Page 1 MediaPack™ MP‐11x & MP‐124 VoIP Media Gateway SIP Protocol User’s Manual Version 6.2 February 2011 Document # LTRT‐65415 ...
Page 3: Table Of Contents
SIP User's Manual Contents Table of Contents Overview ......................21 1.1 Gateway Description ....................21 1.2 MediaPack Features ....................23 1.2.1 MP-11x Hardware Features ..................23 1.2.2 MP-124 Hardware Features ..................23 1.3 SIP Overview ......................24 ...
Page 4 MediaPack Series 3.3.2.6 SIP Definitions ..................110 3.3.2.7 Coders and Profiles ................117 3.3.2.8 GW and IP to IP ..................124 3.3.2.9 SAS ....................... 161 3.4 Maintenance Tab ....................166 3.4.1 Maintenance ......................166 ...
Page 5 SIP User's Manual Contents Restoring Factory Default Settings ............... 215 6.1 Restoring Defaults using CLI ................215 6.2 Restoring Defaults using an ini File ..............216 6.3 Restoring Defaults using Hardware Reset Button ..........216 Auxiliary Configuration Files ................. 217 ...
Page 6 MediaPack Series 8.3.1.1 SAS Outbound Mode ................280 8.3.1.2 SAS Redundant Mode................282 8.3.2 SAS Routing ......................284 8.3.2.1 SAS Routing in Normal State ..............284 8.3.2.2 SAS Routing in Emergency State ............286 ...
Page 7 SIP User's Manual Contents 10.1.8 DHCP Parameters ....................343 10.1.9 NTP and Daylight Saving Time Parameters ............344 10.2 Web and Telnet Parameters ................345 10.2.1 General Parameters ....................345 10.2.2 Web Parameters ....................346 10.2.3 Telnet Parameters ....................348 ...
Page 8 MediaPack Series 10.12.7.3 M etering Tone Parameters ..............465 10.12.8 Telephone Keypad Sequence Parameters............467 10.12.9 General FXO Parameters ..................471 10.12.10 FXS Parameters ....................473 10.12.11 Hunt Groups, Number Manipulation and Routing Parameters ......474 ...
Page 9 SIP User's Manual Contents List of Figures Figure 1-1: Typical MediaPack VoIP Application ................... 22 Figure 3-1: Login Screen ........................29 Figure 3-2: Main Areas of the Web Interface GUI .................. 30 Figure 3-3: "Reset" Displayed on Toolbar ....................31 ...
Page 10 MediaPack Series Figure 3-56: Firewall Settings Page ....................... 90 Figure 3-57: 8021x Settings Page ......................93 Figure 3-58: General Security Settings Page ..................93 Figure 3-59: IP Security Proposals Table ....................94 Figure 3-60: IP Security Associations Table Page ................. 95 ...
Page 11 SIP User's Manual Contents Figure 3-115: IP Interface Status Page ....................184 Figure 3-116: Basic Statistics Page .....................184 Figure 3-117: Calls Count Page ......................185 Figure 3-118: SAS/SBC Registered Users Page .................187 Figure 3-119: Call Routing Status Page ....................188 ...
Page 12 MediaPack Series Figure 8-37: Enabling the SAS Application ..................287 Figure 8-38: Configuring Common Settings ..................289 Figure 8-39: Defining UAs' Proxy Server....................290 Figure 8-40: Enabling Proxy Server for Gateway Application ..............292 Figure 8-41: Defining Proxy Server for Gateway Application ...............292 ...
Page 13 SIP User's Manual Contents List of Tables Table 1-1: Supported MediaPack Series Configurations ............... 21 Table 3-1: Description of Toolbar Buttons ....................31 Table 3-2: ini File Parameter for Welcome Login Message ..............50 Table 3-3: Description of the Areas of the Home Page ................54 ...
Page 14 MediaPack Series Table 10-1: Ethernet Parameters ......................333 Table 10-2: IP Network Interfaces and VLAN Parameters ..............334 Table 10-3: Static Routing Parameters ....................337 Table 10-4: QoS Parameters .......................338 Table 10-5: NAT and STUN Parameters ....................339 Table 10-6: NFS Parameters .......................341 ...
Page 15 SIP User's Manual Contents Table 10-60: Alternative Routing Parameters ..................480 Table 10-61: Number Manipulation Parameters ..................484 Table 10-62: SAS Parameters ......................492 Table 10-63: Auxiliary and Configuration File Parameters ..............496 Table 10-64: Automatic Update of Software and Configuration Files Parameters ......497 ...
Page 16 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 17: February
SIP User's Manual Notices Notice This document describes the AudioCodes MediaPack series MP-11x and MP-124 Voice over IP (VoIP) gateways. Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions.
Page 18: Related Documentation
MediaPack Series Related Documentation Manual Name SIP CPE Release Notes Product Reference Manual for SIP CPE Devices MP-11x & MP-124 SIP Installation Manual MP-11x SIP Fast Track Guide MP-124 AC SIP Fast Track Guide MP-124 DC SIP Fast Track Guide CPE Configuration Guide for IP Voice Mail Warning: The device is supplied as a sealed unit and must only be serviced by...
Page 19 SIP User's Manual Notices Notes: • FXO (Foreign Exchange Office) is the interface replacing the analog telephone and connects to a Public Switched Telephone Network (PSTN) line from the Central Office (CO) or to a Private Branch Exchange (PBX). The FXO is designed to receive line voltage and ringing current, supplied from the CO or the PBX (just like an analog telephone).
Page 20 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 21: Overview
SIP User's Manual 1. Overview Overview This manual provides you with information for configuring and operating the VoIP analog MediaPack series devices listed in the table below: Table 1-1: Supported MediaPack Series Configurations Combined Number of Product Name FXS/FXO Channels MP-124 MP-118 4 + 4...
Page 22: Figure 1-1: Typical Mediapack Voip Application
MediaPack Series The figure below illustrates a typical MediaPack VoIP application. Figure 1-1: Typical MediaPack VoIP Application SIP User's Manual Document #: LTRT-65415...
Page 23: Mediapack Features
SIP User's Manual 1. Overview MediaPack Features This section provides a high-level overview of some of the many device supported features. For more updated information on the device's supported features, refer to the latest MP-11x & MP-124 SIP Release Notes. 1.2.1 MP-11x Hardware Features The MP-11x series hardware features includes the following:...
Page 24: Sip Overview
MediaPack Series SIP Overview Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol used on the gateway for creating, modifying, and terminating sessions with one or more participants. These sessions can include Internet telephone calls, media announcements, and conferences. SIP invitations are used to create sessions and carry session descriptions that enable participants to agree on a set of compatible media types.
Page 25: Configuration Tools
Note: To initialize the device by assigning it an IP address, a firmware file (cmp), and a configuration file (ini file), you can use AudioCodes' BootP/TFTP utility, which accesses the device using the device's MAC address (refer to the Product Reference Manual).
Page 26 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 27: Web-Based Management
SIP User's Manual 3. Web-Based Management Web-Based Management The device's Embedded Web Server (Web interface) provides FCAPS (fault management, configuration, accounting, performance, and security) functionality. The Web interface allows you to remotely configure your device for quick-and-easy deployment, including uploading of software (*.cmp), configuration (*.ini), and auxiliary files, and resetting the device.
Page 28: Getting Acquainted With The Web Interface
MediaPack Series 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). A network connection to the device's Web interface.
Page 29: 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 30: Areas Of The Gui
MediaPack Series 3.1.3 Areas of the GUI The figure below displays the general layout of the Graphical User Interface (GUI) of the Web interface: Figure 3-2: Main Areas of the Web Interface GUI The Web GUI is composed of the following main areas: Title bar: Displays the corporate logo and product name.
Page 31: Toolbar
SIP User's Manual 3. Web-Based Management 3.1.4 Toolbar The toolbar provides command buttons for quick-and-easy access to frequently required commands, as described in the table below: Table 3-1: Description of Toolbar Buttons Icon Button Description Name Submit Applies parameter settings to the device (see ''Saving Configuration'' on page 169).
Page 32: Navigation Tree
MediaPack Series 3.1.5 Navigation Tree The Navigation tree, located in the Navigation pane, displays the menus (pertaining to the menu tab selected on the Navigation bar) used for accessing the configuration pages. The Navigation tree displays a tree-like structure of menus. You can easily drill-down to the required page item level to open its corresponding page in the Work pane.
Page 33: Displaying Navigation Tree In Basic And Full View
SIP User's Manual 3. Web-Based Management 3.1.5.1 Displaying Navigation Tree in Basic and Full View You can view an expanded or reduced Navigation tree display regarding the number of listed menus and submenus. This is relevant when using the configuration tabs (Configuration, Maintenance, and Status &...
Page 34: Showing / Hiding The Navigation Pane
MediaPack Series 3.1.5.2 Showing / Hiding the Navigation Pane The Navigation pane can be hidden to provide more space for elements displayed in the Work pane. This is especially useful when the Work pane displays a page with a table that's wider than the Work pane and to view the all the columns, you need to use scroll bars.
Page 35: Working With Configuration
SIP User's Manual 3. Web-Based Management 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. 3.1.6.1 Accessing Pages The configuration pages are accessed by clicking the required page item in the Navigation...
Page 36: Figure 3-7: Toggling Between Basic And Advanced View
MediaPack Series 3.1.6.2.1 Displaying Basic and Advanced Parameters Some pages provide you with an Advanced Parameter List / Basic Parameter List toggle button that allows you to show or hide advanced parameters (in addition to displaying the basic parameters). This button is located on the top-right corner of the page and has two states: Advanced Parameter List button with down-pointing arrow: click this button to display all parameters.
Page 37: 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 38: Entering Phone Numbers
MediaPack Series 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 39: 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 40: Searching For Configuration Parameters
MediaPack Series To organize the index entries in ascending, consecutive order: Click Compact; the index entries are organized in ascending, consecutive order, starting from index 0. For example, if you added three index entries 0, 4, and 6, then the index entry 4 is re-assigned index number 1 and the index entry 6 is re-assigned index number 2.
Page 41: Working With Scenarios
SIP User's Manual 3. Web-Based Management • ini file parameter name • Link (in green) to its location (page) in the Web interface • Brief description of the parameter In the searched list, click the required parameter (link in green) to open the page in which the parameter appears;...
Page 42: Creating A Scenario
MediaPack Series 3.1.8.1 Creating a Scenario The Web interface allows you to create one Scenario with up to 20 configuration pages, as described in the procedure below: To create a Scenario: On the Navigation bar, click the Scenarios tab; a message box appears, requesting you to confirm creation of a Scenario: Figure 3-14: Scenario Creation Confirm Message Box Note: If a Scenario already exists, the Scenario Loading message box appears.
Page 43: Figure 3-15: Creating A Scenario
SIP User's Manual 3. Web-Based Management Click the Next button located at the bottom of the page; the Step is added to the Scenario and appears in the Scenario Step list: Figure 3-15: Creating a Scenario Repeat steps 5 through 8 to add additional Steps (i.e., pages). When you have added all the required Steps for your Scenario, click the Save &...
Page 44: Accessing A Scenario
MediaPack Series 3.1.8.2 Accessing a Scenario Once you have created the Scenario, you can access it at anytime by following the procedure below: To access the Scenario: On the Navigation bar, select the Scenario tab; a message box appears, requesting you to confirm the loading of the Scenario.
Page 45: Editing A Scenario
SIP User's Manual 3. Web-Based Management To navigate between Scenario Steps, you can perform one of the following: In the Navigation tree, click the required Scenario Step. In an opened Scenario Step (i.e., page appears in the Work pane), use the following navigation buttons: •...
Page 46 MediaPack Series • Add or Remove Parameters: In the Navigation tree, select the required Step; the corresponding page opens in the Work pane. To add parameters, select the check boxes corresponding to the desired parameters; to remove parameters, clear the check boxes corresponding to the parameters that you want removed.
Page 47: Saving A Scenario To A Pc
SIP User's Manual 3. Web-Based Management 3.1.8.4 Saving a Scenario to a PC You can save a Scenario to a PC (as a dat file). This is especially useful when requiring more than one Scenario to represent different environment setups (e.g., where one includes PBX interoperability and another not).
Page 48: Loading A Scenario To The Device
MediaPack Series 3.1.8.5 Loading a Scenario to the Device Instead of creating a Scenario, you can load a Scenario file (data file) from your PC to the device. To load a Scenario to the device: On the Navigation bar, click the Scenarios tab; the Scenario appears in the Navigation tree.
Page 49: Exiting Scenario Mode
SIP User's Manual 3. Web-Based Management Click the Delete Scenario File button; a message box appears requesting confirmation for deletion. Figure 3-20: Message Box for Confirming Scenario Deletion Click OK; the Scenario is deleted and the Scenario mode closes. Note: You can also delete a Scenario using the following alternative methods: •...
Page 50: Creating A Login Welcome Message
MediaPack Series 3.1.9 Creating a Login Welcome Message You can create a Welcome message box (alert message) that appears after each successful login to the device's Web interface. The ini file table parameter WelcomeMessage allows you to create the Welcome message. Up to 20 lines of character strings can be defined for the message.
Page 51: 3.1.10 Getting Help
SIP User's Manual 3. Web-Based Management 3.1.10 Getting Help The Web interface provides you with context-sensitive Online Help. The Online Help provides 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 52: 3.1.11 Logging Off The Web Interface
MediaPack Series 3.1.11 Logging Off the Web Interface You can log off the Web interface and re-access it with a different user account. For detailed information on the Web User Accounts, see User Accounts. To log off the Web interface: On the toolbar, click the Log Off button;...
Page 53: 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 54: Table 3-3: Description Of The Areas Of The Home Page
MediaPack Series The table below describes the areas of the 'Home' page. Table 3-3: Description of the Areas of the Home Page Label Description Alarms Displays the highest severity of an active alarm raised (if any) by the device: Green = no alarms Red = Critical alarm Orange = Major alarm Yellow = Minor alarm...
Page 55: Assigning A Port Name
SIP User's Manual 3. Web-Based Management 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 56: Viewing Analog Port Information
MediaPack Series 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 3-31: Shortcut Menu (e.g.
Page 57: 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 57) VoIP (see "VoIP Settings" on page 78) 3.3.1 System Settings The System menu includes the following:...
Page 58: Configuring Application Settings
MediaPack Series 3.3.1.1 Configuring Application Settings The 'Application Settings' page is used for configuring various application parameters such as Network Time Protocol (NTP), daylight saving time, and Network File System (NFS). For a description of these parameters, see ''Configuration Parameters Reference'' on page 333.
Page 59: Configuring Nfs Settings
SIP User's Manual 3. Web-Based Management 3.3.1.2 Configuring NFS Settings Network File System (NFS) enables the device to access a remote server's shared files and directories, and to handle them as if they're located locally. You can configure up to 16 different NFS file systems.
Page 60: Table 3-4: Nfs Settings Parameters
MediaPack Series Table 3-4: 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 61: Configuring Syslog Settings
SIP User's Manual 3. Web-Based Management 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 350. For viewing Syslog messages in the Web interface, see Viewing Syslog Messages on page 180.
Page 62: Configuring Regional Settings
MediaPack Series 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 63: Figure 3-37: Certificates Signing Request Page
SIP User's Manual 3. Web-Based Management To replace the device's self-signed certificate: Your network administrator should allocate a unique DNS name for the device (e.g., dns_name.corp.customer.com). This DNS name is used to access the device and therefore, must be listed in the server certificate. If the device is operating in HTTPS mode, then set the HTTPSOnly parameter to 'HTTP and HTTPS' (0) - see ''Configuring Web Security Settings'' on page 69.
Page 64: Figure 3-38: Ike Table Listing Loaded Certificate Files
MediaPack Series When the certificate has successfully loaded, save the configuration (see ''Saving Configuration'' on page 169) and restart the device; the Web interface uses the provided certificate. If the device was originally operating in HTTPS mode and you disabled it in Step 2, then return it to HTTPS by setting the parameter 'Secured Web Connection (HTTPS)' to 'HTTPS Only' (1) - see ''Configuring Web Security Settings'' on page 69.
Page 65: Client Certificates
SIP User's Manual 3. Web-Based Management 3.3.1.5.2 Client Certificates By default, Web servers using SSL provide one-way authentication. The client is certain that the information provided by the Web server is authentic. When an organizational PKI is used, two-way authentication may be desired: both client and server should be authenticated using X.509 certificates.
Page 66: Management Settings
MediaPack Series 3.3.1.5.3 Self-Signed Certificates The device is shipped with an operational, self-signed server certificate. The subject name for this default certificate is 'ACL_nnnnnnn', where nnnnnnn denotes the serial number of the device. However, this subject name may not be appropriate for production and can be changed while still using self-signed certificates.
Page 67: Table 3-5: Web User Accounts Access Levels And Privileges
SIP User's Manual 3. Web-Based Management Table 3-5: Web User Accounts Access Levels and Privileges Numeric Access Level Privileges Representation* Security Administrator Read / write privileges for all pages. read / write privileges for all pages except Administrator security-related pages, which are read-only. No access to security-related and file-loading pages;...
Page 68: Figure 3-39: Web User Accounts Page (For Users With 'Security Administrator' Privileges)
MediaPack Series To change the Web user accounts attributes: Open the 'Web User Accounts' page (Configuration tab > System menu > Web User Accounts). Figure 3-39: WEB User Accounts Page (for Users with 'Security Administrator' Privileges) Note: If you are logged into the Web interface as the Security Administrator, both Web user accounts are displayed on the 'Web User Accounts' page (as shown above).
Page 69: Figure 3-40: Web Security Settings Page
SIP User's Manual 3. Web-Based Management Click Change Password; if you are currently logged into the Web interface with this account, the 'Enter Network Password' dialog box appears, requesting you to enter the new password. Notes: • For security, it's recommended that you change the default user name and password.
Page 70: Figure 3-41: Telnet/Ssh Settings Page
MediaPack Series 3.3.1.6.3 Configuring Telnet and SSH Settings The 'Telnet/SSH Settings' page is used to define Telnet and Secure Shell (SSH). For a description of these parameters, see ''Web and Telnet Parameters'' on page 345. To define Telnet and SSH: Open the 'Telnet/SSH Settings' page (Configuration tab >...
Page 71: Figure 3-43: Web & Telnet Access List Table
SIP User's Manual 3. Web-Based Management To add an authorized IP address, in the 'Add an authorized IP address' field, enter the required IP address, and then click Add New Entry; the IP address you entered is added as a new entry to the 'Web & Telnet Access List' table. Figure 3-43: Web &...
Page 72: Figure 3-44: Radius Parameters Page
MediaPack Series 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 333. To configure RADIUS: Open the ‘RADIUS Settings' page (Configuration tab >...
Page 73: Figure 3-45: Snmp Community String Page
SIP User's Manual 3. Web-Based Management 3.3.1.6.6 SNMP Settings The SNMP submenu includes the following items: SNMP Community Settings (see ''Configuring SNMP Community Strings'' on page 73) SNMP Trap Destinations (see ''Configuring SNMP Trap Destinations'' on page 74) SNMP Trusted Managers (see ''Configuring SNMP Trusted Managers'' on page 75) SNMP V3 Users (see ''Configuring SNMP V3 Users'' on page 76) 3.3.1.6.6.1 Configuring SNMP Community Strings The 'SNMP Community String' page allows you to configure up to five read-only and up to...
Page 74: Figure 3-46: Snmp Trap Destinations Page
MediaPack Series Table 3-7: SNMP Community String Parameters Description Parameter Description Read Only [SNMPReadOnlyCommunityString_x]: Up to five Community String read-only community strings (up to 19 characters each). The default string is 'public'. Read / Write [SNMPReadWriteCommunityString_x]: Up to five read / write community strings (up to 19 characters each). The default string is 'private'.
Page 75: Figure 3-47: Snmp Trusted Managers
SIP User's Manual 3. Web-Based Management Parameter Description IP Address IP address of the remote host used as an SNMP Manager. [SNMPManagerTableIP_x] The device sends SNMP traps to these IP addresses. Enter the IP address in dotted-decimal notation, e.g., 108.10.1.255. Trap Port Defines the port number of the remote SNMP Manager.
Page 76: Figure 3-48: Snmp V3 Setting Page
MediaPack Series 3.3.1.6.6.4 Configuring SNMP V3 Users The 'SNMP v3 Users' page allows you to configure authentication and privacy for up to 10 SNMP v3 users. To configure the SNMP v3 users: Open the 'SNMP v3 Users' page (Maintenance tab > System menu > Management submenu >...
Page 77 SIP User's Manual 3. Web-Based Management Parameter Description Privacy Key Privacy key. Keys can be entered in the form of a text password or [SNMPUsers_PrivKey] long hex string. Keys are always persisted as long hex strings and keys are localized. Group The group with which the SNMP v3 user is associated.
Page 78: Voip Settings
MediaPack Series 3.3.2 VoIP Settings The VoIP menu includes the following main submenus: Network (see ''Network'' on page 78) Security (see ''Security'' on page 89) Media (see ''Media'' on page 98) Applications Enabling (see "Enabling Applications" on page 102) Control Network (see ''Control Network'' on page 103) SIP Definitions (see ''SIP Definitions'' on page 110) Coders And Profiles (see ''Coders and Profiles'' on page 117) GW and IP to IP (see ''GW and IP to IP'' on page 124)
Page 79: Figure 3-49: Ip Settings Page
SIP User's Manual 3. Web-Based Management Notes: • For a detailed description and examples of network interfaces configuration, see ''Network Configuration'' on page 316. • When adding more than one interface, ensure that you enable VLANs using the 'VLAN Mode' (VlANMode) parameter. •...
Page 80: Figure 3-50: Confirmation Message For Accessing The Multiple Interface Table
MediaPack Series Under the 'Multiple Interface Settings' group, click the Multiple Interface Table button; a confirmation message box appears: Figure 3-50: Confirmation Message for Accessing the Multiple Interface Table Click OK to confirm; the 'Multiple Interface Table' page appears: In the 'Add Index' field, enter the desired index number for the new interface, and then click Add Index;...
Page 81 SIP User's Manual 3. Web-Based Management Parameter Description [5] Media + Control = Only Media and Call Control applications are allowed on the interface. [6] OAMP + Media + Control = All application types are allowed on the interface. Notes: A single OAMP interface (and only one) must be configured.
Page 82 MediaPack Series Parameter Description way (e.g., defining two interfaces with 10.0.0.1/8 and 10.50.10.1/24 is invalid). Each interface must have its own address space. Web/EMS: Gateway Defines the IP address of the default gateway for this [InterfaceTable_Gateway] interface. Notes: A default gateway can be defined for each interface. The default gateway's IP address must be in the same subnet as the interface address.
Page 83: Figure 3-51: Ip Routing Table Page
SIP User's Manual 3. Web-Based Management To configure static IP routing: Open the 'IP Routing Table' page (Configuration tab > VoIP menu > Network submenu > IP Routing Table). Figure 3-51: IP Routing Table Page In the 'Add a new table entry' table, add a new static routing rule according to the parameters described in the table below.
Page 84 MediaPack Series Parameter Description Gateway IP Address The IP address of the router (next hop) to which the packets [StaticRouteTable_Gateway] are sent if their destination matches the rules in the adjacent columns. Note: The Gateway address must be in the same subnet as the IP address of the interface over which you configure this static routing rule.
Page 85: Figure 3-52: Qos Settings Page
SIP User's Manual 3. Web-Based Management To configure QoS: Open the 'QoS Settings' page (Configuration tab > VoIP menu > Network submenu > QoS Settings). Figure 3-52: QoS Settings Page Configure the QoS parameters as required. Click the Submit button to save your changes. Save the changes to flash memory (see ''Saving Configuration'' on page 169).
Page 86: Figure 3-53: Dns Settings Page
MediaPack Series 3.3.2.1.4 DNS The DNS submenu includes the following items: DNS Settings (refer to ''Configuring DNS Settings'' on page 86) Internal DNS Table (refer to ''Configuring the Internal DNS Table'' on page 87) Internal SRV Table (refer to ''Configuring the Internal SRV Table'' on page 88) 3.3.2.1.4.1 Configuring DNS Settings The 'DNS Settings' page defines the VoIP Domain Name System (DNS) server IP addresses.
Page 87: Figure 3-54: Internal Dns Table Page
SIP User's Manual 3. Web-Based Management 3.3.2.1.4.2 Configuring the Internal DNS Table The 'Internal DNS Table' page, similar to a DNS resolution translates up to 20 host (domain) names into IP addresses (e.g., when using the 'Tel to IP Routing' for Tel-to-IP call routing).
Page 88: Figure 3-55: Internal Srv Table Page
MediaPack Series 3.3.2.1.4.3 Configuring the Internal SRV Table The 'Internal SRV Table' page resolves host names to DNS A-Records. Three different A- Records can be assigned to each host name. Each A-Record contains the host name, priority, weight, and port. Notes: •...
Page 89: Security
SIP User's Manual 3. Web-Based Management 3.3.2.2 Security The Security Settings submenu allows you to configure various security settings. This menu contains the following page items: Firewall Settings (see ''Configuring Firewall Settings'' on page 89) 802.1x Settings (see "Configuring 802.1x Settings" on page 92) General Security Settings (see ''Configuring General Security Settings'' on page 93) IPSec Proposal Table (see "Configuring IP Security Associations Table"...
Page 90: Figure 3-56: Firewall Settings Page
MediaPack Series To add firewall rules: Open the 'Firewall Settings' page (Configuration tab > VoIP menu > Security submenu > Firewall Settings). Figure 3-56: Firewall Settings Page In the 'Add' field, enter the index of the access rule that you want to add, and then click Add;...
Page 91: Table 3-12: Internal Firewall Parameters
SIP User's Manual 3. Web-Based Management To delete a rule: Select the radio button of the entry you want to activate. Click the Delete Rule button; the rule is deleted. To save the changes to flash memory, see ''Saving Configuration'' on page 169. Table 3-12: Internal Firewall Parameters Parameter Description...
Page 92 MediaPack Series Parameter Description Packet Size Maximum allowed packet size. [AccessList_Packet_Size] The valid range is 0 to 65535. Note: When filtering fragmented IP packets, this field relates to the overall (re-assembled) packet size, and not to the size of each fragment. Byte Rate Expected traffic rate (bytes per second).
Page 93: Figure 3-57: 8021X Settings Page
SIP User's Manual 3. Web-Based Management To configure the 802.1x parameters: Open the '802.1x Settings' page (Configuration tab > VoIP menu > Security submenu > 802.1x Settings). Figure 3-57: 8021x Settings Page 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 169.
Page 94: Figure 3-59: Ip Security Proposals Table
MediaPack Series 3.3.2.2.4 Configuring IP Security Proposal Table The 'IP Security Proposals Table' page is used to configure Internet Key Exchange (IKE) with up to four proposal settings. Each proposal defines an encryption algorithm, an authentication algorithm, and a Diffie-Hellman group identifier. The same set of proposals applies to both Main mode and Quick mode.
Page 95: Figure 3-60: Ip Security Associations Table Page
SIP User's Manual 3. Web-Based Management If no proposals are defined, the default settings (shown in the following table) are applied. Table 3-14: Default IPSec/IKE Proposals Proposal Encryption Authentication DH Group Proposal 0 3DES SHA1 Group 2 (1024 bit) Proposal 1 3DES Group 2 (1024 bit) Proposal 2...
Page 96: Table 3-15: Ip Security Associations Table Configuration Parameters
MediaPack Series Table 3-15: IP Security Associations Table Configuration Parameters Parameter Name Description Operational Mode Defines the IPSec mode of operation. [IPsecSATable_IPsecMode] [0] Transport (default) [1] Tunnel Remote Endpoint Addr Defines the IP address or DNS host name of the peer. [IPsecSATable_RemoteEndpointAddres Note: This parameter is applicable only if the sOrName]...
Page 97 SIP User's Manual 3. Web-Based Management Parameter Name Description Note: Main mode negotiation is a processor-intensive operation; for best performance, do not set this parameter to less than 28,800 (i.e., eight hours). The default value is 0 (i.e., unlimited). IPSec SA Lifetime (sec) Determines the duration (in seconds) for which the [IPsecSATable_Phase2SaLifetimeInSec] negotiated IPSec SA (Quick mode) is valid.
Page 98: Media
MediaPack Series 3.3.2.3 Media The Media submenu allows you to configure the device's channel parameters and contains the following items: Voice Settings (see ''Configuring Voice Settings'' on page 98) Fax/Modem/CID Settings (see "Configuring Fax/Modem/CID Settings" on page 99) RTP/RTCP Settings (see ''Configuring RTP/RTCP Settings'' on page 100) General Media Settings (see ''Configuring General Media Settings'' on page 101) Analog Settings (see "Configuring Analog Settings"...
Page 99: Figure 3-62: Fax/Modem/Cid Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.3.2 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 333. To configure the fax, modem, and CID parameters: Open the 'Fax/Modem/CID Settings' page (Configuration tab >...
Page 100: Figure 3-63: Rtp/Rtcp Settings Page
MediaPack Series 3.3.2.3.3 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 333.
Page 101: Figure 3-64: General Media Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.3.4 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 333. To configure general media parameters: Open the 'General Media Settings' page (Configuration tab >...
Page 102: Applications Enabling
MediaPack Series 3.3.2.3.6 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 333. To configure media security: Open the 'Media Security' page (Configuration tab > VoIP menu > Media submenu > Media Security).
Page 103: Control Network
SIP User's Manual 3. Web-Based Management To enable an application: Open the 'Applications Enabling' page (Configuration tab > VoIP menu > Applications Enabling submenu > Applications Enabling). Figure 3-67: Applications Enabling Page Save the changes to the device's flash memory and then reset the device (see ''Saving Configuration'' on page 169).
Page 104: Figure 3-68: Ip Group Table Page
MediaPack Series To configure IP Groups: Open the 'IP Group Table' page (Configuration tab > VoIP menu > Control Network submenu > IP Group Table). Figure 3-68: IP Group Table Page Configure the IP group parameters according to the table below. Click the Submit button to save your changes.
Page 105 SIP User's Manual 3. Web-Based Management Parameter Description Contact User Defines the user part for the From, To, and Contact headers of [IPGroup_ContactUser] SIP REGISTER messages, and the user part for the Contact header of INVITE messages that are received from the IP Group and forwarded by the device to another IP Group.
Page 106: Figure 3-69: Proxy Sets Table Page
MediaPack Series 3.3.2.5.2 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 10 Proxy Sets, each with a unique ID number and up to five Proxy server addresses.
Page 107: Table 3-17: Proxy Sets Table Parameters
SIP User's Manual 3. Web-Based Management Table 3-17: Proxy Sets Table Parameters Parameter Description Web: Proxy Set ID The Proxy Set identification number. EMS: Index The valid range is 0 to 9. The Proxy Set ID 0 is used as the default [ProxySet_Index] Proxy Set.
Page 108 MediaPack Series Parameter Description ProxyDNSQueryType is set to 1 or 2. Transport Type The transport type per Proxy server. [ProxyIp_TransportType] [0] UDP [1] TCP [2] TLS [-1] = Undefined Note: If no transport type is selected, the value of the global parameter SIPTransportType is used (see ''Configuring SIP General Parameters'' on page 110).
Page 109 SIP User's Manual 3. Web-Based Management Parameter Description and ProxyKeepAliveTime) tags each entry as 'offline' or 'online'. Load balancing is only performed on Proxy servers that are tagged as 'online'. All outgoing messages are equally distributed across the list of IP addresses.
Page 110: Sip Definitions
MediaPack Series 3.3.2.6 SIP Definitions The SIP Definitions submenu allows you to configure various SIP call control settings. This menu contains the following page items: General Parameters (see ''Configuring SIP General Parameters'' on page 110) Advanced Parameters (see ''Configuring Advanced Parameters'' on page 112) Account Table (see "Configuring Account Table"...
Page 111: Figure 3-70: 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 3-70: SIP General Parameters Page Configure the parameters as required. Click the Submit button to save your changes.
Page 112: Figure 3-71: Advanced Parameters Page
MediaPack Series 3.3.2.6.2 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 333. To configure advanced general protocol parameters: Open the 'Advanced Parameters' page (Configuration tab >...
Page 113: Figure 3-72: Account Table Page
SIP User's Manual 3. Web-Based Management 3.3.2.6.3 Configuring Account Table The 'Account Table' page allows you to define up to 10 Accounts per Hunt Group (Served Hunt Group) for registration and/or digest authentication (user name and password) to a destination IP address (Serving IP Group). The Account table can be used, for example, to register to an Internet Telephony Service Provider (ITSP) on behalf of an IP-PBX to which the device is connected.
Page 114: Table 3-18: Account Table Parameters Description
MediaPack Series Table 3-18: Account Table Parameters Description Parameter Description Served Trunk Group The Hunt Group ID for which you want to register and/or [Account_ServedTrunkGroup] authenticate to a destination IP Group (i.e., Serving IP Group). For Tel-to-IP calls, the Served Hunt Group is the source Hunt Group from where the call originated.
Page 115 SIP User's Manual 3. Web-Based Management Parameter Description Register Enables registration. [Account_Register] [0] No = Don't register [1] Yes = Enables registration When enabled, the device sends REGISTER requests to the Serving IP Group. In addition, to activate registration, you also need to set the parameter 'Registration Mode' to 'Per Account' in the 'Hunt Group Settings' table for the specific Hunt Group.
Page 116: Figure 3-73: Proxy & Registration Page
MediaPack Series To configure the Proxy and registration parameters: Open the 'Proxy & Registration' page (Configuration tab > VoIP menu > SIP Definitions submenu > Proxy & Registration). Figure 3-73: Proxy & Registration Page Configure the parameters as required. Click the Submit button to save your changes. Click the Register or Un-Register...
Page 117: Coders And Profiles
SIP User's Manual 3. Web-Based Management 3.3.2.6.5 Configuring Accounting Settings The 'RADIUS Parameters' page allows you to configure the RADIUS parameters. For a description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 333. To configure the RADIUS parameters: Open the 'RADIUS Parameters' page (Configuration tab >...
Page 118: Figure 3-75: Coders Page
MediaPack Series In addition, you can associate different Profiles per the channels. Notes: • The default values of the parameters in the 'Tel Profile Settings' and 'IP Profile Settings' pages are identical to their default values in their respective primary configuration page ("global" parameter). •...
Page 119 SIP User's Manual 3. Web-Based Management In the 'Payload Type' field, if the payload type (i.e., format of the RTP payload) for the selected coder is dynamic, enter a value from 0 to 120 (payload types of 'well-known' coders cannot be modified). From the 'Silence Suppression' drop-down list, enable or disable the silence suppression option for the selected coder.
Page 120: Figure 3-76: Coder Group Settings Page
MediaPack Series To configure Coder Groups: Open the 'Coder Group Settings' page (Configuration tab > VoIP menu > Coders And Profiles submenu > Coders Group Settings). Figure 3-76: Coder Group Settings Page From the 'Coder Group ID' drop-down list, select a Coder Group ID. From the 'Coder Name' drop-down list, select the first coder for the Coder Group.
Page 121: Figure 3-77: Tel Profile Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.7.3 Configuring Tel Profile The 'Tel Profile Settings' page allows you to define up to nine Tel Profiles. You can assign these Tel Profiles to the device's channels in the <Endpoint Phone Number Table (see Configuring Endpoint Phone Numbers on page 124), and thereby, apply different behaviors to different channels.
Page 122 MediaPack Series From the 'Profile ID' drop-down list, select the Tel Profile identification number you want to configure. In the 'Profile Name' field, enter an arbitrary name that enables you to easily identify the Tel Profile. From the 'Profile Preference' drop-down list, select the priority of the Tel Profile, where '1' is the lowest priority and '20' is the highest.
Page 123: Figure 3-78: Ip Profile Settings Page
SIP User's Manual 3. Web-Based Management To configure IP Profiles: Open the 'IP Profile Settings' page (Configuration tab > VoIP menu > Coders And Profiles submenu > IP Profile Settings). Figure 3-78: IP Profile Settings Page From the 'Profile ID' drop-down list, select an identification number for the IP Profile. In the 'Profile Name' field, enter an arbitrary name that allows you to easily identify the IP Profile.
Page 124: Gw And Ip To Ip
MediaPack Series 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 125: Figure 3-79: Endpoint Phone Number Table Page
SIP User's Manual 3. Web-Based Management To configure the Endpoint Phone Number table: Open the ‘Endpoint Phone Number Table’ page (Configuration tab > VoIP menu > GW and IP to IP submenu > Hunt Group submenu > Endpoint Phone Number). Figure 3-79: Endpoint Phone Number Table Page Configure the endpoint phone numbers according to the table below.
Page 126: Figure 3-80: Hunt Group Settings Page
MediaPack Series Parameter Description Hunt Group ID The Hunt Group ID (1-99) assigned to the corresponding channels. The same Hunt Group ID can be assigned to more than one group of channels. The Hunt Group ID is used to define a group of common channel behaviors that are used for routing IP-to-Tel calls.
Page 127: Table 3-20: Hunt Group Settings Parameters
SIP User's Manual 3. Web-Based Management To save the changes to flash memory, see ''Saving Configuration'' on page 169. An example is shown below of a REGISTER message for registering endpoint "101" using registration Per Endpoint mode. The "SipGroupName" in the Request-URI is defined in the IP Group table (see ''Configuring IP Groups'' on page 103).
Page 128 MediaPack Series Parameter Description configured globally to register all its endpoints (using the parameter ChannelSelectMode), you can exclude some endpoints from being registered by assigning them to a Hunt Group and configuring the Hunt Group registration mode to 'Don't Register'. [5] Per Account = Registrations are sent (or not) to an IP Group, according to the settings in the Account table (see ''Configuring Account Table'' on page 113).
Page 129: Figure 3-81: General Settings Page
SIP User's Manual 3. Web-Based Management 3.3.2.8.2 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 129) Manipulation tables (see ''Configuring Number Manipulation Tables'' on page 129): •...
Page 130 MediaPack Series IP-to-Tel calls: • Destination Phone Number Manipulation Table for IP-to-Tel Calls (NumberMapIP2Tel ini file parameter) - up to 100 entries • Source Phone Number Manipulation Table for IP-to-Tel Calls (SourceNumberMapIP2Tel ini file parameter) - up to 20 entries The device searches a matching manipulation rule starting from the first entry (i.e., top of the table).
Page 131: Figure 3-82: Source Phone Number Manipulation Table For Tel-To-Ip Calls
SIP User's Manual 3. Web-Based Management To configure number manipulation rules: Open the required 'Number Manipulation' page (Configuration tab > VoIP menu > GW and IP to IP submenu > Manipulations submenu > Dest Number IP->Tel, Dest Number Tel->IP, Source Number IP->Tel, or Source Number Tel->IP); the relevant Manipulation table page is displayed (e.g., 'Source Phone Number Manipulation Table for Tel IP Calls' page).
Page 132: Table 3-21: Number Manipulation Parameters Description
MediaPack Series Table 3-21: Number Manipulation Parameters Description Parameter Description Source Trunk Group The source Hunt Group ID for Tel-to-IP calls. To denote all Hunt Groups, leave this field empty. Notes: The value -1 indicates that this field is ignored in the rule. This parameter is available only in the 'Source Phone Number Manipulation Table for Tel ->...
Page 133 SIP User's Manual 3. Web-Based Management Parameter Description Web: Suffix to Add The number or string that you want added to the end of the telephone EMS: Prefix/Suffix To Add number. For example, if you enter '00' and the phone number is 1234, the new number is 123400.
Page 134: Figure 3-83: Redirect Number Tel To Ip Page
MediaPack Series 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 3-83: Redirect Number Tel to IP Page The figure below shows an example configuration in which the redirect prefix "555"...
Page 135: Figure 3-84: Phone Context Table Page
SIP User's Manual 3. Web-Based Management Parameter Description Web/EMS: Number of The number of digits that you want to retain from the right of the phone Digits to Leave number. Web: Presentation Determines whether Caller ID is permitted: EMS: Is Presentation Not Configured = Privacy is determined according to the Caller ID Restricted table (see ''Configuring Caller Display Information'' on page 155).
Page 136: Table 3-23: Phone-Context Parameters Description
MediaPack Series 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. • You can also configure the Phone Context table using the ini file table parameter PhoneContext (see ''Number Manipulation and Routing Parameters'' on page 474).
Page 137: Figure 3-85: Routing General Parameters Page
SIP User's Manual 3. Web-Based Management 3.3.2.8.3 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 137) Tel to IP Routing (see ''Configuring Tel to IP Routing'' on page 138) IP to Trunk Group Routing (see ''Configuring IP to Hunt Group Routing Table'' on page 142) Alternative Routing Reasons (see ''Configuring Alternative Routing Reasons'' on page...
Page 138 MediaPack Series 3.3.2.8.3.2 Configuring Tel to IP Routing The 'Tel to IP Routing' page allows you to configure up to 50 Tel-to-IP call routing rules. The device uses these rules to route calls (from the Tel ) to IP destinations. This table provides two main areas for defining a routing rule: Matching Characteristics: User-defined characteristics of the incoming call.
Page 139: Figure 3-86: Tel To Ip Routing Page
SIP User's Manual 3. Web-Based Management • Ping to the initial destination is unavailable, poor QoS (delay or packet loss, calculated according to previous calls) is detected, or a DNS host name is unresolved. For detailed information on Alternative Routing, see ''Configuring Alternative Routing (Based on Connectivity and QoS'' on page 247).
Page 140: Table 3-24: Tel-To-Ip Routing Table Parameters
MediaPack Series From the 'Routing Index' drop-down list, select the range of entries that you want to add. Configure the routing rules according to the table below. Click the Submit button to apply your changes. To save the changes to flash memory, see ''Saving Configuration'' on page 169. Table 3-24: Tel-to-IP Routing Table Parameters Parameter Description...
Page 141: Parameter Description
SIP User's Manual 3. Web-Based Management Parameter Description If the string 'ENUM' is specified for the destination IP address, an ENUM query containing the destination phone number is sent to the DNS server. The ENUM reply includes a SIP URI used as the Request-URI in the outgoing INVITE and for routing (if a proxy is not used).
Page 142: Figure 3-87: Inbound Ip Routing Table Page
MediaPack Series Parameter Description Web/EMS: Charge Code Optional Charge Code assigned to the routing rule. For configuring Charge Codes, see Configuring Charge Codes Table on page 151. Note: This parameter is applicable only to FXS interfaces. 3.3.2.8.3.3 Configuring IP to Hunt Group Routing Table The 'IP to Hunt Group Routing Table' page allows you to configure up to 24 inbound (IP-to- Tel / Hunt Group) call routing rules.
Page 143: Table 3-25: Ip-To-Tel Routing Table Description
SIP User's Manual 3. Web-Based Management • Rule 2: If the incoming IP call destination phone prefix is between 501 and 502, and source phone prefix is 101, the call is assigned settings configured for IP Profile ID 1 and routed to Hunt Group ID 2. •...
Page 144: Configuring Alternative Routing Reasons
MediaPack Series Parameter Description 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. Notes: You can configure from where the source IP address is obtained, using the parameter SourceIPAddressInput.
Page 145: Figure 3-88: Reasons For Alternative Routing Page
SIP User's Manual 3. Web-Based Management Notes: • To enable alternative routing using the IP-to-Tel routing table, set the parameter RedundantRoutingMode to 1 (default). • The reasons for alternative routing for Tel-to-IP calls also apply for Proxies (if the parameter RedundantRoutingMode is set to 2). •...
Page 146: Figure 3-89: Forward On Busy Trunk Destination Page
MediaPack Series 3.3.2.8.3.5 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, if an unavailable FXS/FXO Hunt Group exists. This feature can be used, for example, to forward the call to another FXS/FXO device.
Page 147: Figure 3-90: Dtmf & Dialing Page
SIP User's Manual 3. Web-Based Management 3.3.2.8.4 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 147) Supplementary Services (see ''Configuring Supplementary Services'' on page 148) 3.3.2.8.4.1 Configuring DTMF and Dialing The 'DTMF &...
Page 148: Figure 3-91: Supplementary Services Page
MediaPack Series 3.3.2.8.4.2 Configuring Supplementary Services The 'Supplementary Services' page is used to configure parameters associated with supplementary services. For a description of the parameters appearing on this page, see ''Configuration Parameters Reference'' on page 333. For an overview on supplementary services, see ''Working with Supplementary Services'' on page 254.
Page 149 SIP User's Manual 3. Web-Based Management Click the Submit button to save your changes, or click the Subscribe to MWI or Unsubscribe to MWI buttons to save your changes and to subscribe / unsubscribe to the MWI server. To save the changes to flash memory, see ''Saving Configuration'' on page 169. 3.3.2.8.5 Analog Gateway The Analog Gateway submenu allows you to configure analog settings.
Page 150: Figure 3-92: Keypad Features Page
MediaPack Series 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 3-92: Keypad Features Page Configure the keypad features as required. For a description of these parameters, see ''Configuration Parameters Reference'' on page 333.
Page 151: Figure 3-93: Metering Tones Page
SIP User's Manual 3. Web-Based Management 3.3.2.8.5.2 Configuring Metering Tones The FXS interfaces can generate 12/16 KHz metering pulses toward the Tel side (e.g., for connection to a pay phone or private meter). Tariff pulse rate is determined according to the device's Charge Codes table.
Page 152: Figure 3-94: Charge Codes Table Page
MediaPack Series 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 151).
Page 153: Figure 3-95: Fxo Settings Page
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 3-95: FXO Settings Page Configure the parameters as required.
Page 154 MediaPack Series 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). In the 'User Name' and 'Password' fields corresponding to a port, enter the user name and password respectively.
Page 155: Figure 3-96: Automatic Dialing Page
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 3-96: Automatic Dialing Page In the 'Destination Phone Number' field corresponding to a port, enter the telephone number that you want automatically dialed.
Page 156: Figure 3-97: Caller Display Information Page
MediaPack Series Notes: • When FXS ports receive 'Private' or 'Anonymous' strings in the From header, they don't send the calling name or number to the Caller ID display. • If Caller ID name is detected on an FXO line (EnableCallerID = 1), it is used instead of the Caller ID name defined on this page.
Page 157: Figure 3-98: Call Forward Table Page
SIP User's Manual 3. Web-Based Management 3.3.2.8.5.8 Configuring Call Forward The 'Call Forwarding Table' page allows you to forward (redirect) IP-to-Tel calls (using SIP 302 response) originally destined to specific device ports, to other device ports or to an IP destination.
Page 158: Figure 3-99: Caller Id Permissions Page
MediaPack Series 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. 3.3.2.8.5.9 Configuring Caller ID Permissions The 'Caller ID Permissions' page allows you to enable or disable (per port) the Caller ID generation (for FXS interfaces) and detection (for FXO interfaces).
Page 159: Figure 3-100: Call Waiting Page
SIP User's Manual 3. Web-Based Management 3.3.2.8.5.10 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 148).
Page 160: Figure 3-101: Voice Mail Settings Page
MediaPack Series 3.3.2.8.6 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 160) 3.3.2.8.6.1 Configuring Voice Mail Parameters The 'Voice Mail Settings' page allows you to configure the voice mail parameters. For a description of these parameters, see ''Configuration Parameters Reference'' on page 333.
Page 161: Sas
SIP User's Manual 3. Web-Based Management 3.3.2.9 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 161), from which you can also access the 'IP2IP Routing Table' page for configuring SAS routing rules (see ''Configuring IP2IP Routing Table (SAS)'' on page 163).
Page 162: Figure 3-102: Sas Configuration Page
MediaPack Series To configure SAS: Open the 'SAS Configuration' page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability). Figure 3-102: 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 163: Table 3-27: Sas Ip2Ip Routing Table Parameters
SIP User's Manual 3. Web-Based Management 3.3.2.9.2 Configuring IP2IP Routing Table (SAS) The 'IP2IP Routing Table' page allows you to configure up to 120 SAS routing rules (for Normal and Emergency modes). The device routes the SAS call (received SIP INVITE message) once a rule in this table is matched.
Page 164 MediaPack Series Parameter Description Destination Host The host part of the incoming SIP INVITE’s destination URI [IP2IPRouting_DestHost] (usually the Request URI). If this rule is not required, leave the field empty. The asterisk (*) symbol can be used to depict any destination host. The default is "*".
Page 165 SIP User's Manual 3. Web-Based Management Parameter Description Destination Address The destination IP address (or domain name, e.g., [IP2IPRouting_DestAddress] domain.com) to where the call is sent. Notes: This parameter is applicable only if the parameter 'Destination Type' is set to 'Dest Address' [1]. When using domain names, enter a DNS server IP address or alternatively, define these names in the 'Internal DNS Table' (see ''Configuring the Internal SRV...
Page 166: Maintenance Tab
MediaPack Series 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 166) Software Update (see ''Software Update'' on page 170) 3.4.1 Maintenance The Maintenance menu allows you to perform various maintenance procedures.
Page 167: Figure 3-104: Reset Confirmation Message Box
SIP User's Manual 3. Web-Based Management 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). Perform a graceful shutdown, i.e., device reset starts only after a user-defined time (i.e., timeout) or after no more active traffic exists (the earliest thereof).
Page 168: Figure 3-105: Device Lock Confirmation Message Box
MediaPack Series 3.4.1.1.2 Locking and Unlocking the Device The Lock and Unlock options allow you to lock the device so that it doesn't accept any new calls. This is useful when, for example, you are uploading new software files to the device and you don't want any traffic to interfere with the process.
Page 169: Saving Configuration
SIP User's Manual 3. Web-Based Management 3.4.1.1.3 Saving Configuration The 'Maintenance Actions' page allows you to save (burn) the current parameter configuration (including loaded auxiliary files) to the device's non-volatile memory (i.e., flash). The parameter modifications that you make throughout the Web interface's pages are temporarily saved (to the volatile memory - RAM) when you click the Submit button on these pages.
Page 170: Software Update
MediaPack Series 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 170) Software Upgrade Key (see ''Loading Software Upgrade Key'' on page 172) Software Upgrade Wizard (see ''Software Upgrade Wizard'' on page 175) Configuration File (see ''Backing Up and Loading Configuration File'' on page 178)
Page 171: Figure 3-106: Load Auxiliary Files Page
SIP User's Manual 3. Web-Based Management Notes: • You can schedule automatic loading of updated auxiliary files using HTTP/HTTPS, FTP, or NFS (for more details, refer to the Product Reference Manual). • For a detailed description on auxiliary files, see ''Auxiliary Configuration Files'' on page 217.
Page 172: Loading Software Upgrade Key
You can load a Software Upgrade Key using one of the following management tools: Web interface BootP/TFTP configuration utility (see Loading via BootP/TFTP on page 175) 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 173: Figure 3-107: Software Upgrade Key Status Page
SIP User's Manual 3. Web-Based Management To load a Software Upgrade Key: Open the 'Software Upgrade Key Status' page (Maintenance tab > Software Update menu > Software Upgrade Key). Figure 3-107: 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 comply with your requirements: In the 'Current Key' field, copy the string of text and paste it into any standard text...
Page 174: Figure 3-108: Software Upgrade Key With Multiple S/N Lines
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 175: Software Upgrade Wizard
3.4.2.2.1 Loading via BootP/TFTP The procedure below describes how to load a Software Upgrade Key to the device using AudioCodes' BootP/TFTP Server utility (for a detailed description on the BootP utility, refer to the Product Reference Manual). To load a Software Upgrade Key file using BootP/TFTP: Place the Software Upgrade Key file (typically, a *.txt file) in the same folder in which...
Page 176: Figure 3-109: 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 177 SIP User's Manual 3. Web-Based Management Click the Browse button, navigate to the cmp file, and then click Send File; a progress bar appears displaying the status of the loading process. When the cmp file is successfully loaded to the device, a message appears notifying you of this. If you want to load only a cmp file, then click the Reset button to reset the device with the newly loaded cmp file, utilizing the existing configuration (ini) and...
Page 178: Backing Up And Loading Configuration File
MediaPack Series After the device resets, the 'End Process' screen appears displaying the burned configuration files: Figure 3-110: 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 179: Figure 3-111: Configuration File Page
SIP User's Manual 3. Web-Based Management To save / load the ini 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 180: Status & Diagnostics Tab
The 'Message Log' page displays Syslog debug messages sent by the device. You can select the Syslog messages in this page, and then copy and paste them into a text editor such as Notepad. This text file (txt) can then be sent to AudioCodes Technical Support for diagnosis and troubleshooting.
Page 181: Figure 3-112: Message Log Page
SIP User's Manual 3. Web-Based Management To activate the Message Log: Activate and configure the device's Syslog client. Open the 'Message Log' page (Status & Diagnostics tab > System Status menu > Message Log); the 'Message Log' page is displayed and the log is activated. Figure 3-112: Message Log Page The displayed logged messages are color coded as follows: •...
Page 182: Viewing Device Information
The 'Device Information' page displays the device's specific hardware and software product information. This information can help you expedite troubleshooting. Capture the page and e-mail it to AudioCodes Technical Support personnel to ensure quick diagnosis and effective corrective action. This page also displays any loaded files used by the device (stored in the RAM) and allows you to remove them.
Page 183: Carrier-Grade Alarms
SIP User's Manual 3. Web-Based Management To view Ethernet port information: Open the ‘Ethernet Port Information’ page (Status & Diagnostics tab > System Status menu > Ethernet Port Information). Figure 3-114: Ethernet Port Information Page Table 3-29: Ethernet Port Information Parameters Parameter Description Port Duplex Mode...
Page 184: Voip Status
MediaPack Series 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 184) Performance Statistics (see ''Viewing Performance Statistics'' on page 184) IP to Tel Calls Count (see ''Viewing Call Counters'' on page 185) Tel to IP Calls Count (see ''Viewing Call Counters'' on page 185)
Page 185: Viewing Call Counters
SIP User's Manual 3. Web-Based Management 3.5.2.3 Viewing Call Counters The 'IP to Tel Calls Count' and 'Tel to IP Calls Count' pages provide you with statistical information on incoming (IP-to-Tel) and outgoing (Tel-to-IP) calls. The statistical information is updated according to the release reason that is received after a call is terminated (during the same time as the end-of-call Call Detail Record or CDR message is sent).
Page 186 MediaPack Series Counter Description GWAPP_NORMAL_CALL_CLEAR increments the 'Number of Failed Calls due to No Answer' counter. The rest of the release reasons increment the 'Number of Failed Calls due to Other Failures' counter. Percentage of The percentage of established calls from attempted calls. Successful Calls (ASR) Number of Calls Indicates the number of calls that failed as a result of a busy line.
Page 187: Viewing Sas/Sbc Registered Users
SIP User's Manual 3. Web-Based Management 3.5.2.4 Viewing SAS/SBC Registered Users The 'SAS/SBC Registered Users' page displays a list of registered SAS users recorded in the device's database. To view the registered users: Open the 'SAS/SBC Registered Users' page (Status & Diagnostics tab > VoIP Status menu >...
Page 188: Viewing Call Routing Status
MediaPack Series 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. To view the call routing status: Open the 'Call Routing Status' page (Status &...
Page 189: Viewing Registration Status
SIP User's Manual 3. Web-Based Management 3.5.2.6 Viewing Registration Status The 'Registration Status' page displays whether the device, its endpoints, SIP Accounts, and BRI endpoints are registered to a SIP Registrar/Proxy server. To view Registration status: Open the 'Registration Status' page (Status & Diagnostics tab > VoIP Status menu >...
Page 190: Viewing Ip Connectivity
MediaPack Series 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 'Tel to IP Routing' page (see ''Configuring Tel to IP Routing'' on page 138). Notes: •...
Page 191 SIP User's Manual 3. Web-Based Management Column Name Description Quality Status Determines the QoS (according to packet loss and delay) of the IP address. Unknown = Recent quality information isn't available. Poor Notes: This parameter is applicable only if the parameter 'Alt Routing Tel to IP Mode' is set to 'QoS' or 'Both' (AltRoutingTel2IPMode = 2 or 3).
Page 192 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 193: Ini File-Based Management
The ini file can be loaded to the device using the following methods: Web interface (see ''Backing Up and Loading Configuration File'' on page 178) AudioCodes' BootP/TFTP utility (refer to the Product Reference Manual) Any standard TFTP server The ini file configuration parameters are saved in the device's non-volatile memory when the file is loaded to the device.
Page 194: Configuring Ini File Table Parameters
MediaPack Series 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 195 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 196: General Ini File Formatting Rules
MediaPack Series 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 197: Secured Encoded Ini File
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 198 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 199: 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 gateways within VoP networks, covering all areas vital for the efficient operation, administration, management and provisioning (OAM&P) of AudioCodes' families of gateways.
Page 200: Securing Ems-Device Communication
MediaPack Series 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 added.
Page 201: Changing Ssh Login Password
SIP User's Manual 5. EMS-Based Management [ IPsecSATable ] FORMAT IPsecSATable_Index = IPsecSATable_RemoteEndpointAddressOrName, IPsecSATable_AuthenticationMethod, IPsecSATable_SharedKey, IPsecSATable_SourcePort, IPsecSATable_DestPort, IPsecSATable_Protocol, IPsecSATable_Phase1SaLifetimeInSec, IPsecSATable_Phase2SaLifetimeInSec, IPsecSATable_Phase2SaLifetimeInKB, IPsecSATable_DPDmode, IPsecSATable_IPsecMode, IPsecSATable_RemoteTunnelAddress, IPsecSATable_RemoteSubnetIPAddress, IPsecSATable_RemoteSubnetPrefixLength, IPsecSATable_InterfaceName; IPsecSATable 1 = <IP address>, 0, <IKE password>, 0, 0, 0, 28800, 28800, 0, 0, 0, 0.0.0.0, 0.0.0.0, 16, ; [ \IPsecSATable ] EnableIPSec = 1 where:...
Page 202: Adding The Device In Ems
MediaPack Series Adding the Device in EMS Once you have defined the IPSec communication protocol for communicating between EMS and the device and configured the device's IP address (refer to the Installation Manual), you can add the device in the EMS. Adding the device to the EMS includes the following main stages: Adding a Region Defining the device's IP address (and other initial settings)
Page 203: Figure 5-3: Adding A Region
SIP User's Manual 5. EMS-Based Management 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 204: Configuring Basic Sip Parameters
MediaPack Series 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 205: Configuring Advanced Ipsec/Ike Parameters
SIP User's Manual 5. EMS-Based Management Double-click each field to enter values. Right-click the new entry, and then select Unlock Rows. Click Apply and close the active window. If a Proxy Server is not implemented, map outgoing telephone calls to IP addresses. Open the 'SIP Routing' frame (Configuration pane >...
Page 206: Provisioning Sip Srtp Crypto Offered Suites
MediaPack Series 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 207: Provisioning Sip Mlpp Parameters
SIP User's Manual 5. EMS-Based Management Provisioning SIP MLPP Parameters This section describes how to configure the MLPP (Multi-Level Precedence and Preemption) parameters using the EMS. To configure the MLPP parameters: In the Navigation pane, select VoIP > SIP, and then in the Configuration pane, select SIP Advanced Configuration;...
Page 208: Configuring The Device To Operate With Snmpv3
MediaPack Series Configuring the Device to Operate with SNMPv3 This section describes the SNMPv3 configuration process: Configuring SNMPv3 using SSH Configuring SNMPv3 using EMS (non-configured SNMPv3 System) Configuring SNMPv3 using EMS (pre-configured SNMPv3 System) Note: After configuring SNMPv3, ensure that you disable IPSec. 5.8.1 Configuring SNMPv3 using SSH The procedure below describes how to configure SNMPv3 using SSH.
Page 209: Configuring Ems To Operate With A Pre-Configured Snmpv3 System
SIP User's Manual 5. EMS-Based Management To end the PuTTY configuration session, type a full-stop (“.”) on an empty line; the device responds with the following: INI File replaced To save the configuration to the non-volatile memory, type sar; the device reboots with IPSec enabled.
Page 210: Configuring Snmpv3 To Operate With Non-Configured Snmpv3 System
MediaPack Series 5.8.3 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 211: Cloning Snmpv3 Users
SIP User's Manual 5. EMS-Based Management 5.8.4 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 212: Upgrading The Device's Software
MediaPack Series 5.10 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 213: Figure 5-13: 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 214 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 215: 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 215) Loading an empty ini file (see ''Restoring Defaults using an ini File'' on page 216) Using the hardware Reset button (see Restoring Defaults using Hardware Reset Button on page 216) Restoring Defaults using CLI...
Page 216: Restoring Defaults Using An Ini File
MediaPack Series 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 178). The only settings that are not restored to default are the management (OAMP) LAN IP address and the Web interface's login user name and password.
Page 217: 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 220) Prerecorded Tones (see ''Prerecorded Tones File'' on page Dial Plan (see Dial Plan File on page 223) User Information (see ''User Information File'' on page 224)
Page 218 MediaPack Series 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 219 SIP User's Manual 7. Auxiliary Configuration Files • First Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the first cadence on-off cycle. For continuous tones, this parameter defines the detection period. For burst tones, it defines the tone's duration. •...
Page 220: Distinctive Ringing
MediaPack Series 7.1.1 Distinctive Ringing Distinctive Ringing is applicable only to FXS interfaces. Using the Distinctive Ringing section of the Call Progress Tones auxiliary file, you can create up to 16 Distinctive Ringing patterns. Each ringing pattern configures the ringing tone frequency and up to four ringing cadences.
Page 221 SIP User's Manual 7. Auxiliary Configuration Files An example of a ringing burst definition is shown below: #Three ringing bursts followed by repeated ringing of 1 sec on and 3 sec off. [NUMBER OF DISTINCTIVE RINGING PATTERNS] Number of Ringing Patterns=1 [Ringing Pattern #0] Ring Type=0 Freq [Hz]=25...
Page 222: Prerecorded Tones File
Resolution: 8-bit Channels: mono Once created, the PRT file can then be loaded to the device using AudioCodes' BootP/TFTP utility or the Web interface (see ''Loading Auxiliary Files'' on page 170). The prerecorded tones are played repeatedly. This allows you to record only part of the tone and then play the tone for the full duration.
Page 223: Dial Plan File
SIP User's Manual 7. Auxiliary Configuration Files Dial Plan File The Dial Plan file contains a list of up to eight dial plans, supporting a total of up to 8,000 user-defined, distinct prefixes (e.g. area codes, international telephone number patterns) for the PSTN to which the device is connected.
Page 224: User Information File
MediaPack Series 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 225: Figure 7-1: Example Of A User Information File
SIP User's Manual 7. Auxiliary Configuration Files Item Description Maximum Size (Characters) A string that represents the user name for SIP Username registration. A string that represents the password for SIP Password registration. Note: For FXS ports, when the device is required to send a new request with the ‘Authorization’...
Page 226 MediaPack Series The calling number of outgoing Tel-to-IP calls is translated to a "Global phone number" only after Tel-to-IP manipulation rules (if defined) are performed. The Display Name is used in the From header in addition to the "Global phone number". The called number of incoming IP-to-Tel calls is translated to a PBX extension only after IP-to-Tel manipulation rules (if defined) are performed.
Page 227: Ip Telephony Capabilities
SIP User's Manual 8. IP Telephony Capabilities IP Telephony Capabilities This section describes the device's main IP telephony capabilities. 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.
Page 228: Gateway And Ip-To-Ip
MediaPack Series Gateway and IP-to-IP This section describes various Gateway and IP-to-IP application features. 8.2.1 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 228) Digit mapping (see ''Digit Mapping'' on page 229) External Dial Plan file containing dial plans (see ''External Dial Plan File'' on page 230)
Page 229: Digit Mapping
SIP User's Manual 8. IP Telephony Capabilities 8.2.1.2 Digit Mapping The device collects digits until a match is found in the user-defined digit pattern (e.g., for closed numbering schemes). The device stops collecting digits and starts sending the digits (collected number) when any one of the following scenarios occur: Maximum number of digits is received.
Page 230: External Dial Plan File
MediaPack Series Notes: • If you want the device to accept/dial any number, ensure that the digit map contains the rule "xx.T"; otherwise, dialed numbers not defined in the digit map are rejected. • If you are using an external Dial Plan file for dialing plans (see ''External Dial Plan File'' on page 230), the device first attempts to locate a matching digit pattern in the Dial Plan file, and if not found, then attempts to locate a matching digit pattern in the Digit Map (configured by the...
Page 231 SIP User's Manual 8. IP Telephony Capabilities An example of a Dial Plan file with indices (in ini-file format before conversion to binary *.dat) is shown below: [ PLAN1 ] ; Area codes 02, 03, - phone numbers include 7 digits. 02,7 03,7 ;...
Page 232: Manipulating Number Prefix
MediaPack Series 8.2.2 Manipulating Number Prefix The device supports a notation for adding a prefix where part of the prefix is first extracted from a user-defined location in the original destination or source number. This notation is entered in the 'Prefix to Add' field in the Number Manipulation tables (see ''Manipulation'' on page 129): x[n,l]y...
Page 233: Configuring Dtmf Transport Types
SIP User's Manual 8. IP Telephony Capabilities 8.2.3 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 234: Fxs And Fxo Capabilities
MediaPack Series Notes: • The device is always ready to receive DTMF packets over IP in all possible transport modes: INFO messages, NOTIFY, and RFC 2833 (in proper payload type) or as part of the audio stream. • To exclude RFC 2833 Telephony event parameter from the device's SDP, set RxDTMFOption to 0 in the ini file.
Page 235: Figure 8-2: Call Flow For One-Stage Dialing
SIP User's Manual 8. IP Telephony Capabilities 8.2.4.2.1 FXO Operations for IP-to-Tel Calls The FXO device provides the following operating modes for IP-to-Tel calls: One-stage dialing (see ''One-Stage Dialing'' on page 235) • Waiting for dial tone (see ''Two-Stage Dialing'' on page 236) •...
Page 236: Figure 8-3: Call Flow For Two-Stage Dialing
MediaPack Series Note: The ini file parameter IsWaitForDialTone must be disabled for this mode. 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 237 SIP User's Manual 8. IP Telephony Capabilities 8.2.4.2.1.3 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 238: Figure 8-4: Call Flow For Automatic Dialing
MediaPack Series 8.2.4.2.2 FXO Operations for Tel-to-IP Calls The FXO device provides the following FXO operating modes for Tel-to-IP calls: Automatic Dialing (see ''Automatic Dialing'' on page 238) Collecting Digits Mode (see ''Collecting Digits Mode'' on page 239) FXO Supplementary Services (see ''FXO Supplementary Services'' on page 239) •...
Page 239: Figure 8-5: Call Flow For Collecting Digits
SIP User's Manual 8. IP Telephony Capabilities 8.2.4.2.2.2 Collecting Digits Mode When automatic dialing is not defined, the device collects the digits. The SIP call flow diagram below illustrates the Collecting Digits Mode. Figure 8-5: Call Flow for Collecting Digits 8.2.4.2.2.3 FXO Supplementary Services The FXO supplementary services include the following: Hold / Transfer toward the Tel side: The ini file parameter LineTransferMode must...
Page 240 MediaPack Series 8.2.4.2.3 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 240) Calls terminated before call establishment (see ''Call Termination before Call Establishment'' on page 241) Ring detection timeout (see ''Ring Detection Timeout'' on page 241) 8.2.4.2.3.1 Calls Termination by PBX...
Page 241 SIP User's Manual 8. IP Telephony Capabilities 8.2.4.2.3.2 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 242: Remote Pbx Extension Between Fxo And Fxs Devices
MediaPack Series 8.2.4.3 Remote PBX Extension Between FXO and FXS Devices Remote PBX extension offers a company the capability of extending the "power" of its local PBX by allowing remote phones (remote offices) to connect to the company's PBX over the IP network (instead of via PSTN).
Page 243 SIP User's Manual 8. IP Telephony Capabilities 8.2.4.3.1 Dialing from Remote Extension (Phone at FXS) The procedure below describes how to dial from the 'remote PBX extension' (i.e., phone connected to the FXS interface). To make a call from the FXS interface: Off-hook the phone and wait for the dial tone from the PBX.
Page 244: Figure 8-7: Mwi For Remote Extensions
MediaPack Series Upon detection of an MWI message, the FXO device sends a SIP NOTIFY message to the IP side. When receiving this NOTIFY message, the remote FXS device generates an MWI signal toward its Tel side. Figure 8-7: MWI for Remote Extensions 8.2.4.3.4 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...
Page 245: Figure 8-10: Automatic Dialing For Fxs Ports
SIP User's Manual 8. IP Telephony Capabilities In the ‘Automatic Dialing’ page (see ''Configuring Automatic Dialing'' on page 154), enter the phone numbers of the FXO device in the ‘Destination Phone Number’ fields. When a phone connected to Port #1 off-hooks, the FXS device automatically dials the number ‘200’.
Page 246: Figure 8-13: Fxo Automatic Dialing Configuration
MediaPack Series Figure 8-13: FXO Automatic Dialing Configuration In the ‘Tel to IP Routing’ page, enter 10 in the ‘Destination Phone Prefix’ field, and the IP address of the FXS device (10.1.10.3) in the field ‘IP Address’. Figure 8-14: FXO Tel-to-IP Routing Configuration In the ‘FXO Settings’...
Page 247: Configuring Alternative Routing (Based On Connectivity And Qos)
SIP User's Manual 8. IP Telephony Capabilities 8.2.5 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.
Page 248: Fax And Modem Capabilities
MediaPack Series 8.2.6 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 248) Fax and modem transport modes (see ''Fax/Modem Transport Modes'' on page 248) V.152 support (see ''V.152 Support'' on page 253) 8.2.6.1 Fax/Modem Operating Modes...
Page 249 SIP User's Manual 8. IP Telephony Capabilities When fax transmission ends, the reverse switching from fax relay to voice is automatically performed at both the local and remote endpoints. You can change the fax rate declared in the SDP, using the parameter FaxRelayMaxRate (this parameter doesn’t affect the actual transmission rate).
Page 250 MediaPack Series 8.2.6.2.2 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: Echo Canceller = off Silence Compression = off Echo Canceller Non-Linear Processor Mode = off...
Page 251 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 252 MediaPack Series The parameters defining payload type for the proprietary AudioCodes’ Bypass mode FaxBypassPayloadType and ModemBypassPayloadType are not used with NSE Bypass. When configured for NSE mode, the device includes in its SDP the following line: a=rtpmap:100 X-NSE/8000 (where 100 is the NSE payload type) The Cisco gateway must include the following definition: "modem passthrough nse...
Page 253: V.152 Support
SIP User's Manual 8. IP Telephony Capabilities V22ModemTransportType = 0 V23ModemTransportType = 0 V32ModemTransportType = 0 V34ModemTransportType = 0 BellModemTransportType = 0 Additional configuration parameters: • CodersGroup • DJBufOptFactor • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission.
Page 254: Fax Transmission Behind Nat
MediaPack Series 0 0 IN IPV4 <IPAdressA> t=0 0 p=+1 c=IN IP4 <IPAddressA m=audio <udpPort A> RTP/AVP 18 0 a=ptime:10 a=rtpmap:96 PCMU/8000 a=gpmd: 96 vbd=yes In the example above, V.152 implementation is supported (using the dynamic payload type 96 and G.711 u-law as the VBD codec) as well as the voice codecs G.711 μ-law and G.729.
Page 255: Call Hold And Retrieve
SIP User's Manual 8. IP Telephony Capabilities To activate these supplementary services, enable each service’s corresponding parameter using the Web interface or ini file. Notes: • All call participants must support the specific supplementary service that is used. • When working with certain application servers (such as BroadSoft’s BroadWorks) in client server mode (the application server controls all supplementary services and keypad features by itself), the device's supplementary services must be disabled.
Page 256: Figure 8-15: Double Hold Sip Call Flow
MediaPack Series The device also supports "double call hold" for FXS interfaces where the called party, which has been placed on-hold by the calling party, can then place the calling party on hold as well and make a call to another destination. The flowchart below provides an example of this type of call hold: Figure 8-15: Double Hold SIP Call Flow The flowchart above describes the following "double"...
Page 257: Call Pickup
SIP User's Manual 8. IP Telephony Capabilities B ends call with D. B retrieves call with A. Notes: • If a party that is placed on hold (e.g., B in the above example) is called by another party (e.g., D), then the on-hold party receives a Call Waiting tone instead of the Held tone.
Page 258: Call Transfer
MediaPack Series 8.2.7.4 Call Transfer The device supports the following call transfer types: Consultation Transfer (see ''Consultation Call Transfer'' on page 258) Blind Transfer (see ''Blind Call Transfer'' on page 258) Notes: • Call transfer is initiated by sending REFER with REPLACES. •...
Page 259: Call Forward
SIP User's Manual 8. IP Telephony Capabilities 8.2.7.5 Call Forward The following methods of call forwarding are supported: Immediate: incoming call is forwarded immediately and unconditionally. Busy: incoming call is forwarded if the endpoint is busy. No Reply: incoming call is forwarded if it isn't answered for a specified time. On Busy or No Reply: incoming call is forwarded if the port is busy or when calls are not answered after a specified time.
Page 260: Figure 8-16: Call Forward Reminder With Application Server
MediaPack Series 8.2.7.5.1 Call Forward Reminder Ring The device supports the Call Forward Reminder Ring feature for FXS interfaces, whereby the device's FXS endpoint emits a short ring burst (only if in onhook state) when a third- party Application Server (e.g., softswitch) forwards an incoming call to another destination. This is important in that it notifies (audibly) the FXS endpoint user that a call forwarding service is currently being performed.
Page 261: Call Waiting
SIP User's Manual 8. IP Telephony Capabilities Content-Type: "application/simservs+xml" Message body is the XML body and contains the “dial-tone-pattern” set to "special- condition-tone" (<ss:dial-tone-pattern>special-condition-tone</ss:dial-tone-pattern>), which is the special tone indication. For cancelling the special dial tone and playing the regular dial tone, the received SIP NOTIFY message must contain the following headers: From and To: contain the same information, indicating the specific endpoint Event: ua-profile...
Page 262: Message Waiting Indication
MediaPack Series 8.2.7.7 Message Waiting Indication The device supports Message Waiting Indication (MWI) according to IETF Internet-Draft draft-ietf-sipping-mwi-04, including SUBSCRIBE (to MWI server). Note: For a detailed description on IP voice mail configuration, refer to the IP Voice Mail CPE Configuration Guide. The FXS device can accept an MWI NOTIFY message that indicates waiting messages or that the MWI is cleared.
Page 263 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 264 MediaPack Series 8.2.7.8.3 Caller ID on the IP Side Caller ID is provided by the SIP From header containing the caller's name and "number", for example: From: “David” <SIP:101@10.33.2.2>;tag=35dfsgasd45dg If Caller ID is restricted (received from Tel or configured in the device), the From header is set to: From: “anonymous”...
Page 265: Three-Way Conferencing
The device supports the following conference modes (configured by the parameter 3WayConferenceMode): Conferencing controlled by an external AudioCodes Conference (media) server: The Conference-initiating INVITE sent by the device uses the ConferenceID concatenated with a unique identifier as the Request-URI. This same Request-URI is set as the Refer-To header value in the REFER messages that are sent to the two remote parties.
Page 266: 8.2.7.10 Multilevel Precedence And Preemption
MediaPack Series To enable three-way conferencing, the following parameters need to be configured: Enable3WayConference ConferenceCode = '!' (default, which is the hook flash button) HookFlashCode 3WayConferenceMode (conference mode) MaxInBoardConferenceCalls (if on-board conferencing) 3WayConfNoneAllocateablePorts (if on-board conferencing) FlashKeysSequenceStyle = 1 or 2 (makes a three-way call conference using the Flash button + 3) Note: For local, on-board three-way conferencing on MP-112, in addition to...
Page 267 SIP User's Manual 8. IP Telephony Capabilities MLPP Precedence Level Precedence Level in Resource- DSCP Configuration Parameter Priority SIP Header flash-override MLPPFlashOverRTPDSCP 9 (highest) flash-override-override MLPPFlashOverOverRTPDSCP Precedence Ring Tone: You can assign a ring tone (in the CPT file) that is played when a Precedence call is received from the IP side.
Page 268: Sip Call Routing Examples
MediaPack Series The device initiates the release procedures for the B-channel associated with the call request and maps the preemption cause to PRI Cause = #8 ‘Preemption’. This value indicates that the call is being preempted. For PRI, it also indicates that the B-channel is not reserved for reuse. The device sends a SIP 200 OK in response to the received BYE, before the SIP end instrument can proceed with the higher precedence call.
Page 269 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/MediaPack/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 270 F4 200 OK (10.8.201.161 >> 10.8.201.108): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: <sip:6000@10.8.201.108>;tag=1c5354 To: <sip:2000@10.8.201.161>;tag=1c7345 Call-ID: 534366556655skKw-6000--2000@10.8.201.108 CSeq: 18153 INVITE Contact: <sip:2000@10.8.201.161;user=phone> Server: Audiocodes-Sip-Gateway/MediaPack/v.6.00.010.006 Supported: 100rel,em Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE, NOTIFY,PRACK,REFER,INFO Content-Type: application/sdp Content-Length: 206 o=AudiocodesGW 30221 87035 IN IP4 10.8.201.161 s=Phone-Call c=IN IP4 10.8.201.10...
Page 271: Sip Authentication Example
Upon receipt of this request, the Registrar/Proxy returns a 401 Unauthorized response: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.1.200 From: <sip:122@10.2.2.222 >;tag=1c17940 To: <sip:122@10.2.2.222 > Call-ID: 634293194@10.1.1.200 Cseq: 1 REGISTER Date: Mon, 30 Jul 2001 15:33:54 GMT Server: Columbia-SIP-Server/1.17 Content-Length: 0 WWW-Authenticate: Digest realm="audiocodes.com", nonce="11432d6bce58ddf02e3b5e1c77c010d2", stale=FALSE, algorithm=MD5 Version 6.2 February 2011...
Page 272 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 273 SIP User's Manual 8. IP Telephony Capabilities Upon receiving this request and if accepted by the Proxy, the proxy returns a 200 OK response closing the REGISTER transaction: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.200 From: <sip: 122@10.1.1.200>;tag=1c23940 To: <sip: 122@10.1.1.200> Call-ID: 654982194@10.1.1.200 Cseq: 1 REGISTER Date: Thu, 26 Jul 2001 09:34:42 GMT...
Page 274: Establishing A Call Between Two Devices
8.2.8.3 Establishing a Call between Two Devices This section provides an example on configuring two AudioCodes' devices with FXS interfaces for establishing call communication. After configuration, you can make calls between telephones connected to the same device and between the two devices.
Page 275: Sip Trunking Between Enterprise And Itsps
This section provides an example of an elaborate routing scheme for SIP trunking between an Enterprise and two Internet Telephony Service Providers (ITSP), using AudioCodes device. Scenario: In this example, an Enterprise has deployed the device with eight FXS interfaces.
Page 276: Figure 8-22: Configuring Proxy Set Id #1 In The Proxy Sets Table
MediaPack Series In the 'Proxy Sets Table' page (see ''Configuring Proxy Sets Table'' on page 106), configure two Proxy Sets and for each, enable Proxy Keep-Alive (using SIP OPTIONS) and 'round robin' load-balancing method: • Proxy Set #1 includes two IP addresses of the first ITSP (ITSP 1) - 10.33.37.77 and 10.33.37.79 - and using UDP.
Page 277: Figure 8-24: Assigning Channels To Hunt Groups
SIP User's Manual 8. IP Telephony Capabilities In the ‘Endpoint Phone Number Table’ page, configure Hunt Group ID #1 for channels 1-4, and Hunt Group ID #2 for channels 5-8. Figure 8-24: Assigning Channels to Hunt Groups In the 'Hunt Group Settings' page, configure 'Per Account' registration for Hunt Group ID #1 (without serving IP Group) and associate it with IP Group #1;...
Page 278: Mapping Pstn Release Cause To Sip Response
MediaPack Series In the 'IP to Hunt Group Routing Table' page, configure that INVITEs with "ITSP1" as the hostname in the From URI are routed to Hunt Group #1, and INVITEs with "ITSP2" as the hostname in the From URI are routed to Hunt Group #2. In addition, configure calls received from ITSP1 as associated with IP Group #1.
Page 279: Stand-Alone Survivability (Sas) Application
SIP User's Manual 8. IP Telephony Capabilities 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 servers (or even WAN connection and access Internet modem), the enterprise typically loses its internal telephony service at any branch, between its offices, and with the external environment.
Page 280: Sas Outbound Mode
MediaPack Series 8.3.1.1 SAS Outbound Mode This section describes the SAS outbound mode, which includes the following states: Normal state (see ''Normal State'' on page 280) Emergency state (see ''Emergency State'' on page 281) 8.3.1.1.1 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 281: Figure 8-31: Sas Outbound Mode In Emergency State (Example)
SIP User's Manual 8. IP Telephony Capabilities 8.3.1.1.2 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). When the device receives calls, it searches its SAS registration database to locate the destination address (according to AOR or Contact).
Page 282: Sas Redundant Mode
MediaPack Series 8.3.1.2 SAS Redundant Mode In SAS redundant mode, the enterprise's UAs register with the external proxy and establish calls directly through it, without traversing SAS (or the device per se'). Only when connection with the proxy fails, do the UAs register with SAS, serving now as the UAs redundant proxy.
Page 283: Figure 8-33: Sas Redundant Mode In Emergency State (Example)
SIP User's Manual 8. IP Telephony Capabilities 8.3.1.2.2 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 8-33: SAS Redundant Mode in Emergency State (Example) 8.3.1.2.3 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 284: Sas Routing
MediaPack Series 8.3.2 SAS Routing This section provides flowcharts describing the routing logic for SAS in normal and emergency states. 8.3.2.1 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 8-34: Flowchart of INVITE from UA's in SAS Normal State SIP User's Manual...
Page 285: Figure 8-35: Flowchart Of Invite From Primary Proxy In Sas Normal State
SIP User's Manual 8. IP Telephony Capabilities The flowchart below displays the routing logic for SAS in normal state for INVITE messages received from the external proxy: Figure 8-35: Flowchart of INVITE from Primary Proxy in SAS Normal State Version 6.2 February 2011...
Page 286: Sas Routing In Emergency State
MediaPack Series 8.3.2.2 SAS Routing in Emergency State The flowchart below shows the routing logic for SAS in emergency state: Figure 8-36: Flowchart for SAS Emergency State SIP User's Manual Document #: LTRT-65415...
Page 287: Sas Configuration
The SAS application is available only if the device is installed with the SAS Software Upgrade Key. If your device is not installed with the SAS feature, contact your AudioCodes representative. To enable the SAS application: Open the 'Applications Enabling' page (Configuration tab > VoIP menu >...
Page 288: Configuring Common Sas Parameters
MediaPack Series 8.3.3.1.2 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). The SAS Proxy Set ID defines the address of the UAs' external proxy.
Page 289: Figure 8-38: Configuring Common Settings
SIP User's Manual 8. IP Telephony Capabilities Figure 8-38: 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 290: Configuring Sas Outbound Mode
MediaPack Series In the 'Proxy Address' field, enter the IP address of the external proxy server. 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.
Page 291: Configuring Sas Redundant Mode
SIP User's Manual 8. IP Telephony Capabilities 8.3.3.3 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 288. Note: The VoIP CPEs (such as IP phones or residential gateways) need to be defined so that their primary proxy is the external proxy, and their redundant proxy destination addresses and port is the same as that configured for the...
Page 292: Figure 8-40: Enabling Proxy Server For Gateway Application
MediaPack Series 8.3.3.4.1 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 & Registration' page (Configuration tab > VoIP menu > SIP Definitions submenu >...
Page 293: Figure 8-42: Disabling User=Phone In Sip Url
SIP User's Manual 8. IP Telephony Capabilities 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 294: Figure 8-44: Defining Proxy Servers For Gateway Application
MediaPack Series In the second 'Proxy Address' field, enter the IP address and port of the device (in the format x.x.x.x:port). This is the same port as defined in the 'SAS Local UDP/TCP/TLS Port' field (see ''Configuring Common SAS Parameters'' on page 288).
Page 295: Advanced Sas Configuration
Contact: <sip: 976653434@10.10.10.10:5050>;expires=180 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE, UPDATE Expires: 180 User-Agent: Audiocodes-Sip-Gateway-/v. Content-Length: 0 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) Version 6.2...
Page 296: Figure 8-46: Manipulating User Part In Incoming Register
MediaPack Series Contact: 976653434@10.10.10.10 The procedure below describes how to configure the manipulation example scenario above (relevant ini parameter is SASRegistrationManipulation): To manipulate incoming Request-URI user part of REGISTER message: Open the 'SAS Configuration' page (Configuration tab > VoIP menu > SAS > Stand Alone Survivability).
Page 297: Figure 8-47: Manipulating Invite Destination Number
SIP User's Manual 8. IP Telephony Capabilities For example, in SAS emergency state, assume an incoming INVITE has a destination number "7001234" which is destined to a user whose registered in the SAS database as "552155551234". In this scenario, the received destination number needs to be manipulated to the number "552155551234".
Page 298: Figure 8-48: Blocking Unregistered Sas Users
MediaPack Series 8.3.3.5.3 SAS Routing Based on SAS Routing Table SAS routing based on rules configured in the SAS Routing table is applicable for SAS in the following states: SAS in normal state, if the SASSurvivabilityMode parameter is set to 4 SAS in emergency state, if the SASSurvivabilityMode parameter is not set to 4 The SAS routing rule destination can be an IP Group, IP address, Request-URI, or ENUM query.
Page 299: Figure 8-49: Configuring Sas Emergency Numbers
SIP User's Manual 8. IP Telephony Capabilities 8.3.3.5.5 Configuring SAS Emergency Calls You can configure SAS to route emergency calls (such as 911 in North America) directly to the PSTN (through its FXO interface). Therefore, even during a communication failure with the external proxy, enterprise UAs can still make emergency calls.
Page 300: Viewing Registered Sas Users
MediaPack Series 8.3.3.5.6 Adding SIP Record-Route Header to SIP INVITE You can configure SAS to add the SIP Record-Route header to SIP requests (e.g. INVITE) received from the enterprise UAs. SAS then sends the request with this header to the proxy.
Page 301: General
SIP User's Manual 8. IP Telephony Capabilities General 8.4.1 Event Notification using X-Detect Header The device supports the sending of notifications to a remote party notifying the occurrence (or detection) of certain events on the media stream. Event detection and notifications is performed using the SIP X-Detect message header and only when establishing a SIP dialog.
Page 302: Table 8-5: Special Information Tones (Sits) Reported By The Device
MediaPack Series Table 8-5: Special Information Tones (SITs) Reported by the device Special Description First Tone Second Tone Third Tone Information Frequency Frequency Frequency Tones (SITs) Duration Duration Duration Name (Hz) (ms) (Hz) (ms) (Hz) (ms) No circuit found 985.2 1428.5 1776.7 Operator intercept...
Page 303 SIP User's Manual 8. IP Telephony Capabilities Below is an example of SIP messages using the X-Detect header: INVITE sip:101@10.33.2.53;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906 Max-Forwards: 70 From: "anonymous" <sip:anonymous@anonymous.invalid>;tag=1c25298 To: <sip:101@10.33.2.53;user=phone> Call-ID: 11923@10.33.2.53 CSeq: 1 INVITE Contact: <sip:100@10.33.2.53> X- Detect: Request=CPT,FAX SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.2.53;branch=z9hG4bKac5906 From: "anonymous"...
Page 304: Supported Radius Attributes
MediaPack Series 8.4.2 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 8-6: Supported RADIUS Attributes Attribute Attribute Value Purpose Example Number Name Format Request Attributes String...
Page 305 SIP User's Manual 8. IP Telephony Capabilities 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 Start Acc...
Page 306: Call Detail Record
MediaPack Series Below is an example of RADIUS Accounting, where the non-standard parameters are preceded with brackets. Accounting-Request (361) user-name = 111 acct-session-id = 1 nas-ip-address = 212.179.22.213 nas-port-type = 0 acct-status-type = 2 acct-input-octets = 4841 acct-output-octets = 8800 acct-session-time = 1 acct-input-packets = 122 acct-output-packets = 220...
Page 307 SIP User's Manual 8. IP Telephony Capabilities Field Name Description DestIp Destination IP Address Source Phone Number Type Source Phone Number Plan SrcPhoneNum Source Phone Number Source Number Before Manipulation SrcNumBeforeMap Destination Phone Number Type Destination Phone Number Plan DstPhoneNum Destination Phone Number DstNumBeforeMap Destination Number Before Manipulation...
Page 308: Release Reasons In Cdr
MediaPack Series 8.4.3.2 Release Reasons in CDR The possible reasons for call termination which is represented in the CDR field TrmReason are listed below: "REASON N/A" "RELEASE_BECAUSE_NORMAL_CALL_DROP" "RELEASE_BECAUSE_DESTINATION_UNREACHABLE" "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"...
Page 309 SIP User's Manual 8. IP Telephony Capabilities "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 310: Rtp Multiplexing (Throughpacket)
MediaPack Series "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" 8.4.4 RTP Multiplexing (ThroughPacket) The device supports a proprietary method to aggregate RTP streams from several channels. This reduces the bandwidth overhead caused by the attached Ethernet, IP, UDP, and RTP headers and reduces the packet/data transmission rate.
Page 311: 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. Ethernet Interface Configuration The device's Ethernet connection can be configured (using the ini file parameter EthernetPhyConfiguration) for one of the following modes: Manual mode: •...
Page 312: Stun
MediaPack Series The following figure illustrates the device's supported NAT architecture. Figure 9-1: Nat Functioning The design of SIP creates a problem for VoIP traffic to pass through NAT. SIP uses IP addresses and port numbers in its message body and the NAT server can’t modify SIP messages and therefore, can’t change local to global addresses.
Page 313: First Incoming Packet Mechanism
SIP User's Manual 9. VoIP Networking Capabilities To enable STUN, perform the following: Enable the STUN feature (by setting the ini file parameter EnableSTUN to 1). Define the STUN server address using one of the following methods: • Define the IP address of the primary and the secondary (optional) STUN servers (using the ini file parameters STUNServerPrimaryIP and STUNServerSecondaryIP).
Page 314: Ip Multicasting
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 333). 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 315: Simple Network Time Protocol Support
SIP User's Manual 9. VoIP Networking Capabilities Simple Network Time Protocol Support The Simple Network Time Protocol (SNTP) client functionality generates requests and reacts to the resulting responses using the NTP version 3 protocol definitions (according to RFC 1305). Through these requests and responses, the NTP client synchronizes the system time to a time source within the network, thereby eliminating any potential issues should the local system clock 'drift' during operation.
Page 316: Network Configuration
MediaPack Series Network Configuration The device allows you to configure up to 16 different IP addresses with associated VLANs, using the Multiple Interface table. Complementing this table is the Routing table, which allows you to define static routing rules for non-local hosts/subnets. This section describes the various network configuration options offered by the device.
Page 317: Overview Of Multiple Interface Table
SIP User's Manual 9. VoIP Networking Capabilities 9.7.1.1 Overview of Multiple Interface Table The Multiple Interfaces scheme allows you to define up to 16 different IP addresses and VLANs in a table format, as shown below: Table 9-1: Multiple Interface Table Index Prefix Default...
Page 318: Table 9-2: Application Types
MediaPack Series 9.7.1.2.2 Application Types Column This column defines the types of applications that are allowed on this interface: OAMP – Operations, Administration, Maintenance and Provisioning applications such as Web, Telnet, SSH, SNMP CONTROL – Call Control protocols (i.e., SIP) MEDIA –...
Page 319: Table 9-3: Configured Default Gateway Example
SIP User's Manual 9. VoIP Networking Capabilities OAMP Interface Address when Booting using BootP/DHCP: When booting using BootP/DHCP protocols, an IP address is obtained from the server. This address is used as the OAMP address for this session, overriding the address configured using the Multiple Interface table.
Page 320: Other Related Parameters
MediaPack Series 9.7.1.3 Other Related Parameters The Multiple Interface table allows you to configure interfaces and their related parameters such as VLAN ID, or interface name. This section lists additional parameters complementing this table functionality. 9.7.1.3.1 Booting using DHCP The DHCPEnable parameter enables the device to boot while acquiring an IP address from a DHCP server.
Page 321: 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) VlanNetworkServiceClassPriority Sets the priority for the Network service class content Sets the priority for the Premium service class content VLANPremiumServiceClassMediaPriority (media traffic) Sets the priority for the Premium service class content...
Page 322: Table 9-6: Traffic/Network Types And Priority
MediaPack Series The mapping of an application to its CoS and traffic type is shown in the table below: Table 9-6: Traffic/Network Types and Priority Application Traffic / Network Types Class-of-Service (Priority) Debugging interface Management Bronze Management Bronze Telnet DHCP Management Network Web server (HTTP)
Page 323: Multiple Interface Table Configuration Summary And Guidelines
SIP User's Manual 9. VoIP Networking Capabilities Table 9-7: Application Type Parameters Parameter Description EnableDNSasOAM Determines the application type for DNS services. [1] = OAMP (default) [0] = Control. Note: For this parameter to take effect, a device reset is required. EnableNTPasOAM Determines the application type for NTP services.
Page 324: Troubleshooting The Multiple Interface Table
MediaPack Series VLANs become available only when booting the device from flash. When booting using BootP/DHCP protocols, VLANs are disabled to allow easier maintenance access. In this scenario, multiple network interface capabilities are unavailable. The 'Native' VLAN ID may be defined using the 'VlanNativeVlanId' parameter. This relates untagged incoming traffic as if reached with a specified VLAN ID.
Page 325: Static Routing Table
SIP User's Manual 9. VoIP Networking Capabilities 9.7.2 Static Routing Table The IP Routing table allows you to configure static routing rules. You may define up to 30 different routing rules, using the ini file, Web interface, and SNMP. 9.7.2.1 Routing Table Overview The IP Routing table consists of the following: Table 9-8: IP Routing Table Layout...
Page 326: Routing Table Configuration Summary And Guidelines
MediaPack Series 9.7.2.2.4 Interface Column This column defines the interface index (in the Multiple Interface table) from which the gateway address is reached. Figure 9-3: Interface Column 9.7.2.2.5 Metric Column The Metric column must be set to 1 for each static routing rule. 9.7.2.2.6 State Column The State column displays the state of each static route.
Page 327: Troubleshooting The Routing Table
SIP User's Manual 9. VoIP Networking Capabilities 9.7.2.4 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 328: Setting Up Voip Networking
MediaPack Series 9.7.3 Setting Up VoIP Networking 9.7.3.1 Using the Web Interface The Web interface is a convenient user interface for configuring the device's network configuration. 9.7.3.2 Using the ini File When configuring the network configuration using the ini File, use a textual presentation of the Interface and Routing Tables, as well as some other parameters.
Page 329: Networking Configuration Examples
SIP User's Manual 9. VoIP Networking Capabilities Values for the Class Of Service parameters are assigned. The DNS application is configured to act as an OAMP application and the NTP application is configured to act as an OAMP application. Notes: •...
Page 330: Table 9-11: Multiple Interface Table - Example2
MediaPack Series The corresponding ini file configuration is shown below: ; Interface Table Configuration: [InterfaceTable] FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes, InterfaceTable_InterfaceMode, InterfaceTable_IPAddress, InterfaceTable_PrefixLength, InterfaceTable_Gateway, InterfaceTable_VlanID, InterfaceTable_InterfaceName; InterfaceTable 0 = 6, 10, 192.168.85.14, 16, 192.168.0.1, 1, myInterface; [\InterfaceTable] ; Routing Table Configuration: [ StaticRouteTable ] FORMAT StaticRouteTable_Index = StaticRouteTable_InterfaceName, StaticRouteTable_Destination, StaticRouteTable_PrefixLength,...
Page 331: Table 9-13: Multiple Interface Table - Example 3
MediaCntrl2 Control Manual VLANs are required. The 'Native' VLAN ID is the same VLAN ID as the AudioCodes Management interface (index 0). One routing rule is required to allow remote management from a host in 176.85.49.0/24: Table 9-14: Routing Table - Example 3...
Page 332 MediaPack Series The corresponding ini file configuration is shown below: ; Interface Table Configuration: [InterfaceTable] FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes, InterfaceTable_InterfaceMode, InterfaceTable_IPAddress, InterfaceTable_PrefixLength, InterfaceTable_Gateway, InterfaceTable_VlanID, InterfaceTable_InterfaceName; InterfaceTable 0 = 0, 10, 192.168.85.14, 16, 192.168.0.1, 1, Mgmt; InterfaceTable 1 = 5, 10, 200.200.85.14, 24, 200.200.85.1, 201, MediaCntrl1;...
Page 333: Configuration Parameters Reference
SIP User's Manual 10. Configuration Parameters Reference Configuration Parameters Reference The device's configuration parameters, default values, and their descriptions are documented in this section. Parameters and values enclosed in square brackets ([...]) represent the ini file parameters and their enumeration values; parameters not enclosed in square brackets represent their corresponding Web interface and/or EMS parameters.
Page 334: 10.1.2 Multiple Network Interfaces And Vlan Parameters
MediaPack Series Parameter Description 802.1xPassword is ignored. Note: The configured mode must match the configuration of the Access server (e.g., RADIUS server). Web: 802.1x Username Username for IEEE 802.1x support. EMS: User Name The valid value is a string of up to 32 characters. The default is an [802.1xUsername] empty string.
Page 335 SIP User's Manual 10. Configuration Parameters Reference Parameter Description 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 336 MediaPack Series Parameter Description VLAN Parameters Web/EMS: VLAN Mode Enables the VLAN functionality. [VLANMode] [0] Disable (default). [1] Enable = VLAN tagging (IEEE 802.1Q) is enabled. Notes: For this parameter to take effect, a device reset is required. To operate with multiple network interfaces, VLANs must be activated.
Page 337: 10.1.3 Static Routing Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.1.3 Static Routing Parameters The static routing parameters are described in the table below. Table 10-3: Static Routing Parameters Parameter Description Static IP Routing Table [StaticRouteTable] You can define up to 30 static IP routing rules for the device. These rules can be associated with IP interfaces defined in the Multiple Interface table (InterfaceTable parameter).
Page 338: Table 10-4: Qos Parameters
MediaPack Series Table 10-4: QoS Parameters Parameter Description Layer-2 Class Of Service (CoS) Parameters (VLAN Tag Priority Field) Web: Network Priority Defines the VLAN priority (IEEE 802.1p) for Network EMS: Network Service Class Priority Class of Service (CoS) content. [VLANNetworkServiceClassPriority] The valid range is 0 to 7.
Page 339: 10.1.5 Nat And Stun Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Bronze QoS Defines the DiffServ value for the Bronze CoS EMS: Bronze Service Class Diff Serv content (OAMP applications). [BronzeServiceClassDiffServ] The valid range is 0 to 63. The default value is 10. 10.1.5 NAT and STUN Parameters The Network Address Translation (NAT) and Simple Traversal of UDP through NAT (STUN) parameters are described in the table below.
Page 340 MediaPack Series Parameter Description NAT Parameters Web/EMS: NAT Traversal Enables or disables the NAT mechanism. [DisableNAT] [0] Enable [1] Disable (default) Note: The compare operation that is performed on the IP address is enabled by default and is configured by the parameter EnableIPAddrTranslation.
Page 341: 10.1.6 Nfs Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.1.6 NFS Parameters The Network File Systems (NFS) configuration parameters are described in the table below. Table 10-6: NFS Parameters Parameter Description [NFSBasePort] Start of the range of numbers used for local UDP ports used by the NFS client.
Page 342: 10.1.7 Dns Parameters
MediaPack Series 10.1.7 DNS Parameters The Domain name System (DNS) parameters are described in the table below. Table 10-7: 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 343: 10.1.8 Dhcp Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description 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 344: 10.1.9 Ntp And Daylight Saving Time Parameters
MediaPack Series Parameter Description EMS: DHCP Speed Factor Determines the DHCP renewal speed. [DHCPSpeedFactor] [0] = Disable [1] = Normal (default) [2] to [10] = Fast When set to 0, the DHCP lease renewal is disabled. Otherwise, the renewal time is divided by this factor. Some DHCP-enabled routers perform better when set to 4.
Page 345: Web And Telnet Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description Daylight Saving Time Parameters Web: Day Light Saving Time Determines whether to enable daylight saving time. EMS: Mode [0] Disable (default) [DayLightSavingTimeEnable] [1] Enable Web: Start Time Defines the date and time when daylight saving begins. EMS: Start The format of the value is mo:dd:hh:mm (month, day, hour, and [DayLightSavingTimeStart]...
Page 346: 10.2.2 Web Parameters
MediaPack Series Parameter Description Web: Use RADIUS for Uses RADIUS queries for Web and Telnet interface authentication. Web/Telnet Login [0] Disable (default). EMS: Web Use Radius [1] Enable. Login When enabled, logging in to the device's Web and Telnet embedded [WebRADIUSLogin] servers is performed through a RADIUS server.
Page 347 SIP User's Manual 10. Configuration Parameters Reference Parameter Description [ResetWebPassword] Resets the username and password of the primary and secondary accounts to their defaults. [0] = Password and username retain their values (default). [1] = Password and username are reset (for the default username and password, see User Accounts).
Page 348: 10.2.3 Telnet Parameters
MediaPack Series 10.2.3 Telnet Parameters The Telnet parameters are described in the table below. Table 10-12: Telnet Parameters Parameter Description Web: Embedded Telnet Server Enables or disables the device's embedded Telnet server. Telnet is EMS: Server Enable disabled by default for security. [TelnetServerEnable] [0] Disable (default) [1] Enable Unsecured...
Page 349 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Enable LAN Determines whether the LAN Watch-Dog feature is enabled. Watchdog [0] Disable = Disable LAN Watch-Dog (default). [EnableLanWatchDog] [1] Enable = Enable LAN Watch-Dog. When LAN Watch-Dog is enabled, the device's overall communication integrity is checked periodically.
Page 350: 10.3.2 Syslog, Cdr And Debug Parameters
MediaPack Series 10.3.2 Syslog, CDR and Debug Parameters The Syslog, CDR and debug parameters are described in the table below. Table 10-14: Syslog, CDR and Debug Parameters Parameter Description Web: Enable Syslog Sends the logs and error message generated by the device to the EMS: Syslog enable Syslog server.
Page 351 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web/EMS: CDR Report Determines whether Call Detail Records (CDR) are sent to the Syslog Level server and when they are sent. [CDRReportLevel] [0] None = CDRs are not used (default). [1] End Call = CDR is sent to the Syslog server at the end of each call.
Page 352 MediaPack Series Parameter Description [19] = local use 3 (local3) [20] = local use 4 (local4) [21] = local use 5 (local5) [22] = local use 6 (local6) [23] = local use 7 (local7) Web: Activity Types to The Activity Log mechanism enables the device to send log messages Report via Activity Log (to a Syslog server) for reporting certain types of Web operations Messages...
Page 353: 10.3.3 Remote Alarm Indication Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.3.3 Remote Alarm Indication Parameters The Remote Alarm Indication (RAI) parameters are described in the table below. Table 10-15: RAI Parameters Parameter Description [EnableRAI] Enables RAI alarm generation if the device's busy endpoints exceed a user-defined threshold.
Page 354: 10.3.5 Bootp Parameters
MediaPack Series Parameter Description EMS: Data Determines the value of the RS-232 data bit. [SerialData] [7] = 7-bit. [8] = 8-bit (default). Note: For this parameter to take effect, a device reset is required. EMS: Parity Determines the value of the RS-232 polarity. [SerialParity] [0] = None (default).
Page 355 SIP User's Manual 10. Configuration Parameters Reference Parameter Description [BootPSelectiveEnable] Enables the Selective BootP mechanism. [1] = Enabled. [0] = Disabled (default). The Selective BootP mechanism (available from Boot version 1.92) enables the device's integral BootP client to filter unsolicited BootP/DHCP replies (accepts only BootP replies that contain the text 'AUDC' in the vendor specific information field).
Page 356: Security Parameters
MediaPack Series 10.4 Security Parameters This subsection describes the device's security parameters. 10.4.1 General Parameters The general security parameters are described in the table below. Table 10-18: General Security Parameters Parameter Description Web: Voice Menu The password for accessing the device's voice menu for configuration and Password status.
Page 357: 10.4.2 Https Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description Notes: This parameter can include up to 50 indices. To configure the firewall using the Web interface and for a description of the parameters of this ini file table parameter, see ''Configuring Firewall Settings'' on page 89.
Page 358: 10.4.3 Srtp Parameters
MediaPack Series Parameter Description [HTTPSRequireClientCertificate] Requires client certificates for HTTPS connection. The client certificate must be preloaded to the device and its matching private key must be installed on the managing PC. Time and date must be correctly set on the device for the client certificate to be verified.
Page 359 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web/EMS: Media Security Determines the device's mode of operation when SRTP is used Behavior (i.e., when the parameter EnableMediaSecurity is set to 1). [MediaSecurityBehaviour] [0] Preferable = The device initiates encrypted calls. If negotiation of the cipher suite fails, an unencrypted call is established.
Page 360: 10.4.4 Tls Parameters
MediaPack Series 10.4.4 TLS Parameters The Transport Layer Security (TLS) parameters are described in the table below. Table 10-21: TLS Parameters Parameter Description Web/EMS: TLS Version Defines the supported versions of SSL/TLS (Secure Socket [TLSVersion] Layer/Transport Layer Security. [0] SSL 2.0-3.0 and TLS 1.0 = SSL 2.0, SSL 3.0, and TLS 1.0 are supported (default).
Page 361: 10.4.5 Ssh Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description terminated. Web: TLS Client Verify Server Determines whether the device, when acting as a client for TLS Certificate connections, verifies the Server certificate. The certificate is EMS: Verify Server Certificate verified with the Root CA information. [VerifyServerCertificate] [0] Disable (default).
Page 362: 10.4.6 Ipsec Parameters
MediaPack Series Parameter Description [0] Disable [1] Enable (default) Note: The last SSH login information is cleared when the device is reset. [SSHMaxSessions] Maximum number of simultaneous SSH sessions. The valid range is 1 to 2. The default is 2 sessions. [SSHRequirePublicKey] Enables or disables RSA public keys for SSH.
Page 363 SIP User's Manual 10. Configuration Parameters Reference Parameter Description [ \IPsecSATable ] For example: IPsecSATable 1 = 0, 10.3.2.73, 0, 123456789, 0, 0, 0, 0, 28800, 3600, ; In the above example, a single IPSec/IKE peer (10.3.2.73) is configured. Pre-shared key authentication is selected, with the pre-shared key set to 123456789.
Page 364: 10.4.7 Ocsp Parameters
MediaPack Series 10.4.7 OCSP Parameters The Online Certificate Status Protocol (OCSP) parameters are described in the table below. Table 10-24: OCSP Parameters Parameter Description EMS: OCSP Enable Enables or disables certificate checking using OCSP. [OCSPEnable] [0] = Disable (default). [1] = Enable. For a description of OCSP, refer to the Product Reference Manual.
Page 365 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: AAA Indications Determines the Authentication, Authorization and Accounting EMS: Indications (AAA) indications. [AAAIndications] [0] None = No indications (default). [3] Accounting Only = Only accounting indications are used. Web: Device Behavior Upon Defines the device's response upon a RADIUS timeout.
Page 366: Snmp Parameters
MediaPack Series 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 Vendor Specific Attributes (VSA) section of the received RADIUS...
Page 367 SIP User's Manual 10. Configuration Parameters Reference Parameter Description [SNMPSysOid] Defines the base product system OID. The default is eSNMP_AC_PRODUCT_BASE_OID_D. Note: For this parameter to take effect, a device reset is required. [SNMPTrapEnterpriseOid] Defines a Trap Enterprise OID. The default is eSNMP_AC_ENTERPRISE_OID. The inner shift of the trap in the AcTrap subtree is added to the end of the OID in this parameter.
Page 368 MediaPack Series Parameter Description Web: IP Address Defines the IP address of the remote host used as an EMS: Address SNMP Manager. The device sends SNMP traps to this IP [SNMPManagerTableIP_x] address. Enter the IP address in dotted-decimal notation, e.g., 108.10.1.255.
Page 369: Sip Media Realm Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description This parameter can include up to 10 indices. For a description of this table's individual parameters and for configuring the table using the Web interface, see ''Configuring SNMP V3 Users'' on page 76. For an explanation on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 10.7...
Page 370: Control Network Parameters
MediaPack Series Parameter Description For a detailed description of all the parameters included in this ini file table parameter and for configuring Media Realms using the Web interface, see Configuring Media Realms. For a description on configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194.
Page 371 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Authentication Table EMS: SIP Endpoints > Authentication [Authentication] This ini file table parameter defines a user name and password for authenticating each device port. The format of this parameter is as follows: [Authentication] FORMAT Authentication_Index = Authentication_UserId, Authentication_UserPassword;...
Page 372 MediaPack Series Parameter Description Web: Account Table EMS: SIP Endpoints > Account This ini file table parameter configures the Account table for [Account] registering and/or authenticating (digest) Hunt Groups (e.g., an IP-PBX) to a Serving IP Group (e.g., an Internet Telephony Service Provider - ITSP).
Page 373 SIP User's Manual 10. Configuration Parameters Reference Parameter Description with the next redundant Proxy (default). [1] Homing = device always tries to work with the primary Proxy server (i.e., switches back to the primary Proxy whenever it's available). Note: To use this Proxy Redundancy mechanism, you need to enable the keep-alive with Proxy option, by setting the parameter EnableProxyKeepAlive to 1 or 2.
Page 374 MediaPack Series Parameter Description Standard mode [0]. When this parameter is set to [2] and the INVITE fails, the device re-routes the call according to the Standard mode [0]. If DNS resolution fails, the device attempts to route the call to the Proxy. If routing to the Proxy also fails, the Redirect/Transfer request is rejected.
Page 375 SIP User's Manual 10. Configuration Parameters Reference Parameter Description returns two domain names and the A-record queries return two IP addresses each, no additional searches are performed. If set to NAPTR [2], an NAPTR query is performed. If it is successful, an SRV query is sent according to the information received in the NAPTR response.
Page 376 MediaPack Series Parameter Description Web/EMS: Mutual Authentication Determines the device's mode of operation when Mode Authentication and Key Agreement (AKA) Digest [MutualAuthenticationMode] Authentication is used. [0] Optional = Incoming requests that don't include AKA authentication information are accepted (default). [1] Mandatory = Incoming requests that don't include AKA authentication information are rejected.
Page 377 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Proxy Set Table EMS: Proxy Set [ProxySet] This ini file table parameter configures the Proxy Set ID table. It is used in conjunction with the ProxyIP ini file table parameter, which defines the IP addresses per Proxy Set ID.
Page 378 MediaPack Series Parameter Description Web: Registrar IP Address The IP address (or FQDN) and port number (optional) of the EMS: Registrar IP Registrar server. The IP address is in dotted-decimal notation, [RegistrarIP] e.g., 201.10.8.1:<5080>. Notes: If not specified, the REGISTER request is sent to the primary Proxy server.
Page 379 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Registration Time Threshold Defines a threshold (in seconds) for re-registration timing. If EMS: Time Threshold this parameter is greater than 0, but lower than the computed [RegistrationTimeThreshold] re-registration timing (according to the parameter RegistrationTimeDivider), the re-registration timing is set to the following: timing set by the Registration server in the SIP Expires header minus the value of the parameter...
Page 380 MediaPack Series Parameter Description Web/EMS: Authentication Mode Determines the device's registration and authentication [AuthenticationMode] method. [0] Per Endpoint = Registration and authentication is performed separately for each endpoint. [1] Per Gateway = Single registration and authentication for the entire device (default). [3] Per FXS = Registration and authentication for FXS endpoints.
Page 381 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web/EMS: Add Empty Authorization Determines whether the SIP Authorization header is included Header in initial registration (REGISTER) requests sent by the device. [EmptyAuthorizationHeader] [0] Disable (default) [1] Enable The Authorization header carries the credentials of a user agent (UA) in a request to a server.
Page 382 MediaPack Series Parameter Description [UsePingPongKeepAlive] Determines whether the carriage-return and line-feed sequences (CRLF) Keep-Alive mechanism, according to RFC 5626 “Managing Client-Initiated Connections in the Session Initiation Protocol (SIP)” is used for reliable, connection- orientated transport types such as TCP. [0] Disable (default) [1] Enable The SIP user agent/client (i.e., device) uses a simple periodic message as a keep-alive mechanism to keep their flow to the...
Page 383: General Sip Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.9 General SIP Parameters The general SIP parameters are described in the table below. Table 10-29: General SIP Parameters Parameter Description Web/EMS: Max SIP Defines the maximum size (in Kbytes) for each SIP message that can Message Length [KB] be sent over the network.
Page 384 MediaPack Series Parameter Description PL=<receive packet loss> JI=<jitter in ms> LA=<latency in ms> Below is an example of the X-RTP-Stat header in a SIP BYE message: BYE sip:302@10.33.4.125 SIP/2.0 Via: SIP/2.0/UDP 10.33.4.126;branch=z9hG4bKac2127550866 Max-Forwards: 70 From: <sip:401@10.33.4.126;user=phone>;tag=1c2113553324 To: <sip:302@company.com>;tag=1c991751121 Call-ID: 991750671245200001912@10.33.4.125 CSeq: 1 BYE X-RTP-Stat: PS=207;OS=49680;;PR=314;OR=50240;PL=0;JI=600;LA=40;...
Page 385 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Minimum Session- Defines the time (in seconds) that is used in the Min-SE header. This Expires header defines the minimum time that the user agent refreshes the EMS: Minimal Session session.
Page 386 MediaPack Series Parameter Description When this parameter is set to 1, 2, or 3, the parameter FaxTransportMode is ignored. 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.
Page 387 SIP User's Manual 10. Configuration Parameters Reference Parameter Description 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. While trying to send a SIP message connection, reuse policy determines whether live connections to the specific destination are re- used.
Page 388 MediaPack Series 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. If a PSTN Redirect number is received, it is added as an additional History-Info header with an appropriate reason.
Page 389 SIP User's Manual 10. Configuration Parameters Reference Parameter Description call) contains “tgrp=<source trunk group ID>;trunk- context=<gateway IP address>”. The <source trunk group ID> is the Hunt Group ID where incoming calls from Tel is received. For IP-Tel calls, the SIP 200 OK device's response contains “tgrp=<destination trunk group ID>;trunk-context=<gateway IP address>”.
Page 390 MediaPack Series Parameter Description [1] Enable A GRUU is a SIP URI that routes to an instance-specific UA and can be reachable from anywhere. There are a number of contexts in which it is desirable to have an identifier that addresses a single UA (using GRUU) rather than the group of UA’s indicated by an Address of Record (AOR).
Page 391 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- MediaPack/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 392 MediaPack Series Parameter Description EMS: Enable P Time Determines whether the 'ptime' attribute is included in the SDP. [EnablePtime] [0] = Remove the 'ptime' attribute from SDP. [1] = Include the 'ptime' attribute in SDP (default). Web/EMS: 3xx Behavior Determines the device's behavior regarding call identifiers when a 3xx [3xxBehavior] response is received for an outgoing INVITE request.
Page 393 SIP User's Manual 10. Configuration Parameters Reference Parameter Description header. If P-Asserted-Identity is selected, the Privacy header is checked and if the Privacy is set to 'id', the calling number is assumed restricted. 'FROM' = Use the source number received in the From header. [SelectSourceHeaderFor Determines the SIP header used for obtaining the called number CalledNumber]...
Page 394 MediaPack Series Parameter Description Web/EMS: Gateway Name Assigns a name to the device (e.g., 'device123.com'). Ensure that the [SIPGatewayName] name you choose is the one with which the Proxy is configured to identify the device. Note: If specified, the device name is used as the host part of the SIP URI in the From header.
Page 395 SIP User's Manual 10. Configuration Parameters Reference Parameter Description EMS: Use URL In Refer To Defines the source for the SIP URI set in the Refer-To header of Header outgoing REFER messages. [UseAORInReferToHeade [0] = Use SIP URI from Contact header of the initial call (default). [1] = Use SIP URI from To/From header of the initial call.
Page 396 MediaPack Series Parameter Description Web/EMS: Progress For Analog (FXS/FXO) interfaces: Indicator to IP [-1] Not Configured (default) = Default values are used. The default [ProgressIndicator2IP] for FXO interfaces is 1; The default for FXS interfaces is 0. [0] No PI = For IP-to-Tel calls, the device sends a 180 Ringing response to IP after placing a call to a phone (FXS) or PBX (FXO).
Page 397 SIP User's Manual 10. Configuration Parameters Reference Parameter Description prefix, removing the "ext=" parameter, and adding the extension number as the suffix (e.g., e622125519100104). Once modified, the device can then manipulate the number further, using the Number Manipulation tables (see ''Number Manipulation and Routing Parameters'' on page 474) to leave only the last 3 digits (for example) for sending to a PBX.
Page 398 MediaPack Series Parameter Description Notes: It is assumed that all Ringback tones are defined in sequence in the CPT file. In case of an MLPP call, the device uses the value of this parameter plus 1 as the index of the Ringback tone in the CPT file (e.g., if this value is set to 1, then the index is 2, i.e., 1 + 1).
Page 399 SIP User's Manual 10. Configuration Parameters Reference 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 400 MediaPack Series Parameter Description used. This parameter is applicable only to FXO interfaces. Out-of-Service (Busy Out) Parameters Web/EMS: Enable Busy Determines whether the Busy Out feature is enabled. [0] Disable = 'Busy out' feature is not used (default). [EnableBusyOut] [1] Enable = 'Busy out' feature is enabled. When Busy Out is enabled and certain scenarios exist, the device performs the following: A reorder tone (configured by the parameter FXSOOSBehavior) is...
Page 401 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Retransmission Parameters Web: SIP T1 The time interval (in msec) between the first transmission of a SIP Retransmission Timer message and the first retransmission of the same message. [msec] The default is 500. EMS: T1 RTX Note: The time interval between subsequent retransmissions of the [SipT1Rtx]...
Page 402: 10.10 Coders And Profile Parameters
MediaPack Series 10.10 Coders and Profile Parameters The profile parameters are described in the table below. Table 10-30: Profile Parameters Parameter Description Web: Coders Table/Coder Group Settings EMS: Coders Group [CodersGroup0] This ini file table parameter defines the device's coders. Up to five groups of [CodersGroup1] coders can be defined, where each group can consist of up to 10 coders.
Page 403 SIP User's Manual 10. Configuration Parameters Reference Parameter Description G.711A- 10, 20 (default), 30, Always Dynamic (0- law_VBD 40, 50, 60, 80, 100, 127) [g711AlawVbd] G.711U- 10, 20 (default), 30, Always Dynamic (0- law_VBD 40, 50, 60, 80, 100, 127) [g711UlawVbd] EG.711 A-law 10 (default), 20, 30...
Page 404 MediaPack Series Parameter Description VBD coders), see ''V.152 Support'' on page 253. For a description of using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194. Web: IP Profile Settings Table EMS: Protocol Definition > IP Profile [IPProfile] This ini file table parameter configures the IP Profile table.
Page 405 SIP User's Manual 10. Configuration Parameters Reference Parameter Description IpProfile_CodersGro Coder Group CodersGroup upID IpProfile_IsFaxUsed Fax Signaling Method IsFaxUsed Dynamic Jitter Buffer DJBufMinDelay IpProfile_JitterBufMi nDelay Minimum Delay IpProfile_JitterBufO Dynamic Jitter Buffer DJBufOptFactor ptFactor Optimization Factor IpProfile_IPDiffServ RTP IP DiffServ PremiumServiceClas sMediaDiffServ IpProfile_SigIPDiffS Signaling DiffServ...
Page 406 MediaPack Series Parameter Description IpProfile_Disconnec Disconnect on Broken DisconnectOnBroken tOnBrokenConnecti Connection Connection IpProfile_FirstTxDtm First Tx DTMF Option TxDTMFOption fOption IpProfile_SecondTx Second Tx DTMF TxDTMFOption DtmfOption Option IpProfile_RxDTMFO Declare RFC 2833 in RxDTMFOption ption Enable Hold EnableHold IpProfile_EnableHol IpProfile_InputGain Input Gain InputGain IpProfile_VoiceVolu Voice Volume...
Page 407 SIP User's Manual 10. Configuration Parameters Reference Parameter Description IpProfile_SBCDivers Diversion Mode ionMode IpProfile_SBCHistor History Info Mode yInfoMode The parameter IpPreference determines the priority of the IP Profile (1 to 20, where 20 is the highest preference). 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 Profiles) of the preferred profile are applied to that call.
Page 408 MediaPack Series Parameter Description TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP, TelProfile_TimeForReorderTone, TelProfile_EnableDIDWink, TelProfile_IsTwoStageDial, TelProfile_DisconnectOnBusyTone, TelProfile_EnableVoiceMailDelay, TelProfile_DialPlanIndex, TelProfile_Enable911PSAP, TelProfile_SwapTelToIpPhoneNumbers, TelProfile_EnableAGC, TelProfile_ECNlpMode; 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 409 SIP User's Manual 10. Configuration Parameters Reference Parameter Description TelProfile_EnableCu Enable Current EnableCurrentDiscon rrentDisconnect Disconnect nect TelProfile_EnableDi Enable Digit Delivery EnableDigitDelivery gitDelivery TelProfile_EnableEC Echo Canceler EnableEchoCanceller TelProfile_MWIAnal MWI Analog Lamp MWIAnalogLamp TelProfile_MWIDispl MWI Display MWIDisplay TelProfile_FlashHoo Flash Hook Period FlashHookPeriod kPeriod TelProfile_EnableEa Enable Early Media...
Page 410: 10.11 Channel Parameters
MediaPack Series Parameter Description parameters take precedence. The parameter EnableVoiceMailDelay is applicable only if voice mail is enabled globally (using the VoiceMailInterface parameter). For a description of using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194. 10.11 Channel Parameters This subsection describes the device's channel parameters.
Page 411 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Answer Detector Silence Currently, not supported. Time [AnswerDetectorSilenceTime] 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.
Page 412: 10.11.2 Coder Parameters
MediaPack Series Parameter Description [ECNLPMode] Defines the echo cancellation Non-Linear Processing (NLP) mode. [0] = NLP adapts according to echo changes (default). [1] = Disables NLP. [2] = Silence output NLP. Note: This parameter can also be configured per Tel Profile, using the TelProfile parameter (see ''Configuring Tel Profiles'' on page 121).
Page 413 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Fax Relay Enhanced Number of times that control packets are retransmitted when Redundancy Depth using the T.38 standard. EMS: Enhanced Relay The valid range is 0 to 4. The default value is 2. Redundancy Depth [FaxRelayEnhancedRedundancy Depth]...
Page 414 MediaPack Series Parameter Description Web/EMS: CNG Detector Mode Determines whether the device detects the fax Calling tone [CNGDetectorMode] (CNG). [0] Disable = The originating device doesn’t detect CNG; the CNG signal passes transparently to the remote side (default). [1] Relay = CNG is detected on the originating side. CNG packets are sent to the remote side according to T.38 (if IsFaxUsed = 1) and the fax session is started.
Page 415 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web/EMS: Modem Bypass Output Defines the modem bypass output gain control. Gain The range is -31 dB to +31 dB, in 1-dB steps. The default is 0 [ModemBypassOutputGain] (i.e., no gain). EMS: NTE Max Duration Maximum time for sending Named Telephony Events (NTEs) to [NTEMaxDuration]...
Page 416 MediaPack Series Parameter Description 'a=rtpmap:100 X-NSE/8000'. To use this feature: The Cisco gateway must include the following definition: 'modem passthrough nse payload-type 100 codec g711alaw'. Set the Modem transport type to Bypass mode (VxxModemTransportType is set to 2) for all modems. Configure the gateway parameter NSEPayloadType = 100.
Page 417: 10.11.4 Dtmf Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: V.34 Modem Transport Type V.90/V.34 Modem Transport Type used by the device. EMS: V34 Transport [0] Disable = Disable (Transparent) [V34ModemTransportType] [1] Enable Relay = N/A [2] Enable Bypass = (default) [3] Events Only = Transparent with Events Note: This parameter can also be configured per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on...
Page 418: 10.11.5 Rtp, Rtcp And T.38 Parameters
MediaPack Series Parameter Description EMS: DTMF Length (msec) Time (in msec) for generating DTMF tones to the PSTN side (if [DTMFDigitLength] TxDTMFOption = 1, 2 or 3). It also configures the duration that is sent in INFO (Cisco) messages. The valid range is 0 to 32767. The default value is 100. EMS: Rx DTMF Relay Hang Defines the Voice Silence time (in msec) after playing DTMF or MF Over Time (msec)
Page 419 SIP User's Manual 10. Configuration Parameters Reference Parameter Description RFC2198PayloadType). Notes: The RTP redundancy dynamic payload type can be included in the SDP, by using the parameter EnableRTPRedundancyNegotiation. This parameter can also be configured per IP Profile, using the IPProfile parameter. Web: Enable RTP Redundancy Determines whether the device includes the RTP Negotiation...
Page 420 MediaPack Series Parameter Description [EnableDetectRemoteMACChange] Changes the RTP packets according to the MAC address of received RTP packets and according to Gratuitous Address Resolution Protocol (GARP) messages. [0] = Nothing is changed. [1] = If the device receives RTP packets with a different source MAC address (than the MAC address of the transmitted RTP packets), then it sends RTP packets to this MAC address and removes this IP entry from the...
Page 421 SIP User's Manual 10. Configuration Parameters Reference Parameter Description The default value is 0 (i.e., RTP multiplexing is disabled). For detailed information on RTP multiplexing, see RTP Multiplexing (ThroughPacket) on page 310. Notes: The value of this parameter on the local device must equal the value of BaseUDPPort on the remote device.
Page 422: 10.12 Gateway And Ip-To-Ip Parameters
MediaPack Series 10.12 Gateway and IP-to-IP Parameters 10.12.1 Fax and Modem Parameters The fax and modem parameters are described in the table below. Table 10-36: 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 423 SIP User's Manual 10. Configuration Parameters Reference 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 424: 10.12.2 Dtmf And Hook-Flash Parameters
MediaPack Series 10.12.2 DTMF and Hook-Flash Parameters The DTMF and hook-flash parameters are described in the table below. Table 10-37: DTMF and Hook-Flash Parameters Parameter Description Hook-Flash Parameters Web/EMS: Hook-Flash Code Defines the digit pattern that when received from the Tel [HookFlashCode] side, indicates a Hook Flash event.
Page 425 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Max. Flash-Hook Detection Period Defines the hook-flash period (in msec) for both Tel and [msec] IP sides (per device). For the IP side, it defines the hook- EMS: Flash Hook Period flash period that is reported to the IP.
Page 426 MediaPack Series Parameter Description Web/EMS: Tx DTMF Option Determines a single or several preferred transmit DTMF [TxDTMFOption] negotiation methods. [0] Not Supported = No negotiation - DTMF digits are sent according to the parameters DTMFTransportType and RFC2833PayloadType (default). [1] INFO (Nortel) = Sends DTMF digits according to IETF Internet-Draft draft-choudhuri-sip-info-digit-00.
Page 427 SIP User's Manual 10. Configuration Parameters Reference Parameter Description This parameter can include up two indices. This parameter can also be configured per IP Profile, using the IPProfile parameter (see ''Configuring IP Profiles'' on page 122). For a description on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194.
Page 428 MediaPack Series Parameter Description number can be as follows: d1005, dpp699, p9p300. To add the 'd' and 'p' digits, use the usual number manipulation rules. To use this feature with FXO interfaces, configure the device to operate in one-stage dialing mode. If this parameter is enabled, it is possible to configure the FXS/FXO interface to wait for dial tone per destination phone number (before or during dialing of...
Page 429: 10.12.3 Digit Collection And Dial Plan Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.12.3 Digit Collection and Dial Plan Parameters The digit collection and dial plan parameters are described in the table below. Table 10-38: Digit Collection and Dial Plan Parameters Parameter Description Web/EMS: Dial Plan Index Determines the Dial Plan index to use in the external Dial Plan [DialPlanIndex] file.
Page 430 MediaPack Series Parameter Description can enter the digit 8. An example of a digit map is shown below: 11xS|00T|[1- 7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T In the example above, the last rule can apply to International numbers: 9 for dialing tone, 011 Country Code, and then any number of digits for the local number ('x.').
Page 431: 10.12.4 Voice Mail Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.12.4 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. Note: Voice Mail is applicable only to FXO interfaces. Table 10-39: Voice Mail Parameters Parameter Description...
Page 432 MediaPack Series Parameter Description (that are received in the Refer-To header). The FXO waits for connection of the transfer call and if speech is detected (e.g., "hello") within approximately 2 seconds, the device completes the call transfer by releasing the line;...
Page 433 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: MWI Source Number Determines the calling party's phone number used in the EMS: MWI Source Name Q.931 MWI Setup message to PSTN. If not configured, [MWISourceNumber] the channel's phone number is used as the calling number.
Page 434: 10.12.5 Supplementary Services Parameters
MediaPack Series Parameter Description Web: Disconnect Call Digit Pattern Determines a digit pattern that when received from the Tel EMS: Tel Disconnect Code side, indicates the device to disconnect the call. [TelDisconnectCode] The valid range is a 25-character string. Web: Digit To Ignore Digit Pattern A digit pattern that if received as Src (S) or Redirect (R) EMS: Digit To Ignore numbers is ignored and not added to that number.
Page 435 SIP User's Manual 10. Configuration Parameters Reference Parameter Description configured in this table is sent in the SIP INVITE message's From header. The format of this parameter is as follows: [CallerDisplayInfo] FORMAT CallerDisplayInfo_Index = CallerDisplayInfo_DisplayString, CallerDisplayInfo_IsCidRestricted; [\CallerDisplayInfo] Where, Index = Port number (where 0 depicts Port 1). DisplayString = Caller ID string (up to 18 characters).
Page 436 MediaPack Series Parameter Description calling number and Display text (from IP) are sent to the device's port. For FXO interfaces, the Caller ID signal is detected and sent to IP in the SIP INVITE message (as 'Display' element). For information on the Caller ID table, see Configuring Caller Display Information on page 155.
Page 437 SIP User's Manual 10. Configuration Parameters Reference 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 When this parameter is enabled, the device sends the Caller ID number (calling number) with the cpc code (received in the SIP...
Page 438 MediaPack Series Parameter Description EMS: Caller ID Timing Mode Determines when Caller ID is generated. [AnalogCallerIDTimingMode] [0] = Caller ID is generated between the first two rings (default). [1] = The device attempts to find an optimized timing to generate the Caller ID according to the selected Caller ID type.
Page 439: Call Waiting Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Use Destination As Determines whether the device includes the Called Party Connected Number Number from outgoing Tel calls (after number manipulation) in [UseDestinationAsConnectedNu the SIP P-Asserted-Identity header. The device includes the mber] SIP P-Asserted-Identity header in 180 Ringing and 200 OK responses for IP-to-Tel calls.
Page 440 MediaPack Series Parameter Description EMS: Send 180 For Call Waiting Determines the SIP response code for indicating Call Waiting. [Send180ForCallWaiting] [0] = Use 182 Queued response to indicate call waiting (default). [1] = Use 180 Ringing response to indicate call waiting. Web: Call Waiting Table EMS: SIP Endpoints >...
Page 441 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Time Between Call Waiting Time (in seconds) between consecutive call waiting indications Indications for call waiting. EMS: Call Waiting Time Between The valid range is 1 to 100. The default value is 10. Indications Note: This parameter is applicable only to FXS ports.
Page 442: Call Forwarding Parameters
MediaPack Series 10.12.5.3 Call Forwarding Parameters The call forwarding parameters are described in the table below. Table 10-42: Call Forwarding Parameters Parameter Description Web: Enable Call Forward Determines whether Call Forward is enabled. [EnableForward] [0] Disable = Disable the Call Forward service. [1] Enable = Enable Call Forward service (using REFER) (default).
Page 443 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Forwarding, use the parameter EnableForward. If the parameter FwdInfo_Destination only contains a telephone number and a Proxy isn't used, the 'forward to' phone number must be specified in the 'Tel to IP Routing' (Prefix ini file parameter).
Page 444: Message Waiting Indication Parameters
MediaPack Series 10.12.5.4 Message Waiting Indication Parameters The message waiting indication (MWI) parameters are described in the table below. Table 10-43: MWI Parameters Parameter Description Web: Enable MWI Enables Message Waiting Indication (MWI). EMS: MWI Enable [0] Disable = Disabled (default). [EnableMWI] [1] Enable = MWI service is enabled.
Page 445 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web/EMS: MWI Server Transport Determines the transport layer used for outgoing SIP dialogs Type initiated by the device to the MWI server. [MWIServerTransportType] [-1] Not Configured (default) [0] UDP [1] TCP [2] TLS Note: When set to ‘Not Configured’, the value of the parameter SIPTransportType is used.
Page 446: Call Hold Parameters
MediaPack Series 10.12.5.5 Call Hold Parameters The call hold parameters are described in the table below. Table 10-44: Call Hold Parameters Parameter Description Web/EMS: Enable Hold Allows users (connected to the device) to place a call on hold. [EnableHold] [0] Disable = Disables the Hold service [1] Enable = Enables the Hold service (default) If the Hold service is enabled, a user can place the call on hold (or remove from hold) using the Hook Flash button.
Page 447: Call Transfer Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description [DisableReminderRing] Disables the reminder ring, which notifies the FXS user of a call on hold or a waiting call when the phone is returned to on-hook position. [0] = (default) The reminder ring feature is active. In other words, if a call is on hold or there is a call waiting, and the phone is changed from offhook to onhook, the phone rings (for a duration defined by the CHRRTimeout parameter) to "remind"...
Page 448: Three-Way Conferencing Parameters
EMS: 3 Way Mode feature is used. [3WayConferenceMode] [0] AudioCodes 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 449 SIP User's Manual 10. Configuration Parameters Reference Parameter Description device) in the Refer-To header value in the REFER messages sent by the device to the remote parties. The remote parties join the conference by sending INVITE messages to the conference using this conference URI. [2] On Board = On-board three-way conference.
Page 450: Emergency Call Parameters
MediaPack Series Parameter Description Web: Establish Conference Code Defines the digit pattern, which upon detection, generates the EMS: Establish Code Conference-initiating INVITE when 3-way conferencing is [ConferenceCode] enabled (Enable3WayConference is set to 1). The valid range is a 25-character string. The default is “!” (Hook-Flash).
Page 451: Call Cut-Through Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.12.5.9 Call Cut-Through Parameters The call cut-through parameters are described in the table below. Table 10-48: Call Cut-Through Parameters Parameter Description Web: Enable Calls Cut Enables FXS endpoints to receive incoming IP calls while the port is in off- Through hook state.
Page 452: Direct Inward Dialing Parameters
MediaPack Series Parameter Description HotLineToneDuration = if Hotline is enabled and the phone (connected to the specific port) is off-hooked and no digit is pressed for this user- defined duration (timeout), the device automatically initiates a call to the user-defined destination phone number. The value range is 0 to 60 seconds, with default as 16.
Page 453 SIP User's Manual 10. Configuration Parameters Reference Parameter Description EMS: NTT DID Signalling Determines the type of DID signaling support for NTT (Japan) modem: Form DTMF- or Frequency Shift Keying (FSK)-based signaling. The devices [NTTDIDSignallingForm] 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 454: Mlpp Parameters
MediaPack Series 10.12.5.12 MLPP Parameters The Multilevel Precedence and Preemption (MLPP) parameters are described in the table below. Table 10-51: MLPP Parameters Parameter Description Web/EMS: Call Priority Mode Enables priority call handling. [CallPriorityMode] [0] Disable = Disable (default). [1] MLPP = MLPP Priority Call handling is enabled. MLPP prioritizes call handling whereby the relative importance of various kinds of communications is strictly defined, allowing higher precedence communication at the expense of lower...
Page 455 SIP User's Manual 10. Configuration Parameters Reference Parameter Description EMS: E911 MLPP Behavior Defines the E911 (or Emergency Telecommunication [E911MLPPBehavior] Services/ETS) MLPP Preemption mode: [0] Standard Mode - ETS calls have the highest priority and preempt any MLPP call (default). [1] Treat as routine mode - ETS calls are handled as routine calls.
Page 456: 10.12.6 Answer And Disconnect Supervision Parameters
MediaPack Series Parameter Description Web/EMS: RTP DSCP for MLPP Defines the RTP DSCP for MLPP Flash-Override precedence Flash Override call level. [MLPPFlashOverRTPDSCP] 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 457 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Send Digit Pattern on Connect Defines a digit pattern to send to the Tel side after a SIP EMS: Connect Code 200 OK is received from the IP side. The digit pattern is a [TelConnectCode] user-defined DTMF sequence that is used to indicate an answer signal (e.g., for billing).
Page 458 MediaPack Series Parameter Description Web: Silence Detection Period [sec] Duration of the silence period (in seconds) after which the EMS: Silence Detection Time Out call is disconnected. [FarEndDisconnectSilencePeriod] The range is 10 to 28,800 (i.e., 8 hours). The default is 120 seconds.
Page 459 SIP User's Manual 10. Configuration Parameters Reference 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 [EnableReversalPolarity] (default).
Page 460 MediaPack Series 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 461: 10.12.7 Tone Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.12.7 Tone Parameters This subsection describes the device's tone parameters. 10.12.7.1 Telephony Tone Parameters The telephony tone parameters are described in the table below. Table 10-53: Tone Parameters Parameter Description Web/EMS: Dial Tone Duration [sec] Duration (in seconds) that the dial tone is played.
Page 462 MediaPack Series Parameter Description Web: Hotline Dial Tone Duration Duration (in seconds) of the Hotline dial tone. If no digits are EMS: Hot Line Tone Duration received during this duration, the device initiates a call to a [HotLineToneDuration] user-defined number (configured in the Automatic Dialing table - TargetOfChannel - see Configuring Automatic Dialing on page 154).
Page 463 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Play Ringback Tone to Tel Enables the play of the ringback tone (RBT) to the Tel side EMS: Play Ring Back Tone To Tel and determines the method for playing the RBT. [PlayRBTone2Tel] [0] Don't Play = RBT is not played.
Page 464: Tone Detection Parameters
MediaPack Series Parameter Description FXSPort_Last = end range of FXS ports. SourcePrefix = prefix of the calling number. DestinationPrefix = prefix of the called number. PriorityIndex = index for Distinctive Ringing and Call Waiting tones (default is 0): Ringing tone index = index in the CPT file for playing the ring tone.
Page 465: Metering Tone Parameters
SIP User's Manual 10. Configuration Parameters Reference Parameter Description EMS: SIT Enable Enables or disables SIT detection according to the ITU-T [SITDetectorEnable] recommendation E.180/Q.35. [0] = Disable (default). [1] = Enable. (Applicable to FXO interfaces): SITDetectorEnable = 1 UserDefinedToneDetectorEnable = 1 DisconnectOnBusyTone = 1 (applicable for Busy, Reorder and SIT tones) Note: For this parameter to take effect, a device reset is...
Page 466 MediaPack Series Parameter Description Web: Analog TTX Voltage Determines the metering signal/pulse voltage level (TTX). Level [0] 0V = 0 Vrms sinusoidal bursts EMS: TTX Voltage Level [1] 0.5V = 0.5 Vrms sinusoidal bursts (default) [AnalogTTXVoltageLevel [2] 1V = 1 Vrms sinusoidal bursts Notes: For this parameter to take effect, a device reset is required.
Page 467: 10.12.8 Telephone Keypad Sequence Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.12.8 Telephone Keypad Sequence Parameters The telephony keypad sequence parameters are described in the table below. Table 10-56: Keypad Sequence Parameters Parameter Description Web/EMS: Call Pickup Key Defines the keying sequence for performing a call pick-up. Call [KeyCallPickup] pick-up allows the FXS endpoint to answer another telephone's incoming call by pressing this user-defined sequence of digits.
Page 468 MediaPack Series Parameter Description Hook Flash Parameters Web: Flash Keys Sequence Style Hook flash keys sequence style for FXS interfaces. [FlashKeysSequenceStyle] [0] 0 = Flash hook (default) - only the phone's Flash button is used, according to the following scenarios: During an existing call, if the user presses the Flash button, the call is put on hold;...
Page 469 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Do Not Disturb Keypad sequence that activates the Do Not Disturb option EMS: CF Do Not Disturb (immediately reject incoming calls). [KeyCFDoNotDisturb] To activate the required forward method from the telephone: Dial the user-defined sequence number on the keypad;...
Page 470 MediaPack Series Parameter Description is added as a prefix to the dialed destination number, by using the parameter KeyBlindTransferAddPrefix. Keypad Feature - Call Waiting Parameters Web: Activate Keypad sequence that activates the Call Waiting option. After EMS: Keypad Features CW the sequence is pressed, a confirmation tone is heard.
Page 471: 10.12.9 General Fxo Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.12.9 General FXO Parameters The general FXO parameters are described in the table below. Table 10-57: 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 472 MediaPack Series Parameter Description 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” response. This parameter is applicable only to FXO interfaces. Web: Time to Wait before Dialing [msec] Determines the delay before the device starts dialing on the FXO line in...
Page 473: Fxs Parameters
SIP User's Manual 10. 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 474: Hunt Groups, Number Manipulation And Routing Parameters
MediaPack Series 10.12.11 Hunt Groups, Number Manipulation and Routing Parameters This subsection describes the device's number manipulation and routing parameters. 10.12.11.1 Hunt Groups and Routing Parameters The routing parameters are described in the table below. Table 10-59: Routing Parameters Parameter Description Web: Endpoint Phone Number Table EMS: SIP Endpoints >...
Page 475 SIP User's Manual 10. Configuration Parameters Reference Parameter Description TrunkGroupSettings_MWIInterrogationType; [\TrunkGroupSettings] 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 476 MediaPack Series Parameter Description Notes: For defining the channel select mode per Hunt Group, see ''Configuring Hunt Group Settings'' on page 126. The phone numbers of the device's channels are defined by the TrunkGroup parameter. Web: Default Destination Number Defines the default destination phone number, which is used if [DefaultNumber] the received message doesn't contain a called party number and no phone number is configured in the 'Endpoint Phone Number...
Page 477 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Use Routing Table for Host Determines whether to use the device's routing table to obtain Names and Profiles the URI host name and optionally, an IP profile (per call) even if EMS: Use Routing Table For Host a Proxy server is used.
Page 478 MediaPack Series Parameter Description Web: IP to Hunt Group Routing Table EMS: SIP Routing > IP to Hunt This ini file table parameter configures the routing of IP calls to [PSTNPrefix] Hunt Groups. The format of this parameter is as follows: [PSTNPrefix] FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix, PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix,...
Page 479 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: IP Security Determines the device's policy on accepting or blocking SIP EMS: Secure Call From IP calls (IP-to-Tel calls). This is useful in preventing unwanted SIP [SecureCallsFromIP] calls, SIP messages, and/or VoIP spam. [0] Disable = The device accepts all SIP calls (default).
Page 480: Alternative Routing Parameters
MediaPack Series Parameter Description Note: After the cic prefix is added, the 'IP to Hunt Group 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 Tel.
Page 481 SIP User's Manual 10. Configuration Parameters Reference Parameter Description ''Configuring Alternative Routing (Based on Connectivity and QoS)'' on page 247. To receive quality information (displayed in the 'Quality Status' and 'Quality Info.' fields in ''Viewing IP Connectivity'' on page 190) per destination, this parameter must be set to 2 or 3.
Page 482 MediaPack Series Parameter Description Web: Reasons for Alternative Tel-to-IP Routing Table EMS: Alt Route Cause Tel to IP This ini file table parameter configures SIP call failure reason [AltRouteCauseTel2IP] 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 'Tel to IP Routing' (if a Proxy is not used) or used as a redundant Proxy...
Page 483 SIP User's Manual 10. Configuration Parameters Reference Parameter Description This parameter can include up to 5 indices. This table can be used for example, in scenarios where the destination is busy and the Release Reason #17 is issued or for other call releases that issue the default Release Reason (#3).
Page 484: Number Manipulation Parameters
MediaPack Series 10.12.11.3 Number Manipulation Parameters The number manipulation parameters are described in the table below. Table 10-61: Number Manipulation Parameters Parameter Description Web: Copy Destination Number to Determines whether the device copies the called number Redirect Number to the outgoing SIP Diversion header for Tel-to-IP calls. EMS: Copy Dest to Redirect Number Therefore, the called number is used as a redirect [CopyDest2RedirectNumber]...
Page 485 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: Add Trunk ID as Prefix Determines whether the port number is added as a prefix EMS: Add Port ID As Prefix to the called number for Tel-to-IP calls. [AddPortAsPrefix] [0] No = port number not added as prefix (default). [1] Yes = port number added as prefix If enabled, the port number (single digit in the range 1 to 8for 8-port devices, two digits in the range 01 to 24 for...
Page 486 MediaPack Series Parameter Description Web: Add Number Plan and Type to RPI Determines whether the TON/PLAN parameters are Header included in the Remote-Party-ID (RPID) header. EMS: Add Ton 2 RPI [0] No [AddTON2RPI] [1] Yes (default) If the Remote-Party-ID header is enabled (EnableRPIHeader = 1) and AddTON2RPI = 1, it's possible to configure the calling and called number type and number plan using the Number Manipulation tables...
Page 487 SIP User's Manual 10. Configuration Parameters Reference Parameter Description the other parameters) is applied to the call. Redirect number manipulation for Tel-to-IP calls is not performed if the CopyDest2RedirectNumber parameter is enabled. This parameter copies the received destination number to the outgoing redirect number.
Page 488 MediaPack Series Parameter Description For a description on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194. Web: Destination Phone Number Manipulation Table for IP to Tel Calls EMS: EMS: SIP Manipulations > Destination IP to Telcom [NumberMapIP2Tel] This ini file table parameter manipulates the destination number of IP-to-Tel calls.
Page 489 SIP User's Manual 10. Configuration Parameters Reference Parameter Description [PerformAdditionalIP2TELDestination Enables additional destination number manipulation for Manipulation] IP-to-Tel calls. The additional manipulation is done on the initially manipulated destination number, and this additional rule is also configured in the manipulation table (NumberMapIP2Tel parameter).
Page 490 MediaPack Series Parameter Description To configure manipulation of source numbers for Tel- to-IP calls using the Web interface, see ''Configuring Number Manipulation Tables'' on page 129). For a description on using ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194.
Page 491 SIP User's Manual 10. Configuration Parameters Reference Parameter Description 194. [PerformAdditionalIP2TELSourceMani Enables additional source number manipulation for IP-to- pulation] Tel calls. The additional manipulation is done on the initially manipulated source number, and this additional rule is also configured in the manipulation table (SourceNumberMapIP2Tel parameter).
Page 492: 10.13 Standalone Survivability Parameters
MediaPack Series 10.13 Standalone Survivability Parameters The Stand-alone Survivability (SAS) parameters are described in the table below. Table 10-62: 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 different SIP entities in the local network and then forwards them to the defined proxy.
Page 493 SIP User's Manual 10. Configuration Parameters Reference Parameter Description header, causing all future dialogs in the session to pass through it as well. When this feature is enabled, the SIP Record-Route header includes the URI "lr" parameter. The presence of this parameter indicates loose routing;...
Page 494 MediaPack Series Parameter Description (default). [1] Always Emergency = The SAS application does not use Keep-Alive messages towards the SASProxySet, instead it always operates in Emergency mode (as if no Proxy in the SASProxySet is available). [2] Ignore Register = Use regular SAS Normal/Emergency logic (same as option [0]), but when in Normal mode incoming REGISTER requests are ignored.
Page 495 SIP User's Manual 10. Configuration Parameters Reference Parameter Description Web: SAS Registration Manipulation Table EMS: Stand-Alone Survivability [SASRegistrationManipulation] This ini file table parameter configures the SAS Registration Manipulation table. This table is used by the SAS application to manipulate the SIP Request-URI user part of incoming INVITE messages and of incoming REGISTER request AoR (To header), before saving it to the registered users database.
Page 496: 10.14 Auxiliary And Configuration Files Parameters
MediaPack Series Parameter Description ''Configuring IP2IP Routing Table (SAS)'' on page 163. For a description on configuring ini file table parameters, see ''Configuring ini File Table Parameters'' on page 194. 10.14 Auxiliary and Configuration Files Parameters This subsection describes the device's auxiliary and configuration files parameters. 10.14.1 Auxiliary/Configuration File Name Parameters The configuration files (i.e., auxiliary files) can be loaded to the device using the Web interface or a TFTP session (see ''Loading Auxiliary Files'' on page 170).
Page 497: 10.14.2 Automatic Update Parameters
SIP User's Manual 10. Configuration Parameters Reference 10.14.2 Automatic Update Parameters The automatic update of software and configuration files parameters are described in the table below. Table 10-64: Automatic Update of Software and Configuration Files Parameters Parameter Description General Automatic Update Parameters [AutoUpdateCmpFile] Enables or disables the Automatic Update mechanism for the cmp file.
Page 498 MediaPack Series Parameter Description Software/Configuration File URL Path for Automatic Update Parameters Specifies the name of the cmp file and the path to the server (IP [CmpFileURL] address or FQDN) from where the device loads a new cmp file and updates itself.
Page 499: 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 500 MediaPack Series Reader's Notes SIP User's Manual Document #: LTRT-65415...
Page 501: Selected Technical Specifications
SIP User's Manual 12. Selected Technical Specifications Selected Technical Specifications The main technical specifications of the MP-11x and MP-124 devices are listed in the table below. Note: All specifications in this document are subject to change without prior notice. Table 12-1: MediaPack Technical Specifications Function Specification Interfaces...
Page 502: Document #: Ltrt
MediaPack Series Function Specification Signaling MP-112: FXS Loop-start Signaling MP-114 & MP-118: FXS, FXO Loop-start MP-124: FXS Loop-start DTMF (TIA 464B) In-band Signaling User-defined and call progress tones Out-of-Band Signaling DTMF Relay (RFC 2833), DTMF via SIP INFO Control SIP (RFC 3261) Provisioning BootP, DHCP, TFTP and HTTP for Automatic Installation Protocols...
Page 503 SIP User's Manual 12. Selected Technical Specifications Function Specification Loop Impedance (including Up to 1500 ohm for the MP-11x, Up to 1600 ohm for the MP-124 phone impedance) Lifeline Supported in all ports of Mixed FXS/FXO and in first port of MP- 114/FXS and MP-118/FXS using special Lifeline cable Bellcore GR-30-CORE Type 1 using Bell 202 FSK modulation, ETSI Caller ID...
Page 504 User’s Manual Ver. 6.2 www.audiocodes.com...

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Mediapack mp-11 series

AudioCodes MediaPack MP-124 User Manual

1 Overview

2 Configuration Tools

3 Web-Based Management

4 INI File-Based Management

5 EMS-Based Management

6 Restoring Factory Default Settings

7 Auxiliary Configuration Files

8 IP Telephony Capabilities

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Summary of Contents for AudioCodes MediaPack MP-124