com>'", 0; Result: Refer-To: <sip:referto@referto.com> Example 2 Rule: Add a Refer-To header with URI headers: MessageManipulations 0 = 1, any, ,header.Refer-to, 0, "'<sips:a8342043f@atlanta.example.com?Replaces=1234560 1%40atlanta.example.com%3bfrom-tag%3d314159%3bto-...
Page 520 Mediant 800 MSBG Below are header manipulation examples: Example 1 Rule: Add a Remote-Party-Id header to the message: MessageManipulations 0 = 1, invite, ,header.REMOTE- PARTY-ID, 0, "'<sip:999@10.132.10.108>;party=calling'", 0; Result: Remote-Party-ID: <sip:999@10.132.10.108>;party=calling;npi=0;ton=0 Example 2 Rule: Create a Remote-Party-Id header using the url in the From header using the + operator to concatenate strings: MessageManipulations 0 = 1, Invite, ,header.REMOTE-...
Page 521 SIP User's Manual 8. IP Telephony Capabilities Below are header manipulation examples: Example 1 Rule: Test the Request-URI transport type. If 1 (TCP), then modify the URL portion of the From header: MessageManipulations 1 = 1, Invite.request, "header.REQUEST-URI.url.user == '101'", header.REMOTE-PARTY-ID.url, 2, 'sip:3200@110.18.5.41;tusunami=0', 0;...
Page 522 Mediant 800 MSBG 8.4.9.2.25 Resource-Priority An example of the header is shown below: Resource-Priority: wps.3 The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Namespace String Read/Write...
Page 523 SIP User's Manual 8. IP Telephony Capabilities Keyword Sub Types Attributes Below are header manipulation examples: Example 1 Rule: Remove the User-Agent header: MessageManipulations 2 = 1, Invite, ,header.user- agent, 1, "''", 0; Result: The header is removed. Example 2 Rule: Change the user agent name in the header: MessageManipulations 3 = 1, Invite,...
Page 524 Mediant 800 MSBG 8.4.9.2.29 Session-Expires An example of the header is shown below: Session-Expires: 480 The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Param Param Read/Write...
Page 525 SIP User's Manual 8. IP Telephony Capabilities Below is a header manipulation example: Rule: Add a Subject header: MessageManipulations 0 = 1, any, , header.Subject, 0, "'A tornado is heading our way!'", 0; Result: Subject: A tornado is heading our way! 8.4.9.2.31 Supported An example of the header is shown below: Supported: early-session...
Page 526 Mediant 800 MSBG Below are header manipulation examples: Example 1 Rule: Set the user phone Boolean to be false in the To header's URL: MessageManipulations 4 = 1, invite.request, , header.to.url.UserPhone, 2, '0', 0; Result: To: <sip:101@10.132.10.128> Example 2 Rule: Change the URL in the To header: MessageManipulations 4 = 1, invite.request, ,...
Page 527 SIP User's Manual 8. IP Telephony Capabilities 8.4.9.2.34 Via An example of the header is shown below: Via: SIP/2.0/UDP 10.132.10.128;branch=z9hG4bKUGOKMQPAVFKTAVYDQPTB The header properties as shown in the table below: Header Level Action Delete Modify List Entries Operations Supported Keyword Sub Types Attributes Alias Boolean...
Page 528 Mediant 800 MSBG Below is a header manipulation example: Rule: Add a Warning header to the message: MessageManipulations 0 = 1, Invite.response.180, ,header.warning, 0, "'Incompatible 380'", 0; Result: Warning: Incompatible 380 8.4.9.2.36 Unknown Header An Unknown header is a SIP header that is not included in this list of supported headers.
Page 529: Table 8-8: Event Structure
SIP User's Manual 8. IP Telephony Capabilities 8.4.9.3 Structure Definitions 8.4.9.3.1 Event Structure The Event structure is used in the Event header (see ''Event'' on page 510). Table 8-8: Event Structure Keyword Sub Types Attributes EventPackage Enum Event Package (see Read/Write ''Event Package'' on page 531)
Page 530: Table 8-11: Reason Structure
Mediant 800 MSBG 8.4.9.3.4 Reason Structure This structure is applicable to the Reason header (see ''Reason'' on page 517). Table 8-11: Reason Structure Keyword Sub Types Reason Enum Reason (see ''Reason (Reason Structure)'' on page 533) Cause Text String 8.4.9.3.5 URL...
Page 531: Enum Definitions
SIP User's Manual 8. IP Telephony Capabilities 8.4.9.4 Enum Definitions 8.4.9.4.1 AgentRole These ENUMs are applicable to the Server or User-Agent headers (see ''Server or User- Agent'' on page 522). Table 8-13: Enum Agent Role AgentRole Value Client Server 8.4.9.4.2 Event Package These ENUMs are applicable to the Server or User-Agent (see ''Server or User-Agent'' on page 522) and Event (see ''Event'' on page 510) headers.
Page 532: Table 8-15: Enum Mlpp Reason Type
Mediant 800 MSBG 8.4.9.4.3 MLPP Reason Type These ENUMs are applicable to the MLPP Structure (see ''MLPP'' on page 529). Table 8-15: Enum MLPP Reason Type Type Value PreEmption Reason MLPP Reason 8.4.9.4.4 Number Plan These ENUMs are applicable to the Remote-Party-Id header (see Remote-Party-Id).
Page 533: Table 8-18: Enum Privacy
SIP User's Manual 8. IP Telephony Capabilities 8.4.9.4.6 Privacy These ENUMs are applicable to the Remote-Party-Id (see Remote-Party-Id) and Diversion (see Diversion) headers. Table 8-18: Enum Privacy Privacy Role Value Full 8.4.9.4.7 Reason (Diversion) These ENUMs are applicable to the Diversion header (see Diversion). Table 8-19: Enum Reason Reason Value...
Page 534 Mediant 800 MSBG Reason Value NOTIFY REFER SUBSCRIBE PRACK UPDATE PUBLISH LAST_REQUEST TRYING_100 RINGING_180 CALL_FORWARD_181 QUEUED_182 SESSION_PROGRESS_183 OK_200 ACCEPTED_202 MULTIPLE_CHOICE_300 MOVED_PERMANENTLY_301 MOVED_TEMPORARILY_302 SEE_OTHER_303 USE_PROXY_305 ALTERNATIVE_SERVICE_380 BAD_REQUEST_400 UNAUTHORIZED_401 PAYMENT_REQUIRED_402 FORBIDDEN_403 NOT_FOUND_404 METHOD_NOT_ALLOWED_405 NOT_ACCEPTABLE_406 AUTHENTICATION_REQUIRED_407 REQUEST_TIMEOUT_408 CONFLICT_409 GONE_410 LENGTH_REQUIRED_411 CONDITIONAL_REQUEST_FAILED_412 REQUEST_TOO_LARGE_413 REQUEST_URI_TOO_LONG_414...
Page 535 SIP User's Manual 8. IP Telephony Capabilities Reason Value UNSUPPORTED_URI_SCHEME_416 UNKNOWN_RESOURCE_PRIORITY_417 BAD_EXTENSION_420 EXTENSION_REQUIRED_421 SESSION_INTERVAL_TOO_SMALL_422 SESSION_INTERVAL_TOO_SMALL_423 ANONYMITY_DISALLOWED_433 UNAVAILABLE_480 TRANSACTION_NOT_EXIST_481 LOOP_DETECTED_482 TOO_MANY_HOPS_483 ADDRESS_INCOMPLETE_484 AMBIGUOUS_485 BUSY_486 REQUEST_TERMINATED_487 NOT_ACCEPTABLE_HERE_488 BAD_EVENT_489 REQUEST_PENDING_491 UNDECIPHERABLE_493 SECURITY_AGREEMENT_NEEDED_494 SERVER_INTERNAL_ERROR_500 NOT_IMPLEMENTED_501 BAD_GATEWAY_502 SERVICE_UNAVAILABLE_503 SERVER_TIME_OUT_504 VERSION_NOT_SUPPORTED_505 MESSAGE_TOO_LARGE_513 PRECONDITION_FAILURE_580 BUSY_EVERYWHERE_600 DECLINE_603 DOES_NOT_EXIST_ANYWHERE_604 NOT_ACCEPTABLE_606 8.4.9.4.9 Reason (Remote-Party-Id) These ENUMs are applicable to the Remote-Party-Id header (see Remote-Party-Id).
Page 536: Table 8-21: Enum Reason (Rpi)
Mediant 800 MSBG Table 8-21: Enum Reason (RPI) Reason Value Busy Immediate No Answer 8.4.9.4.10 Refresher These ENUMs are used in the Session-Expires header (see Session-Expires). Table 8-22: Enum Refresher Refresher String Value 8.4.9.4.11 Screen These ENUMs are applicable to the Remote-Party-Id (see Remote-Party-Id) and Diversion (see Diversion) headers.
Page 537: Actions And Types
SIP User's Manual 8. IP Telephony Capabilities 8.4.9.4.13 TransportType These ENUMs are applicable to the URL Structure (see ''URL'' on page 530) and the Via header (see ''Via'' on page 527). Table 8-25: Enum TransportType TransportType Value SCTP 8.4.9.4.14 Type These ENUMs are applicable to the URL Structure (see ''URL'' on page 530).
Page 538 Mediant 800 MSBG Element Command Command Value Type Remarks Type Type equals to the value. "!=" String Returns true if the body's content not equals to the value. "contains" String Returns true if the string given is found in the body's content.
Page 539 SIP User's Manual 8. IP Telephony Capabilities Element Command Command Value Type Remarks Type Type "Remove" Removes the header from the message, if the header is part of a list only that header is removed. "Add" String Adds a new header to the end of the list.
Page 540 Mediant 800 MSBG Element Command Command Value Type Remarks Type Type "!=" String Returns true if the header's structure's value not equals to the *Structure value. The string given must be able to be parsed to the structure. Action Modify...
Page 541: Syntax
SIP User's Manual 8. IP Telephony Capabilities Element Command Command Value Type Remarks Type Type "!=" Boolean Returns true if the Boolean element not equals to the value. Boolean – can be either "0" or "1". Action "Modify" Boolean Sets the Boolean element to the value.
Page 542 Mediant 800 MSBG Syntax: ( token / "any" ) Examples: ♦ Invite, subscribe – rule applies only to INVITE messages ♦ Unknown – unknown methods are also allowed ♦ Any – no limitation on the method type message-role Description: rule is applied only if this is the message's role Syntax: ( "request"...
Page 543 SIP User's Manual 8. IP Telephony Capabilities message-element: Description: element in the message Syntax: ( "header" / "body" ) "." message-element-name [ "." header-index ] * [ "." ( sub-element / sub-element-param ) ] Examples: • Header.from • Header.via.2.host • Header.contact.header-param.expires •...
Page 544 Mediant 800 MSBG Examples: ♦ expires (contact's header's param) ♦ duration (retry-after header's param) ♦ unknown-param (any unknown param can be added/removed from the header) param Description: Params can be as values for match and action Syntax: "param" "." Param-sub-element "." Param-dir-element "." (Call-Param-...
Page 545 SIP User's Manual 8. IP Telephony Capabilities string Description: string enclosed in double quotes Syntax: quoted-string Examples: ♦ "username" ♦ "123" ♦ "user@host" integer Description: a number Syntax: 1 * DIGIT Example: ♦ action-type: Description: action to be performed on the element Syntax: ( "modify"...
Page 546: Sbc Configuration Example
Mediant 800 MSBG 8.4.10 SBC Configuration Example This section provides basic SBC configuration examples. Note: The examples described in this section are for reference only. Modifications to device configuration should be made to suite your networking environment. 8.4.10.1 General SBC Setup This section provides a basic SBC configuration example scenario.
Page 547: Figure 8-58: Sbc Example Scenario
SIP User's Manual 8. IP Telephony Capabilities The figure below illustrates the example scenario setup: Figure 8-58: SBC Example Scenario 8.4.10.1.1 Step 1: Configure LAN VoIP IP Address The procedure below describes how to configure the VoIP LAN IP addresses (in our example, 10.8.6.86).
Page 548: Figure 8-60: Selecting Wan Interface For Voip Traffic
Mediant 800 MSBG 8.4.10.1.2 Step 2: Assign VoIP Traffic to WAN Interface Once you have defined the WAN IP address (see ''Assigning a WAN IP Address'' on page 29) for the data-routing interface, you then need to associate it with VoIP traffic (i.e., SIP signaling and media / RTP interfaces).
Page 549: Figure 8-61: Applications Enabling
SIP User's Manual 8. IP Telephony Capabilities To enable SBC: Open the 'Applications Enabling' page (Configuration tab > VoIP menu > Applications Enabling submenu > Applications Enabling), and then from the 'Enable SBC Application' drop-down list, select 'Enable': Figure 8-61: Applications Enabling Page Click Submit.
Page 550: Figure 8-63: Lan And Wan Media Realms In Sip Media Realm Table
Mediant 800 MSBG 8.4.10.1.5 Step 5: Configure Multiple SIP and RTP Interfaces The procedure below describes how to configure multiple SIP signaling interfaces and RTP interfaces. The SIP signaling interfaces are defined as SIP Interfaces; the RTP interfaces are defined as Media Realms. These are associated together under one entity termed SRD (Signaling Routing Domain).
Page 551: Figure 8-64: Srds For Lan And Wan In Srd Table
SIP User's Manual 8. IP Telephony Capabilities In the 'Media Realm' field, enter "LanMediaR". Note: This string must be identical (and case-sensitive) as that defined in the 'SIP Media Realm' table (see Step 1.b). Click Apply. Add SRD #1 for WAN: In the 'Name' field, enter "WanSRD".
Page 552: Figure 8-66: Proxy Sets Table
Mediant 800 MSBG 8.4.10.1.6 Step 6: Define Proxy Set for WAN IP-PBX The procedure below describes how to configure a Proxy Set for the WAN hosted IP-PBX. This represents the IP address of the IP-PBX, which in the case study is 100.33.2.26.
Page 553: Figure 8-67: Ip Group 1 (For Enterprise Users) In Ip Group Table
SIP User's Manual 8. IP Telephony Capabilities Enterprise LAN (users) WAN (IP-PBX) To configure IP Groups: Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). Add IP Group #1 for enterprise LAN (users): From the 'Type' drop-down list, select 'USER'.
Page 554: Figure 8-68: Ip Group 2 (For Wan Itsp) In Ip Group Table
Mediant 800 MSBG In the 'SRD' field, enter "1" to associate it with the WAN SRD (defined in ''Step 5: Configure Multiple SIP and RTP Interfaces'' on page 550). From the 'Classify By Proxy Set' drop-down list, select 'Enable' to allow the device to classify incoming calls as this IP Group based on the Proxy Set.
Page 555: Figure 8-69: Ip Group Classification Rule For Lan Users
SIP User's Manual 8. IP Telephony Capabilities In the 'Source IP Group' field, enter '1'. This classifies calls from LAN users as belonging to IP Group 1 (defined in ''Step 7: Define IP Groups'' on page 552). Figure 8-69: IP Group Classification Rule for LAN Users Click Apply.
Page 556: Survivability And Alternative Routing
Mediant 800 MSBG Click Apply. Figure 8-70: IP-to-IP Routing Rules 8.4.10.2 Survivability and Alternative Routing This section provides an example for configuring SBC Survivability for LAN users. This example is based on the scenario described in ''General SBC Setup'' on page 546.
Page 557: Figure 8-71: Survivability Example Setup
SIP User's Manual 8. IP Telephony Capabilities Define an alternative IP-to-IP routing rule for IP Phones IP Group #1 (USER) to IP Phones IP Group #1 (USER). This is the alternative route if the IP-PBX does not respond, whereby the user is searched in the device's users registration database. Figure 8-71: Survivability Example Setup 8.4.10.2.1 Step 1: Enable Proxy Keep-Alive The procedure below describes how to configure the Proxy Keep-Alive mechanism for the...
Page 558: Figure 8-72: Enabling Proxy Keep-Alive
Mediant 800 MSBG From the 'Enable Proxy Keep Alive' drop-down list, select 'Using Options' to enable the Proxy Keep-alive mechanism. Figure 8-72: Enabling Proxy Keep-Alive Click Submit. Save the settings to flash memory ("burn") and reset the device (see ''Saving Configuration'' on page 336).
Page 559: Figure 8-73: Configuring Ip-To-Ip Routing Rules
SIP User's Manual 8. IP Telephony Capabilities From the 'Alternative Route Options' drop-down list, select 'Alt Route Consider Inputs'. Figure 8-73: Configuring IP-to-IP Routing Rules Click Apply. Version 6.2 February 2011...
Page 560: Sbc-To-Pstn Routing
Mediant 800 MSBG 8.4.10.3 SBC-to-PSTN Routing This section describes how to setup the device for SBC-to-PSTN routing. This example is based on the general scenario described in ''General SBC Setup'' on page 546, but in addition assumes the following: The device is connected to the PSTN network by an E1/T1 trunk...
Page 561: Figure 8-75: Configuring Sip Interface For Pstn (Gw)
SIP User's Manual 8. IP Telephony Capabilities Define the UDP, TCP, and TLS ports as 5070, 5070, and 5071 respectively. In the 'SRD' field, enter '0'. This associates the SIP interface with the LAN SRD you defined ''Step 5: Configure Multiple SIP and RTP Interfaces'' on page 550. Figure 8-75: Configuring SIP Interface for PSTN (GW) Click Apply.
Page 562: Figure 8-76: Defining Device As Proxy Set
Mediant 800 MSBG From the 'SRD Index' drop-down list, select '0'. This associates the Proxy Set with the LAN SRD, configured in ''Step 5: Configure Multiple SIP and RTP Interfaces'' on page 550. It allows the device to classify calls by Proxy Set for this SRD.
Page 563: Figure 8-77: Defining Ip Group For Pstn Users
SIP User's Manual 8. IP Telephony Capabilities From the 'Classify By Proxy Set' drop-down list, select 'Disable'. This ensures that the existing Classification rule for 10.8.6.* (defined in ''Step 8: Define Classification Rules for LAN Users'' on page 554) also applies to PSTN users as belonging to IP Group 1 and FXS users can be registered to the device's database.
Page 564 Mediant 800 MSBG 8.4.10.3.4 Step 4: Define IP-to-IP Routing Rules The procedure below describes how to configure IP-to-IP routing rules. The following rules need to be included in the configuration: Existing rules: • Routing from LAN users to IP-PBX (i.e., from IP Group 1 to IP Group 2), as previously defined in ''Step 9: Define IP-to-IP Routing Rules'' on page •...
Page 565: Figure 8-78: Dfining Ip-To-Ip Routing Rules
SIP User's Manual 8. IP Telephony Capabilities Click Apply. Figure 8-78: Dfining IP-to-IP Routing Rules The configured rules are summarized below: Index #1: First choice route for IP Group 1 (i.e., LAN and PSTN users) when calling each other or any user on WAN. The call is sent to IP Group 2 (i.e., WAN IP-PBX) through the device's SBC interface.
Page 566: Figure 8-79: Defining Trunk Groups
Mediant 800 MSBG 8.4.10.3.5 Step 5: Define Trunk Groups for PSTN Users The procedure below describes how to configure and enable the PSTN users. This is done by defining Trunk Groups. You need to configure Trunk Groups for the following PSTN...
Page 567: Figure 8-80: Defining Channel Select Mode
SIP User's Manual 8. IP Telephony Capabilities From the 'Channel Select Mode' drop-down list, select 'By Dest Phone Number'. This setting sends the call to a specific FXS user according to the called (destination) number. From the 'Registration Mode' drop-down list, select 'Per Endpoint'. This allows the FXS users to register to the device's internal database, using IP Group 3 (defined in ''Step 3: Define IP Group for PSTN'' on page 562).
Page 568: Basic Coder Transcoding
Mediant 800 MSBG In the 'Trunk Group ID' field, enter '1'. Click Submit. Add a rule for routing all other calls to FXS users: In the 'Dest Phone Prefix', 'Source Phone Prefix' and Source IP Address' fields, enter an asterisk symbol (*) to indicate any.
Page 569: Figure 8-82: Configuring The Coder Group
SIP User's Manual 8. IP Telephony Capabilities Add an entry for G.711 and another entry for G.729. Figure 8-82: Configuring the Coder Group 8.4.10.4.2 Step 2: Define an IP Profile The procedure below describes how to configure an IP Profile. You need to assign to this IP Profile the Coder Group #1 that you defined in ''Step 1: Define a Coder Group'' on page 568.
Page 570: Advanced Coder Transcoding
Mediant 800 MSBG 8.4.10.4.3 Step 3: Assign IP Profile to LAN Users IP Group The procedure below describes how to assign the IP Profile defined in ''Step 2: Define an IP Profile'' on page to the LAN users IP Group #3. You need to assign to this IP Profile the Coder Group #1 that you defined in ''Step 1: Define a Coder Group'' on page 568.
Page 571: Figure 8-85: Advanced Transcoding Example Scenario
SIP User's Manual 8. IP Telephony Capabilities This example assumes the following: Device deployed at an enterprise, interfacing between enterprise's IP-PBX and an Internet Telephony Service Provider (ITSP) Enterprise deployed with LAN IP phones for VoIP calls LAN IP phones use G.711 coder ITSP uses only G.729 coder Figure 8-85: Advanced Transcoding Example Scenario 8.4.10.5.1 Step 1: Define Coder Groups...
Page 572: Figure 8-87: Defining Coder Group For Itsp
Mediant 800 MSBG Add an entry for G.729. Figure 8-87: Defining Coder Group for ITSP Click Submit. 8.4.10.5.2 Step 2: Define Allowed Coders The procedure below describes how to configure an Allowed Coders Group for ITSP. This ensures that the device uses only the coder G.729 on its outbound leg to the ITSP. This Allowed Coder Group is later assigned to the IP Profile of the ITSP (in ''Step 2: Define an IP Profile'' on page 569).
Page 573: Figure 8-89: Defining Ip Profile For Lan Users
SIP User's Manual 8. IP Telephony Capabilities To configure IP Profiles: Open the 'IP Profile Settings' page (Configuration tab > VoIP menu > Coders And Profiles submenu > IP Profile Settings). Add IP Profile #1 for LAN users: From the 'Profile ID' drop-down list, select '1'. In the 'Profile Name', enter a brief description (e.g., "LAN Users").
Page 574: Figure 8-90: Defining Ip Profile For Itsp
Mediant 800 MSBG Click Submit. Figure 8-90: Defining IP Profile for ITSP 8.4.10.5.4 Step 4: Assign IP Profiles to IP Groups The procedure below describes how to assign the IP Profiles defined in ''Step 3: Define IP Profiles'' on page to the IP Groups for the LAN users and ITSP.
Page 575: Figure 8-91: Assigning Ip Profile To Lan Users Ip Group
SIP User's Manual 8. IP Telephony Capabilities Click Submit. Figure 8-91: Assigning IP Profile to LAN Users IP Group Assign IP Profile #2 to ITSP IP Group #2: From the 'Index' drop-down list, select '2'. From the 'IP Profile ID' drop-down list, select '2'. This is the IP Profile that you defined for the ITSP.
Page 576: Rtp-Srtp Transcoding
Mediant 800 MSBG Click Submit. Figure 8-92: Assigning IP Profile to ITSP IP Group 8.4.10.6 RTP-SRTP Transcoding This section describes how to configure an RTP-SRTP transcoding scenario. This example is based on the previous examples and only describes the configuration of the transcoding feature itself.
Page 577: Figure 8-93: Rtp-Srtp Transcoding Mode For Lan Users
SIP User's Manual 8. IP Telephony Capabilities To configure RTP-SRTP transcoding: Open the 'IP Profile Settings' page (Configuration tab > VoIP menu > Coders And Profiles submenu > IP Profile Settings). Configure RTP-SRTP transcoding mode for IP Profile #1 (i.e., LAN users): From the 'Profile ID' drop-down list, select '1'.
Page 578: Sip Uri Manipulation
Mediant 800 MSBG 8.4.10.7 SIP URI Manipulation This section describes how to configure SBC SIP URI user and host parts manipulation. This example is based on the general scenario described in ''General SBC Setup'' on page 546. This example describes how to configure manipulation of the following: SIP URI host part: For the SIP INVITE sent from any source destination IP Group (i.e.,...
Page 579: Sip Header Manipulation
SIP User's Manual 8. IP Telephony Capabilities 8.4.10.7.2 Step 2: Manipulate SIP URI User Part The procedure below describes how to configure a manipulation rule for the SIP URI user part. In this manipulation, the destination URI user name prefix "976" in SIP INVITE messages sent from the LAN users IP Group #1 is removed (i.e., "976") in the outgoing INVITE to the WAN IP-PBX.
Page 580 Mediant 800 MSBG From: <sip:1000@10.8.5.41>;tag=1c1286571572 To: <sip:FEU8-999-1@WANWAN> Call-ID: 128652844814102010161846@212.25.26.70 CSeq: 1 INVITE Contact: <sip:FEU3-998-2@212.25.26.70:5060> Supported: em,100rel,timer,replaces,path,resource-priority,sdp- anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUB SCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 800/v.6.20A.004 P-Asserted-Identity: <sip:1000@MSBG.com> To configure this SIP message manipulation, you need to do the following: Add a SIP message manipulation rule in the Message Manipulation page (see ''Step 1: Add SIP Message Manipulation Rule'' on page 580).
Page 581: Figure 8-97: Sip Header Manipulation Example
SIP User's Manual 8. IP Telephony Capabilities The manipulation rule is shown in the figure below: Figure 8-97: SIP Header Manipulation Example Click Apply. 8.4.10.8.2 Step 2: Assign Message Manipulation Rule to IP Group The procedure below describes how to assign the configured SIP message manipulation rule (rule #1) to the IP Group belonging to the LAN users.
Page 582: Stand-Alone Survivability (Sas) Application
Mediant 800 MSBG Stand-Alone Survivability (SAS) Application 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...
Page 583: Sas Outbound Mode
SIP User's Manual 8. IP Telephony Capabilities 8.5.1.1 SAS Outbound Mode This section describes the SAS outbound mode, which includes the following states: Normal state (see ''Normal State'' on page 583) Emergency state (see ''Emergency State'' on page 584) 8.5.1.1.1 Normal State In normal state, SAS receives REGISTER requests from the enterprise's UAs and forwards them to the external proxy (i.e., outbound proxy).
Page 584: Sas Redundant Mode
Mediant 800 MSBG 8.5.1.1.2 Emergency State When a connection with the external proxy fails (detected by the device's keep-alive messages), the device enters SAS emergency state. The device serves as a proxy for the UAs, by handling internal call routing of the UAs (within the LAN enterprise).
Page 585: Figure 8-101: Sas Redundant Mode In Normal State (Example)
SIP User's Manual 8. IP Telephony Capabilities Note: In this SAS deployment, the UAs (e.g., IP phones) must support configuration for primary and secondary proxy servers (i.e., proxy redundancy), as well as homing. Homing allows the UAs to switch back to the primary server from the secondary proxy once the connection to the primary server returns (UAs check this using keep-alive messages to the primary server).
Page 586: Sas Routing
Mediant 800 MSBG 8.5.1.2.2 Emergency State If the UAs detect that their primary (external) proxy does not respond, they immediately register to SAS and start routing calls to it. Figure 8-102: SAS Redundant Mode in Emergency State (Example) 8.5.1.2.3 Exiting Emergency and Returning to Normal State Once the connection with the primary proxy is re-established, the following occurs: UAs: switch back to operate with the primary proxy.
Page 587: Sas Routing In Normal State
SIP User's Manual 8. IP Telephony Capabilities 8.5.2.1 SAS Routing in Normal State The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the UAs: Figure 8-103: Flowchart of INVITE from UA's in SAS Normal State Version 6.2 February 2011...
Page 588: Figure 8-104: Flowchart Of Invite From Primary Proxy In Sas Normal State
Mediant 800 MSBG The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the external proxy: Figure 8-104: Flowchart of INVITE from Primary Proxy in SAS Normal State SIP User's Manual Document #: LTRT-12804...
Page 589: Sas Routing In Emergency State
SIP User's Manual 8. IP Telephony Capabilities 8.5.2.2 SAS Routing in Emergency State The flowchart below shows the routing logic for SAS in emergency state: Figure 8-105: Flowchart for SAS Emergency State Version 6.2 February 2011...
Page 590: Sas Configuration
Mediant 800 MSBG 8.5.3 SAS Configuration SAS supports various configuration possibilities, depending on how the device is deployed in the network and the network architecture requirements. This section provides step-by- step procedures on configuring the SAS application, using the device's Web interface.
Page 591 SIP User's Manual 8. IP Telephony Capabilities 8.5.3.1.2 Configuring Common SAS Parameters The procedure below describes how to configure SAS settings that are common to all SAS modes. This includes various SAS parameters as well as configuring the Proxy Set for the SAS proxy (if required).
Page 592: Figure 8-107: Configuring Common Settings
Mediant 800 MSBG Figure 8-107: Configuring Common Settings In the 'SAS Proxy Set' field, enter the Proxy Set used for SAS. The SAS Proxy Set must be defined only for the following SAS modes: • Outbound mode: In SAS normal state, SAS forwards REGISTER and INVITE messages received from the UAs to the proxy servers defined in this Proxy Set.
Page 593: Configuring Sas Outbound Mode
SIP User's Manual 8. IP Telephony Capabilities From the 'Enable Proxy Keep Alive' drop-down list, select ‘Using Options’. This instructs the device to send SIP OPTIONS messages to the proxy for the keep- alive mechanism. Figure 8-108: Defining UAs' Proxy Server Click Submit to apply your settings.
Page 594: Configuring Sas Redundant Mode
Mediant 800 MSBG 8.5.3.3 Configuring SAS Redundant Mode This section describes how to configure the SAS redundant mode. These settings are in addition to the ones described in ''Configuring Common SAS Parameters'' on page 591. Note: The VoIP CPEs (such as IP phones or residential gateways) need to be...
Page 595: Figure 8-109: Enabling Proxy Server For Gateway Application
SIP User's Manual 8. IP Telephony Capabilities 8.5.3.4.1 Gateway with SAS Outbound Mode The procedure below describes how to configure the Gateway application with SAS outbound mode. To configure Gateway application with SAS outbound mode: Define the proxy server address for the Gateway application: Open the 'Proxy &...
Page 596: Figure 8-111: Disabling User=Phone In Sip Url
Mediant 800 MSBG Disable use of user=phone in SIP URL: Open the 'SIP General Parameters' page (Configuration tab > VoIP menu > SIP Definitions submenu > General Parameters). From the 'Use user=phone in SIP URL' drop-down list, select 'No'. This instructs the Gateway application to not use user=phone in the SIP URL and therefore, REGISTER and INVITE messages use SIP URI.
Page 597: Figure 8-113: Defining Proxy Servers For Gateway Application
SIP User's Manual 8. IP Telephony Capabilities From the 'Proxy Set ID' drop-down list, select '0'. In the first 'Proxy Address' field, enter the IP address of the external proxy server. In the second 'Proxy Address' field, enter the IP address and port of the device (in the format x.x.x.x:port).
Page 598: Advanced Sas Configuration
Mediant 800 MSBG 8.5.3.5 Advanced SAS Configuration This section describes the configuration of advanced SAS features that can be optionally implemented in your SAS deployment: Manipulating incoming SAS Request-URI user part of REGISTER message (see ''Manipulating URI user part of Incoming REGISTER'' on page 598)
Page 599: Figure 8-115: Manipulating User Part In Incoming Register
SIP User's Manual 8. IP Telephony Capabilities After manipulation, SAS registers the user in its database as follows: AOR: 976653434@10.33.4.226 Associated AOR: 3434@10.33.4.226 (after manipulation, in which only the four digits from the right of the URI user part are retained) Contact: 976653434@10.10.10.10 The procedure below describes how to configure the manipulation example scenario above (relevant ini parameter is SASRegistrationManipulation):...
Page 600: Figure 8-116: Manipulating Invite Destination Number
Mediant 800 MSBG 8.5.3.5.2 Manipulating Destination Number of Incoming INVITE You can define a manipulation rule to manipulate the destination number in the Request- URI of incoming INVITE messages when SAS is in emergency state. This is required, for example, if the call is destined to a registered user but the destination number in the received INVITE is not the number assigned to the registered user in the SAS registration database.
Page 601 SIP User's Manual 8. IP Telephony Capabilities Notes: • The 'Source IP Group' field must not be configured; leave it at '-1'. • The 'Is Additional Manipulation' field must be set to '0'. • The 'Manipulation Purpose' field must be set to 'Normal'. •...
Page 602: Figure 8-117: Blocking Unregistered Sas Users
Mediant 800 MSBG 8.5.3.5.4 Blocking Calls from Unregistered SAS Users To prevent malicious calls (for example, Service Theft), it is recommended to configure the feature for blocking SIP INVITE messages received from SAS users that are not registered in the SAS database. This applies to SAS in normal and emergency states.
Page 603: Figure 8-118: Configuring Sas Emergency Numbers
SIP User's Manual 8. IP Telephony Capabilities To configure SAS emergency numbers: Open the 'SAS Configuration' page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability). In the ‘SAS Default Gateway IP' field, define the IP address and port (in the format x.x.x.x:port) of the device (Gateway application).
Page 604: Viewing Registered Sas Users
Mediant 800 MSBG Note: This feature is applicable only to SAS outbound mode. When this feature is enabled, the SIP Record-Route header includes the URI "lr" parameter. The presence of this parameter indicates loose routing; the lack of it indicates strict routing.
Page 605: Routing Based On Ldap Active Directory Queries
SIP User's Manual 8. IP Telephony Capabilities Routing Based on LDAP Active Directory Queries The device supports Lightweight Directory Access Protocol (LDAP), allowing the device to make call routing decisions based on information stored on a third-party LDAP server (or Microsoft’s Active Directory-based enterprise directory server).
Page 606: Ad-Based Tel-To-Ip Routing In Microsoft Ocs 2007 Environment
Mediant 800 MSBG 8.6.2 AD-Based Tel-to-IP Routing in Microsoft OCS 2007 Environment Typically, enterprises wishing to deploy Microsoft’s Office Communication Server 2007 (OCS 2007) are faced with a complex, call routing dial plan when migrating users from their existing PBX/IP-PBX to the OCS 2007 platform. As more and more end-users migrate to the new voice system, dialing plan management and PBX link capacity can be adversely impacted.
Page 607: Figure 8-119: Active Directory-Based Routing Rules In Outbound Ip Routing Table
SIP User's Manual 8. IP Telephony Capabilities This feature uses the following parameters to configure the attribute names in the AD used in the LDAP query: AD attribute for Mediation Server: MSLDAPOCSNumAttributeName (the default is "msRTCSIPPrimaryUserAddress") AD attribute for PBX/IP-PBX: MSLDAPPBXNumAttributeName (the default is "telephoneNumber") AD attribute for mobile number: MSLDAPMobileNumAttributeName (the default is "mobile")
Page 608: General
Mediant 800 MSBG General 8.7.1 Transcoding using Third-Party Call Control The device supports transcoding using a third-party call control Application server. This support is provided by the following: Using RFC 4117 (see ''Using RFC 4117'' on page 608) Note: Transcoding can also be implemented using the IP-to-IP (IP2IP) application and SBC application.
Page 609: Supported Radius Attributes
SIP User's Manual 8. IP Telephony Capabilities 8.7.2 Supported RADIUS Attributes The following table provides explanations on the RADIUS attributes included in the communication packets transmitted between the device and a RADIUS Server. Table 8-27: Supported RADIUS Attributes Attribute Attribute Value Purpose Example...
Page 610 Mediant 800 MSBG Attribute Attribute Value Purpose Example Number Name Format The call's terminator: Call- PSTN-terminated call String Yes, No Stop Acc Terminator (Yes); IP-terminated call (No). String 8004567145 Start Acc Destination phone number String 2427456425 Stop Acc Calling Party Number...
Page 611: Call Detail Record
SIP User's Manual 8. IP Telephony Capabilities acct-status-type = 2 acct-input-octets = 4841 acct-output-octets = 8800 acct-session-time = 1 acct-input-packets = 122 acct-output-packets = 220 called-station-id = 201 calling-station-id = 202 // Accounting non-standard parameters: (4923 33) h323-gw-id = (4923 23) h323-remote-address = 212.179.22.214 (4923 1) h323-ivr-out = h323-incoming-conf-id:02102944 600a1899 3fd61009 0e2f3cc5 (4923 30) h323-disconnect-cause = 22 (0x16)
Page 612: Release Reasons In Cdr
Mediant 800 MSBG Field Name Description Destination Phone Number Type Destination Phone Number Plan DstPhoneNum Destination Phone Number DstNumBeforeMap Destination Number Before Manipulation Durat Call Duration Coder Selected Coder Intrv Packet Interval RTP IP Address RtpIp Port Remote RTP Port...
Page 613 SIP User's Manual 8. IP Telephony Capabilities "RELEASE_BECAUSE_DESTINATION_BUSY" "RELEASE_BECAUSE_NOANSWER" "RELEASE_BECAUSE_UNKNOWN_REASON" "RELEASE_BECAUSE_REMOTE_CANCEL_CALL" "RELEASE_BECAUSE_UNMATCHED_CAPABILITIES" "RELEASE_BECAUSE_UNMATCHED_CREDENTIALS" "RELEASE_BECAUSE_UNABLE_TO_HANDLE_REMOTE_REQUEST" "RELEASE_BECAUSE_NO_CONFERENCE_RESOURCES_LEFT" "RELEASE_BECAUSE_CONFERENCE_FULL" "RELEASE_BECAUSE_VOICE_PROMPT_PLAY_ENDED" "RELEASE_BECAUSE_VOICE_PROMPT_NOT_FOUND" "RELEASE_BECAUSE_TRUNK_DISCONNECTED" "RELEASE_BECAUSE_RSRC_PROBLEM" "RELEASE_BECAUSE_MANUAL_DISC" "RELEASE_BECAUSE_SILENCE_DISC" "RELEASE_BECAUSE_RTP_CONN_BROKEN" "RELEASE_BECAUSE_DISCONNECT_CODE" "RELEASE_BECAUSE_GW_LOCKED" "RELEASE_BECAUSE_NORTEL_XFER_SUCCESS" "RELEASE_BECAUSE_FAIL" "RELEASE_BECAUSE_FORWARD" "RELEASE_BECAUSE_ANONYMOUS_SOURCE" "RELEASE_BECAUSE_IP_PROFILE_CALL_LIMIT" "GWAPP_UNASSIGNED_NUMBER" "GWAPP_NO_ROUTE_TO_TRANSIT_NET" "GWAPP_NO_ROUTE_TO_DESTINATION" "GWAPP_CHANNEL_UNACCEPTABLE" "GWAPP_CALL_AWARDED_AND " "GWAPP_PREEMPTION" "PREEMPTION_CIRCUIT_RESERVED_FOR_REUSE" "GWAPP_NORMAL_CALL_CLEAR" "GWAPP_USER_BUSY" "GWAPP_NO_USER_RESPONDING"...
Page 614 Mediant 800 MSBG "GWAPP_CALL_REJECTED" "GWAPP_NUMBER_CHANGED" "GWAPP_NON_SELECTED_USER_CLEARING" "GWAPP_INVALID_NUMBER_FORMAT" "GWAPP_FACILITY_REJECT" "GWAPP_RESPONSE_TO_STATUS_ENQUIRY" "GWAPP_NORMAL_UNSPECIFIED" "GWAPP_CIRCUIT_CONGESTION" "GWAPP_USER_CONGESTION" "GWAPP_NO_CIRCUIT_AVAILABLE" "GWAPP_NETWORK_OUT_OF_ORDER" "GWAPP_NETWORK_TEMPORARY_FAILURE" "GWAPP_NETWORK_CONGESTION" "GWAPP_ACCESS_INFORMATION_DISCARDED" "GWAPP_REQUESTED_CIRCUIT_NOT_AVAILABLE" "GWAPP_RESOURCE_UNAVAILABLE_UNSPECIFIED " "GWAPP_PERM_FR_MODE_CONN_OUT_OF_S" "GWAPP_PERM_FR_MODE_CONN_OPERATIONAL" "GWAPP_PRECEDENCE_CALL_BLOCKED" • "RELEASE_BECAUSE_PREEMPTION_ANALOG_CIRCUIT_RESERVED_FOR_ REUSE" • "RELEASE_BECAUSE_PRECEDENCE_CALL_BLOCKED" "GWAPP_QUALITY_OF_SERVICE_UNAVAILABLE" "GWAPP_REQUESTED_FAC_NOT_SUBSCRIBED" "GWAPP_BC_NOT_AUTHORIZED" "GWAPP_BC_NOT_PRESENTLY_AVAILABLE" "GWAPP_SERVICE_NOT_AVAILABLE" "GWAPP_CUG_OUT_CALLS_BARRED" "GWAPP_CUG_INC_CALLS_BARRED" "GWAPP_ACCES_INFO_SUBS_CLASS_INCONS " "GWAPP_BC_NOT_IMPLEMENTED" "GWAPP_CHANNEL_TYPE_NOT_IMPLEMENTED" "GWAPP_REQUESTED_FAC_NOT_IMPLEMENTED"...
Page 615 SIP User's Manual 8. IP Telephony Capabilities "GWAPP_IDENTIFIED_CHANNEL_NOT_EXIST" "GWAPP_SUSPENDED_CALL_BUT_CALL_ID_NOT_EXIST" "GWAPP_CALL_ID_IN_USE" "GWAPP_NO_CALL_SUSPENDED" "GWAPP_CALL_HAVING_CALL_ID_CLEARED" "GWAPP_INCOMPATIBLE_DESTINATION" "GWAPP_INVALID_TRANSIT_NETWORK_SELECTION" "GWAPP_INVALID_MESSAGE_UNSPECIFIED" "GWAPP_NOT_CUG_MEMBER" "GWAPP_CUG_NON_EXISTENT" "GWAPP_MANDATORY_IE_MISSING" "GWAPP_MESSAGE_TYPE_NON_EXISTENT" "GWAPP_MESSAGE_STATE_INCONSISTENCY" "GWAPP_NON_EXISTENT_IE" "GWAPP_INVALID_IE_CONTENT" "GWAPP_MESSAGE_NOT_COMPATIBLE" "GWAPP_RECOVERY_ON_TIMER_EXPIRY" "GWAPP_PROTOCOL_ERROR_UNSPECIFIED" "GWAPP_INTERWORKING_UNSPECIFIED" "GWAPP_UKNOWN_ERROR" "RELEASE_BECAUSE_HELD_TIMEOUT" Version 6.2 February 2011...
Page 616 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 617: Voip Networking Capabilities
SIP User's Manual 9. VoIP Networking Capabilities VoIP Networking Capabilities This section provides an overview of the device's VoIP networking capabilities. NAT (Network Address Translation) Support Network Address Translation (NAT) is a mechanism that maps a set of internal IP addresses used within a private network to global IP addresses, providing transparent routing to end hosts.
Page 618: No-Op Packets
You can control the payload type with which the No-Op packets are sent. This is performed using the RTPNoOpPayloadType ini parameter (see ''Networking Parameters'' on page 653). 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 619: Multiple Routers Support
SIP User's Manual 9. VoIP Networking Capabilities ♦ Inbound media re-latch during a silence period: If a silence compression RTP packet is received, latching new RTP streams is disabled until a silence timeout expires. Currently, RTP packets with payload types 13 and 19 are considered silence compression packets.
Page 620: Network Configuration
Mediant 800 MSBG When the client receives a response to its request from the identified NTP server, it must be interpreted based on time zone or location offset that the system is to a standard point of reference called the Universal Time Coordinate (UTC). The time offset that the NTP client uses is configurable using the ini file parameter NTPServerUTCOffset, or via an SNMP MIB object (refer to the Product Reference Manual).
Page 621: Overview Of Multiple Interface Table
SIP User's Manual 9. VoIP Networking Capabilities The figure depicts a typical configuration featuring in which the device is configured with three network interfaces for: Operations, Administration, Maintenance, and Provisioning (OAMP) applications Call Control applications Media The Multiple Interfaces scheme allows the configuration of up to 12 different IP addresses, each associated with a unique VLAN ID.
Page 622: Columns Of The Multiple Interface Table
Mediant 800 MSBG 9.5.1.2 Columns of the Multiple Interface Table Each row of the table defines a logical IP interface with its own IP address, subnet mask (represented by Prefix Length), VLAN ID, name, and application types that are allowed on this interface.
Page 623: Table 9-3: Configured Default Gateway Example
SIP User's Manual 9. VoIP Networking Capabilities IPv6 addresses may be assigned in two ways: "IPv6 Manual" (4) "IPv6 Manual Prefix" (3) 9.5.1.2.4 IP Address and Prefix Length Columns These columns allow the user to configure an IPv4/IPv6 IP address and its related subnet mask.
Page 624: Other Related Parameters
Mediant 800 MSBG A separate routing table allows configuring static routing rules. Configuring the following routing rules enable OAMP applications to access peers on subnet 17.17.0.0 through the gateway 192.168.10.1 (which is not the default gateway of the interface), and Media &...
Page 625: Table 9-5: Quality Of Service Parameters
SIP User's Manual 9. VoIP Networking Capabilities Table 9-5: Quality of Service Parameters Parameter Description Layer-2 Class Of Service Parameter (VLAN Tag Priority Field) DiffServ Table This ini file table parameter allows you to configure [DiffServToVlanPriority] DiffServ-to-VLAN Priority mapping. For each packet sent to the LAN, the VLAN Priority of the packet is set according to the DiffServ value in the IP header of the packet.
Page 626: Table 9-7: Application Type Parameters
Mediant 800 MSBG Application Traffic / Network Types Class-of-Service (Priority) Web server (HTTP) Management Bronze Management Bronze SNMP GET/SET Web server (HTTPS) Management Bronze RTP traffic Media Premium media RTCP traffic Media Premium media T.38 traffic Media Premium media Control...
Page 627: Multiple Interface Table Configuration Summary And Guidelines
SIP User's Manual 9. VoIP Networking Capabilities 9.5.1.4 Multiple Interface Table Configuration Summary and Guidelines Multiple Interface table configuration must adhere to the following rules: Up to 12 different interfaces may be defined. The indices used must be in the range between 0 and 11. Each interface must have its own subnet.
Page 628: Troubleshooting The Multiple Interface Table
Mediant 800 MSBG Note: When configuring the device using the Web interface, it is possible to perform a quick validation of the configured Multiple Interface table and VLAN definitions, by clicking the Done button in the Multiple Interface Table Web page.
Page 629: Routing Table Columns
SIP User's Manual 9. VoIP Networking Capabilities 9.5.2.2 Routing Table Columns Each row of the Routing table defines a static routing rule. Traffic destined to the subnet specified in the routing rule is re-directed to the defined gateway, reachable through the specified interface.
Page 630: Routing Table Configuration Summary And Guidelines
Mediant 800 MSBG Figure 9-2: Interface Column 9.5.2.2.5 Metric Column The Metric column must be set to 1 for each static routing rule. 9.5.2.2.6 State Column The State column displays the state of each static route. Possible values are "Active" and "Inactive".
Page 631: Troubleshooting The Routing Table
SIP User's Manual 9. VoIP Networking Capabilities 9.5.2.4 Troubleshooting the Routing Table When adding a new static routing rule, the added rule passes a validation test. If errors are found, the routing rule is rejected and is not added to the IP Routing table. Failed routing validations may result in limited connectivity (or no connectivity) to the destinations specified in the incorrect routing rule.
Page 632 Mediant 800 MSBG FORMAT StaticRouteTable Index = StaticRouteTable InterfaceName, StaticRouteTable_Destination, StaticRouteTable_PrefixLength, StaticRouteTable_Gateway, StaticRouteTable_Description; StaticRouteTable 0 = 0, 10.31.174.0, 24, 192.168.11.1, ; StaticRouteTable 1 = 1, 174.96.151.15, 24, 10.32.174.12, ; StaticRouteTable 2 = 3, 10.35.174.0, 24, 10.34.174.240, ; [ \StaticRouteTable ] ;...
Page 633: Networking Configuration Examples
SIP User's Manual 9. VoIP Networking Capabilities • For packets sent with DiffServ value of 10, set VLAN priority to 2 The DNS and the NTP applications are configured to serve as OAMP applications. Notes: • Lines that begin with a semicolon are considered a remark and are ignored.
Page 634: Table 9-11: Multiple Interface Table - Example 2
Mediant 800 MSBG StaticRouteTable 0 = 0, 201.201.0.0, 16, 192.168.11.10, ; StaticRouteTable 1 = 0, 202.202.0.0, 16, 192.168.11.1, ; [ \StaticRouteTable ] Example 2 - Three VoIP Interfaces, One for each Application Exclusively: the Multiple Interface table is configured with three interfaces, one exclusively for each application type:...
Page 635: Table 9-13: Multiple Interface Table - Example 3
SIP User's Manual 9. VoIP Networking Capabilities Example 3 - Three Interfaces: one exclusively for management (OAMP applications) and two others for Call Control and RTP (Control and Media applications) : Table 9-13: Multiple Interface Table - Example 3 Allowed Interface Prefix Default...
Page 636 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 637: Advanced Pstn Configuration
SIP User's Manual 10. Advanced PSTN Configuration Advanced PSTN Configuration This section discusses advanced PSTN configurations. 10.1 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 638: Configuring Internal Clock As Clock Source
Mediant 800 MSBG 10.1.2 Configuring Internal Clock as Clock Source This section describes how to configure the device to use its internal clock source. The internal clock source is a stratum 4E-compliant clock source. When the device has no line interfaces, the device should be configured in this mode.
Page 639: Fixed Mapping Of Isdn Release Reason To Sip Response
SIP User's Manual 10. Advanced PSTN Configuration 10.2.2 Fixed Mapping of ISDN Release Reason to SIP Response The following table describes the mapping of ISDN release reason to SIP response. Table 10-1: Mapping of ISDN Release Reason to SIP Response ISDN Release Description Description...
Page 640 Mediant 800 MSBG ISDN Release Description Description Reason Response Bearer capability not implemented Not implemented Channel type not implemented 480* Temporarily unavailable Requested facility not implemented 503* Service unavailable Only restricted digital information bearer 503* Service unavailable capability is available...
Page 641: Fixed Mapping Of Sip Response To Isdn Release Reason
SIP User's Manual 10. Advanced PSTN Configuration 10.2.3 Fixed Mapping of SIP Response to ISDN Release Reason The following table describes the mapping of SIP response to ISDN release reason. Table 10-2: Mapping of SIP Response to ISDN Release Reason ISDN Release SIP Response Description...
Page 642: Isdn Overlap Dialing
Mediant 800 MSBG ISDN Release SIP Response Description Description Reason Busy everywhere User busy Decline Call rejected Does not exist anywhere Unallocated number 606* Not acceptable Network out of order * Messages and responses were created because the ‘ISUP to SIP Mapping’ draft does not specify their cause code mapping.
Page 643: Isdn Non-Facility Associated Signaling (Nfas)
SIP User's Manual 10. Advanced PSTN Configuration Relevant parameters (described in ''PSTN Parameters'' on page 783): ISDNRxOverlap ISDNTxOverlap TimeBetweenDigits MaxDigits ISDNInCallsBehavior DigitMapping MinOverlapDigitsForRouting For configuring ISDN overlap dialing using the Web interface, see ''Configuring Trunk Settings'' on page 101. 10.4 ISDN Non-Facility Associated Signaling (NFAS) In regular T1 ISDN trunks, a single 64 kbps channel carries signaling for the other 23 B- channels of that particular T1 trunk.
Page 644: Nfas Interface Id
Mediant 800 MSBG 10.4.1 NFAS Interface ID Several ISDN switches require an additional configuration parameter per T1 trunk that is called ‘Interface Identifier’. In NFAS T1 trunks, the Interface Identifier is sent explicitly in Q.931 Setup / Channel Identification IE for all NFAS trunks, except for the B-channels of the Primary trunk (see note below).
Page 645: Creating An Nfas-Related Trunk Configuration
SIP User's Manual 10. Advanced PSTN Configuration If there is no NFAS Backup trunk, the following configuration should be used: ISDNNFASInterfaceID_0 = 0 ISDNNFASInterfaceID_1 = 2 ISDNNFASInterfaceID_2 = 3 ISDNNFASInterfaceID_3 = 4 ISDNIBehavior = 512 ;This parameter should be added because of ;ISDNNFASInterfaceID coniguration above NFASGroupNumber_0 = 1 NFASGroupNumber_1 = 1...
Page 646: Redirect Number And Calling Name (Display)
Mediant 800 MSBG 10.5 Redirect Number and Calling Name (Display) The following tables define the device's redirect number and calling name (Display) support for various ISDN variants according to NT (Network Termination) / TE (Termination Equipment) interface direction: Table 10-3: Calling Name (Display)
Page 647: Tunneling Applications
SIP User's Manual 11. Tunneling Applications Tunneling Applications This section discusses the device's support for VoIP tunneling applications. 11.1 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 648 Mediant 800 MSBG By utilizing the ‘Profiles’ mechanism (see ''Coders and Profiles'' on page 138), you can configure the TDM Tunneling feature to choose different settings based on a timeslot or groups of timeslots. For example, you can use low-bit-rate vocoders to transport voice and ‘Transparent’...
Page 649 SIP User's Manual 11. Tunneling Applications [TelProfile] FORMAT TelProfile_Index = TelProfile_ProfileName, TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP; TelProfile 1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$; TelProfile 2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$; [\TelProfile] Originating Side: ;E1_TRANSPARENT_31 ProtocolType_0 = 5 ProtocolType_1 = 5...
Page 650: Dsp Pattern Detector
Mediant 800 MSBG 11.1.1 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. The device can be configured to detect up to four different one-byte data patterns.
Page 651 SIP User's Manual 11. 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 652 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 653: Configuration Parameters Reference
SIP User's Manual 12. Configuration Parameters Reference Configuration Parameters Reference The device's VoIP functionality (not data-routing functionality) configuration parameters, default values, and their descriptions are documented in this section. Parameters and values enclosed in square brackets ([...]) represent the ini file parameters and their enumeration values;...
Page 654: Voip Static Routing Parameters
Mediant 800 MSBG Parameter Description Realm' table, which in turn is assigned to an IP Group). Each interface index must be unique. Each interface must have a unique VLAN ID. Each interface must have a unique subnet. Subnets in different interfaces must not overlap (e.g., defining two interfaces with 10.0.0.1/8 and 10.50.10.1/24 is invalid).
Page 655: Quality Of Service Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description defined). The format of this parameter is as follows: [ StaticRouteTable ] FORMAT StaticRouteTable_Index = StaticRouteTable_InterfaceName, StaticRouteTable_Destination, StaticRouteTable_PrefixLength, StaticRouteTable_Gateway, StaticRouteTable_Description; [ \StaticRouteTable ] Notes: The Gateway address must be in the same subnet as configured in the 'Multiple Interface' table for VoIP network interfaces (refer to ''Configuring IP Interface Settings'' on page 83).
Page 656 Mediant 800 MSBG Parameter Description DiffServToVlanPriority 0 = 46, 6; DiffServToVlanPriority 1 = 40, 6; DiffServToVlanPriority 2 = 26, 4; DiffServToVlanPriority 3 = 10, 2; Notes: For this parameter to take effect, a device reset is required. You can configure up to 64 VLAN tag priorities (i.e., indices 0-63).
Page 657: Nat Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.1.4 NAT Parameters The Network Address Translation (NAT) parameters are described in the table below. Table 12-4: NAT Parameters Parameter Description Web/EMS: NAT Traversal Enables or disables the NAT mechanism. [DisableNAT] [0] Enable [1] Disable (default) Web: NAT IP Address Global (public) IP address of the device to enable static NAT between...
Page 658: Nfs Parameters
Mediant 800 MSBG 12.1.5 NFS Parameters The Network File Systems (NFS) configuration parameters are described in the table below. Table 12-5: NFS Parameters Parameter Description [NFSBasePort] Start of the range of numbers used for local UDP ports used by the NFS client.
Page 659: Dns Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.1.6 DNS Parameters The Domain name System (DNS) parameters are described in the table below. Table 12-6: DNS Parameters Parameter Description Web: DNS Primary Server The IP address of the primary DNS server. Enter the IP address in dotted-decimal notation, for example, 10.8.2.255.
Page 660: Dhcp Parameters
Mediant 800 MSBG Parameter Description FORMAT SRV2IP_Index = SRV2IP_InternalDomain, SRV2IP_TransportType, SRV2IP_Dns1, SRV2IP_Priority1, SRV2IP_Weight1, SRV2IP_Port1, SRV2IP_Dns2, SRV2IP_Priority2, SRV2IP_Weight2, SRV2IP_Port2, SRV2IP_Dns3, SRV2IP_Priority3, SRV2IP_Weight3, SRV2IP_Port3; [\SRV2IP] For example: SRV2IP 0 = SrvDomain,0,Dnsname1,1,1,500,Dnsname2,2,2,501,$$,0,0,0; Notes: This parameter can include up to 10 indices. If the Internal SRV table is used, the device first attempts to resolve a domain name using this table.
Page 661: Ntp And Daylight Saving Time Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description [2] to [10] = Fast When set to 0, the DHCP lease renewal is disabled. Otherwise, the renewal time is divided by this factor. Some DHCP-enabled routers perform better when set to 4. Note: For this parameter to take effect, a device reset is required.
Page 662: Web And Telnet Parameters
Mediant 800 MSBG Parameter Description Web: End Time Defines the date and time when daylight saving ends. EMS: End The format of the value is mo:dd:hh:mm (month, day, hour, and [DayLightSavingTimeEnd] minutes). Web/EMS: Offset Daylight saving time offset (in minutes).
Page 663: Web Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.2.2 Web Parameters The Web parameters are described in the table below. Table 12-10: Web Parameters Parameter Description [DisableWebTask] Disables or enables device management through the Web interface. [0] = Enable Web management (default). [1] = Disable Web management.
Page 664: Telnet Parameters
Mediant 800 MSBG Parameter Description [WelcomeMessage] This ini file table parameter configures the Welcome message that appears after a Web interface login. The format of this parameter is as follows: [WelcomeMessage ] FORMAT WelcomeMessage_Index = WelcomeMessage_Text [\WelcomeMessage] For Example: FORMAT WelcomeMessage_Index = WelcomeMessage_Text WelcomeMessage 1 = "**********************************"...
Page 665: Debugging And Diagnostics Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.3 Debugging and Diagnostics Parameters This subsection describes the device's debugging and diagnostic parameters. 12.3.1 General Parameters The general debugging and diagnostic parameters are described in the table below. Table 12-12: General Debugging and Diagnostic Parameters Parameter Description EMS: Enable Diagnostics...
Page 666: Syslog, Cdr And Debug Parameters
Mediant 800 MSBG Parameter Description [0] = Lifeline is activated upon power failure (default). [1] = Lifeline is activated upon power failure or when the link is down (physically disconnected). [2] = Lifeline is activated upon power failure, when the link is down, or upon network failure (logical link disconnected).
Page 667 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web/EMS: Syslog Server IP The IP address (in dotted-decimal notation) of the computer on which Address the Syslog server is running. The Syslog server is an application [SyslogServerIP] designed to collect the logs and error messages generated by the device.
Page 668 Mediant 800 MSBG Parameter Description The Syslog debug level automatically changes between level 5, level 1, and level 0, depending on the device's CPU consumption so that VoIP traffic isn’t affected. Syslog messages are bundled into a single UDP packet, after which they are sent to a Syslog server (bundling size is determined by the MaxBundleSyslogLength parameter).
Page 669: Remote Alarm Indication Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description (2) 'General Security Settings' (3) 'Configuration File' (5) 'Software Upgrade Key' (7) 'Web Access List' (8) 'Web User Accounts' [naa] Non Authorized Access = Attempt to access the Web interface with a false or empty user name or password. [spc] Sensitive Parameters Value Change = Changes made to sensitive parameters: (1) IP Address...
Page 670: Serial Parameters
Mediant 800 MSBG Parameter Description [RAILowThreshold] Low threshold percentage of total calls that are active (busy endpoints). When the percentage of the device's busy endpoints falls below this low threshold, the device sends an SNMP acBoardCallResourcesAlarm alarm trap with a 'cleared' alarm status.
Page 671: Security Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description EMS: Flow Control Determines the value of the RS-232 flow control. [SerialFlowControl] [0] = None (default). [1] = Hardware. Note: For this parameter to take effect, a device reset is required. 12.4 Security Parameters This subsection describes the device's security parameters.
Page 672: Https Parameters
Mediant 800 MSBG 12.4.2 HTTPS Parameters The Secure Hypertext Transport Protocol (HTTPS) parameters are described in the table below. Table 12-17: HTTPS Parameters Parameter Description Web: Secured Web Connection Determines the protocol used to access the Web interface. (HTTPS) [0] HTTP and HTTPS (default).
Page 673: Srtp Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Notes: For this parameter to take effect, a device reset is required. For a description on implementing client certificates, see ''Client Certificates'' on page 71. 12.4.3 SRTP Parameters The Secure Real-Time Transport Protocol (SRTP) parameters are described in the table below.
Page 674: Tls Parameters
Mediant 800 MSBG Parameter Description encryption with a 128-bit key and HMAC-SHA1 message authentication with a 80-bit tag. [2] AES_CM_128_HMAC_SHA1_32 = device uses AES-CM encryption with a 128-bit key and HMAC-SHA1 message authentication with a 32-bit tag. Web: Disable Authentication On...
Page 675 SIP User's Manual 12. Configuration Parameters Reference Parameter Description the client certificate to establish the TLS connection. Notes: For this parameter to take effect, a device reset is required. The SIPS certificate files can be changed using the parameters HTTPSCertFileName and HTTPSRootFileName.
Page 676: Ssh Parameters
Mediant 800 MSBG 12.4.5 SSH Parameters The Secure Shell (SSH) parameters are described in the table below. Table 12-20: SSH Parameters Parameter Description Web/EMS: SSH Server Enable Enables or disables the device's embedded SSH server. [SSHServerEnable] [0] Disable (default) [1] Enable Web/EMS: SSH Server Port Defines the port number for the embedded SSH server.
Page 677: Ocsp Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.4.6 OCSP Parameters The Online Certificate Status Protocol (OCSP) parameters are described in the table below. Table 12-21: OCSP Parameters Parameter Description EMS: OCSP Enable Enables or disables certificate checking using OCSP. [OCSPEnable] [0] = Disable (default).
Page 678 Mediant 800 MSBG Parameter Description [1] At Connect & Release = Sent at call connect and release. [2] At Setup & Release = Sent at call setup and release. Web: AAA Indications Determines the Authentication, Authorization and Accounting EMS: Indications (AAA) indications.
Page 679: Snmp Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: RADIUS VSA Vendor ID Defines the vendor ID that the device accepts when parsing a [RadiusVSAVendorID] RADIUS response packet. The valid range is 0 to 0xFFFFFFFF. The default value is 5003. Web: RADIUS VSA Access Defines the code that indicates the access level attribute in the Level Attribute...
Page 680 Mediant 800 MSBG Parameter Description [ifAlias] The textual name of the interface. The value is equal to the ifAlias SNMP MIB object. The valid range is a string of up to 64 characters. EMS: Keep Alive Trap Port The port to which the keep-alive traps are sent.
Page 681 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: SNMP Trap Destination Parameters EMS: Network > SNMP Managers Table Note: Up to five SNMP trap managers can be defined. SNMP Manager Determines the validity of the parameters (IP address and port [SNMPManagerIsUsed_x] number) of the corresponding SNMP Manager used to receive SNMP traps.
Page 682: Sip Media Realm Parameters
Mediant 800 MSBG Parameter Description Web: SNMP V3 Table EMS: SNMP V3 Users This ini file table parameter configures SNMP v3 users. The [SNMPUsers] format of this parameter is as follows: [SNMPUsers] FORMAT SNMPUsers_Index = SNMPUsers_Username, SNMPUsers_AuthProtocol, SNMPUsers_PrivProtocol, SNMPUsers_AuthKey, SNMPUsers_PrivKey, SNMPUsers_Group;...
Page 683: Control Network Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description For example, CpMediaRealm 1 = Mrealm1, Voice, , 6600, 20, 6790; CpMediaRealm 2 = Mrealm2, Voice, , 6800, 10, 6890; Notes: For this parameter to take effect, a device reset is required. This table can include up to 64 indices (where 0 is the first index).
Page 684 Mediant 800 MSBG Parameter Description IPGroup_OutboundManSet, IPGroup_ContactName; [\IPGroup] For example: IPGroup 1 = 0, "dol gateway", 1, firstIPgroup, , 0, -1, 0, 0, -1, 0, mrealm1, 1, 1, ; IPGroup 2 = 0, "abc server", 2, secondIPgroup, , 0, -1, 0, 0, -1, 0, mrealm2, 1, 2, ;...
Page 685 SIP User's Manual 12. Configuration Parameters Reference Parameter Description individual parameter Password) are used for authentication. Authentication is typically used for FXS interfaces, but can also be used for FXO interfaces. For configuring the Authentication table using the Web interface, see Configuring Authentication on page 183. For an explanation on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 368.
Page 686 Mediant 800 MSBG Parameter Description Web/EMS: Proxy Name Defines the Home Proxy domain name. If specified, this name [ProxyName] is used as the Request-URI in REGISTER, INVITE, and other SIP messages, and as the host part of the To header in INVITE messages.
Page 687 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [0] Standard = INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI, according to the Refer-To header in the REFER message, or Contact header in the 3xx response (default). [1] Proxy = Sends a new INVITE to the Proxy.
Page 688 Mediant 800 MSBG Parameter Description performed. Note: To enable NAPTR/SRV queries for Proxy servers only, use the parameter ProxyDNSQueryType. Web: Proxy DNS Query Type Enables the use of DNS Naming Authority Pointer (NAPTR) [ProxyDNSQueryType] and Service Record (SRV) queries to discover Proxy servers.
Page 689 SIP User's Manual 12. Configuration Parameters Reference Parameter Description The OPTIONS Request-URI host part contains either the device's IP address or a string defined by the parameter SIPGatewayName. The device uses the OPTIONS request as a keep-alive message to its primary and redundant Proxies (i.e., the parameter EnableProxyKeepAlive is set to 1).
Page 690 Mediant 800 MSBG Parameter Description Note: Challenge Caching is used with all proxies and not only with the active one. Web: Proxy IP Table EMS: Proxy IP [ProxyIP] This ini file table parameter configures the Proxy Set table with Proxy Set IDs, each with up to five Proxy server IP addresses (or fully qualified domain name/FQDN).
Page 691 SIP User's Manual 12. Configuration Parameters Reference Parameter Description This table parameter can include up to 32 indices (0-31). For configuring the Proxy Set IDs and their IP addresses, use the parameter ProxyIP. For configuring the Proxy Set ID table using the Web interface and for a detailed description of the parameters of this ini file table, see ''Configuring Proxy Sets Table'' on page 126.
Page 692 Mediant 800 MSBG Parameter Description Web/EMS: Registrar Transport Determines the transport layer used for outgoing SIP dialogs Type initiated by the device to the Registrar. [RegistrarTransportType] [-1] Not Configured (default) [0] UDP [1] TCP [2] TLS Note: When set to ‘Not Configured’, the value of the parameter SIPTransportType is used.
Page 693 SIP User's Manual 12. Configuration Parameters Reference Parameter Description outbound proxy server failure). The remote SIP UA abandons a call and the only provisional response the device has received for the call is 100 Trying (indicative of a home proxy server failure, i.e., the failure of a proxy in the route after the outbound proxy).
Page 694 Mediant 800 MSBG 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. EMS: Set OOS On Registration Fail [0] Disable (default) [OOSOnRegistrationFail] [1] Enable If the registration is per endpoint (i.e., AuthenticationMode is...
Page 695 SIP User's Manual 12. Configuration Parameters Reference Parameter Description response - set to an empty value For example: Authorization: Digest username=alice_private@home1.net, realm=”home1.net”, nonce=””, response=”e56131d19580cd833064787ecc” Note: This registration header is according to the IMS 3GPP TS24.229 and PKT-SP-24.220 specifications. Web: Add initial Route Header Determines whether the SIP Route header is included in initial registration or re-registration (REGISTER) requests sent by the [InitialRouteHeader]...
Page 696: Network Application Parameters
Mediant 800 MSBG Parameter Description [PingPongKeepAliveTime] Defines the periodic interval (in seconds) after which a “ping” (double-CRLF) keep-alive is sent to a proxy/registrar, using the CRLF Keep-Alive mechanism. The default range is 5 to 2,000,000. The default is 120. The device uses the range of 80-100% of this user-defined value as the actual interval.
Page 697 SIP User's Manual 12. Configuration Parameters Reference Parameter Description SIPInterface 0 = Voice, 2, 5060, 5060, 5061, 1; SIPInterface 1 = Voice, 2, 5070, 5070, 5071, 2; SIPInterface 2 = Voice, 0, 5090, 5000, 5081, 2; Notes: This table can include up to 32 indices (where 0 is the first index). Each SIP Interface must have a unique signaling port (i.e., no two SIP Interfaces can share the same port - no port overlapping).
Page 698: General Sip Parameters
Mediant 800 MSBG Parameter Description Uses the NATTranslation parameter to define NAT per interface. If NAT is not configured (by any of the above-mentioned methods), the device sends the packet according to its IP address defined in the Multiple Interface table.
Page 699 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Number of received and sent voice packets Number of received and sent voice octets Received packet loss, jitter (in ms), and latency (in ms) The X-RTP-Stat header contains the following fields: PS=<voice packets sent>...
Page 700 Mediant 800 MSBG Parameter Description set the parameter ProgressIndicator2IP to 1. If it is equal to 0, 180 Ringing response is sent. For Digital interfaces: Sending a 183 response depends on the ISDN Progress Indicator (PI). It is sent only if PI is set to 1 or 8 in the received Proceeding or Alerting PRI messages.
Page 701 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web/EMS: Session Determines the SIP method used for session-timer updates. Expires Method [0] Re-INVITE = Uses Re-INVITE messages for session-timer [SessionExpiresMethod] updates (default). [1] UPDATE = Uses UPDATE messages. Notes: The device can receive session-timer refreshes using both methods. The UPDATE message used for session-timer is excluded from the SDP body.
Page 702 Mediant 800 MSBG Parameter Description When this parameter is set to 0, T.38 might still be used without the control protocol's involvement. To completely disable T.38, set FaxTransportMode to a value other than 1. This parameter can also be configured per IP Profile (using the IPProfile parameter).
Page 703 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Mode [0] = Disable (default) - all TCP connections (except those that are [ReliableConnectionPersi set to a proxy IP) are released if not used by any SIP stentMode] dialog\transaction. [1] = Enable - TCP connections to all destinations are persistent and not released unless the device reaches 70% of its maximum TCP resources.
Page 704 Mediant 800 MSBG Parameter Description Web: Enable History-Info Enables usage of the History-Info header. Header [0] Disable (default) EMS: Enable History Info [1] Enable [EnableHistoryInfo] User Agent Client (UAC) Behavior: Initial request: The History-Info header is equal to the Request-URI.
Page 705 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [2] No Reply = Call forwarding no reply [9] DTE Out of Order = Call forwarding DTE out of order [10] Deflection = Call deflection [15] Systematic/Unconditional = Call forward unconditional Web: Use Tgrp Information Determines whether the SIP 'tgrp' parameter is used.
Page 706 Mediant 800 MSBG Parameter Description the Bearer Capability IE contains “Speech”, the INVITE in this case does not contain tgrp and trunk-context parameters. [4] Hotline Extended = Interworks the ISDN Setup message’s hotline "OffHook Indicator" Information Element (IE) to SIP INVITE’s Request-URI and Contact headers.
Page 707 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Group ID 56: INVITE sip:123456@192.168.1.2;dtg=56;user=phone SIP/2.0 Note: If the Hunt Group is not found based on the 'dtg' parameter, the 'Inbound IP Routing Table' is used instead for routing the call to the appropriate Hunt Group.
Page 708 EMS: User Agent Display value>/software version' is used, for example: Info User-Agent: myproduct/v.6.00.010.006 [UserAgentDisplayInfo] If not configured, the default string, '<AudioCodes product- name>/software version' is used, for example: User-Agent: Audiocodes-Sip-Gateway-Mediant 800 MSBG/v.6.00.010.006 The maximum string length is 50 characters. Note: The software version number and preceding forward slash (/) cannot be modified.
Page 709 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Multiple Determines whether the 'mptime' attribute is included in the outgoing Packetization Time Format SDP. EMS: Multi Ptime Format [0] None = Disabled (default) [MultiPtimeFormat] [1] PacketCable = includes the 'mptime' attribute in the outgoing SDP - PacketCable-defined format The 'mptime' attribute enables the device to define a separate Packetization period for each negotiated coder in the SDP.
Page 710 Mediant 800 MSBG Parameter Description Web/EMS: Enable P- Determines the device usage of the P-Associated-URI header. This Associated-URI Header header can be received in 200 OK responses to REGISTER requests. [EnablePAssociatedURIH When enabled, the first URI in the P-Associated-URI header is used in eader] subsequent requests as the From/P-Asserted-Identity headers value.
Page 711 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Forking Timeout The timeout (in seconds) that is started after the first SIP 2xx response [ForkingTimeOut] has been received for a User Agent when a Proxy server performs call forking (Proxy server forwards the INVITE to multiple SIP User Agents). The device sends a SIP ACK and BYE in response to any additional SIP 2xx received from the Proxy within this timeout.
Page 712 Mediant 800 MSBG Parameter Description [1] = The device's IP address is used as the URI host part instead of "anonymous.invalid". This parameter may be useful, for example, for service providers who identify their SIP Trunking customers by their source phone number or IP address, reflected in the From header of the SIP INVITE.
Page 713 SIP User's Manual 12. Configuration Parameters Reference Parameter Description If the Via header includes the 'rport' parameter without a port value, the destination port of the response is the source port of the incoming request. If the Via header includes 'rport' with a port value (e.g., rport=1001), the destination port of the response is the port indicated in the 'rport' parmeter.
Page 714 Mediant 800 MSBG Parameter Description [EnableRekeyAfter181] Enables the device to send a Re-INVITE with a new (different) SRTP key (in the SDP) upon receipt of a SIP 181 response ("call is being forwarded"). [0] = Disable (default) [1] = Enable Note: This parameter is applicable only if SRTP is used.
Page 715 SIP User's Manual 12. Configuration Parameters Reference Parameter Description For a list of SIP responses-Q.931 release cause mapping, see ''Release Reason Mapping'' on page 638. Determines the device's interworking of Alerting messages from PRI to [IgnoreAlertAfterEarlyMe dia] SIP. [0] = Disabled (default). [1] = Enabled.
Page 716 Mediant 800 MSBG Parameter Description [IgnoreRemoteSDPMKI] Determines whether the device ignores the Master Key Identifier (MKI) if present in the SDP received from the remote side. [0] Disable (default) [1] Enable [TrunkStatusReportingM Determines whether the device responds to SIP OPTIONS if all the ode] trunks pertaining to Trunk Group #1 are down or busy.
Page 717 SIP User's Manual 12. Configuration Parameters Reference 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 718 Mediant 800 MSBG Parameter Description [2] Transmit Only= Send RTP only [3] Receive Only= Receive RTP only Notes: To configure the RTP Only mode per trunk, use the RTPOnlyModeForTrunk_ID parameter. If per trunk configuration (using the RTPOnlyModeForTrunk_ID parameter) is set to a value other than the default, the RTPOnlyMode parameter value is ignored.
Page 719 SIP User's Manual 12. Configuration Parameters Reference Parameter Description and SITQ850CauseForRO parameters. Web/EMS: SIT Q850 Determines the Q.850 cause value specified in the SIP Reason header Cause For NC that is included in a 4xx response when SIT-NC (No Circuit Found [SITQ850CauseForNC] Special Information Tone) is detected from the PSTN for IP-to-Tel calls.
Page 720 Mediant 800 MSBG Parameter Description Notes: For Analog interfaces: The FXSOOSBehavior parameter determines the behavior of the FXS endpoints when a Busy Out or Graceful Lock occurs. For Analog interfaces: FXO endpoints during Busy Out and Lock are inactive. For Analog interfaces: See the LifeLineType parameter for complementary optional behavior.
Page 721: C Oders And Profile Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: SIP T2 The maximum interval (in msec) between retransmissions of SIP Retransmission Timer messages. [msec] The default is 4000. EMS: T2 RTX Note: The time interval between subsequent retransmissions of the [SipT2Rtx] same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.
Page 722 Mediant 800 MSBG Parameter Description [0] Disabled (default) [1] Enabled For example, below are defined two Coder Groups (0 and 1): [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = g711Alaw64k, 20, 0, 255, 0;...
Page 723 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Each coder type can appear only once per Coder Group. Only the packetization time of the first coder in the defined coder list is declared in INVITE/200 OK SDP, even if multiple coders are defined.
Page 724 Mediant 800 MSBG Parameter Description IpProfile_DisconnectOnBrokenConnection, IpProfile_FirstTxDtmfOption, IpProfile_SecondTxDtmfOption, IpProfile_RxDTMFOption, IpProfile_EnableHold, IpProfile_InputGain, IpProfile_VoiceVolume, IpProfile_AddIEInSetup, IpProfile_SBCExtensionCodersGroupID, IpProfile_MediaIPVersionPreference, IpProfile_TranscodingMode, IpProfile_SBCAllowedCodersGroupID, IpProfile_SBCAllowedCodersMode, IpProfile_SBCMediaSecurityBehaviour, IpProfile_SBCRFC2833Behavior, IpProfile_SBCAlternativeDTMFMethod, IpProfile_SBCAssertIdentity, IpProfile_AMDSensitivityParameterSuit, IpProfile_AMDSensitivityLevel, IpProfile_AMDMaxGreetingTime, IpProfile_AMDMaxPostSilenceGreetingTime, IpProfile_SBCDiversionMode, IpProfile_SBCHistoryInfoMode; [\IPProfile] For example: IPProfile 1 = ITSP, 1, 0, 0, 10, 10, 46, 40, 0, 0, 0, 0, 2, 0, 0, 0, 0, -1, 1, 0, 0, -1, 1, 4, -1, 1, 1, 0, 0, , -1, 0, 0, -1, 0, 0, 0, 0, -1, 0, 8, 300, 400, -1, -1;...
Page 725 SIP User's Manual 12. Configuration Parameters Reference Parameter Description pression IpProfile_RTPRed RTP Redundancy RTPRedundancyDe undancyDepth Depth IpProfile_RemoteB Remote RTP Base RemoteBaseUDPP aseUDPPort UDP Port IpProfile_CNGmod CNG Detector CNGDetectorMode Mode IpProfile_VxxTran Modems Transport V21ModemTranspo sportType Type rtType; V22ModemTranspo rtType; V23ModemTranspo rtType; V32ModemTranspo rtType;...
Page 726 Mediant 800 MSBG Parameter Description IpProfile_InputGai Input Gain InputGain IpProfile_VoiceVol Voice Volume VoiceVolume IpProfile_AddIEInS Add IE In SETUP AddIEinSetup etup IpProfile_SBCExte Extension Coders SBCExtensionCode nsionCodersGrou Group ID rsGroupID Media IP Version MediaIPVersionPref IpProfile_MediaIPV Preference erence ersionPreference IpProfile_Transco Transcoding Mode TranscodingMode...
Page 727 SIP User's Manual 12. Configuration Parameters Reference Parameter Description parameters (i.e., parameters configurable in both IP and Tel Profiles) of the preferred profile are applied to that call. If the Tel and IP Profiles are identical, the Tel Profile parameters take precedence.
Page 728 Mediant 800 MSBG Parameter Description TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP, TelProfile_TimeForReorderTone, TelProfile_EnableDIDWink, TelProfile_IsTwoStageDial, TelProfile_DisconnectOnBusyTone, TelProfile_EnableVoiceMailDelay, TelProfile_DialPlanIndex, TelProfile_Enable911PSAP, TelProfile_SwapTelToIpPhoneNumbers, TelProfile_EnableAGC, TelProfile_ECNlpMode; TelProfile_DigitalCutThrough; [\TelProfile] For example: TelProfile 1 = ITSP_audio, 1, 0, 0, 10, 10, 46, 40, -11, 0, 0, 0, 0, 0, 1, 0, 0, 700, 0, -1, 255, 0, 1, 1, 1, -1, 1, 0, 0, 0, 0;...
Page 729 SIP User's Manual 12. Configuration Parameters Reference Parameter Description TelProfile_EnableC Enable Current EnableCurrentDisco urrentDisconnect Disconnect nnect TelProfile_EnableD Enable Digit EnableDigitDelivery igitDelivery Delivery TelProfile_EnableE Echo Canceler EnableEchoCancell TelProfile_MWIAna MWI Analog Lamp MWIAnalogLamp TelProfile_MWIDis MWI Display MWIDisplay play Flash Hook Period FlashHookPeriod TelProfile_FlashHo okPeriod TelProfile_EnableE...
Page 730 Mediant 800 MSBG Parameter Description DisconnectOnBusyTone, and Enable911PSAP. The parameter IpPreference determines the priority of the Tel Profile (1 to 20, where 20 is the highest preference). If both IP and Tel Profiles apply to the same call, the coders and common parameters (i.e., parameters configurable in both IP and Tel...
Page 731: C Hannel Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.11 Channel Parameters This subsection describes the device's channel parameters. 12.11.1 Voice Parameters The voice parameters are described in the table below. Table 12-29: 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 732 Mediant 800 MSBG Parameter Description Web: Answer Detector Currently, not supported. Redirection [AnswerDetectorRedirection] Web: Answer Detector Sensitivity Determines the Answer Detector sensitivity. EMS: Sensitivity The range is 0 (most sensitive) to 2 (least sensitive). The default [AnswerDetectorSensitivity] is 0. Web: Silence Suppression...
Page 733: Coder Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Note: This parameter can also be configured per Tel Profile, using the TelProfile parameter (see ''Configuring Tel Profiles'' on page 141). [EchoCancellerAggressiveNLP] Enables or disables the Aggressive NLP at the first 0.5 second of the call.
Page 734: Fax And Modem Parameters
Mediant 800 MSBG 12.11.3 Fax and Modem Parameters The fax and modem parameters are described in the table below. Table 12-31: Fax and Modem Parameters Parameter Description Web: Fax Transport Mode Fax transport mode used by the device. EMS: Transport Mode...
Page 735 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [10] 26400bps = 26.4 kbps [11] 28800bps = 28.8 kbps [12] 31200bps = 31.2 kbps [13] 33600bps = 33.6 kbps Notes: The rate is negotiated between both sides (i.e., the device adapts to the capabilities of the remote side).
Page 736 Mediant 800 MSBG Parameter Description Web: Fax/Modem Bypass Number of (20 msec) coder payloads that are used to generate a Packing Factor fax/modem bypass packet. EMS: Packetization Period The valid range is 1, 2, or 3 coder payloads. The default value is [FaxModemBypassM] 1 coder payload.
Page 737 SIP User's Manual 12. Configuration Parameters Reference Parameter Description EMS: Enable Inband Network Enables or disables in-band network detection related to Detection fax/modem. [EnableFaxModemInbandNetw [0] = Disable (default) orkDetection] [1] = Enable When this parameter is enabled on Bypass and transparent with events mode (VxxTransportType is set to 2 or 3), a detection of an Answer Tone from the network triggers a switch to bypass mode in addition to the local Fax/Modem tone detections.
Page 738 Mediant 800 MSBG Parameter Description Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 143). Web: V.22 Modem Transport V.22 Modem Transport Type used by the device. Type [0] Disable = Disable (Transparent)
Page 739: Dtmf Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.11.4 DTMF Parameters The dual-tone multi-frequency (DTMF) parameters are described in the table below. Table 12-32: DTMF Parameters Parameter Description Web/EMS: DTMF Transport Determines the DTMF transport type. Type [0] DTMF Mute = Erases digits from voice stream and doesn't [DTMFTransportType] relay to remote.
Page 740: Rtp, Rtcp And T.38 Parameters
Mediant 800 MSBG 12.11.5 RTP, RTCP and T.38 Parameters The RTP, RTCP and T.38 parameters are described in the table below. Table 12-33: RTP/RTCP and T.38 Parameters Parameter Description Web: Dynamic Jitter Buffer Minimum Minimum delay (in msec) for the Dynamic Jitter Buffer.
Page 741 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [1] Enable When enabled, the device includes in the SDP message the RTP payload type "RED" and the payload type configured by the parameter RFC2198PayloadType. a=rtpmap:<PT> RED/8000 Where <PT> is the payload type as defined by RFC2198PayloadType.
Page 742 Mediant 800 MSBG Parameter Description Notes: For this parameter to take effect, a device reset is required. If the device is located in a network subnet which is connected to other gateways using a router that uses Virtual Router Redundancy Protocol (VRRP) for redundancy, then set this parameter to 0 or 2.
Page 743: G Ateway And Ip-To-Ip Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12 Gateway and IP-to-IP Parameters 12.12.1 Fax and Modem Parameters The fax and modem parameters are described in the table below. Table 12-34: Fax and Modem Parameters Parameter Description EMS: T38 Use RTP Port Defines the port (with relation to RTP port) for sending and [T38UseRTPPort] receiving T.38 packets.
Page 744 Mediant 800 MSBG Parameter Description Web/EMS: Fax CNG Mode Determines the device's behavior upon detection of a CNG tone. [FaxCNGMode] [0] = Does not send a SIP Re-INVITE upon detection of a fax CNG tone when the parameter CNGDetectorMode is set to 1 (default).
Page 745: Dtmf And Hook-Flash Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.2 DTMF and Hook-Flash Parameters The DTMF and hook-flash parameters are described in the table below. Table 12-35: DTMF and Hook-Flash Parameters Parameter Description Hook-Flash Parameters Web/EMS: Hook-Flash Code For analog interfaces: Defines the digit pattern that when [HookFlashCode] received from the Tel side, indicates a Hook Flash event.
Page 746 Mediant 800 MSBG Parameter Description Web: Min. Flash-Hook Detection Defines the minimum time (in msec) for detection of a hook- Period [msec] flash event. Detection is guaranteed for hook-flash periods of EMS: Min Flash Hook Time at least 60 msec (when setting the minimum time to 25). Hook- [MinFlashHookTime] flash signals that last a shorter period of time are ignored.
Page 747 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 143). Web/EMS: Tx DTMF Option Determines a single or several preferred transmit DTMF [TxDTMFOption] negotiation methods.
Page 748 Mediant 800 MSBG Parameter Description Web/EMS: Tx DTMF Option Table This ini file table parameter configures up to two preferred [TxDTMFOption] transmit DTMF negotiation methods. The format of this parameter is as follows: [TxDTMFOption] FORMAT TxDTMFOption_Index = TxDTMFOption_Type; [\TxDTMFOption] For example: TxDTMFOption 0 = 1;...
Page 749 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Enable Digit Delivery to Tel Enables the Digit Delivery feature, which sends DTMF digits of EMS: Enable Digit Delivery the called number to the device's port (analog)/B-channel [EnableDigitDelivery] (digital) (phone line) after the call is answered (i.e., line is off- hooked for FXS, or seized for FXO) for IP-to-Tel calls.
Page 750: Digit Collection And Dial Plan Parameters
Mediant 800 MSBG Parameter Description used for the received DTMF packets. If negotiation isn't used, this payload type is used for receive and for transmit. Determines whether to replace the number sign (#) with the [ReplaceNumberSignWithEscape escape character (%23) in outgoing SIP messages for Tel-to- Char] IP calls.
Page 751 SIP User's Manual 12. Configuration Parameters Reference Parameter Description the TelProfile parameter. For a detailed description of the Dial Plan file, see ''External Dial Plan File'' on page 415. [Tel2IPSourceNumberMapping Defines the Dial Plan index in the external Dial Plan file for the DialPlanIndex] Tel-to-IP Source Number Mapping feature.
Page 752 Mediant 800 MSBG Parameter Description Web: Max Digits in Phone Num Defines the maximum number of collected destination number EMS: Max Digits in Phone digits that can be received (i.e., dialed) from the Tel side (analog) Number when Tel-to-IP ISDN overlap dialing is performed (digital). When...
Page 753: Voice Mail Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.4 Voice Mail Parameters The voice mail parameters are described in the table below. For detailed information on the Voice Mail application, refer to the CPE Configuration Guide for Voice Mail. Table 12-37: Voice Mail Parameters Parameter Description Web/EMS: Voice Mail Interface...
Page 754 Mediant 800 MSBG Parameter Description 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 the Redirect number reason.
Page 755 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: MWI Source Number Determines the calling party's phone number used in the Q.931 EMS: MWI Source Name MWI Setup message to PSTN. If not configured, the channel's [MWISourceNumber] phone number is used as the calling number. Digit Patterns The following digit pattern parameters apply only to voice mail applications that use the DTMF communication method.
Page 756 Mediant 800 MSBG Parameter Description Web: Internal Call Digit Pattern Determines the digit pattern used by the PBX to indicate an EMS: Digit Pattern Internal Call internal call. [DigitPatternInternalCall] The valid range is a 120-character string. Web: External Call Digit Pattern...
Page 757: Supplementary Services Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.5 Supplementary Services Parameters This subsection describes the device's supplementary telephony services parameters. 12.12.5.1 Caller ID Parameters The caller ID parameters are described in the table below. Table 12-38: Caller ID Parameters Parameter Description Web: Caller ID Permissions Table EMS: SIP Endpoints >...
Page 758 Mediant 800 MSBG Parameter Description From header. The format of this parameter is as follows: [CallerDisplayInfo] FORMAT CallerDisplayInfo_Index = CallerDisplayInfo_DisplayString, CallerDisplayInfo_IsCidRestricted, CallerDisplayInfo_Module, CallerDisplayInfo_Port; [\CallerDisplayInfo] Where, DisplayString = Caller ID string (up to 18 characters). IsCidRestricted = [0] Allowed = sends the defined caller ID string when a Tel-to-IP call is made using the corresponding device port (default).
Page 759 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web/EMS: Enable Caller ID Determines whether Caller ID is enabled. [EnableCallerID] [0] Disable = Disable the Caller ID service (default). [1] Enable = Enable the Caller ID service. If the Caller ID service is enabled, then for FXS interfaces, calling number and Display text (from IP) are sent to the device's port.
Page 760 Mediant 800 MSBG 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) [1] Enable = Interworking of CPC is performed...
Page 761 SIP User's Manual 12. Configuration Parameters Reference Parameter Description generate the Caller ID according to the selected Caller ID type. Notes: This parameter is applicable only to FXS interfaces. If this parameter is set to 1 and used with distinctive ringing, the Caller ID signal doesn't change the distinctive ringing timing.
Page 762: C All Waiting Parameters
Mediant 800 MSBG Parameter Description response includes the P-Asserted-Identity with Caller ID. The device interworks (in some ISDN variants), the Connected Party number and name from Q.931 Connect message to SIP 200 OK with the P-Asserted-Identity header. In the opposite...
Page 763 SIP User's Manual 12. Configuration Parameters Reference Parameter Description The device's Call Progress Tones (CPT) file must include a Call Waiting Ringback tone (caller side) and a Call Waiting tone (called side, FXS only). The EnableHold parameter must be enabled on both the calling and the called side.
Page 764 Mediant 800 MSBG Parameter Description EnableCallWaiting. The device's CPT file must include a 'call waiting Ringback' tone (caller side) and a 'call waiting' tone (called side, FXS interfaces only). The EnableHold parameter must be enabled on both the calling and the called sides.
Page 765: C All Forwarding Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description sequence in the CPT file. SIP Alert-Info header examples: Alert-Info:<Bellcore-dr2> Alert-Info:<http://…/Bellcore-dr2> (where "dr2" defines call waiting tone #2) The SIP INFO message is according to Broadsoft's application server definition. Below is an example of such an INFO message: INFO sip:06@192.168.13.2:5060 SIP/2.0 Via:SIP/2.0/UDP...
Page 766 Mediant 800 MSBG Parameter Description [1] On Busy = Forward incoming calls when the port is busy. [2] Unconditional = Always forward incoming calls. [3] No Answer = Forward incoming calls that are not answered within the time specified in the 'Time for No Reply Forward' field.
Page 767: M Essage Waiting Indication Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: AS Subscribe IPGroupID Defines the IP Group ID that contains the Application server for [ASSubscribeIPGroupID] Subscription. The valid value range is 1 to 8. The default is -1 (i.e., not configured).
Page 768 Mediant 800 MSBG Parameter Description the TelProfile parameter). Web: Subscribe to MWI Enables subscription to an MWI server. EMS: Enable MWI Subscription [0] No = Disables MWI subscription (default). [EnableMWISubscription] [1] Yes = Enables subscription to an MWI server (defined by the parameter MWIServerIP address).
Page 769: C All Hold Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description required. EMS: Bellcore VMWI Type One Selects the Bellcore VMWI sub-standard. Standard [0] = Between rings (default). [BellcoreVMWITypeOneStandard] [1] = Not ring related. Note: For this parameter to take effect, a device reset is required.
Page 770: C All Transfer Parameters
Mediant 800 MSBG Parameter Description Web: Call Hold Reminder Defines the duration (in seconds) that the Call Hold Reminder Ring is Ring Timeout played. If a user hangs up while a call is still on hold or there is a call...
Page 771 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Transfer Prefix Defines the string that is added as a prefix to the EMS: Logical Prefix For Transferred transferred/forwarded called number when the REFER/3xx Call message is received. [xferPrefix] Notes: The number manipulation rules apply to the user part of the Refer-To and/or Contact URI before it is sent in the INVITE message.
Page 772: T Hree-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 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 773: E Mergency Call Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description more than one three-way conference may be supported (up to six). Web: Establish Defines the digit pattern, which upon detection, generates the Conference Code Conference-initiating INVITE when 3-way conferencing is enabled EMS: Establish Code (Enable3WayConference is set to 1).
Page 774: C All Cut-Through Parameters
Mediant 800 MSBG Parameter Description Web: Emergency Calls Determines the time (in minutes) that the device waits before tearing- Regret Timeout down an emergency call (defined by the parameter EMS: Emergency Regret EmergencyNumbers). Until this time expires, an emergency call can...
Page 775: Automatic Dialing Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Notes: If this parameter is disabled and the PSTN side remains in off-hook state after the IP call ends the call, the device releases the call after 60 seconds. A special CAS table can be used to report call status events (Active/Idle) to the PSTN side during Cut Through mode.
Page 776: Direct Inward Dialing Parameters
Mediant 800 MSBG Parameter Description hooked for over 10 seconds: TargetOfChannel 0 = 911, 1, 1, 1 ,10; (phone number "911" is automatically dialed for Port 1 of Module 1 after being off-hooked for 10 seconds) Notes: This is parameter is applicable to FXS and FXO interfaces.
Page 777: Mlpp Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description EMS: Enable DID This ini file table parameter enables support for Japan NTT 'Modem' DID. [EnableDID] FXS interfaces can be connected to Japan's NTT PBX using 'Modem' DID lines. These DID lines are used to deliver a called number to the PBX.
Page 778 Mediant 800 MSBG Parameter Description [2] Emergency = Preemption of IP-to-Tel E911 emergency calls. If the device receives an E911 call and there are unavailable channels to receive the call, the device terminates one of the channel calls and sends the E911 call to that channel.
Page 779 SIP User's Manual 12. Configuration Parameters Reference Parameter Description In this scenario, the character string is sent without translation to a numerical value. Web: MLPP DiffServ Defines the DiffServ value (differentiated services code EMS: Diff Serv point/DSCP) used in IP packets containing SIP messages that [MLPPDiffserv] are related to MLPP calls.
Page 780 Mediant 800 MSBG Parameter Description "000000". Note: This parameter is applicable only to the MLPP NI-2 ISDN variant with CallPriorityMode set to 1. EMS: E911 MLPP Behavior Defines the E911 (or Emergency Telecommunication Services/ETS) MLPP Preemption mode: [E911MLPPBehavior] [0] Standard Mode - ETS calls have the highest priority and preempt any MLPP call (default).
Page 781: Isdn Bri Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Immediate precedence call Immediate level. [MLPPImmediateRTPDSCP] The valid range is -1 to 63. The default is -1. Note: If set to -1, the DiffServ value is taken from the global parameter PremiumServiceClassMediaDiffServ or as defined for IP Profiles per call (using the parameter IPProfile).
Page 782 Mediant 800 MSBG Parameter Description Notes: For an explanation on each of the table's parameters and for configuring the table using the Web interface, see ''Configuring ISDN Supplementary Services'' on page 191. For an explanation on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 368.
Page 783: Pstn Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.6 PSTN Parameters This subsection describes the device's PSTN parameters. 12.12.6.1 General Parameters The general PSTN parameters are described in the table below. Table 12-51: General PSTN Parameters Parameter Description Web/EMS: Protocol Type Defines the PSTN protocol for a the Trunks.
Page 784 [54] BRI QSIG = QSIG over BRI [55] BRI FRENCH VN6 ISDN = VN6 over BRI [56] BRI NTT = BRI ISDN Japan (Nippon Telegraph) Note: For supported protocols, please contact your AudioCodes representative. [ProtocolType_x] Same as the description for the parameter ProtocolType, but for a specific trunk ID (where x denotes the Trunk ID and 0 is the first trunk).
Page 785 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [ISDNJapanNTTTimerT3JA] T3_JA timer (in seconds). This parameter overrides the internal PSTN T301 timeout on the Users Side (TE side). If an outgoing call from the device to ISDN is not answered during this timeout, the call is released.
Page 786 Mediant 800 MSBG 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 787: T Dm Bus And Clock Timing Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.6.2 TDM Bus and Clock Timing Parameters The TDM Bus parameters are described in the table below. Table 12-52: TDM Bus and Clock Timing Parameters Parameter Description TDM Bus Parameters Web/EMS: PCM Law Select Determines the type of PCM companding law in input/output TDM [PCMLawSelect] bus.
Page 788 Mediant 800 MSBG Parameter Description Web: TDM Bus Fallback Clock Selects the fallback clock source on which the device Source synchronizes in the event of a clock failure. EMS: TDM Bus Fallback Clock [4] Network (default) [TDMBusFallbackClock] [8] H.110_A [9] H.110_B...
Page 789: C As Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.6.3 CAS Parameters The Common Channel Associated (CAS) parameters are described in the table below. Note that CAS is not applicable to BRI interfaces. Table 12-53: CAS Parameters Parameter Description Web: CAS Transport Type Controls the ABCD signaling transport type over IP.
Page 790 Mediant 800 MSBG Parameter Description Web: Dial Plan The CAS Dial Plan name that is used on a specific trunk (where EMS: Dial Plan Name x denotes the trunk ID). [CASTrunkDialPlanName_x] The range is up to 11 characters. For example, the below configures E1_MFCR2 trunk with a...
Page 791 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [CASTablesNum] Indicates how many CAS protocol configurations files are loaded. The valid range is 1 to 8. Note: For this parameter to take effect, a device reset is required. CAS State Machines Parameters Note: For configuring the 'CAS State Machine' table using the Web interface, see ''Configuring CAS State Machines'' on page 99.
Page 792: I Sdn Parameters
Mediant 800 MSBG 12.12.6.4 ISDN Parameters The ISDN parameters are described in the table below. Table 12-54: ISDN Parameters Parameter Description Web: ISDN Termination Side Selects the ISDN termination side. EMS: Termination Side [0] User side = ISDN User Termination Equipment (TE)
Page 793 SIP User's Manual 12. Configuration Parameters Reference Parameter Description NFAS Parameters Note: These parameters are applicable to PRI interfaces. Web: NFAS Group Number Indicates the NFAS group number (NFAS member) for the EMS: Group Number selected trunk, where x depicts the Trunk ID. [NFASGroupNumber_x] 0 = Non-NFAS trunk (default) 1 to 12 = NFAS group number...
Page 794 Mediant 800 MSBG Parameter Description [1] = PI 1 is added to a sent ISDN Setup message - call is not end-to-end ISDN. [3] = PI 3 is added to a sent ISDN Setup message - calling equipment is not ISDN.
Page 795 SIP User's Manual 12. Configuration Parameters Reference Parameter Description ISDNInCallsBehavior = 67584 (i.e., 2048 + 65536). [ISDNInCallsBehavior_x] Same as the description for the parameter ISDNInCallsBehavior, but per trunk (i.e., where x depicts the Trunk ID). Web/EMS: Q.931 Layer Response Bit-field used to determine several behavior options that Behavior influence the behaviour of the Q.931 protocol.
Page 796 Mediant 800 MSBG 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 797 SIP User's Manual 12. Configuration Parameters Reference Parameter Description remote user has cleared the call, especially in the case of a long distance voice call. [32] CHAN ID 16 ALLOWED = Applies only to ETSI E1 lines (30B+D). Enables handling the differences between the newer QSIG standard (ETS 300-172) and other ETSI- based standards (ETS 300-102 and ETS 300-403) in the conversion of B-channel ID values into timeslot values:...
Page 798 Mediant 800 MSBG Parameter Description Note: This option is applicable only to the Korean variant. [128] DIAL WITH KEYPAD = The device uses the Keypad IE to store the called number digits instead of the CALLED_NB IE. Note: This option is applicable only to the Korean variant (Korean network).
Page 799: Isdn And Cas Interworking Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description [PSTNExtendedParams] Bit map for special PSTN behavior parameters: [0] (default) = For QSIG "Networking Extensions". This bit (bit #0) is responsible for the InvokeId size: If this bit is not set (default), then the InvokeId size is one byte.
Page 800 Mediant 800 MSBG Parameter Description [MinOverlapDigitsForRouting] IP calls. The valid value range is 0 to 49. The default is 1. Note: This parameter is applicable when the ISDNRxOverlap parameter is set to [2]. Web/EMS: ISDN Overlap IP to Tel Enables ISDN overlap dialing for IP-to-Tel calls. This feature is Dialing part of ISDN-to-SIP overlap dialing according to RFC 3578.
Page 801 SIP User's Manual 12. Configuration Parameters Reference Parameter Description a timer expires (e.g., for open numbering schemes). If a match is found (or the timer expires), the digit collection process is terminated even if Sending Complete is not received. For enabling ISDN overlap dialing for IP-to-Tel calls, use the ISDNTxOverlap parameter.
Page 802 Mediant 800 MSBG Parameter Description Notes: For this feature to function, you must set the parameter ISDNDuplicateQ931BuffMode to 128 (i.e., duplicate all messages). ISDN tunneling is applicable for all ISDN variants as well as QSIG. Web/EMS: Enable ISDN Tunneling Enables ISDN Tunneling to the Tel side.
Page 803 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Notes: This parameter is applicable to Euro ISDN variants - from TE (user) to NT (network). This parameter is applicable also to QSIG BRI. If the parameter is disabled, the device plays a Held tone to the Tel side when a SIP request with 0.0.0.0 or "inactive"...
Page 804 Mediant 800 MSBG Parameter Description [ISUBNumberOfDigits] Specifies the number of digits (from the end) that the device takes from the called number (received from the IP) for the isub number (in the sent ISDN Setup message). This feature is only applicable for IP-to-ISDN calls.
Page 805 SIP User's Manual 12. Configuration Parameters Reference Parameter Description before it reaches the Connect state; otherwise, the Disconnect message is sent immediately and no tones are played. Web: Play Ringback Tone to Trunk Enables the playing of a ringback tone (RBT) to the trunk side and per trunk (where ID depicts the trunk number and 0 is the [PlayRBTone2Trunk_ID] first trunk).
Page 806 Mediant 800 MSBG Parameter Description play an RBT. No PI is sent in the ISDN Alert message (unless the parameter ProgressIndicator2ISDN_ID is configured differently). In this case, the PBX/PSTN should play an RBT tone to the originating terminal by itself.
Page 807 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Digital Out-Of-Service Determines the method for setting digital trunks to Out-Of- Behavior Service state per device. For a description, refer to the [DigitalOOSBehavior] parameter DigitalOOSBehaviorFor Trunk_ID. Note: To configure the method for setting Out-Of-Service state per trunk, use the parameter DigitalOOSBehaviorForTrunk_ID.
Page 808 Mediant 800 MSBG Parameter Description SipResponse = SIP Response IsdnReleaseCause = Q.850 Release Cause For example: CauseMapSIP2ISDN 0 = 480,50; CauseMapSIP2ISDN 0 = 404,3; When a SIP response is received (from the IP side), the device searches this mapping table for a match. If the SIP response is found, the Q.850 Release Cause assigned to it is sent to the...
Page 809 SIP User's Manual 12. Configuration Parameters Reference Parameter Description according to the parameter PlayRBTone2Tel (default). [0] No PI = PI is not sent to ISDN. [1] PI = 1; [8] PI = 8: The PI value is sent to PSTN in Q.931/Proceeding and Alerting messages.
Page 810 Mediant 800 MSBG Parameter Description [TrunkPSTNAlertTimeout_ID] Setup message is sent to the Tel side (IP-to-Tel call establishment) and a Connect message is received. If Alerting is received, the timer is restarted. In the ini file parameter, ID depicts the trunk number, where 0 is the first trunk.
Page 811 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Selection (TNS) IE. If enabled, the CIC code (received in an INVITE Request-URI) is included in a TNS IE in the ISDN Setup message. For example: INVITE sip:555666;cic=2345@100.2.3.4 sip/2.0. Notes: This feature is supported only for SIP-to-ISDN calls. The parameter AddCicAsPrefix can be used to add the CIC as a prefix to the destination phone number for routing IP-to- Tel calls.
Page 812 Mediant 800 MSBG Parameter Description Profile ID in the 'Inbound IP Routing Table' (PSTNPrefix). When IP Profiles are used for configuring different IE data for Hunt Groups, this parameter is ignored. Web: Enable User-to-User IE for Enables ISDN PRI-to-SIP interworking.
Page 813 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web/EMS: Trunk Transfer Mode Determines the trunk transfer method (for all trunks) when a [TrunkTransferMode] SIP REFER message is received. The transfer method depends on the Trunk's PSTN protocol (configured by the parameter ProtocolType) and is applicable only when one of these protocols are used: PSTN Protocol...
Page 814 Mediant 800 MSBG Parameter Description [3] = Supports CAS Normal transfer. When a SIP REFER message is received, the device performs a Blind Transfer by executing a CAS Wink, dialing the Refer-to number to the switch, and then releasing the call.
Page 815 SIP User's Manual 12. 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 816 Mediant 800 MSBG Parameter Description Web: Enable QSIG Transfer Determines whether the device interworks QSIG Facility Update messages with callTranferComplete invoke application protocol [EnableQSIGTransferUpdate] data unit (APDU) to SIP UPDATE messages with P-Asserted- Identity and optional Privacy headers. This feature is supported for IP-to-Tel and Tel-to-IP calls.
Page 817: Answer And Disconnect Supervision Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.8 Answer and Disconnect Supervision Parameters The answer and disconnect supervision parameters are described in the table below. Table 12-56: Answer and Disconnect Parameters Parameter Description Web: Answer Supervision Enables the sending of SIP 200 OK upon detection of speech, EMS: Enable Voice Detection fax, or modem.
Page 818 Mediant 800 MSBG Parameter Description [DisconnectOnBrokenConnection] [1] Yes (default) Notes: The timeout is configured by the BrokenConnectionEventTimeout parameter. This feature is applicable only if the RTP session is used without Silence Compression. If Silence Compression is enabled, the device doesn't detect a broken RTP connection.
Page 819 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [2] Voice/Energy Detectors = N/A. [3] All = N/A. Note: For this parameter to take effect, a device reset is required. [FarEndDisconnectSilenceThresh Threshold of the packet count (in percentages) below which is old] considered silence by the device.
Page 820 Mediant 800 MSBG Parameter Description Polarity (Current) Reversal for Call Release (Analog Interfaces) Parameters Web: Enable Polarity Reversal Enables the polarity reversal feature for call release. EMS: Enable Reversal Polarity [0] Disable = Disable the polarity reversal service (default). [EnableReversalPolarity] [1] Enable = Enable the polarity reversal service.
Page 821: Tone Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description EMS: Current Disconnect Duration The duration (in msec) of the current disconnect pulse. [CurrentDisconnectDuration] The range is 200 to 1500. The default is 900. Notes: This parameter is applicable for FXS and FXO interfaces. The FXO interface detection window is 100 msec below the parameter's value and 350 msec above the parameter's value.
Page 822 Mediant 800 MSBG Parameter Description [sec] interfaces, to an ISDN terminal). For digital interfaces: This parameter is applicable for overlap [TimeForDialTone] dialing when ISDNInCallsBehavior is set to 65536. The dial tone is played if the ISDN Setup message doesn't include the called number.
Page 823 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web/EMS: Reorder Tone For Analog: The duration (in seconds) that the device plays a Duration [sec] Busy or Reorder tone duration before releasing the line. [TimeForReorderTone] The valid range is 0 to 254. The default is 0 seconds. Typically, after playing a Reorder/Busy tone for the specified duration, the device starts playing an Offhook Warning tone.
Page 824 Mediant 800 MSBG Parameter Description when a SIP 180/183 response is received. [2] Prefer IP = RBT is played to the Tel side only if a 180/183 response without SDP is received. If 180/183 with SDP message is received, the device cuts through the voice channel and doesn't play RBT (default).
Page 825 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [1] Play. Notes: For Blind transfer, the local RBT is played to first call PSTN party when the second leg receives the ISDN Alerting or Progress message. For Consulted transfer, the local RBT is played when the second leg receives ISDN Alerting or Progress message if the Progress message is received after a SIP REFER.
Page 826: T One Detection Parameters
Mediant 800 MSBG Parameter Description ToneIndex 1 = 1, 2, 20*, , 3; Notes: You can define up to 50 indices. This parameter is applicable only to FXS interfaces. Typically, the Ringing and/or Call Waiting tone played is indicated in the SIP Alert-Info header field of the received INVITE message.
Page 827 SIP User's Manual 12. Configuration Parameters Reference Parameter Description ISDNDisconnectOnBusyTone = 1 (applicable for Busy, Reorder and SIT tones) Another parameter for handling the SIT tone is SITQ850Cause, which determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when a SIT tone is detected on an IP-to-Tel call.
Page 828: M Etering Tone Parameters
Mediant 800 MSBG 12.12.9.3 Metering Tone Parameters The metering tone parameters are described in the table below. Table 12-59: Metering Tone Parameters Parameter Description Web: Generate Metering Determines the method used to configure the metering tones that are Tones generated to the Tel side.
Page 829: Telephone Keypad Sequence Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description 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; ChargeCode 3 = 0,60,1; ChargeCode 0 = 6, 3, 1, 12, 2, 1, 18, 5, 2, 0, 2, 1;...
Page 830 Mediant 800 MSBG Parameter Description destination "123", the device collects and sends all the dialed digits, including the prefix string, as "9123" to the IP destination number. Note: This parameter is applicable only to FXS interfaces. Hook Flash Parameters Web: Flash Keys Sequence Style Hook flash keys sequence style for FXS interfaces.
Page 831 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: On Busy or No Answer Keypad sequence that activates the forward on 'busy or no EMS: CF Busy Or No Answer answer' option. [KeyCFBusyOrNoAnswer] Web: Do Not Disturb Keypad sequence that activates the Do Not Disturb option EMS: CF Do Not Disturb (immediately reject incoming calls).
Page 832 Mediant 800 MSBG Parameter Description a mismatch, a reorder tone is played to the phone. Notes: This parameter is applicable to FXO and FXS interfaces (but for FXO the Web interface does not display this parameter). It is possible to configure whether the KeyBlindTransfer code is added as a prefix to the dialed destination number, by using the parameter KeyBlindTransferAddPrefix.
Page 833: General Fxo Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.12.11 General FXO Parameters The general FXO parameters are described in the table below. Table 12-61: General FXO Parameters Parameter Description Web: FXO Coefficient Type Determines the FXO line characteristics (AC and DC) according to USA EMS: Country Coefficients or TBR21 standard.
Page 834 Mediant 800 MSBG Parameter Description Notes: The correct dial tone parameters must be configured in the CPT file. The device may take 1 to 3 seconds to detect a dial tone (according to the dial tone configuration in the CPT file). If the dial tone is not detected within 6 seconds, the device releases the call and sends a SIP 500 "Server Internal Error”...
Page 835: Fxs Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Rings before Determines the number of rings before the device starts detecting Detecting Caller ID Caller ID. EMS: Rings Before Caller [0] 0 = Before first ring. [1] 1 = After first ring (default). [RingsBeforeCallerID] [2] 2 = After second ring.
Page 836: Hunt Groups, Number Manipulation And Routing Parameters
Mediant 800 MSBG 12.12.13 Hunt Groups, Number Manipulation and Routing Parameters This subsection describes the device's number manipulation and routing parameters. 12.12.13.1 Hunt Groups and Routing Parameters The routing parameters are described in the table below. Table 12-63: Routing Parameters...
Page 837 SIP User's Manual 12. Configuration Parameters Reference Parameter Description For example: TrunkGroupSettings 0 = 1, 0, 5, branch-hq, user, 1, 255; TrunkGroupSettings 1 = 2, 1, 0, localname, user1, 2, 255; Notes: This parameter can include up to 24 indices. The parameter MWIInterrogationType is not applicable.
Page 838 Mediant 800 MSBG Parameter Description cyclic ascending method (i.e., selects the channel after the last allocated channel). This option is applicable only to digital interfaces. For example, if the Hunt Group includes two physical trunks, 0 and 1: For the first incoming call, the first channel of Trunk 0 is allocated.
Page 839 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Name is received from the Tel side, the IP Display Name remains empty (default). [1] Yes = If a Tel Display Name is received, the Tel Source Number is used as the IP Source Number and the Tel Display Name is used as the IP Display Name.
Page 840 Mediant 800 MSBG Parameter Description PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort, PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID, PREFIX_DestSRD; [\PREFIX] For example: PREFIX 0 = *, domain.com, *, 0, 255, $$, -1, , 1, , -1, -1, -1; PREFIX 1 = 20, 10.33.37.77, *, 0, 255, $$, -1, , 2, , 0, -1;...
Page 841 SIP User's Manual 12. Configuration Parameters Reference Parameter Description asterisk ('*') wildcard to represent any number between 0 and 255. For example, 10.8.8.* represents all addresses between 10.8.8.0 and 10.8.8.255. If the source IP address (SourceAddress) includes an FQDN, DNS resolution is performed according to the parameter DNSQueryType.
Page 842 Mediant 800 MSBG Parameter Description first checks the 'Outbound IP Routing Table' before making a call through the Proxy. If the number is not allowed (i.e., number isn't listed in the table or a call restriction routing rule of IP address 0.0.0.0 is applied), the call is released.
Page 843: Alternative Routing Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description sip:5550001;cic=+16789@172.18.202.60:5060;user=phone SIP/2.0 Note: After the cic prefix is added, the 'Inbound IP Routing Table' can be used to route this call to a specific Hunt Group. The Destination Number IP to Tel Manipulation table must be used to remove this prefix before placing the call to the ISDN.
Page 844 Mediant 800 MSBG Parameter Description host name is not resolved (default). Notes: QoS is quantified according to delay and packet loss calculated according to previous calls. QoS statistics are reset if no new data is received within two minutes. For...
Page 845 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Reasons for Alternative Tel-to-IP Routing Table EMS: Alt Route Cause Tel to IP [AltRouteCauseTel2IP] This ini file table parameter configures SIP call failure reason values received from the IP side. If an IP call is released as a result of one of these reasons, the device attempts to locate an alternative IP route (address) for the call in the 'Outbound IP Routing Table' (if a Proxy is not used) or used as a redundant...
Page 846 Mediant 800 MSBG Parameter Description 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 Hunt Group (e.g., all...
Page 847: Number Manipulation Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description Call Forward upon Busy Trunk on page 12.12.13.3 Number Manipulation Parameters The number manipulation parameters are described in the table below. Table 12-65: Number Manipulation Parameters Parameter Description Web: Set Redirect number Defines the value of the Redirect Number screening indicator in Screening Indicator to TEL ISDN Setup messages.
Page 848 Mediant 800 MSBG Parameter Description the CopyDest2RedirectNumber parameter set to 1, to the IP- to-Tel Routing table (PSTNPrefix parameter). Even if there is no SIP Diversion or History header in the incoming INVITE message, the outgoing Q.931 Setup message will contain a redirect number.
Page 849 SIP User's Manual 12. Configuration Parameters Reference 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 Determines whether the Numbering Plan Indicator (NPI) and Calling Number Type of Numbering (TON) are added to the Calling Number for...
Page 850 Mediant 800 MSBG Parameter Description [1] = Swap calling and called numbers Note: This parameter can also be configured per Tel Profile, using the TelProfile parameter. Web/EMS: Add Prefix to Redirect Defines a string prefix that is added to the Redirect number Number received from the Tel side.
Page 851 SIP User's Manual 12. Configuration Parameters Reference Parameter Description RedirectPrefix, and SourceAddress) matches the IP-to-Tel call, then the redirect number manipulation rule (defined by the other parameters) is applied to the call. The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 852 Mediant 800 MSBG Parameter Description Web: Destination Phone Number Manipulation Table for Tel to IP Calls EMS: SIP Manipulations > Destination Telcom to IPs This ini file table parameter manipulates the destination number [NumberMapTel2IP] of Tel-to-IP calls. The format of this parameter is as follows:...
Page 853 SIP User's Manual 12. Configuration Parameters Reference Parameter Description NumberMapIp2Tel_NumberType, NumberMapIp2Tel_NumberPlan, NumberMapIp2Tel_RemoveFromLeft, NumberMapIp2Tel_RemoveFromRight, NumberMapIp2Tel_LeaveFromRight, NumberMapIp2Tel_Prefix2Add, NumberMapIp2Tel_Suffix2Add, NumberMapIp2Tel_IsPresentationRestricted; [\NumberMapIp2Tel] For example: NumberMapIp2Tel 0 = 01,034,10.13.77.8,$$,0,$$,2,$$,667,$$; NumberMapIp2Tel 1 = 10,10,1.1.1.1,255,255,3,0,5,100,$$,255; Notes: This table parameter can include up to 100 indices. The manipulation rules are done in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add.
Page 854 Mediant 800 MSBG Parameter Description Web: Source Phone Number Manipulation Table for Tel to IP Calls EMS: SIP Manipulations > Source Telcom to IP This ini file table parameter manipulates the source phone [SourceNumberMapTel2IP] number for Tel-to-IP calls. The format of this parameter is as...
Page 855 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: Source Phone Number Manipulation Table for IP to Tel Calls EMS: EMS: SIP Manipulations > Source IP to Telcom [SourceNumberMapIP2Tel] This ini file table parameter manipulates the source number for IP-to-Tel calls.
Page 856 Mediant 800 MSBG Parameter Description For the 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 857: Ldap Parameters
SIP User's Manual 12. Configuration Parameters Reference Parameter Description PhoneContext 2 = 9,1,na.e164.host.com Notes: This parameter can include up to 20 indices. Several entries with the same NPI-TON or Phone-Context are allowed. In this scenario, a Tel-to-IP call uses the first match. To configure the Phone Context table using the Web interface, see ''Mapping NPI/TON to SIP Phone-Context'' on page 160.
Page 858: S Bc Parameters
Mediant 800 MSBG Parameter Description Web: LDAP Server Max Respond Defines the time (in seconds) that the device waits for LDAP Time server responses. [LDAPServerMaxRespondTime] The valid value range is 0 to 86400. The default is 3000. [LDAPDebugMode] Determines whether to enable the LDAP task debug messages.
Page 859 SIP User's Manual 12. Configuration Parameters Reference Parameter Description then you can select the WAN VLAN on which you want to run these SIP signaling and/or media interfaces. Therefore, for each outgoing SIP packet, the device sends it on the defined outgoing WAN interface;...
Page 860 Mediant 800 MSBG Parameter Description not passed on and is rejected. If the received header’s original value is less than this parameter's value, the header’s value is decremented before being sent on. If the received header’s original value is greater than the parameter's value, the header’s value is replaced by the...
Page 861 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [SBCReferBehavior] Defines how the device handles REFER requests. [0] = Refer-To header is unchanged (default). [1] = Uses the database for Refer-To as described below. When enabled, the device handles REFERs as follows: Before passing on the REFER request, the device changes the host part to the device's IP address and adds a special prefix ("T~&R_") to the Contact user part.
Page 862 Mediant 800 MSBG Parameter Description which the device then sends to the correct destination. Notes: When this parameter is changed from 1 to 0, new 3xx Contact headers remain unchanged. However, requests with the special prefix continue using the device's database to locate the new destination.
Page 863 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [4] Both = The device provides temporary and public GRUU to users. (Currently not supported.) This parameter allows the device to act as a GRUU server for its SIP UA clients, providing them with public GRUU’s, according to RFC 5627.
Page 864 Mediant 800 MSBG Parameter Description is as follows: Identifying a No Media Anchoring call, according to configuration and the call’s properties (such as source, destination, IP Group, and SRD). Handing the identified No Media Anchoring call. You can enable No Media Anchoring per SRD, where...
Page 865 SIP User's Manual 12. Configuration Parameters Reference Parameter Description DSP allocation. SRTP to SRTP does not require DSP allocation. Note: This parameter can only be configured as an IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 143). [SBCRFC2833Behavior] Determines RFC 2833 SDP offer\answer negotiation.
Page 866 Mediant 800 MSBG Parameter Description [0] Don't Care = History-Info header is not handled. (default) [1] Add = Diversion header converted to a History-Info header. [2] Remove = History-Info header removed from the SIP dialog and the conversion to the Diversion header depends on the settings of the SBCDiversionMode parameter.
Page 867 SIP User's Manual 12. Configuration Parameters Reference Parameter Description For a detailed description of the Allowed Coders feature, see ''Coder Restrictions Control'' on page 499. Web: Allowed Audio Coders Table This ini file table parameter allows you to define up to 5 [AllowedCodersGroup0] [AllowedCodersGroup1] Allowed Coders Groups, each with up to 10 coders.
Page 868 Mediant 800 MSBG Parameter Description Web: Message Manipulations Table EMS: Message Manipulations This ini file table parameter defines manipulation rules for SIP [MessageManipulations] header messages. The format of this parameter is as follows: [ MessageManipulations] FORMAT MessageManipulations_Index = MessageManipulations_ManSetID, MessageManipulations_MessageType,...
Page 869 SIP User's Manual 12. Configuration Parameters Reference Parameter Description For example, the below configuration allows a maximum of 10 concurrent SIP INVITEs for IP Group 1: SBCAdmissionControl 1 = 0, 1, -1, 1, 0, 10, -1, 0, 0; Notes: For a detailed description of the table's individual parameters and for configuring the table using the Web interface, see ''Configuring Admission Control'' on page 195.
Page 870 Mediant 800 MSBG Parameter Description IP2IPRouting_RequestType, IP2IPRouting_DestType, IP2IPRouting_DestIPGroupID, IP2IPRouting_DestSRDID, IP2IPRouting_DestAddress, IP2IPRouting_DestPort, IP2IPRouting_DestTransportType, IP2IPRouting_AltRouteOptions; [\IP2IPRouting] For example: IP2IPRouting 1 = 1, *, *, *, *, 3, 0, -1, -1, , 0, -1, 0; Notes: This table can include up to 120 indices (where 0 is the first index).
Page 871 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: IP to IP Inbound Manipulation Table EMS: IP to IP Inbound Manipulation [IPInboundManipulation] This ini file table parameter configures the IP to IP Inbound Manipulation table. This table allows you to manipulate the SIP URI user part (source and/or destination) of the inbound SIP dialog message.
Page 872 Mediant 800 MSBG Parameter Description Web: IP to IP Outbound Manipulation Table EMS: IP to IP Outbound Manipulation This ini file table parameter configures the IP to IP Outbound [IPOutboundManipulation] Manipulation table. This table allows you to manipulate the SIP URI user part (source and/or destination) of the outbound SIP dialog message.
Page 873: S Tandalone Survivability Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.14 Standalone Survivability Parameters The Stand-alone Survivability (SAS) parameters are described in the table below. Table 12-68: SAS Parameters Parameter Description Web: Enable SAS Enables the Stand-Alone Survivability (SAS) feature. EMS: Enable [0] Disable Disabled (default) [EnableSAS] [1] Enable = SAS is enabled When enabled, the device receives the registration requests from...
Page 874 Mediant 800 MSBG Parameter Description The Record-Route header is inserted in a request by a SAS proxy to force future requests in the dialog session to be routed through the SAS agent. Each traversed proxy in the path can insert this header, causing all future dialogs in the session to pass through it as well.
Page 875 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: SAS Survivability Mode Determines the Survivability mode used by the SAS application. EMS: Survivability Mode [0] Standard = Incoming INVITE and REGISTER requests are [SASSurvivabilityMode] forwarded to the defined Proxy list of SASProxySet in Normal mode and handled by the SAS application in Emergency mode (default).
Page 876 Mediant 800 MSBG Parameter Description [SASEmergencyPrefix] Defines a prefix that is added to the Request-URI user part of the INVITE message that is sent by the device's SAS agent when in Emergency mode to the default gateway or to any other destination (using the 'IP2IP Routing' table).
Page 877 SIP User's Manual 12. Configuration Parameters Reference Parameter Description Web: SAS IP-to-IP Routing Table [IP2IPRouting] This ini file table parameter configures the IP-to-IP Routing table for SAS routing rules. The format of this parameter is as follows: [IP2IPRouting] FORMAT IP2IPRouting_Index = IP2IPRouting_SrcIPGroupID, IP2IPRouting_SrcUsernamePrefix, IP2IPRouting_SrcHost, IP2IPRouting_DestUsernamePrefix, IP2IPRouting_DestHost, IP2IPRouting_DestType, IP2IPRouting_DestIPGroupID,...
Page 878: I P Media Parameters
Mediant 800 MSBG 12.15 IP Media Parameters The IP media parameters are described in the table below. Table 12-69: IP Media Parameters Parameter Description Automatic Gain Control (AGC) Parameters Web: Enable AGC Activates the AGC mechanism. The AGC mechanism adjusts the...
Page 879 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [25] 25 = 20.00 dB/sec [26] 26 = 25.00 dB/sec [27] 27 = 30.00 dB/sec [28] 28 = 35.00 dB/sec [29] 29 = 40.00 dB/sec [30] 30 = 50.00 dB/sec [31] 31 = 70.00 dB/sec Web: AGC Redirection Determines the AGC direction.
Page 880 Mediant 800 MSBG Parameter Description Pattern Detection Parameters Note: For an overview on the pattern detector feature for TDM tunneling, see DSP Pattern Detector on page 650. Web: Enable Pattern Detector Enables or disables the activation of the Pattern Detector (PD).
Page 881: A Uxiliary And Configuration Files Parameters
SIP User's Manual 12. Configuration Parameters Reference 12.16 Auxiliary and Configuration Files Parameters This subsection describes the device's auxiliary and configuration files parameters. 12.16.1 Auxiliary/Configuration File Name Parameters The configuration files (i.e., auxiliary files) can be loaded to the device using the Web interface (see ''Loading Auxiliary Files'' on page 337).
Page 882: Automatic Update Parameters
Mediant 800 MSBG Parameter Description Web: Dial Plan The Dial Plan name (up to 11-character strings) that is used on a EMS: Dial Plan Name specific trunk (denoted by x). [CasTrunkDialPlanName_x] Web: Dial Plan File The name (and path) of the Dial Plan file (defining dial plans). This...
Page 883 SIP User's Manual 12. Configuration Parameters Reference Parameter Description [AUPDCheckIfIniChanged ] Determines whether the Automatic Update mechanism performs CRC checking to determine if the ini file has changed prior to processing. [0] = Do not check CRC. The ini file is loaded whenever the server provides it.
Page 884 Mediant 800 MSBG Parameter Description [PrtFileURL] Specifies the name of the Prerecorded Tones file and the path to the server (IP address or FQDN) on which it is located. For example: http://server_name/file, https://server_name/file. Note: The maximum length of the URL address is 99 characters.
Page 885: Sip Software Package
MIB files, and Utilities) from AudioCodes Web site at www.audiocodes.com/downloads (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 886 Mediant 800 MSBG Reader's Notes SIP User's Manual Document #: LTRT-12804...
Page 887: Technical Specifications
SIP User's Manual 14. Technical Specifications Technical Specifications The device's technical specifications are listed in the table below. Table 14-1: Technical Specifications Function Specification Interfaces PSTN Capacity Voice interfaces: 8 analog PSTN interfaces, 4 FXS and 4 The configuration is fixed and is not field upgradable or changeable Future support for up to 12 analog PSTN interfaces, 4 BRI ports and single E1/T1/J1 span module Digital Interfaces* (Optional) Single span E1/T1/ using RJ-48c connectors...
Page 888 Mediant 800 MSBG Function Specification Signaling E1/T1: Digital – PSTN Protocols PRI: ETSI/Euro ISDN, ANSI NI2, 4/5ESS, DMS 100, QSIG (basic and supplementary), Japan INS1500, VN3, VN4, VN6, Australian Telecom, New Zealand Telecom, Hong Kong Variant, Korean Variant CAS: - T1 CAS (protocol type 2) (MF-R1\DTMF) – supports various variants supplied as state machine such as E&M family,...
Page 889 SIP User's Manual 14. Technical Specifications Function Specification Translation of RTP, SRTP* Support SIP trunk with multi-ITSP (Registrations to ITSPs is invoked independently) Topology hiding Call Admission Control Call Black/White list IPsec Data Security (Optional) ESP – Tunnel mode Encryption Authentication IKE mode –...
Page 890 User's Manual Ver. 6.2 www.audiocodes.com...

AudioCodes Mediant 800 MSBG User Manual

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