User's Manual Version 5.
SIP User's Manual Contents Table of Contents 1 Overview ........................................................................................................... 15 1.1 Gateway Description ..............................................................................................15 1.2 MediaPack Features ..............................................................................................16 1.2.1 1.2.2 1.3 MP-11x Hardware Features .......................................................
MediaPack Series 3.4.2 3.4.3 3.4.4 3.4.5 3.5 Management Tab .................................................................................................198 3.5.1 3.5.2 3.6 Management Configuration................................................................................... 199 3.5.1.1 Configuring the Management Settings .................................................. 199 3.5.1.2 Configuring the Regional Settings ......................................................... 206 3.5.1.
SIP User's Manual 4.2.4 4.3 4.4 6 Reference for ini File Parameters .........................................................................236 Networking Parameters ........................................................................................ 236 System Parameters............................................................................................... 244 Web and Telnet Parameters .................................................................................
MediaPack Series 7.4.2 7.4.1.3 Call Termination (Disconnect Supervision) on FXO Devices ................ 328 7.4.1.4 DID Wink................................................................................................ 329 Telephone-to-IP Calls ........................................................................................... 330 7.4.2.1 Automatic Dialing ................................................................................... 330 7.4.2.2 Collecting Digits Mode .....................
SIP User's Manual 8.7 IP QoS via Differentiated Services (DiffServ) .......................................................369 8.8 VLANS and Multiple IPs .......................................................................................370 8.8.1 8.8.2 8.8.3 9 Contents Multiple IPs............................................................................................................ 370 IEEE 802.1p/Q (VLANs and Priority) ....................................................................
MediaPack Series List of Figures Figure 1-1: Typical MediaPack VoIP Application................................................................................... 16 Figure 3-1: Enter Network Password Screen ........................................................................................ 22 Figure 3-2: Main Areas of the Web Interface GUI ................................................................................. 23 Figure 3-3: "Reset" Displayed on Toolbar .....................................
SIP User's Manual Contents Figure 3-57: SIP General Parameters Page ........................................................................................ 101 Figure 3-58: Proxy & Registration Page .............................................................................................. 112 Figure 3-59: Proxy Sets Table Page.................................................................................................... 120 Figure 3-60: Coders Page ...............................................
MediaPack Series Figure 3-115: Call Routing Status Page .............................................................................................. 226 Figure 3-116: Registration Status Page............................................................................................... 227 Figure 3-117: SAS Registered Users Page ......................................................................................... 227 Figure 3-118: IP Connectivity Page ..................................................
SIP User's Manual Contents List of Tables Table 1-1: Supported MediaPack Series Configurations....................................................................... 15 Table 3-1: Description of Toolbar Buttons ............................................................................................. 24 Table 3-2: ini File Parameters for Changing Logo Image ...................................................................... 45 Table 3-3: ini File Parameters for Replacing Logo with Text .............
MediaPack Series Table 3-55: SNMP V3 Users Parameters ............................................................................................ 205 Table 3-56: Auxiliary Files Descriptions............................................................................................... 210 Table 3-57: Ethernet Port Information Parameters .............................................................................. 220 Table 3-58: Call Counters Description ...................................................
SIP User's Manual Notices Notice This document describes the AudioCodes MediaPack series 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.
MediaPack Series Related Documentation Document # Manual Name LTRT-523xx (where xx is the document version) Product Reference Manual LTRT-656xx MP-11x & MP-124 SIP Release Notes LTRT-598xx MP-11x & MP-124 SIP Installation Manual LTRT-529xx MP-11x SIP Fast Track Guide LTRT-532xx MP-124 SIP Fast Track Guide LTRT-665xx CPE Configuration Guide for IP Voice Mail Warning: The device is supplied as a sealed unit and must only be serviced by qualified service personnel.
SIP User's Manual 1 1.
MediaPack Series The figure below illustrates a typical MediaPack VoIP application. Figure 1-1: Typical MediaPack VoIP Application 1.2 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.
SIP User's Manual 1.2.2 1. Overview MP-124 Hardware Features The MP-124 hardware features include the following: 1.3 MP-124 19-inch, 1U rugged enclosure provides up to 24 analog FXS ports, using a single 50-pin Telco connector. LEDs on the front panel that provide information on the device's operating status and the network interface. Reset button on the front panel for restarting the MP-124 and for restoring the MP-124 parameters to their factory default settings.
MediaPack Series Reader’s Notes SIP User's Manual 18 Document #: LTRT-65411
SIP User's Manual 2 2. Configuration Concepts Configuration Concepts You can configure the device's parameters (including upgrading the software, and uploading configuration and auxiliary files), using the following tools: An HTTP-based Embedded Web Server (Web interface), using any standard Web browser (described in ''Web-based Management'' on page 21). A configuration file referred to as the ini file (refer to ''ini File Configuration'' on page 231).
MediaPack Series Reader’s Notes SIP User's Manual 20 Document #: LTRT-65411
SIP User's Manual 3 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 configuration (software upgrade) and auxiliary files, and resetting the device.
MediaPack Series ¾ To access the Web interface, take these 4 steps: 1. Open a standard Web browser application. 2. In the Web browser's Uniform Resource Locator (URL) field, specify the device's IP address (e.g., http://10.1.10.10); the Web interface's 'Enter Network Password' dialog box appears, as shown in the figure below: Figure 3-1: Enter Network Password Screen 3. In the 'User Name' and 'Password' fields, enter the case-sensitive, user name and password. 4.
SIP User's Manual 3.3 3. Web-Based Management Getting Acquainted with the Web Interface 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. For replacing the logo with another image or text, refer to ''Replacing the Corporate Logo'' on page 43.
MediaPack Series 3.3.1 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 Name Submit Description Applies parameter settings to the device (refer to ''Saving Configuration'' on page 209). Note: This icon is grayed out when not applicable to the currently opened page. Saves parameter settings to flash memory (refer to ''Saving Configuration'' on page 209).
SIP User's Manual 3.3.2 3. Web-Based Management 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.
MediaPack Series ¾ To navigate to a page, take these 2 steps: 1. 2. 3.3.2.1 Navigate to the required page item, by performing the following: • Drilling-down using the plus signs to expand the menus and submenus • Drilling-up using the minus signs to collapse the menus and submenus Select the required page item; the page opens in the Work pane.
SIP User's Manual 3.3.2.2 3. Web-Based Management 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. The arrow button located just below the Navigation bar is used to hide and show the Navigation pane.
MediaPack Series 3.3.3.1 Accessing Pages The configuration pages are accessed by clicking the required page item in the Navigation tree. ¾ To open a configuration page in the Work pane, take these 2 steps: 1.
SIP User's Manual 3. Web-Based Management 3.3.3.2.1 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.
MediaPack Series 3.3.3.2.2 Showing / Hiding Parameter Groups Some pages provide groups of parameters, which can be hidden or shown. To toggle between hiding and showing a group, simply click the group name button that appears above each group. The button appears with a down-pointing or up-pointing arrow, indicating that it can be collapsed or expanded when clicked, respectively.
SIP User's Manual 3.3.3.3 3. Web-Based Management Modifying and Saving Parameters When you change parameter values on a page, the Edit symbol appears to the right of these parameters. This is especially useful for indicating the parameters that you have currently modified (before applying the changes). After you save your parameter modifications (refer to the procedure described below), the Edit symbols disappear.
MediaPack Series If you enter an invalid parameter value (e.g., not in the range of permitted values) and then click Submit, a message box appears notifying you of the invalid value. In addition, the parameter value reverts to its previous value and is highlighted in red, as shown in the figure below: Figure 3-10: Value Reverts to Previous Valid Value 3.3.3.
SIP User's Manual 3. Web-Based Management ¾ To add an entry to a table, take these 2 steps: 1. In the 'Add' field, enter the desired index entry number, and then click Add; an index entry row appears in the table: Figure 3-11: Adding an Index Entry to a Table 2. Click Apply to save the index entry. Notes: • Before you can add another index entry, you must ensure that you have applied the previously added index entry (by clicking Apply).
MediaPack Series ¾ To organize the index entries in ascending, consecutive order, take the following step: 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. Figure 3-12: Compacting a Web Interface Table ¾ To delete an existing index table entry, take these 3 steps: 3.3.4 1.
SIP User's Manual 3. Web-Based Management 2. In the 'Search' field, enter the parameter name or sub-string of the parameter name that you want to search. If you have performed a previous search for such a parameter, instead of entering the required string, you can use the 'Search History' drop-down list to select the string (saved from a previous search). 3. Click Search; a list of located parameters based on your search appears in the Navigation pane. Each searched result displays the following: 4.
MediaPack Series 3.3.5.1 Creating a Scenario The Web interface allows you to create one Scenario with up to 20 configuration pages, as described in the procedure below: ¾ To create a Scenario, take these 10 steps: 1. 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. 2.
SIP User's Manual 7. 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 8. Repeat steps 5 through 8 to add additional Steps (i.e., pages). 9. When you have added all the required Steps for your Scenario, click the Save & Finish button located at the bottom of the Navigation tree; a message box appears informing you that the Scenario has been successfully created.
MediaPack Series 3.3.5.2 Accessing a Scenario Once you have created the Scenario, you can access it at anytime by following the procedure below: ¾ To access the Scenario, take these 2 steps: 1. On the Navigation bar, select the Scenario tab; a message box appears, requesting you to confirm the loading of the Scenario. Figure 3-16: Scenario Loading Message Box 2.
SIP User's Manual 3. Web-Based Management In an opened Scenario Step (i.e., page appears in the Work pane), use the following navigation buttons: • Next: opens the next Step listed in the Scenario. • Previous: opens the previous Step listed in the Scenario. Note: If you reset the device while in Scenario mode, after the device resets, you are returned once again to the Scenario mode. 3.3.5.3 Editing a Scenario You can modify a Scenario anytime by adding or removing Steps (i.e.
MediaPack Series • • • 3.3.5.4 Edit the Step Name: a. In the Navigation tree, select the required Step. b. In the 'Step Name' field, modify the Step name. c. In the page, click Next. Edit the Scenario Name: a. In the 'Scenario Name' field, edit the Scenario name. b. In the displayed page, click Next. Remove a Step: a. In the Navigation tree, select the required Step; the corresponding page opens in the Work pane. b. In the page, clear all the check boxes corresponding to the parameters. c.
SIP User's Manual 3.3.5.5 3. Web-Based Management 3. Click the Get Scenario File button; the 'File Download' window appears. 4. Click Save, and then in the 'Save As' window navigate to the folder to where you want to save the Scenario file. When the file is successfully downloaded to your PC, the 'Download Complete' window appears. 5. Click Close to close the 'Download Complete' window.
MediaPack Series 3.3.5.6 Deleting a Scenario You can delete the Scenario by using the Delete Scenario File button, as described in the procedure below: ¾ To delete the Scenario, take these 4 steps: 1. On the Navigation bar, click the Scenarios tab; a message box appears, requesting you to confirm: Figure 3-19: Scenario Loading Message Box 2. Click OK; the Scenario mode appears in the Navigation tree. 3.
SIP User's Manual 3.3.5.7 3. Web-Based Management Exiting Scenario Mode When you want to close the Scenario mode after using it for device configuration, follow the procedure below: ¾ To close the Scenario mode, take these 2 steps: 1. Simply click any tab (besides the Scenarios tab) on the Navigation bar, or click the Cancel Scenarios button located at the bottom of the Navigation tree; a message box appears, requesting you to confirm exiting Scenario mode, as shown below.
MediaPack Series 3.3.6.1.1 Replacing the Corporate Logo with an Image You can replace the logo that appears in the Web interface's Title bar, using either the Web interface or the ini file. ¾ To replace the default logo with a different image via the Web interface, take these 7 steps: 1. Access the device's Web interface (refer to ''Accessing the Web Interface'' on page 21). 2. In the URL field, append the case-sensitive suffix ‘AdminPage’ to the IP address (e.g., http://10.1.229.
SIP User's Manual 3. Web-Based Management Tip: If you encounter any problem during the loading of the file or you want to restore the default image, click the Restore Default Images button. ¾ To replace the default logo with a different image using the ini file, take these 3 steps: 1. Place your corporate logo image file on the TFTP server in the same folder where the device’s ini file is located. 2. Configure the ini file parameters as described in the table below.
MediaPack Series 3.3.6.2 Customizing the Product Name You can customize the product name (text) that appears in the Title bar, using the ini file parameters listed in the table below. (For a description on using the ini file, refer to ''Modifying an ini File'' on page 235.) Table 3-4: ini File Parameters for Customizing Product Name Parameter Description UseProductName UserProductName 3.3.6.3 Defines whether or not to change the product name: [0] = Don’t change the product name (default).
SIP User's Manual 3.3.7 3. Web-Based Management 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. ¾ To view the Help topic for a currently opened page, take these 4 steps: 1. Using the Navigation tree, open the required page for which you want Help. 2.
MediaPack Series 3.3.8 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. By default, the 'Home' page is displayed when you access the device's Web interface. When you are configuring the device (in a configuration page), you can always return to the 'Home' page, by simply clicking the Home icon on the toolbar.
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MediaPack Series 3.3.8.2 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. ¾ To view detailed port information, take these 3 steps: 1. Click the port for which you want to view port settings; the shortcut menu appears. Figure 3-30: Shortcut Menu when Clicking Port – Port Settings (e.g. MP-11x) 2. From the shortcut menu, click Port Settings; the 'Basic Channel Information' page appears.
SIP User's Manual 3. Web-Based Management Figure 3-32: Shortcut Menu when Clicking Port – Reset Channel (e.g. MP-11x) 3.3.9 Logging Off the Web Interface You can log off the Web interface and re-access it with a different user account. For detailed information on the Web User Accounts, refer to User Accounts. ¾ To log off the Web interface, take these 2 steps: 1. On the toolbar, click the Log Off appears: button; the 'Log Off' confirmation message box Figure 3-33: Log Off Confirmation Box 2.
MediaPack Series 3.4 Configuration Tab The Configuration tab on the Navigation bar displays all menus related to device configuration.
SIP User's Manual 3. Web-Based Management ¾ To configure the IP settings parameters, take these 4 steps: 1. Open the 'IP Settings' page (Configuration tab > Network Settings menu > IP Settings page item). Figure 3-35: IP Settings Page 2. Configure the IP parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209.
MediaPack Series Parameter Description Single IP Settings IP Address IP address of the device. Enter the IP address in dotted-decimal notation, for example, 10.8.201.1. Notes: Subnet Mask A warning message is displayed (after clicking Submit) if the entered value is incorrect. After changing the IP address, you must reset the device. Subnet mask of the device. Enter the subnet mask in dotted-decimal notation, for example, 255.255.0.0.
SIP User's Manual 3. Web-Based Management Parameter Description Multiple Interface Settings Multiple Interface Table button to open the 'Multiple Interface Click the right-pointing arrow Table' page. For a description of configuring multiple IP interfaces, refer to ''Configuring the Multiple Interface Table'' on page 55. VLAN (For detailed information on the device's VLAN implementation, refer to ''VLANS and Multiple IPs'' on page 370.) VLAN Mode [VlANMode] Enables the VLAN functionality.
MediaPack Series ¾ To configure the multiple IP interface table, take these 7 steps: 1. Open the 'IP Settings' page (refer to ''Configuring the IP Settings'' on page 52). 2. Under the Multiple Interface Settings group, click the right-arrow Multiple Interface Table; a confirmation message box appears: button alongside Figure 3-36: Confirmation Message for Accessing the Multiple Interface Table 3. Click OK to confirm; the 'Multiple Interface Table' page appears: Figure 3-37: Interface Table Page 4.
SIP User's Manual 3. Web-Based Management Table 3-8: Multiple Interface Table Parameters Description Parameter Description Table parameters Index Index of each interface. The range is 0-3. Note: Each interface index must be unique. Types of applications that are allowed on the specific interface. Application Type [0] OAM = Only Operations, Administration, Maintenance and Provisioning (OAMP) applications (e.g., Web, Telnet, SSH, and SNMP) are allowed on the interface. [1] Media = Only Media (i.
MediaPack Series Parameter Description interface address. For configuring additional routing rules for other interfaces, refer to ''Configuring the IP Routing Table'' on page 63. Defines the VLAN ID for each interface. When using VLANs, the VLAN ID must be unique for each interface. Incoming traffic tagged with this VLAN ID is routed to the corresponding interface, and outgoing traffic from that interface is tagged with this VLAN ID.
SIP User's Manual 3. Web-Based Management Figure 3-38: Application Settings Page 2. Configure the Applications parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-9: Application Settings Parameters Parameter Description NTP Settings (For detailed information on Network Time Protocol (NTP), refer to ''Simple Network Time Protocol Support'' on page 369.
MediaPack Series Parameter NTP Update Interval [NTPUpdateInterval] Description Defines the time interval (in seconds) that the NTP client requests for a time update. The default interval is 86400 (i.e., 24 hours). The range is 0 to 214783647. Note: AudioCodes does not recommend setting this parameter to beyond one month (i.e., 2592000 seconds). Telnet Settings Embedded Telnet Server [TelnetServerEnable] Enables or disables the device's embedded Telnet server.
SIP User's Manual 3. Web-Based Management Parameter Description Notes: For defining the STUN server domain name, use the ini file parameter STUNServerDomainName (refer to ''Networking Parameters'' on page 236). This parameter cannot be changed on-the-fly and requires a device reset. STUN Server Primary IP [STUNServerPrimaryIP] Defines the IP address of the primary STUN server. The valid range is the legal IP addresses. The default value is 0.0.0.0.
MediaPack Series 3.4.1.4 Configuring the NFS Settings Network File System (NFS) enables the device to access a remote server's shared files and directories, and to handle them as if they're located locally. You can configure up to five different NFS file systems. As a file system, the NFS is independent of machine types, OSs, and network architectures. NFS is used by the device to load the cmp, ini, and auxiliary files, using the Automatic Update mechanism (refer to Automatic Update Mechanism).
SIP User's Manual 3. Web-Based Management Table 3-10: Network Settings -- NFS Settings Parameters Parameter Description Index The row index of the remote file system. The valid range is 0 to 4. 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. Root Path Path to the root of the remote file system in the format: /[path]. For example, '/audio'. NFS version used to access the remote file system. NFS Version [2] NFS Version 2.
MediaPack Series ¾ To configure static IP routing, take these 3 steps: 1. Open the 'IP Routing Table' page (Configuration tab > Network Settings menu > IP Routing Table page item). Figure 3-40: IP Routing Table Page 2. In the 'Add a new table entry' group, add a new static routing rule according to the parameters described in the table below. 3. Click Add New Entry; the new routing rule is added to the IP routing table.
SIP User's Manual 3. Web-Based Management Parameter Description Multiple Interface Table'' on page 55). Metric [RoutingTableHopsCountColumn] The maximum number of allowed routers (hops) between the device and destination. Note: This parameter must be set to 1 for the routing rule to be valid. Routing entries with Hop Count equals 0 are local routes set automatically by the device. Interface [RoutingTableInterfacesColumn] Specifies the interface (network type) to which the routing rule is applied.
MediaPack Series Table 3-12: QoS Settings Parameters Parameter Description Priority Settings Network Priority [VLANNetworkServiceClassPriority] Defines the priority for Network Class of Service (CoS) content. The valid range is 0 to 7. The default value is 7. Media Premium Priority [VLANPremiumServiceClassMediaPriority] Defines the priority for the Premium CoS content and media traffic. The valid range is 0 to 7. The default value is 6.
SIP User's Manual 3.4.2 3. Web-Based Management Media Settings The Media Settings menu allows you to configure the device's channel parameters. These parameters are applied to all the device's channels.
MediaPack Series 2. Configure the Voice parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-13: Media Settings, Voice Settings Parameters Parameter Description Voice Volume [VoiceVolume] Voice gain control (in decibels). This parameter sets the level for the transmitted (IP-to-Tel) signal. The valid range is -32 to 31 dB. The default value is 0 dB.
SIP User's Manual 3. Web-Based Management Parameter Description DTMF Volume (-31 to 0 dB) [DTMFVolume] DTMF gain control value (in decibels) to the or analog side. The valid range is -31 to 0 dB. The default value is -11 dB. Enable Answer Detector [EnableAnswerDetector] N/A. Answer Detector Activity Delay [AnswerDetectorActivityDelay] N/A. Answer Detector Silence Time [AnswerDetectorSilenceTime] N/A. Answer Detector Redirection [AnswerDetectorRedirection] N/A.
MediaPack Series 3.4.2.2 Configuring the Fax / Modem / CID Settings The 'Fax/Modem/CID Settings' page is used for configuring fax, modem, and Caller ID (CID) parameters. ¾ To configure the fax, modem, and CID parameters, take these 4 steps: 1. Open the 'Fax/Modem/CID Settings' page (Configuration tab > Media Settings menu > Fax/Modem/CID Settings page item). Figure 3-43: Fax/Modem/CID Settings Page 2. Configure the fax, Modem, and CID parameters according to the table below. 3.
SIP User's Manual 3. Web-Based Management Parameter Caller ID Transport Type [CallerIDTransportType] Caller ID Type [CallerIDType] Description Determines the device's behavior for Caller ID detection. [0] Disable = Caller ID is not detected - DTMF digits remain in the voice stream. [1] Relay = Caller ID is detected - DTMF digits are erased from the voice stream. [3] Mute = Caller ID is detected - DTMF digits are erased from the voice stream (default).
MediaPack Series Parameter V.23 Modem Transport Type [V23ModemTransportTy pe] V.32 Modem Transport Type [V32ModemTransportTy pe] Description V.23 Modem Transport Type used by the device. [0] Disable = Disable (Transparent) [1] Enable Relay = N/A [2] Enable Bypass = (default) [3] Events Only = Transparent with Events V.32 Modem Transport Type used by the device.
SIP User's Manual 3. Web-Based Management Parameter Description Fax/Modem Bypass Coder Type [FaxModemBypassCode rType] Coder used by the device when performing fax/modem bypass. Usually, high-bit-rate coders such as G.711 should be used. [0] G.711Alaw= G.711 A-law 64 (default). [1] G.711Mulaw = G.711 μ-law. Fax/Modem Bypass Packing Factor [FaxModemBypassM] Number of (20 msec) coder payloads that are used to generate a fax/modem bypass packet. The valid range is 1, 2, or 3 coder payloads.
MediaPack Series 3.4.2.3 Configuring the RTP / RTCP Settings The 'RTP/RTCP Settings' page allows you to configure the Real-Time Transport Protocol (RTP) and Real-Time Transport (RTP) Control Protocol (RTCP) parameters. ¾ To configure the RTP / RTCP parameters, take these 4 steps: 1. Open the 'RTP/RTCP Settings' page (Configuration tab > Media Settings menu > RTP / RTCP Settings page item). Figure 3-44: RTP / RTCP Settings Page 2. Configure the RTP / RTCP parameters according to the table below. 3.
SIP User's Manual 3. Web-Based Management Parameter Description Packing Factor [RTPPackingFactor] N/A. Controlled internally by the device according to the selected coder. Basic RTP Packet Interval [BasicRTPPacketInterval] N/A. Controlled internally by the device according to the selected coder. RTP Directional Control [RTPDirectionControl] N/A. Controlled internally by the device according to the selected coder. RFC 2833 TX Payload Type [RFC2833TxPayloadType] N/A.
MediaPack Series Parameter Description message is generated: 'invalid local RTP port'. For detailed information on the default RTP/RTCP/T.38 port allocation, refer to the Product Reference Manual. Remote RTP Base UDP Port [RemoteBaseUDPPort] Determines the lower boundary of UDP ports used for RTP, RTCP and T.38 by a remote device. If this parameter is set to a non-zero value, ThroughPacket™ (RTP multiplexing) is enabled.
SIP User's Manual 3. Web-Based Management Figure 3-45: General Media Settings Page 2. Configure the general media parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-16: Media Settings Parameters Parameter Description Max Echo Canceller Length N/A Enable Continuity Tones N/A. 3.4.2.
MediaPack Series Parameter Max. Flash-Hook Detection Period [msec] [FlashHookPeriod] Description Defines the hook-flash period (in msec) for both analog and IP sides. For the IP side, it defines the hook-flash period that is reported to the IP. For the analog side, it defines the following: FXS interfaces: Maximum hook-flash detection period. A longer signal is considered an off-hook or on-hook event.
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MediaPack Series 3.4.3.1 Configuring the Web User Accounts To prevent unauthorized access to the Web interface, two Web user accounts are available (primary and secondary) with assigned user name, password, and access level. When you login to the Web interface, you are requested to provide the user name and password of one of these Web user accounts. If the Web session is idle (i.e.
SIP User's Manual 3. Web-Based Management ¾ To change the Web user accounts attributes, take these 4 steps: 1. Open the 'Web User Accounts' page (Configuration tab > Security Settings menu > Web User Accounts page item). Figure 3-48: 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).
MediaPack Series 2. To change the password of an account, perform the following: a. In the field 'Current Password', enter the current password. b. In the fields 'New Password' and 'Confirm New Password', enter the new password (maximum of 19 case-sensitive characters). c. 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: 3.4.3.
SIP User's Manual 3. Web-Based Management ¾ To add authorized IP addresses for Web and Telnet interfaces access, take these 4 steps: 1. Open the 'Web & Telnet Access List' page (Configuration tab > Security Settings menu > Web & Telnet Access List page item). Figure 3-49: Web & Telnet Access List Page - Add New Entry 2.
MediaPack Series 3.4.3.3 Configuring the Firewall Settings The device provides an internal firewall, allowing you (the security administrator) to define network traffic filtering rules. You can add up to 50 ordered firewall rules. For each packet received on the network interface, the table is scanned from the top down until a matching rule is found. This rule can either deny (block) or permit (allow) the packet. Once a rule in the table is located, subsequent rules further down the table are ignored.
SIP User's Manual 3. Web-Based Management ¾ To activate a de-activated rule, take these 2 steps: 1. In the 'Edit Rule' column, select the de-activated rule that you want to activate. 2. Click the Activate button; the rule is activated. ¾ To de-activate an activated rule, take these 2 steps: 1. In the 'Edit Rule' column, select the activated rule that you want to de-activate.. 2. Click the DeActivate button; the rule is de-activated. ¾ To delete a rule, take these 3 steps: 1.
MediaPack Series Parameter Description Action Upon Match [AccessList_Allow_Type] Action upon match (i.e., 'Allow' or 'Block'). Match Count [AccessList_MatchCount] A read-only field providing the number of packets accepted / rejected by the specific rule. 3.4.3.
SIP User's Manual 3. Web-Based Management 3. In the 'Subject Name' field, enter the DNS name, and then click Generate CSR. A textual certificate signing request that contains the SSL device identifier is displayed. 4. Copy this text and send it to your security provider. The security provider (also known as Certification Authority or CA) signs this request and then sends you a server certificate for the device. 5. Save the certificate to a file (e.g., cert.txt).
MediaPack Series ¾ To apply the loaded certificate for IPsec negotiations, take these 2 steps: 1. Open the ‘IKE Table’ page (refer to ''Configuring the IKE Table'' on page 97); the 'Loaded Certificates Files' group lists the newly uploaded certificates, as shown below: Figure 3-53: IKE Table Listing Loaded Certificate Files 2. Click the Apply button to load the certificates; future IKE negotiations are now performed using the new certificates. 3.4.3.4.
SIP User's Manual 3. Web-Based Management 4. When the operation is complete, HTTPSRequireClientCertificates to 1. set the 5. Save the configuration (refer to ''Saving Configuration'' on page 209), and then restart the device. ini file parameter When a user connects to the secured Web server: If the user has a client certificate from a CA that is listed in the Trusted Root Certificate file, the connection is accepted and the user is prompted for the system password.
MediaPack Series 3.4.3.5 Configuring the General Security Settings The 'General Security Settings' page is used to configure various security features. ¾ To configure the general security parameters, take these 4 steps: 1. Open the 'General Security Settings' page (Configuration tab > Security Settings menu > General Security Settings page item). Figure 3-54: General Security Settings Page 2. Configure the General Security parameters according to the table below. 3.
SIP User's Manual 3. Web-Based Management Table 3-22: General Security Parameters Parameter HTTP Authentication Mode [WebAuthMode] Description Determines the authentication mode for the Web interface. [0] Basic Mode = Basic authentication (clear text) is used (default). [1] Digest When Possible = Digest authentication (MD5) is used. [2] Basic if HTTPS, Digest if HTTP = Digest authentication (MD5) is used for HTTP, and basic authentication is used for HTTPS.
MediaPack Series Parameter Description RADIUS Authentication Server IP Address [RADIUSAuthServerIP] IP address of the RADIUS authentication server. RADIUS Authentication Server Port [RADIUSAuthPort] Port number of the RADIUS authentication server. The default value is 1645. RADIUS Shared Secret [SharedSecret] 'Secret' used to authenticate the device to the RADIUS server. Should be a cryptographically strong password.
SIP User's Manual 3. Web-Based Management Parameter Dead Peer Detection Mode [IPSecDPDMode] Description [0] Disable = IPSec is disabled (default). [1] Enable = IPSec is enabled. Enables the Dead Peer Detection (DPD) 'keep-alive' mechanism (according to RFC 3706) to detect loss of peer connectivity. [0] Disabled (default). [1] Periodic = message exchanges at regular intervals. [2] On Demand = message exchanges as needed (i.e., before sending data to the peer).
MediaPack Series Parameter Description server or client for the TLS connection. When a remote certificate is received and this parameter is not disabled, the SubjectAltName value is compared with the list of available Proxies. If a match is found for any of the configured Proxies, the TLS connection is established. The comparison is performed if the SubjectAltName is either a DNS name (DNSName) or an IP address.
SIP User's Manual 3. Web-Based Management ¾ To configure the IPSec SPD table, take these 5 steps: 1. Open the ‘IPSec Table’ page (Configuration tab > Security Settings menu > IPSec Table page item). Figure 3-55: IPSec Table Page 2. From the ‘Policy Index’ drop-down list, select the rule you want to edit (up to 20 policy rules can be configured). 3. Configure the IPSec SPD parameters according to the table below. 4. Click the button Create; the IPSec rule is applied on-the-fly to the device. 5.
MediaPack Series Table 3-24: IPSec SPD Table Configuration Parameters Parameter Name IPSec Mode [IPSecMode] Remote Tunnel IP Address [IPSecPolicyRemoteTunnelIPAddress] Description Defines the IPSec mode of operation. [0] Transport (Default) [1] Tunneling Defines the IP address of the remote IPSec tunneling device. Note: This parameter is available only if the parameter IPSecMode is set to Tunneling (1).
SIP User's Manual 3. Web-Based Management Parameter Name Description IKE Second Phase Parameters (Quick Mode) SA Lifetime (sec) [PsecPolicyLifeInSec] Determines the time (in seconds) that the SA negotiated in the second IKE session (quick mode) is valid. After the time expires, the SA is re-negotiated. The default value is 28,800 (i.e., 8 hours). SA Lifetime (KB) [IPSecPolicyLifeInKB] Determines the lifetime (in kilobytes) that the SA negotiated in the second IKE session (quick mode) is valid.
MediaPack Series ¾ To configure the IKE table, take these 5 steps: 1. Open the ‘IKE Table’ page (Configuration tab > Security Settings menu > IKE Table page item). Figure 3-56: IKE Table Page 2. From the ‘Policy Index’ drop-down list, select the peer you want to edit (up to 20 peers can be configured). 3. Configure the IKE parameters according to the table below. Up to two IKE main mode proposals (Encryption / Authentication / DH group combinations) can be defined.
SIP User's Manual 3. Web-Based Management The parameters described in the following table are used to configure the first phase (main mode) of the IKE negotiation for a specific peer. A different set of parameters can be configured for each of the 20 available peers. Table 3-26: IKE Table Configuration Parameters Parameter Name Authentication Method [IkePolicyAuthenticationMe thod] Description Determines the authentication method for IKE.
MediaPack Series Parameter Name First to Fourth Proposal Authentication Type [IKEPolicyProposalAuthenti cation_X ] First to Fourth Proposal DH Group [IKEPolicyProposalDHGrou p_X] 3.4.4 Description [2] Triple DES-CBC [3] AES-CBC Not Defined (default) Determines the authentication protocol used in the main mode negotiation for up to four proposals. For the ini file parameter, X depicts the proposal number (0 to 3).
SIP User's Manual 3. Web-Based Management 3.4.4.1.1 SIP General Parameters The 'SIP General Parameters' page is used to configure general SIP parameters. ¾ To configure the general SIP protocol parameters, take these 4 steps: 1. Open the 'SIP General Parameters' page (Configuration tab > Protocol Configuration menu > Protocol Definition submenu > SIP General Parameters page item). Figure 3-57: SIP General Parameters Page Version 5.
MediaPack Series 2. Configure the parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-27: SIP General Parameters (Protocol Definition) Parameter PRACK Mode [PRACKMode] Description PRACK (Provisional Acknowledgment) mechanism mode for 1xx SIP reliable responses.
SIP User's Manual 3. Web-Based Management Parameter Description Note that to send a 183 response, you must also set the parameter ProgressIndicator2IP to 1. If it is equal to 0, 180 Ringing response is sent. 183 Message Behavior [SIP183Behaviour] Defines the response of the device upon receipt of a SIP 183 response. [0] Progress = A 183 response (without SDP) does not cause the device to play a ringback tone (default).
MediaPack Series Parameter Fax Signaling Method [IsFaxUsed] Description Determines the SIP signaling method for establishing and transmitting a fax session after a fax is detected. [0] No Fax = No fax negotiation using SIP signaling. Fax transport method is according to the parameter FaxTransportMode (default). [1] T.38 Relay = Initiates T.38 fax relay. [2] G.711 Transport = Initiates fax / modem using the coder G.711 Alaw/μ-law with adaptations (refer to Note below).
SIP User's Manual 3. Web-Based Management Parameter SIP Transport Type [SIPTransportType] Description Determines the default transport layer for outgoing SIP calls initiated by the device. [0] UDP (default) [1] TCP [2] TLS (SIPS) Notes: It's recommended to use TLS for communication with a SIP Proxy and not for direct device-to-device communication. For received calls (i.e., incoming), the device accepts all these protocols.
MediaPack Series Parameter Use user=phone in From Header [IsUserPhoneInFrom] Use Tel URI for Asserted Identity [UseTelURIForAssertedI D] Description Determines whether to add 'user=phone' string in the From header. [0] No = Doesn't use 'user=phone' string in From header (default). [1] Yes = 'user=phone' string is part of the From header. Determines the format of the URI in the P-Asserted-Identity and PPreferred-Identity headers. [0] Disable = 'sip:' (default). [1] Enable = 'tel:'.
SIP User's Manual 3. Web-Based Management Parameter Enable History-Info Header [EnableHistoryInfo] Description Enables usage of the History-Info header. [0] Disable = Disable (default) [1] Enable = Enable 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.
MediaPack Series Parameter Use Display Name as Source Number [UseDisplayNameAsSou rceNumber] Description Determines the use of Source Number and Display Name for IP-to-Tel calls. [0] No = If IP Display Name is received, the IP Source Number is used as the Tel Source Number and the IP Display Name is used as the Tel Display Name. If no Display Name is received from IP, the Tel Display Name remains empty (default).
SIP User's Manual 3. Web-Based Management Parameter Description parameter isn't included. If a 'tgrp' value is specified in incoming messages, it is ignored. Enable GRUU [EnableGRUU] [2] Send and Receive = The functionality of outgoing SIP messages is identical to the functionality described in option (1). In addition, for incoming SIP messages, if the Request-URI includes a 'tgrp' parameter, the device routes the call according to that value (if possible).
MediaPack Series Parameter Description SDP Session Owner [SIPSDPSessionOwner] Determines the value of the Owner line ('o' field) in outgoing SDP messages. The valid range is a string of up to 39 characters. The default value is 'AudiocodesGW'. For example: o=AudiocodesGW 1145023829 1145023705 IN IP4 10.33.4.126 Subject [SIPSubject] Defines the value of the Subject header in outgoing INVITE messages. If not specified, the Subject header isn't included (default).
SIP User's Manual 3. Web-Based Management Parameter Description [1] Enable. Note: P-Associated-URIs in registration responses is handled only if the device is registered per endpoint. Source Number Preference [SourceNumberPreferen ce] Determines the SIP header used to determine the Source Number in incoming INVITE messages. “” (empty string) = Use device's internal logic for header preference (default). “FROM” = Use the Source Number received in the From header.
MediaPack Series 3.4.4.1.2 Proxy & Registration Parameters The 'Proxy & Registration' page allows you to configure parameters that are associated with Proxy and Registration. Note: To view whether the device or its endpoints have registered to a SIP Registrar/Proxy server, refer to ''Registration Status'' on page 226. ¾ To configure the Proxy & Registration parameters, take these 4 steps: 1.
SIP User's Manual 3. Web-Based Management Table 3-28: Proxy & Registration Parameters Parameter Description Proxy Parameters Use Default Proxy [IsProxyUsed] Enables the use of a SIP Proxy server. [0] No = Proxy isn't used - the internal routing table is used instead (default). [1] Yes = Proxy is used. Parameters relevant to Proxy configuration are displayed.
MediaPack Series Parameter Prefer Routing Table [PreferRouteTable] Use Routing Table for Host Names and Profiles [AlwaysUseRouteTable] Description Determines if the device's internal routing table takes precedence over a Proxy for routing calls. [0] No = Only a Proxy server is used to route calls (default). [1] Yes = The device checks the routing rules in the 'Tel to IP Routing' table for a match with the Tel-to-IP call. Only if a match is not found is a Proxy used.
SIP User's Manual 3. Web-Based Management Parameter Description 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. When this parameter is set to [2], the XferPrefix parameter can be used to define different routing rules for redirected calls. This parameter is disregarded if the parameter AlwaysSendToProxy is set to 1.
MediaPack Series Parameter Description Registration Time [RegistrationTime] Defines the time interval (in seconds) for registering to a Proxy server. The value is used in the Expires header. In addition, this parameter defines the time interval between Keep-Alive messages when the parameter EnableProxyKeepAlive is set to 2 (REGISTER). Typically, the device registers every 3,600 sec (i.e., one hour). The device resumes registration according to the parameter RegistrationTimeDivider.
SIP User's Manual 3. Web-Based Management Parameter Gateway Registration Name [GWRegistrationName] Description Defines the user name that is used in the From and To headers in REGISTER messages. If no value is specified (default) for this parameter, the UserName parameter is used instead. Note: This parameter is applicable only for single registration per device (i.e., AuthenticationMode is set to 1). When the device registers each channel separately (i.e.
MediaPack Series Parameter Description an SRV query is sent according to the information received in the NAPTR response. If the NAPTR query fails, an SRV query is performed according to the configured transport type. If the Proxy IP address parameter contains a domain name with port definition (e.g., ProxyIP = domain.com:5080), the device performs a regular DNS A-record query. If a specific Transport Type is defined, a NAPTR query is not performed.
SIP User's Manual 3. Web-Based Management Parameter Description Cnonce [Cnonce] Cnonce string used by the SIP server and client to provide mutual authentication. (Free format, i.e., 'Cnonce = 0a4f113b'). The default is 'Default_Cnonce'. Authentication Mode [AuthenticationMode] Determines the device's registration and authentication method. [0] Per Endpoint = Registration and Authentication separately for each endpoint.
MediaPack Series Parameter Mutual Authentication Mode [MutualAuthenticationMod e] Description Determines the device's mode of operation when Authentication and Key Agreement (AKA) Digest 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. 3.4.4.1.
SIP User's Manual 3. Web-Based Management 2. From the Proxy Set ID drop-down list, select an ID for the desired group. 3. Configure the Proxy parameters according to the following table. 4. Click the Submit button to save your changes. 5. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-29: Proxy Sets Table Parameters Parameter Proxy Set ID Description The Proxy Set identification number. The valid range is 0 to 5 (i.e.
MediaPack Series Parameter Description REGISTER). Transport Type To use Proxy Redundancy, you must specify one or more redundant Proxies. When a port number is specified (e.g., domain.com:5080), DNS NAPTR/SRV queries aren't performed, even if ProxyDNSQueryType is set to 1 or 2. The transport type per Proxy server.
SIP User's Manual 3. Web-Based Management Parameter Description defined interval, as configured by the parameter RegistrationTime. Any response from the Proxy, either success (200 OK) or failure (4xx response) is considered as if the Proxy is communicating correctly. Notes: This parameter must be set to 'Using OPTIONS' when Proxy redundancy is used. When this parameter is set to 'Using REGISTER', the homing redundancy mode is disabled.
MediaPack Series The coders supported by the device are listed in the table below: Table 3-30: Supported Coders Coder Name Packetization Time Rate G.711 A-law [g711Alaw64k] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 G.711 U-law [g711Ulaw64k] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 G.729 [g729] 10, 20 (default), 30, 40, 50, 60, 80, 100 Always 8 G.723.1 [g7231] 30 (default), 60, 90 G.
SIP User's Manual 3. Web-Based Management 3. From the 'Packetization Time' drop-down list, select the packetization time (in msec) for the selected coder. The packetization time determines how many coder payloads are combined into a single RTP packet. 4. From the 'Rate' drop-down list, select the bit rate (in kbps) for the selected coder. 5.
MediaPack Series Figure 3-61: DTMF & Dialing Page 2. Configure the DTMF and dialing parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-31: DTMF and Dialing Parameters Parameter Max Digits in Phone Num [MaxDigits] Description Defines the maximum number of collected destination number digits that can be received (i.e., dialed) from the Tel side.
SIP User's Manual 3. Web-Based Management Parameter Description relay packets. Therefore, it is always correct to include the 'telephonyevent' parameter as default in the SDP. However, some devices use the absence of the 'telephony-event' in the SDP to decide to send DTMF digits in-band using G.711 coder. If this is the case, you can set RxDTMFOption to 0. 1st to 5th Tx DTMF Option [TxDTMFOption] Determines a single or several preferred transmit DTMF negotiation methods.
MediaPack Series Parameter Hook-Flash Option [HookFlashOption] Description Determines the supported hook-flash Transport Type (i.e., method by which hook-flash is sent and received). [0] Not Supported = Hook-Flash indication isn't sent (default). [1] INFO = Send proprietary INFO message with Hook-Flash indication. [4] RFC 2833 [5] INFO (Lucent) = Send proprietary INFO message with HookFlash indication.
SIP User's Manual 3. Web-Based Management Parameter Enable Special Digits [IsSpecialDigits] Description Determines whether the asterisk (*) and pound (#) digits can be used. [0] Disable = Use '*' or '#' to terminate number collection (refer to the parameter UseDigitForSpecialDTMF). (Default.) [1] Enable = Allows '*' and '#' for telephone numbers dialed by a user or for the endpoint telephone number.
MediaPack Series 3.4.4.2.1 Advanced Parameters The 'Advanced Parameters' page allows you to configure general control protocol parameters. ¾ To configure the advanced general protocol parameters, take these 4 steps: 1. Open the 'Advanced Parameters' page (Configuration tab > Protocol Configuration menu > SIP Advanced Parameters submenu > Advanced Parameters page item). Figure 3-62: Advanced Parameters Page 2. Configure the parameters according to the table below. 3.
SIP User's Manual 3. Web-Based Management Table 3-32: Advanced Parameters Description Parameter Description General IP Security [SecureCallsFromIP] Determines whether the device accepts SIP calls received from only IP addresses defined in the 'Tel to IP Routing' table (refer to ''Tel to IP Routing Table'' on page 160). This is useful in preventing unwanted SIP calls or messages and/or VoIP spam. [0] Disable = device accepts all SIP calls (default).
MediaPack Series Parameter Description times. For example (for FXS/FXO interfaces), the called number can be as follows: d1005, dpp699, p9p300. To add the 'd' and 'p' digits, use the usual number manipulation rules. RTP Only Mode [RTPOnlyMode] Enable DID Wink [EnableDIDWink] To use this feature with FXO interfaces, configure the device to operate in one-stage dialing mode.
SIP User's Manual 3. Web-Based Management Parameter Description If the polarity reversal service is enabled, the FXS interface changes the line polarity on call answer and then changes it back on call release. The FXO interface sends a 200 OK response when polarity reversal signal is detected (applicable only to one-stage dialing) and releases a call when a second polarity reversal signal is detected.
MediaPack Series Parameter Description Silence Detection Period [sec] [FarEndDisconnectSile ncePeriod] Duration of silence period (in seconds) prior to call disconnection. The range is 10 to 28,800 (i.e., 8 hours). The default is 120 seconds. Silence Detection Method [FarEndDisconnectSile nceMethod] Silence detection method. Enable Fax Re-Routing [EnableFaxReRouting] [0] None = Silence detection option is disabled. [1] Packets Count = According to packet count.
SIP User's Manual 3. Web-Based Management Parameter Description [1] 1 = Flow debugging is enabled. [2] 2 = Flow and device interface debugging are enabled. [3] 3 = Flow, device interface, and stack interface debugging are enabled. [4] 4 = Flow, device interface, stack interface, and session manager debugging are enabled. [5] 5 = Flow, device interface, stack interface, session manager, and device interface expanded debugging are enabled.
MediaPack Series Parameter Description Default Release Cause [DefaultReleaseCause] Default Release Cause (to IP) for IP-to-Tel calls when the device initiates a call release and an explicit matching cause for this release isn't found. The default release cause is NO_ROUTE_TO_DESTINATION (3). Other common values include NO_CIRCUIT_AVAILABLE (34), DESTINATION_OUT_OF_ORDER (27), etc. Notes: The default release cause is described in the Q.
SIP User's Manual 3. Web-Based Management Parameter Enable Calls Cut Through [CutThrough] Description Enables users to receive incoming IP calls while the port is in off-hook state. [0] Disable = Disabled (default). [1] Enable = Enabled. If enabled, the FXS interface answers the call and 'cuts through' the voice channel if there is no other active call on the port, even if the port is in offhook state.
MediaPack Series Parameter Description Emergency Calls Emergency Numbers [EmergencyNumbers] Defines a list of numbers which are defined as 'emergency numbers'. When one of these numbers is dialed, the outgoing INVITE message includes the Priority and Resource-Priority headers. If the user sets the phone on-hook, the call is not disconnected, but instead a Hold ReINVITE request is sent to the remote party. Only if the remote party disconnects the call (i.e.
SIP User's Manual 3. Web-Based Management Figure 3-63: Supplementary Services Page 2. Configure the supplementary services parameters according to the table below. 3. 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. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Version 5.
MediaPack Series Table 3-33: Supplementary Services Parameters Parameter Enable Hold [EnableHold] Description Allows users (connected to the device) to place a call on hold. [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. On receiving a Hold request, the remote party is placed on hold and hears the hold tone.
SIP User's Manual 3. Web-Based Management Parameter Enable Call Forward [EnableForward] Description Determines whether Call Forward is enabled. [0] Disable = Disable the Call Forward service. [1] Enable = Enable Call Forward service (using REFER) (default). For FXS interfaces, the 'Call Forward' table (refer to “Call Forward” on page 178) must be defined to use the Call Forward service. Note: To use this service, the devices at both ends must support this option.
MediaPack Series Parameter Description If the Caller ID service is enabled, then for FXS interfaces, calling number and Display text (from IP) are sent to the device's port. 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, refer to “Caller ID” on page 177. To disable/enable caller ID generation per port, refer to “Call Forward” on page 178.
SIP User's Manual 3. Web-Based Management Parameter MWI Analog Lamp [MWIAnalogLamp] Description Enables visual display of MWI. [0] Disable = Disable (default). [1] Enable = Enables visual Message Waiting Indication by supplying line voltage of approximately 100 VDC to activate the phone's lamp. Note: This parameter is applicable only for FXS interfaces. MWI Display [MWIDisplay] Determines whether MWI information is sent to the phone display.
MediaPack Series Parameter MWI Subscribe Retry Time [SubscribeRetryTime] Description Subscription retry time (in seconds) after last subscription failure. The default is 120 seconds. The range is 10 to 7200. Conference Parameters Enable 3-Way Conference [Enable3WayConference ] Enables or disables the 3-Way Conference feature.
SIP User's Manual 3. Web-Based Management ¾ To configure the Metering tones, take these 4 steps: 1. Open the 'Metering Tones' page (Configuration tab > Protocol Configuration menu > SIP Advanced Parameters submenu > Metering Tones page item). Figure 3-64: Metering Tones Page 2. Configure the Metering tones parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to the flash memory, refer to ''Saving Configuration'' on page 209.
MediaPack Series 3.4.4.2.4 Charge Codes Table The 'Charge Codes Table' page is used to configure the metering tones (and their time interval) that the FXS interfaces generate to the Tel side. To associate a charge code to an outgoing Tel-to-IP call, use the 'Tel to IP Routing' table. Notes: • The 'Charge Codes Table' page is available only for FXS interfaces.
SIP User's Manual 3. Web-Based Management 3.4.4.2.5 Keypad Features The 'Keypad Features' page (applicable only to FXS interfaces) enables you to activate and deactivate the following features directly from the connected telephone's keypad: Call Forward (refer to ''Call Forward'' on page 178) Caller ID Restriction (refer to ''Caller ID'' on page 177) Hotline (refer to ''Automatic Dialing'' on page 175) Notes: • The 'Keypad Features' page is available only for FXS interfaces.
MediaPack Series Table 3-35: Keypad Features Parameters Description Parameter Description Forward (Note: The forward type and number can be viewed in the 'Call Forward' table - refer to ''Call Forward'' on page 178.) Unconditional [KeyCFUnCond] Keypad sequence that activates the immediate call forward option. No Answer [KeyCFNoAnswer] Keypad sequence that activates the forward on no answer option. On Busy [KeyCFBusy] Keypad sequence that activates the forward on busy option.
SIP User's Manual 3. Web-Based Management Parameter Description of the destination phone number. For both options, after the phone number is collected, it's sent to the transferee in a SIP REFER request (without a Replaces header). The call is then terminated and a confirmation tone is played to the phone. If the phone number collection fails due to a mismatch, a reorder tone is played to the phone.
MediaPack Series ¾ To configure the Stand-Alone Survivability parameters, take these 4 steps: 1. Open the 'SAS Configuration' page (Configuration tab > Protocol Configuration menu > SIP Advanced Parameters submenu > Stand-Alone Survivability page item). Figure 3-67: SAS Configuration Page 2. Configure the parameters according to the table below. 3. Click the Submit button to apply your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209.
SIP User's Manual 3. Web-Based Management Parameter Description SAS Registration Time [SASRegistrationTime] Determines the value of the SIP Expires header that is sent in a 200 OK response to an incoming REGISTER message when in SAS 'Emergency Mode'. The valid range is 0 to 2,000,000. The default value is 20. Short Number Length [SASShortNumberLength] This parameter is obsolete; instead, use the parameter SASRegistrationManipulation.
MediaPack Series The number manipulation is configured in the following tables: For Tel-to-IP calls: • Destination Phone Number Manipulation Table for Tel-to-IP Calls (NumberMapTel2IP ini file parameter) • Source Phone Number Manipulation Table for Tel-to-IP Calls (SourceNumberMapTel2IP ini file parameter) For IP-to-Tel calls: • Destination Phone Number Manipulation Table for IP-to-Tel Calls (NumberMapIP2Tel ini file parameter) • Source Phone Number Manipulation Table for IP-to-Tel Calls (Sourc
SIP User's Manual 3. Web-Based Management • When the source number is 1001876, it is changed to 587623. • When the source number is 1234510012001, it is changed to 20018. • When the source number is 3122, it is changed to 2312. 2.
MediaPack Series Parameter Description originating from this Source IP Group is sent to the Serving IP Group. In this scenario, this table is used only if the parameter PreferRouteTable is set to 1. Destination Prefix [_DestinationPrefix] Destination (called) telephone number prefix. An asterisk (*) represents any number. Source Prefix [_SourcePrefix] Source (calling) telephone number prefix. An asterisk (*) represents any number.
SIP User's Manual 3. Web-Based Management 3.4.4.3.1 Dialing Plan Notation The dialing plan notation applies to the Number Manipulation tables, 'Tel to IP Routing' table (refer to ''Tel to IP Routing Table'' on page 160), and 'IP to Hunt Group Routing' table (refer to ''IP to Trunk Group Routing'' on page 163). The dialing notation applies to digits entered for the destination and source prefixes to represent multiple numbers. Table 3-38: Dialing Plan Notations Notation [n-m] [n,m,...
MediaPack Series ¾ To configure the Phone-Context tables, take these 4 steps: 1. Open the 'Phone Context Table' page (Configuration tab > Protocol Configuration menu > Manipulation Tables submenu > Phone Context Table page item). Figure 3-69: Phone Context Table Page 2. Configure the Phone Context table according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209.
SIP User's Manual 3. Web-Based Management Parameter Description Level 0 Regional (Local) [4]. Phone Context 3.4.4.4 If you selected E.164 Public as the NPI, you can select Unknown [0], International [1], National [2], Network Specific [3], Subscriber [4], or Abbreviated [6]. The Phone-Context SIP URI parameter. Configuring the Routing Tables The Routing Tables submenu allows you to configure the device's call routing.
MediaPack Series 2. Configure the general parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-40: Routing General Parameters Description Parameter Description Add Hunt Group ID as Prefix [AddTrunkGroupAsPrefix] Determines whether the device's Hunt Group ID is added as a prefix to the destination phone number for Tel-to-IP calls.
SIP User's Manual 3. Web-Based Management Parameter Enable Alt Routing Tel to IP [AltRoutingTel2IPEnable] Description Enables the Alternative Routing feature for Tel-to-IP calls. [0] Disable = Disables the Alternative Routing feature (default). [1] Enable = Enables the Alternative Routing feature. [2] Status Only = The Alternative Routing feature is disabled, but read-only information on the Quality of Service of the destination IP addresses is provided.
MediaPack Series Parameter Description Max Allowed Packet Loss for Alt Routing [%] [IPConnQoSMaxAllowedPL] Packet loss percentage at which the IP connection is considered a failure and Alternative Routing mechanism is activated. The range is 1 to 20%. The default value is 20%. Max Allowed Delay for Alt Routing [msec] [IPConnQoSMaxAllowedDel ay] Transmission delay (in msec) at which the IP connection is considered a failure and Alternative Routing mechanism is activated. The range is 100 to 1000.
SIP User's Manual 3. Web-Based Management Assign Profiles to destination addresses (also when a Proxy is used). Alternative Routing (when a Proxy isn't used): an alternative IP destination for telephone number prefixes is available. To associate an alternative IP address to a called telephone number prefix, assign it with an additional entry (with a different IP address), or use an FQDN that resolves into two IP addresses.
MediaPack Series 3. Configure the Tel to IP Routing table according to the table below. 4. Click the Submit button to save your changes. 5. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-41: Tel to IP Routing Table Parameters Description Parameter Tel to IP Routing Mode [RouteModeTel2IP] Description Determines whether to route Tel calls to IP before or after manipulation of destination number.
SIP User's Manual 3. Web-Based Management Parameter Description Port [PREFIX_DestPort] The destination port to where you want to route the Tel-to-IP call. Transport Type [PREFIX_TransportType] The transport layer type for sending the Tel-to-IP calls: [-1] Not Configured [0] UDP [1] TCP [2] TLS Note: When 'Not Configured' is selected, the transport type defined by the parameter SIPTransportType (refer to ''SIP General Parameters'' on page 101) is used.
MediaPack Series 3.4.4.4.3 IP to Trunk Group Routing Table The 'IP to Hunt Group Routing Table' page provides a table for routing incoming IP calls to groups of channels (FXS/FXO endpoints)called Hunt Groups. Hunt Group ID's are assigned to the device's channels in the 'Endpoint Phone Number' page (refer to “Configuring the Endpoint Phone Numbers” on page 181). You can add up to 24 IP-to-Hunt Group routing rules in the table.
SIP User's Manual 3. Web-Based Management Figure 3-72: IP to Hunt Group Routing Page 2. From the 'Routing Index' drop-down list, select the range of entries that you want to add. 3. Configure the table according to the table below. 4. Click the Submit button to save your changes. 5. To save the changes so they are available after a power failure, refer to ''Saving Configuration'' on page 209.
MediaPack Series Parameter Description Dest. Phone Prefix [PstnPrefix_DestPrefix] Represents a called telephone number prefix. The prefix can be 1 to 49 digits long. Note: For notations representing multiple numbers, refer to ''Dialing Plan Notation'' on page 155. Source Phone Prefix [PstnPrefix_SourcePrefix] Represents a calling telephone number prefix. The prefix can be 1 to 49 digits long. Note: For notations representing multiple numbers, refer to ''Dialing Plan Notation'' on page 155.
SIP User's Manual 3. Web-Based Management ¾ To configure the internal DNS table, take these 6 steps: 1. Open the 'Internal DNS Table' page (Configuration tab > Protocol Configuration menu > Routing Tables submenu > Internal DNS Table page item). Figure 3-73: Internal DNS Table Page 2. In the 'Domain Name' field, enter the host name to be translated. You can enter a string of up to 31 characters long. 3.
MediaPack Series ¾ To configure the Internal SRV table, take these 9 steps: 1. Open the 'Internal SRV Table' page (Configuration tab > Protocol Configuration menu > Routing Tables submenu > Internal SRV Table page item). Figure 3-74: Internal SRV Table Screen 2. In the 'Domain Name' field, enter the host name to be translated. You can enter a string of up to 31 characters long. 3. From the 'Transport Type' drop-down list, select a transport type. 4.
SIP User's Manual 3. Web-Based Management Notes: • The reasons for alternative routing for Tel-to-IP calls only apply when a Proxy isn't used. • For Tel-to-IP calls, the device sends the call to an alternative route only after the call has failed and the device has subsequently attempted twice to establish the call unsuccessfully.
MediaPack Series You can assign different Profiles (behavior) per call, using the call routing tables: 'Tel to IP Routing' page (refer to ''Tel to IP Routing Table'' on page 160) 'IP to Hunt Group Routing' page (refer to ''IP to Trunk Group Routing'' on page 163), In addition, you can associate different Profiles per the device's channels. Each Profile contains a set of parameters such as coders, T.
SIP User's Manual 3. Web-Based Management ¾ To configure coder groups, take these 11 steps: 1. Open the 'Coder Group Settings' page (Configuration tab > Protocol Configuration menu > Profile Definitions submenu > Coder Group Settings page item). Figure 3-76: Coder Group Settings Page 2. From the 'Coder Group ID' drop-down list, select a coder group ID. 3. From the 'Coder Name' drop-down list, select the first coder for the coder group. 4.
MediaPack Series ¾ To configure Tel Profiles, take these 9 steps: 1. Open the 'Tel Profile Settings' page (Configuration tab > Protocol Configuration menu > Profile Definitions submenu > Tel Profile Settings page item). Figure 3-77: Tel Profile Settings Screen 2. From the 'Profile ID' drop-down list, select the Tel Profile identification number you want to configure. 3. In the 'Profile Name' field, enter an arbitrary name that enables you to easily identify the Tel Profile. 4.
SIP User's Manual 3. Web-Based Management 6. From the 'Coder Group' drop-down list, select the Coder Group (refer to ''Coder Group Settings'' on page 170) or the device's default coder (refer to ''Coders'' on page 123) to which you want to assign the Profile. 7. Repeat steps 2 through 6 to configure additional Tel Profiles (optional). 8. Click the Submit button to save your changes. 9. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. 3.4.4.5.
MediaPack Series 2. From the 'Profile ID' drop-down list, select an identification number for the IP Profile. 3. In the 'Profile Name' field, enter an arbitrary name that allows you to easily identify the IP Profile. 4. 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.
SIP User's Manual 3. Web-Based Management 3.4.4.6.1 Authentication The 'Authentication' page defines a user name and password for authenticating each device port. Authentication is typically used for FXS interfaces, but can also be used for FXO interfaces. Notes: • The 'Authentication Mode' parameter (refer to ''Proxy & Registration Parameters'' on page 112) determines whether authentication is performed per port or for the entire device.
MediaPack Series 3.4.4.6.2 Automatic Dialing The 'Automatic Dialing' page allows you to define a telephone number that is automatically dialed when an FXS or FXO port is used (e.g., off-hooked). Notes: • After a ring signal is detected on an 'Enabled' FXO port, the device initiates a call to the destination number without seizing the line. The line is seized only after the call is answered. • After a ring signal is detected on a 'Disabled' or 'Hotline' FXO port, the device seizes the line.
SIP User's Manual • 3. Web-Based Management Hotline [2]: When a phone is off-hooked and no digit is dialed for a user-defined interval (Hotline Dial Tone Duration - refer to ''DTMF & Dialing Parameters'' on page 125), the number in the 'Destination Phone Number' field is automatically dialed (applies to FXS and FXO interfaces). 4. Click the Submit button to save your changes. 5. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. 3.4.4.6.
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. • When the 'Presentation' field is set to 'Restricted', the Caller ID is sent to the remote side using only the P-Asserted-Identity and P-PreferredIdentity headers (AssertedIdMode).
SIP User's Manual 3. Web-Based Management 2. Configure the Call Forward parameters for each port according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-44: Call Forward Table Parameter Description Determines the scenario for forwarding a call. Forward Type Forward to Phone Number [0] Deactivate = Don't forward incoming calls (default).
MediaPack Series ¾ To configure Caller ID Permissions per port, take these 4 steps: 1. Open the 'Caller ID Permissions' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu > Caller ID Permissions page item). Figure 3-83: Caller ID Permissions Page 2. From the 'Caller ID' drop-down list, select one of the following: • 'Enable': Enables Caller ID generation (FXS) or detection (FXO) for the specific port.
SIP User's Manual 3. Web-Based Management ¾ To configure Call Waiting, take these 4 steps: 1. Open the 'Caller Waiting' page (Configuration tab > Protocol Configuration menu > Endpoint Settings submenu > Call Waiting page item). Figure 3-84: Call Waiting Page 2. 3.4.4.7 From the 'Call Waiting Configuration' drop-down list corresponding to the port you want to configure for call waiting, select one of the following options: • 'Enable': Enables call waiting for the specific port.
MediaPack Series Figure 3-85: Endpoint Phone Number Table Page 2. Configure the endpoint phone numbers according to the table below. You must enter a number in the 'Phone Number' fields for each port that you want to use. 3. Click the Submit button to save your changes, or click the Register or Un-Register buttons to save your changes and to register / unregister to a Proxy / Registrar. 4. To save the changes to the flash memory, refer to ''Saving Configuration'' on page 209.
SIP User's Manual 3.4.4.8 3. Web-Based Management Configuring the Hunt and IP Groups The Hunt/IP Group menu allows you to configure groups of channels. This submenu includes the following page items: Hunt Group Settings (refer to ''Configuring the Hunt Group Settings'' on page 183) IP Group Table (refer to ''Configuring the IP Groups'' on page 186) Account Table (refer to ''Configuring the Account Table'' on page 188) 3.4.4.8.
MediaPack Series Parameter Channel Select Mode [TrunkGroupSettings_ChannelS electMode] Registration Mode [TrunkGroupSettings_Registrati onMode] SIP User's Manual Description The method in which IP-to-Tel calls are assigned to channels pertaining to a Hunt Group: [0] By Dest Phone Number = Selects the device's channel according to the called number defined in the 'Endpoint Phone Number' (refer to “Configuring the Endpoint Phone Numbers” on page 181).
SIP User's Manual 3. Web-Based Management Parameter Description 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 (refer to ''Configuring the Account Table'' on page 188).
MediaPack Series 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 taken from the IP Group table. REGISTER sip:SipGroupName SIP/2.0 Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac862428454 From: ;tag=1c862422082 To: Call-ID: 9907977062512000232825@10.33.37.78 CSeq: 3 REGISTER Contact:
SIP User's Manual 3. Web-Based Management 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Table 3-47: IP Group Parameters Description Parameter Description Description Brief string description of the IP Group. The value range is a string of up to 29 characters. The default is an empty field. Proxy Set ID Selects the Proxy Set ID (defined in ''Proxy Sets Table'' on page 120) to associate with the IP Group.
MediaPack Series Parameter Description Always Use Route Table This parameter is disregarded if the parameter AlwaysSendToProxy is set to 1. Determines the Request URI host name in outgoing INVITE messages. Disable (default). Enable = The device uses the IP address (or domain name) defined in the 'Tel to IP Routing' table (''Tel to IP Routing Table'' on page 160) as the Request URI host name in outgoing INVITE messages, instead of the value entered in the 'SIP Group Name' field. 3.4.4.8.
SIP User's Manual 3. Web-Based Management Table 3-48: Account Parameters Description Parameter Description Served Trunk Group The Hunt Group ID for which the device performs registration and/or authentication to a destination IP Group (i.e., Serving IP Group). For Tel-to-IP calls, the Served Trunk Group is the source Hunt Group from where the call initiated.
MediaPack Series Parameter Description REGISTER sip:audiocodes SIP/2.0 Via: SIP/2.0/UDP 10.33.37.78;branch=z9hG4bKac1397582418 From: ;tag=1c1397576231 To: Call-ID: 1397568957261200022256@10.33.37.78 CSeq: 1 REGISTER Contact: ;expires=3600 Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-MP-118 FXS_FXO/v.5.40A.008.
SIP User's Manual 3. Web-Based Management Figure 3-89: Voice Mail Settings Page 2. Configure the voice mail parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209. Version 5.
MediaPack Series Table 3-49: Voice Mail Parameters Parameter Description General Voice Mail Interface [VoiceMailInterface] Line Transfer Mode [LineTransferMode] Enables the voice mail application on the device and determines the communication method used between the PBX and the device. [0] None (default) [1] DTMF [2] SMDI Determines the call transfer method used by the device. [0] None = IP (default). [1] Blind = PBX blind transfer.
SIP User's Manual 3. Web-Based Management Parameter Description Forward on Do Not Disturb Digit Pattern (Internal) [DigitPatternForwardOnDND] Determines the digit pattern used by the PBX to indicate 'call forward on do not disturb' when the original call is received from an internal extension. The valid range is a 120-character string.
MediaPack Series Parameter Description MWI Suffix Pattern [MWISuffixCode] Determines the digit code used by the device as a suffix for 'MWI On Digit Pattern' and 'MWI Off Digit Pattern'. This suffix is added to the generated DTMF string after the extension number. The valid range is a 25-character string. MWI Source Number [MWISourceNumber] Determines the calling party's phone number used in the Q.931 MWI SETUP message to PSTN.
SIP User's Manual 3. Web-Based Management Table 3-50: RADIUS Parameters Description Parameter Enable RADIUS Access Control EnableRADIUS Description Enables or disables the RADIUS application. [0] Disable = disables RADIUS application (default) [1] Enable = enables RADIUS application Accounting Server IP Address [RADIUSAccServerIP] IP address of the RADIUS accounting server. Accounting Port [RADIUSAccPort] Port of the RADIUS accounting server. The default value is 1646.
MediaPack Series ¾ To configure the FXO parameters, take these 4 steps: 1. Open the 'FXO Settings' page (Configuration tab > Advanced Applications menu > FXO Settings page item). Figure 3-91: FXO Settings Page 2. Configure the FXO parameters according to the table below. 3. Click the Submit button to save your changes. 4. To save the changes to flash memory, refer to ''Saving Configuration'' on page 209.
SIP User's Manual 3. Web-Based Management Parameter Description Notes: Time to Wait before Dialing [msec] [WaitForDialTime] The correct dial tone parameters should be configured in the Call Progress Tones file. The device may take 1 to 3 seconds to detect a dial tone (according to the dial tone configuration in the Call Progress Tones file).
MediaPack Series Parameter Description Typically, this feature is used only when early media (EnableEarlyMedia) is used to establish the voice path before the call is answered. Note: This feature is applicable only for one-stage dialing. Rings before Detecting Caller ID [RingsBeforeCallerID] Determines the number of rings before the device starts detecting Caller ID. [0] 0 = Before first ring. [1] 1 = After first ring (default). [2] 2 = After second ring.
SIP User's Manual 3.5.1 3. Web-Based Management Management Configuration The Management Configuration menu allows you to configure the device's management parameters. This menu contains the following page items: 3.5.1.
MediaPack Series Table 3-52: Management Settings Parameters Parameter Description Syslog Settings Syslog Server IP Address [SyslogServerIP] IP address (in dotted-decimal notation) of the computer you are using to run the Syslog server. The Syslog server is an application designed to collect the logs and error messages generated by the device. Default IP address is 0.0.0.0. For information on Syslog, refer to the Product Reference Manual.
SIP User's Manual 3. Web-Based Management Parameter Description page 205). Enable SNMP [DisableSNMP] [0] Enable = SNMP is enabled (default). [1] Disable = SNMP is disabled and no traps are sent. Trap Manager Host Name [SNMPTrapManagerHostName] Defines an FQDN of a remote host that is used as an SNMP manager.
MediaPack Series 3.5.1.1.1 Configuring the SNMP Trap Destinations Table The 'SNMP Trap Destinations' page allows you to configure up to five SNMP trap managers. ¾ To configure the SNMP Trap Destinations table, take these 5 steps: 1. Access the 'Management Settings' page, as described in ''Configuring the Management Settings'' on page 199. 2. In the 'SNMP Trap Destinations' field, click the right-pointing arrow 'SNMP Trap Destinations' page appears.
SIP User's Manual 3. Web-Based Management Parameter Description Trap Enable [SNMPManagerTrapSendin gEnable_x] Activates or de-activates the sending of traps to the corresponding SNMP Manager. [0] Disable = Sending is disabled. [1] Enable = Sending is enabled (default). 3.5.1.1.
MediaPack Series Table 3-54: SNMP Community Strings Parameters Description Parameter Description Community String Trap Community String [SNMPTrapCommunityString] Read Only [SNMPReadOnlyCommunityString_x]: Up to five 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'.
SIP User's Manual 3. Web-Based Management Table 3-55: SNMP V3 Users Parameters Parameter Description Index [SNMPUsers_Index] The table index. The valid range is 0 to 9. User Name [SNMPUsers_Username] Name of the SNMP v3 user. This name must be unique. Authentication Protocol [SNMPUsers_AuthProtocol] Authentication protocol of the SNMP v3 user. Privacy Protocol [SNMPUsers_PrivProtocol] [0] None (default) [1] MD5 [2] SHA-1 Privacy protocol of the SNMP v3 user.
MediaPack Series 2. In the 'SNMP Trusted Managers' field, click the right-pointing arrow 'SNMP Trusted Managers' page appears. button; the Figure 3-96: SNMP Trusted Managers 3.5.1.2 3. Select the check box corresponding to the SNMP Trusted Manager that you want to enable and for whom you want to define an IP address. 4. Define an IP address in dotted-decimal notation. 5. Click the Submit button to apply your changes. 6. To save the changes, refer to ''Saving Configuration'' on page 209.
SIP User's Manual 3.5.1.3 3.
MediaPack Series 3. Under the 'Reset Configuration' group, from the 'Graceful Option' drop-down list, select one of the following options: • 'Yes': Reset starts only after the user-defined time in the 'Shutdown Timeout' field (refer to Step 4) expires or after no more active traffic exists (the earliest thereof). In addition, no new traffic is accepted. • 'No': Reset starts regardless of traffic, and any existing traffic is terminated at once. 4.
SIP User's Manual • 3. Web-Based Management 'No': The device is 'locked' regardless of traffic. Any existing traffic is terminated immediately. Note: These options are only available if the current status of the device is in the Unlock state. 3. In the 'Lock Timeout' field (relevant only if the parameter 'Graceful Option' in the previous step is set to 'Yes'), enter the time (in seconds) after which the device locks. Note that if no traffic exists and the time has not yet expired, the device locks. 4.
MediaPack Series Notes: 3.5.2 • Saving configuration to the non-volatile memory may disrupt traffic on the device. To avoid this, disable all new traffic before saving, by performing a graceful lock (refer to ''Locking and Unlocking the Device'' on page 208). • Throughout the Web interface, parameters preceded by the lightning symbol are not applied on-the-fly to the device and require that you reset the device (refer to ''Resetting the Device'' on page 207) for them to take effect.
SIP User's Manual 3. Web-Based Management File Type Description Prerecorded Tones The dat PRT file enhances the device's capabilities of playing a wide range of telephone exchange tones that cannot be defined in the Call Progress Tones file. User Info The User Information file maps PBX extensions to IP numbers. This file can be used to represent PBX extensions as IP phones in the global 'IP world'. ¾ To load an auxiliary file to the device using the Web interface, take these 6 steps: 1.
MediaPack Series Notes: • Saving an auxiliary file to flash memory may disrupt traffic on the device. To avoid this, disable all traffic on the device by performing a graceful lock (refer to ''Locking and Unlocking the Device'' on page 208). • You can schedule automatic loading of updated auxiliary files using HTTP, HTTPS, FTP, or NFS (refer to the Product Reference Manual). You can also load the Auxiliary files using the ini file.
SIP User's Manual 3. Web-Based Management Notes: • Before you can load an ini or any auxiliary file, you must first load a cmp file. • When you activate the wizard, the rest of the Web interface is unavailable. After you load the desired files, access to the full Web interface is restored. • You can schedule automatic loading of cmp, ini, and auxiliary files using HTTP, HTTPS, FTP, or NFS. (Refer to the Product Reference Manual). ¾ To use the Software Upgrade Wizard, take these 11 steps: 1.
MediaPack Series Figure 3-103: Load a CMP file Page Note: At this stage, you can quit the Software Update Wizard, by clicking Cancel , without requiring a device reset. However, once you start uploading a cmp file, the process must be completed with a device reset. 4. Click the Browse button, navigate to the cmp file, and then click Send File; the cmp file is loaded to the device and you're notified as to a successful loading, as shown below.
SIP User's Manual 3. Web-Based Management Figure 3-104: CMP File Loaded Successfully Message 5. Click one of the following buttons: • • Reset; the device resets with the newly loaded cmp, and utilizing the current configuration and auxiliary files. Next; the 'Load an ini File' wizard page opens. Note that as you progress by clicking Next, the relevant file name corresponding to the applicable Wizard page is highlighted in the file list on the left. 6. Version 5.
MediaPack Series Figure 3-105: Load an ini File Page 7. 8. You can now choose to either: • Click Reset; the device resets, utilizing the new cmp and ini file you loaded up to now as well as utilizing the other auxiliary files. • Click Back; the 'Load a cmp file' page is opened again. • Click Next; the next page opens for loading the next consecutive auxiliary file listed in the Wizard. Follow the same procedure as for loading the ini file (Step 6) for loading the auxiliary files.
SIP User's Manual 9. 3. Web-Based Management In the 'FINISH' page, complete the upgrade process by clicking Reset; the device 'burns' the newly loaded files to flash memory and then resets t.he device. After the device resets, the 'End Process' screen appears displaying the burned configuration files (refer to the figure below). Figure 3-106: End Process Wizard Page 10.
MediaPack Series ¾ To save and restore the ini file, take these 3 steps: 1. Open the 'Configuration File' page (Management tab > Software Update menu > Configuration File). Figure 3-108: Configuration File Page 2. 3. 3.6 To save the ini file to a PC, perform the following: a. Click the Save INI File button; the 'File Download' dialog box opens. b.
SIP User's Manual 3.6.1 3. Web-Based Management Status & Diagnostics The Status & Diagnostics menu is used to view and monitor the device's channels, Syslog messages, hardware and software product information, and to assess the device's statistics and IP connectivity information. This menu includes the following page items: 3.6.1.
MediaPack Series The displayed logged messages are color coded as follows: 3. • Yellow - fatal error message • Blue - recoverable error message (i.e., non-fatal error) • Black - notice message To clear the page of Syslog messages, in the Navigation tree, click the page item Message Log again; the page is cleared and new messages begin appearing. ¾ To stop the Message Log, take this step: 3.6.1.2 Close the page by accessing any another page in the Web interface.
SIP User's Manual 3. Web-Based Management ¾ To view the 'Active IP Interfaces' page, take this step: Open the 'Active IP Interfaces' page (Status & Diagnostics tab > Status & Diagnostics menu > Active IP Interfaces page item). Figure 3-111: Active IP Interfaces Page 3.6.1.4 Viewing Device Information The 'Device Information' page displays the device's specific hardware and software product information. This information can help you to expedite troubleshooting.
MediaPack Series The 'Board Type' field number depicts the following devices: MP-118 = 56 MP-114 = 57 MP-112 = 58 MP-124 FXS = 3 ¾ To delete any of the loaded files, take this step: 3.6.1.5 Click the Delete button corresponding to the files that you want to delete. Deleting a file takes effect only after the device is reset (refer to ''Resetting the Device'' on page 207). Viewing Performance Statistics The 'Performance Statistics' page provides read-only, device performance statistics.
SIP User's Manual 3.6.1.6 3. Web-Based Management Viewing Active Alarms The 'Active Alarms' page displays a list of currently active alarms.
MediaPack Series 3.6.2.1 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). The release reason can be viewed in the 'Termination Reason' field in the CDR message.
SIP User's Manual 3. Web-Based Management Counter Description to No Answer' counter. The rest of the release reasons increment the 'Number of Failed Calls due to Other Failures' counter. Percentage of Successful Calls (ASR) The percentage of established calls from attempted calls. Number of Calls Terminated due to a Busy Line Indicates the number of calls that failed as a result of a busy line.
MediaPack Series 3.6.2.2 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. ¾ To view the call routing status, take this step: Open the 'Call Routing Status' page (Status & Diagnostics tab > Gateway Statistics menu > Calls Routing Status page item).
SIP User's Manual 3.6.2.3 3. Web-Based Management Registration Status The 'Registration Status' page displays whether the device or endpoints are registered to a SIP Registrar/Proxy server. ¾ To view Registration status, take this step: Open the 'Registration Status' page (Status & Diagnostics tab > Gateway Statistics menu > Registration Status page item).
MediaPack Series Table 3-60: SAS Registered Users Parameters Column Name Description Address of Record An address-of-record (AOR) is a SIP or SIPS URI that points to a domain with a location service that can map the URI to another URI (Contact) where the user might be available. Contact SIP URI that can be used to contact that specific instance of the User Agent for subsequent requests. 3.6.2.
SIP User's Manual 3. Web-Based Management Table 3-61: IP Connectivity Parameters Column Name IP Address Description The IP address can be one of the following: IP address defined as the destination IP address in the 'Tel to IP Routing' table. IP address resolved from the host name defined as the destination IP address in the 'Tel to IP Routing' table . Host Name Host name (or IP address) as defined in the 'Tel to IP Routing' table .
MediaPack Series Reader’s Notes SIP User's Manual 230 Document #: LTRT-65411
SIP User's Manual 4 4. ini File Configuration ini File Configuration As an alternative to configuring the device using the Web interface (as described in ''WebBased Management'' on page 21), you can configure the device by loading an ini file containing user-defined parameters.
MediaPack Series 4.2 The ini File Structure The ini file can contain any number of parameters. The ini file can contain the following types of parameters: 4.2.
SIP User's Manual 4.2.3 4. ini File Configuration Structure of ini File Table Parameters You can use anini file to configure table parameters, which include several parameters (table columns) grouped according to the applications they configure (e.g., NFS and IPSec). When loading an ini file to the device, it's recommended to include only tables that belong to applications that are to be configured (dynamic tables of other applications are empty, but static tables are not).
MediaPack Series The following displays an example of the structure of an ini file table parameter. [Table Title] ; This is the title of the table. FORMAT Item_Index = Item_Name1, Item_Name2, Item_Name3; ; This is the Format line. Item 0 = value1, value2, value3; Item 1 = value1, $$, value3; ; These are the Data lines. [\Table_Title] ; This is the end-of-the-table-mark. Refer to the following notes: Indices (in both the Format and the Data lines) must appear in the same order.
SIP User's Manual 4.2.4 4. ini File Configuration Example of an ini File Below is an example of an ini file for the VoIP device. ;Channel Params DJBufMinDelay = 75 RTPRedundancyDepth = 1 IsProxyUsed = 1 ProxyIP = 192.168.122.179 [CoderName] FORMAT CoderName_Index = CoderName_Type, CoderName_PacketInterval, CoderName_rate, CoderName_PayloadType, CoderName_Sce; CoderName 1= g7231,90 [\CoderName] CallProgressTonesFilename = 'CPUSA.dat' SaveConfiguration = 1 4.
MediaPack Series 4.4 Reference for ini File Parameters This subsection lists all the ini file parameters. References to their descriptions in the Web interface are provided except for those ini file parameters that can only be configured using the ini file. 4.4.1 Networking Parameters The networking-related ini file configuration parameters are described in the table below.
SIP User's Manual 4. ini File Configuration Parameter DNS2IP Description This ini file table parameter configures the internal DNS table for resolving host names into IP addresses. Up to four different IP addresses (in dotted-decimal notation) can be assigned to a host name.
MediaPack Series Parameter Description For an explanation on using ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233. EnableSTUN For a description of this parameter, refer to ''Configuring the Application Settings'' on page 58. STUNServerPrimaryIP For a description of this parameter, refer to ''Configuring the Application Settings'' on page 58.
SIP User's Manual 4. ini File Configuration Parameter EnableUDPPortTranslatio n Description [0] = Disable UDP port translation (default). [1] = Enable UDP port translation. When enabled, the device compares the source UDP port of the first incoming packet, to the remote UDP port stated in the opening of the channel. If the two UDP ports don't match, the NAT mechanism is activated.
MediaPack Series Parameter SyslogOutputMethod Description Determines the method used for Syslog messages. [0] = Send all Syslog messages to the defined Syslog server (default). [1] = Send all Syslog messages using the Debug Recording mechanism. [2] = Send only Error and Warning level Syslog messages using the Debug Recording mechanism. For a detailed description on Debug Recording, refer to Debug Recording (DR).
SIP User's Manual 4. ini File Configuration Parameter Description VLANNativeVLANID For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52. VLANOamVLANID For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52. VLANControlVLANID For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52. VLANMediaVLANID For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52.
MediaPack Series Parameter Description LocalControlIPAddress For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52. LocalControlSubnetMask For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52. LocalControlDefaultGW For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52. LocalOAMIPAddress For a description of this parameter, refer to ''Configuring the IP Settings'' on page 52.
SIP User's Manual 4. ini File Configuration Parameter Description PremiumServiceClassMed iaDiffServ For a description of this parameter, refer to ''Configuring the QoS Settings'' on page 65. PremiumServiceClassCon trolDiffServ For a description of this parameter, refer to ''Configuring the QoS Settings'' on page 65. GoldServiceClassDiffServ For a description of this parameter, refer to ''Configuring the QoS Settings'' on page 65.
MediaPack Series 4.4.2 System Parameters The system-related ini file configuration parameters are described in the table below. Table 4-2: System ini File Parameters Parameter EnableDiagnostics Description Checks the correct functionality of the different hardware components on the device. On completion of the check, if the test fails, the device sends information on the test results of each hardware component to the Syslog server. [0] = Rapid and Enhanced self-test mode (default).
SIP User's Manual 4. ini File Configuration Parameter Description (1) ini parameters (AdminPage) (2) 'General Security Settings' (3) 'Configuration File' (4) 'IPSec/IKE' tables (5) 'Software Upgrade Key' (6) 'Internal Firewall' (7) 'Web Access List' (8) 'Web User Accounts' [NAA] (Non Authorized Access) = Attempt to access the Web interface with a false / empty user name or password.
MediaPack Series Parameter Description acBoardCallResourcesAlarm Alarm Trap with a 'major' Alarm Status. The range is 0 to 100. The default value is 90. Note: The percentage of busy endpoints is calculated by dividing the number of busy endpoints by the total number of “enabled” endpoints. RAILowThreshold Low threshold percentage of total calls that are active (busy endpoints).
SIP User's Manual 4. ini File Configuration Parameter Description http://192.168.0.1/filename http://192.8.77.13/config https://:@/ Notes: PrtFileURL When using HTTP or HTTPS, the date and time of the ini file are validated. Only more recently-dated ini files are loaded. The optional string '' is replaced with the device's MAC address. Therefore, the device requests an ini file name that contains its MAC address.
MediaPack Series Parameter Description [0] = The immediate restart mechanism is disabled (default). [1] = The device immediately restarts after an ini file with this parameter set to 1 is loaded. BootP and TFTP Parameters The BootP parameters are special 'Hidden' parameters. Once defined and saved in the flash memory, they are used even if they don't appear in the ini file. BootPRetries Note: This parameter only takes effect from the next reset of the device.
SIP User's Manual 4. ini File Configuration Parameter Description If enabled, the device uses the vendor specific information field in the BootP request to provide device-related initial startup information such as blade type, current IP address, software version, etc. For a full list of the vendor specific Information fields, refer to the Product Reference Manual. The BootP/TFTP configuration utility displays this information in the 'Client Info' column (refer to the Product Reference Manual).
MediaPack Series 4.4.3 Web and Telnet Parameters The Web- and Telnet-related ini file configuration parameters are described in the table below. Table 4-3: Web and Telnet ini File Parameters Parameter WebAccessList_x Description Defines up to ten IP addresses that are permitted to access the device's Web interface and Telnet interfaces. Access from an undefined IP address is denied. This security feature is inactive (i.e., the device can be accessed from any IP address) when the table is empty.
SIP User's Manual 4. ini File Configuration Parameter Description DisableWebConfig For a description on using ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233. Determines whether the entire Web interface is in read-only mode. [0] = Enables modifications of parameters (default). [1] = Web interface in read-only mode. When in read-only mode, parameters can't be modified.
MediaPack Series Parameter UseWebLogo Description [0] = Logo image is used (default). [1] = Text string is used instead of a logo image. If enabled, AudioCodes' default logo (or any other logo defined by the LogoFileName parameter) is replaced with a text string defined by the WebLogoText parameter. WebLogoText Text string that replaces the logo image. The string can be up to 15 characters. LogoWidth Width (in pixels) of the logo image.
SIP User's Manual 4. ini File Configuration Parameter Description SIPSRequireClientCertificat e For a description of this parameter, refer to ''Configuring the General Security Settings'' on page 90. PeerHostNameVerification Mode For a description of this parameter, refer to ''Configuring the General Security Settings'' on page 90. VerifyServerCertificate For a description of this parameter, refer to ''Configuring the General Security Settings'' on page 90.
MediaPack Series Parameter Description format of this parameter is as follows: [IPSEC_SPD_TABLE] Format SPD_INDEX = IPSecMode, IPSecPolicyRemoteIPAddress, IPSecPolicySrcPort, IPSecPolicyDStPort,IPSecPolicyProtocol, IPSecPolicyLifeInSec, IPSecPolicyLifeInKB, IPSecPolicyProposalEncryption_X, IPSecPolicyProposalAuthentication_X, IPSecPolicyKeyExchangeMethodIndex, IPSecPolicyLocalIPAddressType, IPSecPolicyRemoteTunnelIPAddress, IPsecPolicyRemoteSubnetMask; [\IPSEC_SPD_TABLE] For example: [IPSEC_SPD_TABLE] For
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MediaPack Series Parameter HTTPSRootFileName Description Defines the name of the HTTPS trusted root certificate file to be loaded via TFTP. The file must be in base64-encoded PEM (Privacy Enhanced Mail) format. The valid range is a 47-character string. Note: This parameter is only relevant when the device is loaded via BootP/TFTP. For information on loading this file via the Web interface, refer to the Product Reference Manual.
SIP User's Manual 4. ini File Configuration Parameter Description For a description of this parameter, refer to ''Configuring the Firewall Settings'' on page 84. AccessList_MatchCount 4.4.5 For a description of configuring with ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233. RADIUS Parameters The RADIUS-related ini file configuration parameters are described in the table below.
MediaPack Series Parameter Description RadiusLocalCacheTimeout For a description of this parameter, refer to ''Configuring the General Security Settings'' on page 90. RadiusVSAVendorID For a description of this parameter, refer to ''Configuring the General Security Settings'' on page 90. RadiusVSAAccessAttribute For a description of this parameter, refer to ''Configuring the General Security Settings'' on page 90. 4.4.
SIP User's Manual 4. ini File Configuration Parameter Description SNMP Trap Parameters SNMPManagerTableIP_x For a description of this parameter, refer to ''Configuring the SNMP Managers Table'' on page 201. SNMPManagerTrapPort_x For a description of this parameter, refer to ''Configuring the SNMP Managers Table'' on page 201. SNMPManagerTrapUser_x This parameter can be set to the name of any configured SNMPV3 user to associate with this trap destination.
MediaPack Series 4.4.7 SIP Configuration Parameters The SIP-related ini file configuration parameters are described in the table below. Table 4-7: SIP ini File Parameters Parameter ReliableConnectionPersistent Mode Description Determines whether all TCP/TLS connections are set as persistent and therefore, not released. [0] = Disable (default) - all TCP connections (except those that are set to a proxy IP) are released if not used by any SIP dialog\transaction.
SIP User's Manual 4. ini File Configuration Parameter Description SIPReroutingMode For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112. EnableProxyKeepAlive For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112. ProxyKeepAliveTime For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112.
MediaPack Series Parameter Description ProxySet_IsProxyHotSwap; ProxySet 0 = 0, 60, 0, 0; ProxySet 1 = 1, 60, 1, 0; [\ProxySet] Notes: This table parameter can include up to 6 indices (0-5). For configuring the Proxy Sets, refer to the ini file parameter ProxyIP. For configuring the Proxy Set ID table using the Web interface and for a description of the parameters of this ini file table, refer to ''Proxy Sets Table'' on page 120.
SIP User's Manual 4. ini File Configuration Parameter Description UseGatewayNameForOptions For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112. IsProxyHotSwap For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112. HotSwapRtx For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112.
MediaPack Series Parameter Description Parameters'' on page 112. RegisterOnInviteFailure For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112. RegistrationTimeThreshold For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112. ZeroSDPHandling Determines the device's response to an incoming SDP with an IP address of 0.0.0.0 in the Connection line. [0] Sets the IP address of the outgoing SDP Connection line to 0.0.0.
SIP User's Manual 4. ini File Configuration Parameter IPGroup Description This ini file table parameter configures the IP Group table.
MediaPack Series Parameter Description UseTelURIForAssertedID For a description of this parameter, refer to ''SIP General Parameters'' on page 101. EnableRPIheader For a description of this parameter, refer to ''SIP General Parameters'' on page 101. IsUserPhone For a description of this parameter, refer to ''SIP General Parameters'' on page 101. IsUserPhoneInFrom For a description of this parameter, refer to ''SIP General Parameters'' on page 101.
SIP User's Manual 4. ini File Configuration Parameter EnableRport Description Enables / disables the usage of the 'rport' parameter in the Via header. [0] = Enabled. [1] = Disabled (default). The device adds an 'rport' parameter to the Via header of each outgoing SIP message. The first Proxy that receives this message sets the 'rport' value of the response to the actual port from which the request was received.
MediaPack Series Parameter Description EnableEarlyMedia For a description of this parameter, refer to ''SIP General Parameters'' on page 101. EnableTransfer For a description of this parameter, refer to ''Supplementary Services'' on page 138. XferPrefix For a description of this parameter, refer to ''Supplementary Services'' on page 138. EnableMicrosofExt Modifies the called number for numbers received with Microsoft's proprietary "ext=xxx" parameter in the SIP INVITE URI user part.
SIP User's Manual 4. ini File Configuration Parameter Description Module = Module number. For example: [CallWaitingPerPort] CallWaitingPerPort 0 = 0,1,1$$; CallWaitingPerPort 1 = 1,2,1$$; [\CallWaitingPerPort] If enabled, when an FXS interface receives a call on a busy endpoint, it responds with a 182 response (and not with a 486 busy). The device plays a call waiting indication signal. When hook-flash is detected, the device switches to the waiting call.
MediaPack Series Parameter Description (by the device) in the Refer-To header value in the REFER messages sent by the device to the remote parties. The remote parties join the conference by sending INVITE messages to the media server using this conference URI. Enable3WayConference For a description of this parameter, refer to “Supplementary Services” on page 138. ConferenceCode For a description of this parameter, refer to “Supplementary Services” on page 138.
SIP User's Manual 4. ini File Configuration Parameter Description RTPOnlyMode For a description of this parameter, refer to ''Advanced Parameters'' on page 129. TimeoutBetween100And18x Defines the timeout (in msec) between receiving a 100 Trying response and a subsequent 18x response. If a 18x response is not received before this timer expires, the call is disconnected. The valid range is 0 to 32,000. The default value is 0 (i.e., no timeout).
MediaPack Series Parameter Description digits in-band (transparent of RFC 2833) in addition to out-of-band DTMF messages. Note: Usually this mode is not recommended. FirstCallRBTId For a description of this parameter, refer to ''Advanced Parameters'' on page 129. EnableReasonHeader For a description of this parameter, refer to ''SIP General Parameters'' on page 101. 3xxBehavior For a description of this parameter, refer to ''SIP General Parameters'' on page 101.
SIP User's Manual 4. ini File Configuration Parameter Description Notes: SITDetectorEnable SourceIPAddressInput You can omit either the username or password using the sign '$$'. If omitted, the port's phone number is used for authentication. The indexing of this ini file table parameter starts at 1. To configure the authentication username and password using the Web interface, refer to Authentication on page 174.
MediaPack Series Parameter SASBindingMode SASEnableENUM SASRegistrationManipulation Description Determines the SAS application database binding mode. [0] URI = If the incoming AoR in the INVITE requests is using a ‘tel:’ URI or ‘user=phone’ is defined, the binding is performed according to the user part of the URI only. Otherwise, the binding is according to the entire URI, i.e., User@Host (default). [1] User Part only = The binding is always performed according to the User Part only.
SIP User's Manual 4. ini File Configuration Parameter Description includes up to five groups of coders (consisting of up to five coders per group) that can be associated with IP or Tel profiles ('Coder Group Settings' page in the Web interface -- refer to ''Coder Group Settings'' on page 170). The first group of coders (indices 0 through 4) is the default coder list and default coder group.
MediaPack Series Parameter IPProfile Description This ini file table parameter configures the IP profiles table.
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MediaPack Series Parameter Description SMDI For a description of this parameter, refer to “Configuring the Voice Mail (VM) Parameters” on page 190. SMDITimeOut For a description of this parameter, refer to “Configuring the Voice Mail (VM) Parameters” on page 190. LineTransferMode For a description of this parameter, refer to ''Configuring the Voice Mail (VM) Parameters'' on page 190.
SIP User's Manual 4.4.9 4. ini File Configuration PSTN Parameters The PSTN-related ini file configuration parameters are described in the table below. Table 4-9: PSTN ini File Parameters Parameter Description CallPriorityMode For a description of this parameter, refer to ''Supplementary Services'' on page 138. MLPPDiffserv For a description of this parameter, refer to ''Supplementary Services'' on page 138.
MediaPack Series 4.4.10 Analog Telephony Parameters The analog telephony-related ini file configuration parameters are described in the table below. Table 4-10: Analog Telephony ini File Parameters Parameter Prefix2ExtLine Description Defines a string prefix (e.g., '9' dialed for an external line) that when identified causes the device's FXS port to play a secondary dial tone and then restart digit collection. The valid range is a 1-character string. The default is an empty string.
SIP User's Manual 4. ini File Configuration Parameter Description Notes: TargetOfChannel The parameter can appear up to 25 times (i.e., up to 25 different metering rules can be defined). To configure the Charge Codes table using the Web interface, refer to “Charge Codes Table”. For an explanation on configuration using ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233.
MediaPack Series Parameter CallerDisplayInfo Description This ini file table parameter enables the device to send Caller ID information to IP when a call is made. The format of this parameter is as follows: [CallerDisplayInfo] FORMAT CallerDisplayInfo_Index = CallerDisplayInfo_DisplayString, CallerDisplayInfo_IsCidRestricted, CallerDisplayInfo_Port, CallerDisplayInfo_Module; [\CallerDisplayInfo] Where, DisplayString = Caller ID string.
SIP User's Manual 4. ini File Configuration Parameter Description FwdInfo 3 = 3,2005,30,2,$$; [\FwdInfo] Notes: EnableCallerID The indexing of this parameter starts at 1. The device ports starts at 0. This parameter can appear up to 24 times for MP-124. To configure the Call Forward table using the Web interface, refer to ''Call Forward'' on page 178. For an explanation on ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233.
MediaPack Series Parameter Description Parameters'' on page 129. FXSOOSBehavior For a description of this parameter, refer to ''Advanced Parameters'' on page 129. NumberOfWaitingIndications For a description of this parameter, refer to ''Supplementary Services'' on page 138. TimeBetweenWaitingIndicati ons For a description of this parameter, refer to ''Supplementary Services'' on page 138.
SIP User's Manual 4. ini File Configuration Parameter Description KeyCFBusy For a description of this parameter, refer to ''Keypad Features'' on page 147. KeyCFBusyOrNoAnswer For a description of this parameter, refer to ''Keypad Features'' on page 147. KeyCFDoNotDisturb For a description of this parameter, refer to ''Keypad Features'' on page 147. KeyCFDeact For a description of this parameter, refer to ''Keypad Features'' on page 147.
MediaPack Series Parameter Description BlindTransferDisconnectTim eout Defines the duration (in milliseconds) for which the device waits for a disconnection from the Tel side after the Blind Transfer Code (KeyBlindTransfer) has been identified. When this timer expires, a SIP REFER message is sent toward the IP side. If this parameter is set to 0, the REFER message is immediately sent. The valid range is 0 to 1,000,000. The default is 0.
SIP User's Manual 4. ini File Configuration Parameter Description [0] = FSK-based signaling (default) [1] = DTMF-based signaling Note: This parameter is applicable only to FXS interfaces. EnableDID This ini file table parameter enables support for Japan NTT 'Modem' Direct Inward Dialing (DID). FXS interfaces can be connected to Japan's NTT PBX using 'Modem' DID lines. These DID lines are used to deliver a called number to the PBX.
MediaPack Series Parameter Description CurrentDisconnectDuration is 200 msec, then the detection range is 100 to 500 msec. CurrentDisconnectDefaultThr eshold Determines the line voltage threshold which, when reached, is considered a current disconnect detection. The valid range is 0 to 20 Volts. The default value is 4 Volts. Note: Applicable only to FXO interfaces.
SIP User's Manual 4. ini File Configuration 4.4.11 Number Manipulation and Routing Parameters The number manipulation and routing-related ini file configuration parameters are described in the table below. Table 4-11: Number Manipulation and Routing ini File Parameters Parameter TrunkGroup Description This ini file table parameter defines the device's endpoints and assigns them to Hunt Groups.
MediaPack Series Parameter Description TrunkGroupSettings 0 = 1, 0, 5, audiocodes, user, 1; TrunkGroupSettings 1 = 2, 1, 0, localname, user1, 2; [\TrunkGroupSettings] Notes: This parameter can include up to 24 indices. For configuring HuntGroup Settings using the Web interface, refer to ''Configuring Hunt Group Settings'' on page 183. For a description on using ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233.
SIP User's Manual 4. ini File Configuration Parameter PSTNPrefix Description The destination and source phone prefixes (PREFIX_DestinationPrefix and PREFIX_SourcePrefix respectively) can be a single number or a range of numbers. Parameters can be skipped using two dollar ($$) symbols, for example: Prefix = $$,10.2.10.2,202,1. The destination IP address (PREFIX_DestAddress) can be either in dotted-decimal notation or FQDN.
MediaPack Series Parameter Description destination number, source number,and source IP address. The source IP address (SourceAddress) can include the 'x' wildcard to represent single digits. For example: 10.8.8.xx represents all IP addresses between 10.8.8.10 and 10.8.8.99. The source IP address (SourceAddress) can include the asterisk ('*') wildcard to represent any number between 0 and 255. For example, 10.8.8.* represents all addresses between 10.8.8.0 and 10.8.8.255.
SIP User's Manual 4. ini File Configuration Parameter Description For example: [NumberMapTel2Ip] NumberMapTel2Ip 0 = 01,$$,*,0,0,2,$$,$$,971,$$,$$,$$,$$; NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; [\NumberMapTel2Ip] Notes: NumberMapIP2Tel This table parameter can include up to 100 indices. The parameters SourceAddress and IsPresentationRestricted are not applicable. Set these to $$.
MediaPack Series Parameter Description NumberMapIp2Tel_NumberPlan are not applicable. Set these to $$. SourceNumberMapTel2I P RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, and LeaveFromRight are applied if the called and calling numbers match the DestinationPrefix, SourcePrefix, and SourceAddress conditions. The manipulation rules are executed in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and Suffix2Add.
SIP User's Manual 4. ini File Configuration Parameter SourceNumberMapIP2Te l Description The manipulation rules are executed in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and Suffix2Add. Parameters can be skipped by using two dollar signs ('$$'). An asterisk ('*') represents all IP addresses. IsPresentationRestricted is set to 'Restricted' only if 'Asserted Identity Mode' is set to 'P-Asserted'.
MediaPack Series Parameter Description represents all the addresses between 10.8.8.0 and 10.8.8.255. To configure manipulation of source numbers for IP-to-Tel calls using the Web interface, refer to ''Configuring the Number Manipulation Tables'' on page 151). For a description on using ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233. SecureCallsFromIP For a description of this parameter, refer to ''Advanced Parameters'' on page 129.
SIP User's Manual 4. ini File Configuration Parameter Description interface, refer to ''Reasons for Alternative Routing'' on page 168. FilterCalls2IP For an explanation on usng ini file table parameters, refer to ''Structure of ini File Table Parameters'' on page 233. For a description of this parameter, refer to ''Advanced Parameters'' on page 129. Alternative Routing Parameters RedundantRoutingMode For a description of this parameter, refer to ''Proxy & Registration Parameters'' on page 112.
MediaPack Series Parameter Description PhoneContext 2 = 9,1,na.e164.host.com [\PhoneContext] Notes: This parameter can include up to 20 indices. Several entries with the same NPI-TON or Phone-Context are allowed. In this scenario, a Tel-to-IP call uses the first match. Phone-Context '+' is a unique as it doesn't appear in the RequestURI as a Phone-Context parameter. Instead, it's added as a prefix to the phone number. The '+' isn't removed from the phone number in the IP-to-Tel direction.
SIP User's Manual 4. ini File Configuration Parameter Description FaxModemBypassM For a description of this parameter, refer to ''Configuring the Fax / Modem / CID Settings'' on page 69. FaxModemNTEMode Determines whether the device sends RFC 2833 ANS/ANSam events upon detection of fax and/or modem answer tones (i.e., CED tone). [0] = Disabled (default). [1] = Enabled. Note: This parameter is applicable only when the fax or modem transport type is set to bypass or Transparent-with-Events.
MediaPack Series Parameter FaxModemBypassBasicRT PPacketInterval Description Determines the basic frame size that is used during fax / modem bypass sessions. [0] = Determined internally (default) [1] = 5 msec (not recommended) [2] = 10 msec [3] = 20 msec Note: When set for 5 msec (1), the maximum number of simultaneous channels supported is 120. FaxModemBypassDJBufMi nDelay Determines the Jitter Buffer delay (in milliseconds) during fax and modem bypass session.
SIP User's Manual 4. ini File Configuration Parameter Description V21ModemTransportType For a description of this parameter, refer to ''Configuring the Fax / Modem / CID Settings'' on page 69. V22ModemTransportType For a description of this parameter, refer to ''Configuring the Fax / Modem / CID Settings'' on page 69. V23ModemTransportType For a description of this parameter, refer to ''Configuring the Fax / Modem / CID Settings'' on page 69.
MediaPack Series Parameter Description selected coder is G.729. EnableEchoCanceller For a description of this parameter, refer to ''Configuring the Voice Settings'' on page 67. ECNLPMode Defines the echo cancellation Non-Linear Processing (NLP) mode. EchoCancellerAggressiveN LP [0] = NLP adapts according to echo changes (default). [1] = Disables NLP. Enables or disables the Aggressive Non-Linear Processor (NLP) in the first 0.5 second of the call.
SIP User's Manual 4. ini File Configuration Parameter Description UDTDetectorFrequencyDevi ation Defines the deviation (in Hz) allowed for the detection of each signal frequency. The valid range is 1 to 50. The default value is 50. CPTDetectorFrequencyDevi ation Defines the deviation (in Hz) allowed for the detection of each CPT signal frequency. The valid range is 1 to 30. The default value is 10. MGCPDTMFDetectionPoint [0] = DTMF event is reported on the end of a detected DTMF digit.
MediaPack Series 4.4.13 Auxiliary / Configuration Files Parameters The configuration files (i.e., auxiliary files) can be loaded to the device using the Web interface or a TFTP session (refer to ''Auxiliary Files'' on page 210). Before you load them to the device, you need to specify these files in the ini file and whether they must be stored in the non-volatile memory.
SIP User's Manual 5 5. Default Settings Default Settings You can restore the device's factory default settings or define your own default settings for the device. 5.1 Defining Default Settings The device is shipped with factory default configuration values stored on its non-volatile memory (flash). However, you can define your own default values instead of using the factory defaults. This is performed using an ini file that includes the header '[ClientDefaults]'.
MediaPack Series Reader’s Notes SIP User's Manual 306 Document #: LTRT-65411
SIP User's Manual 6 6. Auxiliary Configuration Files Auxiliary Configuration Files This section describes the auxiliary files (with the dat file extension), which are loaded, in addition to the ini file, to the device. You can load the auxiliary files to the device using one of the following methods: 6.
MediaPack Series You can specify several tones of the same type. These additional tones are used only for tone detection. Generation of a specific tone conforms to the first definition of the specific tone. For example, you can define an additional dial tone by appending the second dial tone's definition lines to the first tone definition in the ini file. The device reports dial tone detection if either of the two tones is detected.
SIP User's Manual 6. Auxiliary Configuration Files • Second Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the second cadence on-off cycle. Can be omitted if there isn't a second cadence. • Second Signal Off Time [10 msec]: 'Signal Off' period (in 10 msec units) for the second cadence on-off cycle. Can be omitted if there isn't a second cadence. • Third Signal On Time [10 msec]: 'Signal On' period (in 10 msec units) for the third cadence ON-OFF cycle.
MediaPack Series 6.2 Configuring the Distinctive Ringing Section of the ini File Distinctive Ringing is only applicable to FXS interface. Using the distinctive ringing section of this 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. The same ringing frequency is used for all the ringing pattern cadences.
SIP User's Manual 6.2.1 6. Auxiliary Configuration Files Examples of Ringing Signals Below is an example of a ringing burst: #Three ringing bursts followed by repeated ringing of 1 sec on and 3 sec off.
MediaPack Series Note: The Prerecorded tones are used only for generation of tones. Detection of tones is performed according to the CPT file. The PRT is a *.dat file containing a set of prerecorded tones that can be played by the device. Up to 40 tones (totaling approximately 10 minutes) can be stored in a single PRT file on the device's flash memory.
SIP User's Manual 6. Auxiliary Configuration Files This means, for example, that changing impedance matching or hybrid balance doesn't require hardware modifications, so that a single device is able to meet requirements for different markets. The digital design of the filters and gain stages also ensures high reliability, no drifts (over temperature or time) and simple variations between different line types.
MediaPack Series An example of a User Information file is shown in the figure below: Figure 6-1: Example of a User Information File Note: The last line in the User Information file must end with a carriage return (i.e., by pressing the key).
SIP User's Manual 7 7. IP Telephony Capabilities IP Telephony Capabilities This section describes the device's IP telephony capabilities. 7.1 Stand-Alone Survivability (SAS) Feature The device's Stand-Alone Survivability (SAS) feature ensures telephony communication continuity (survivability) for enterprises using hosted IP services (such as IP Centrex) or IPPBX in cases of failure of these entities.
MediaPack Series 7.1.1 Configuring SAS For configuring the device to operate with SAS, perform the following configurations: 7.1.2 IsProxyUsed = 1 ProxyIP 0 =
SIP User's Manual 7. IP Telephony Capabilities To configure support for emergency calls, configure the parameters below. The device and the SAS feature are configured independently. If the device and the SAS agent use different proxies, then the device's proxy server is defined using the 'Use Default Proxy' parameter, while the SAS proxy agent is defined using the 'Proxy Set' table and SASProxySet parameter. 7.
MediaPack Series • TxDTMFOption = 2 (ini file); '1st to 5th Tx DTMF Option' field = 'NOTIFY' (Web interface -- refer to ''DTMF & Dialing Parameters'' on page 125) Note that in this mode, DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0 ('DTMF Transport Type' field = 'DTMF Mute' -- Web interface)]. Using RFC 2833 relay with Payload type negotiation: DTMF digits are carried to the remote side as part of the RTP stream in accordance with RFC 2833 standard.
SIP User's Manual 7. IP Telephony Capabilities 7.3 Fax and Modem Capabilities 7.3.1 Fax/Modem Operating Modes The device supports two modes of operations: 7.3.2 Fax / modem negotiation that isn’t performed during the establishment of the call. VBD mode for V.152 implementation (refer to ''Supporting V.152 Implementation'' on page 325): fax / modem capabilities are negotiated between the device and the remote endpoint at the establishment of the call.
MediaPack Series 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). In addition, you can enable or disable Error Correction Mode (ECM) fax mode using the FaxRelayECMEnable parameter. When using T.
SIP User's Manual 7. IP Telephony Capabilities The network packets generated and received during the bypass period are regular voice RTP packets (per the selected bypass coder), but with a different RTP payload type (according to the parameters FaxBypassPayloadType and ModemBypassPayloadType). During the bypass period, the coder uses the packing factor, which is defined by the parameter FaxModemBypassM.
MediaPack Series 7.3.2.3 Fax / Modem NSE Mode In this mode, fax and modem signals are transferred using Cisco-compatible Pass-through bypass mode. Upon detection of fax or modem answering tone signal, the terminating device sends three to six special NSE RTP packets (using NSEpayloadType, usually 100). These packets signal the remote device to switch to G.711 coder (according to the parameter FaxModemBypassCoderType). After a few NSE packets are exchanged between the devices, both devices start using G.
SIP User's Manual 7. IP Telephony Capabilities V34ModemTransportType = 0 BellModemTransportType = 0 Additional configuration parameters: • CoderName • DJBufOptFactor • EnableSilenceCompression • EnableEchoCanceller Note: This mode can be used for fax, but is not recommended for modem transmission. Instead, use the modes Bypass (refer to ''Fax/Modem Bypass Mode'' on page 320) or Transparent with Events (refer to ''Fax / Modem Transparent with Events Mode'' on page 323) for modem. 7.3.2.
MediaPack Series After a few seconds upon detection of fax V.21 preamble or super G3 fax signals, the device sends a second Re-INVITE enabling the echo canceller (the echo canceller is disabled only on modem transmission). A ‘gpmd’ attribute is added to the SDP according to the following format: For G.711A-law: a=gpmd:0 vbd=yes;ecan=on (or off, for modems) For G.
SIP User's Manual 7.3.3.1 7. IP Telephony Capabilities Using Bypass Mechanism for V.34 Fax Transmission In this proprietary scenario, the device uses bypass (or NSE) mode to transmit V.34 faxes, enabling the full utilization of its speed. Configure the following parameters to use bypass mode for both T.30 and V.
MediaPack Series When in VBD mode for V.152 implementation, support is negotiated between the device and the remote endpoint at the establishment of the call. During this time, initial exchange of call capabilities is exchanged in the outgoing SDP. These capabilities include whether VBD is supported and associated RTP payload types ('gpmd' SDP attribute), supported codecs, and packetization periods for all codec payload types ('ptime' SDP attribute).
SIP User's Manual 7.4.1.1 7. IP Telephony Capabilities One-Stage Dialing One-stage dialing is when the FXO device receives an IP-to-Tel call, off-hooks the PBX line connected to the telephone, and then immediately dials the destination telephone number. In other words, the IP caller doesn't dial the PSTN number upon hearing a dial tone.
MediaPack Series 7.4.1.2 Two-Stage Dialing Two-stage dialing is when the IP caller is required to dial twice. The caller initially dials to the FXO device and only after receiving a dial tone from the PBX (via the FXO device), dials the destination telephone number. Figure 7-3: Call Flow for Two-Stage Dialing Two-stage dialing implements the Dialing Time feature. Dialing Time allows you to define the time that each digit can be separately dialed.
SIP User's Manual 7. IP Telephony Capabilities Detection of Reorder, Busy, Dial, and Special Information Tone (SIT) tones: The call is immediately disconnected after a Reorder, Busy, Dial, or SIT tone is detected on the Tel side (assuming the PBX / CO generates this tone). This method requires the correct tone frequencies and cadence to be defined in the Call Progress Tones file.
MediaPack Series 7.4.2 Telephone-to-IP Calls The FXO device provides the following FXO operating modes for Tel-to-IP calls: 7.4.2.
SIP User's Manual 7. IP Telephony Capabilities Figure 7-5: Collecting Digits Mode 7.4.2.3 Ring Detection Timeout The operation of Ring Detection Timeout depends on the following: No automatic dialing and Caller ID is enabled: if the second ring signal doesn’t arrive for Ring Detection Timeout, the device doesn’t initiate a call to the IP.
MediaPack Series The blind transfer call process is as follows: 7.5 • FXO receives a REFER request from the IP side • FXO sends a hook-flash to the PBX, dials the digits (that are received in the Refer-To header), and then drops the line (on-hook). Note that the time between flash to dial is according to the WaitForDialTime parameter. • PBX performs the transfer internally Hold / Transfer toward the IP side: The FXO device doesn't initiate hold / transfer as a response to input from the Tel side.
SIP User's Manual 7. IP Telephony Capabilities Below is an example of SIP messages implementing 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" ;tag=1c25298 To: Call-ID: 11923@10.33.2.53 CSeq: 1 INVITE Contact: X- Detect: Request=CPT,FAX SIP/2.0 200 OK Via: SIP/2.0/UDP 10.33.2.
MediaPack Series Notes: 7.7 • RTP Multiplexing must be enabled on both devices. • When VLANs are imlemented, the RTP Multiplexing mechanism is not supported. Dynamic Jitter Buffer Operation Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames.
SIP User's Manual 7. IP Telephony Capabilities condition occurs, the correction is performed by dropping several voice packets simultaneously or by adding several BFI packets simultaneously, so that the Jitter Buffer returns to its normal condition. 7.8 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.
MediaPack Series 7.9 IPConnQoSMaxAllowedPL IPConnQoSMaxAllowedDelay Mapping PSTN Release Cause to SIP Response The device's FXO interface interoperates between the SIP network and the PSTN/PBX. This interoperability includes the mapping of PSTN/PBX Call Progress Tones to SIP 4xx or 5xx responses for IP-to-Tel calls. The converse is also true: for Tel-to-IP calls, the SIP 4xx or 5xx responses are mapped to tones played to the PSTN/PBX.
SIP User's Manual Attribute Number Attribute Name 7. IP Telephony Capabilities VSA No. Address 26 H323-ConfID 26 H323-SetupTime 26 H323-CallOrigin Purpose Value Format Example gateway AAA1 Acc Up to 32 octets Start Acc Stop Acc String Start Acc Stop Acc 24 H.
MediaPack Series Attribute Number Attribute Name VSA No. Type Purpose Value Format Example (start or stop) Note: ‘start’ isn’t supported on the Calling Card application. AAA1 Acc Stop Acc Start Acc Stop Acc 41 Acct-DelayTime No.
SIP User's Manual 7. IP Telephony Capabilities calling-station-id = 202 // Accounting non-standard parameters: (4923 33) h323-gw-id = (4923 23) h323-remote-address = 212.179.22.214 (4923 1) h323-ivr-out = h323-incoming-conf-id:02102944 600a1899 3fd61009 0e2f3cc5 (4923 30) h323-disconnect-cause = 22 (0x16) (4923 27) h323-call-type = VOIP (4923 26) h323-call-origin = Originate (4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5 7.
MediaPack Series Field Name Description Intrv Packet Interval RtpIp RTP IP Address Port Remote RTP Port TrmSd Initiator of Call Release (IP, Tel, Unknown) TrmReason Termination Reason Fax Fax Transaction during the Call InPackets Number of Incoming Packets OutPackets Number of Outgoing Packets PackLoss Local Packet Loss RemotePackLoss Number of Outgoing Lost Packets UniqueId unique RTP ID SetupTime Call Setup Time ConnectTime Call Connect Time ReleaseTime Call Release Time RTP
SIP User's Manual 7. IP Telephony Capabilities be any string. Otherwise, the "servername" is equal to "RegistrarIP" (either FQDN or numerical IP address), if configured. Otherwise, the "servername" is equal to "ProxyName" if configured. The "ProxyName" can be any string. Otherwise, the "servername" is equal to "ProxyIP" (either FQDN or numerical IP address). The parameter GWRegistrationName can be any string. This parameter is used only if registration is per device.
MediaPack Series 7.13 Configuration Examples 7.13.1 SIP Call Flow The SIP call flow (shown in the following figure), describes SIP messages exchanged between two devices during a simple call. In this call flow example, device (10.8.201.158) with phone number ‘6000’ dials device (10.8.201.161) with phone number ‘2000’. Figure 7-6: SIP Call Flow F1 (10.8.201.108 >> 10.8.201.10 INVITE): INVITE sip:1000@10.8.201.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.
SIP User's Manual 7. IP Telephony Capabilities F2 (10.8.201.10 >> 10.8.201.108 TRYING): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd From: ;tag=1c5354 To: Call-ID: 534366556655skKw-8000--1000@10.8.201.108 Server: Audiocodes-Sip-Gateway/MediaPack/v.5.40.010.006 CSeq: 18153 INVITE Content-Length: 0 F3 (10.8.201.10 >> 10.8.201.108 180 RINGING): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.8.201.
MediaPack Series ACK sip:1000@10.8.201.10;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ From: ;tag=1c5354 To: ;tag=1c7345 Call-ID: 534366556655skKw-8000--1000@10.8.201.108 User-Agent: Audiocodes-Sip-Gateway/MediaPack/v.5.40.010.006 CSeq: 18153 ACK Supported: 100rel,em Content-Length: 0 Note: Phone ‘8000’ goes on-hook and device 10.8.201.108 sends a BYE to device 10.8.201.10. Voice path is established. F6 (10.8.201.108 >> 10.8.201.
SIP User's Manual 7. IP Telephony Capabilities REGISTER sip:10.2.2.222 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200 From: ;tag=1c17940 To: Call-ID: 634293194@10.1.1.200 User-Agent: Audiocodes-Sip-Gateway/MediaPack/v.5.40.010.006 CSeq: 1 REGISTER Contact: sip:122@10.1.1.200: Expires:3600 2. Upon receipt of this request, the Registrar/Proxy returns 401 Unauthorized response. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.1.200 From:
MediaPack Series 6. Final stage: • The A1 result: The nonce from the proxy response is ‘11432d6bce58ddf02e3b5e1c77c010d2’. • The A2 result: The equation to be evaluated is ‘A1:11432d6bce58ddf02e3b5e1c77c010d2:A2’. • The MD5 algorithm is run on this equation. The outcome of the calculation is the response needed by the device to register with the Proxy. • The response is ‘b9c45d0234a5abf5ddf5c704029b38cf’. At this time, a new REGISTER request is issued with the following response: REGISTER sip:10.2.
SIP User's Manual 7. IP Telephony Capabilities ¾ To configure the two devices for call communication, take these 4 steps: 1. For the first device (10.2.37.10), in the ‘Endpoint Phone Number Table' page (refer to Configuring the Endpoint Phone Numbers on page 181), assign the phone numbers 101 to 104 to the device's endpoints. Figure 7-7: Assigning Phone Numbers to Device 10.2.37.10 2. For the second device (10.2.37.
MediaPack Series 7.13.4 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). This is as if the remote office is located in the head office (where the PBX is installed). PBX extensions are connected through FXO ports to the IP network, instead of being connected to individual telephone stations.
SIP User's Manual 7. IP Telephony Capabilities 7.13.4.1 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 device). ¾ To make a call from the FXS device, take these 3 steps: 1. Off-hook the phone and wait for the dial tone from the PBX. This is as if the phone is connected directly to the PBX.
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 7-11: MWI for Remote Extensions 7.13.4.4 Call Waiting for Remote Extensions When the FXO device detects a Call Waiting indication (FSK data of the Caller Id CallerIDType2) from the PBX, it sends a proprietary INFO message, which includes the caller identification to the FXS device.
SIP User's Manual 2. 7. IP Telephony Capabilities In the ‘Automatic Dialing’ page (refer to ''Automatic Dialing'' on page 175), 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’. Figure 7-14: Automatic Dialing Configuration 3. In the ‘Tel to IP Routing’ page (refer to ''Tel to IP Routing Table'' on page 160), enter 20 for the destination phone prefix, and 10.1.10.
MediaPack Series 2. In the ‘Automatic Dialing’ page, enter the phone numbers of the FXS device in the ‘Destination Phone Number’ fields. When a ringing signal is detected at Port #1, the FXO device automatically dials the number ‘100’. Figure 7-17: Automatic Dialing Configuration 3. 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 7-18: FXO Tel-to-IP Routing Configuration 4.
SIP User's Manual 7. IP Telephony Capabilities Figure 7-19: SIP Trunking Example Architecture ¾ To configure call routing between Enterprise and two ITSPs using the device, take these 9 steps: 1. Enable the device to register to a Proxy / Registrar server, using the parameter IsRegisterNeeded in the 'Proxy & Registration' page (refer to ''Proxy & Registration Parameters'' on page 112). 2.
MediaPack Series • Proxy Set #2 includes two IP addresses of the second ITSP (ITSP 2) - 10.8.8.40 and 10.8.8.10 - and using TCP. The figure below displays the configuration of Proxy Set ID #1. Perform similar configuration for Proxy Set ID #2, but using different IP addresses. Figure 7-20: Configuring Proxy Set ID #1 in the Proxy Sets Table Page 3. In the 'IP Group Table' page (refer to ''Configuring the IP Groups'' on page 186), configure the two IP Groups #1 and #2.
SIP User's Manual 7. IP Telephony Capabilities Figure 7-23: Configuring Hunt Groups Settings 6. In the 'Authentication' page (refer to Authentication on page 174), for channels 5-8 (i.e., Hunt Group ID #2), define for each channel the registration (authentication) user name and password. Figure 7-24: Configuring Username and Password for Channels 5-8 in Authentication Page 7.
MediaPack Series Figure 7-27: Configuring Tel-to-IP Routing 7.14 Working with Supplementary Services The device supports the following supplementary services: Call Hold and Retrieve (refer to ''Call Hold and Retrieve'' on page 356). Consultation / Alternate (refer to “Consultation / Alternate” on page 359). Call Transfer (refer to ''Call Transfer'' on page 359). Call Forward: 3xx Redirect Responses (refer to “Call Forward” on page 360).
SIP User's Manual 7. IP Telephony Capabilities a=sendonly in the SDP according to the parameter HoldFormat. Receiving Hold / Retrieve: When an active call receives a Re-INVITE message with either the IP address 0.0.0.0 or the ‘inactive’ string in SDP, the device stops sending RTP and plays a local Held tone. When an active call receives a Re-INVITE message with the ‘sendonly’ string in SDP, the device stops sending RTP and listens to the remote party.
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.
SIP User's Manual 7. IP Telephony Capabilities The previous flowchart describes the following "double" call hold scenario: 1. A calls B and establishes a voice path. 2. A places B on hold; A hears a Dial tone and B hears a Held tone. 3. A calls C and establishes a voice path. 4. B places A on hold; B hears a Dial tone. 5. B calls D and establishes a voice path. 6. A ends call with C; A hears a Held tone. 7. B ends call with D. 8. B retrieves call with A.
MediaPack Series • After A completes dialing C, A can perform the transfer by on-hooking the A phone. • After the transfer is complete, B and C parties are engaged in a call. The transfer can be initiated at any of the following stages of the call between A and C: • Just after completing dialing C phone number - transfer from setup. • While hearing Ringback – transfer from alert. • While speaking to C - transfer from active.
SIP User's Manual 7. IP Telephony Capabilities Notes: • When call forward is initiated, the device sends a SIP 302 response with a contact that contains the phone number from the forward table and its corresponding IP address from the routing table (or when a proxy is used, the proxy’s IP address). • For receiving call forward, the device handles SIP 3xx responses for redirecting calls with a new contact. 7.14.
MediaPack Series To configure MWI, set the following parameters: EnableMWI (or using the Web interface, refer to ''Supplementary Services'' on page 138) MWIServerIP (or using the Web interface, refer to ''Supplementary Services'' on page 138) MWIAnalogLamp (or using the Web interface, refer to ''Supplementary Services'' on page 138) MWIDisplay (or using the Web interface, refer to ''Supplementary Services'' on page 138) StutterToneDuration (or using the Web interface, refer to ''Supplemen
SIP User's Manual 7. IP Telephony Capabilities Enable or disable (per port) the caller ID generation (for FXS) and detection (for FXO) using the ‘Generate / Detect Caller ID to Tel’ table (EnableCallerID). If a port isn’t configured, its caller ID generation / detection are determined according to the global parameter EnableCallerID. EnableCallerIDTypeTwo: disables / enables the generation of Caller ID type 2 when the phone is off-hooked (used for call waiting).
MediaPack Series 7. Capture the RTP using Wireshark (you can also use DSP trace) and send the file to AudioCodes. 7.14.7.3 Caller ID on the IP Side Caller ID is provided by the From header containing the caller's name and "number", for example: From: “David” ;tag=35dfsgasd45dg If Caller ID is restricted (received from Tel or configured in the device), the From header is set to: From: “anonymous”
SIP User's Manual 8. Networking Capabilities 8 Networking Capabilities 8.
MediaPack Series The following figure illustrates the device's supported NAT architecture. Figure 8-1: 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. Two different streams traverse through NAT: signaling and media.
SIP User's Manual 8. Networking Capabilities To enable STUN, perform the following: Enable the STUN feature using either the Web interface (refer to ''Configuring the Application Settings'' on page 58) or the ini file (set EnableSTUN to 1).
MediaPack Series You can control the activation of No-Op packets by using the ini file parameter NoOpEnable. If No-Op packet transmission is activated, you can control the time interval in which No-Op packets are sent in the case of silence (i.e., no RTP or T.38 traffic). This is performed using the ini file parameter NoOpInterval. For a description of the RTP No-Op ini file parameters, refer to ''Networking Parameters'' on page 236.
SIP User's Manual 8. Networking Capabilities When a set of routers operating within the same subnet serve as devices to that network and intercommunicate using a dynamic routing protocol, the routers can determine the shortest path to a certain destination and signal the remote host the existence of the better route. Using multiple router support, the device can utilize these router messages to change its next hop and establish the best path.
MediaPack Series The device can be configured to set a different DiffServ value to IP packets according to their class-of-service: Network, Premium Media, Premium Control, Gold, and Bronze. The DiffServ parameters are described in ''Networking Parameters'' on page 236. For the mapping of an application to its class-of-service, refer to ''IEEE 802.1p/Q (VLANs and Priority)'' on page 370. 8.8 VLANS and Multiple IPs 8.8.
SIP User's Manual 8. Networking Capabilities Traffic type tagging can be used to implement Layer 2 VLAN security. By discriminating traffic into separate and independent domains, the information is preserved within the VLAN. Incoming packets received from an incorrect VLAN are discarded. The traffic tagging mechanism is as follows: Outgoing packets (from the device to the switch): All outgoing packets are tagged, each according to its interface (Control, Media or OAMP).
MediaPack Series Notes: • For security, the VLAN mechanism is activated only when the device is loaded from the flash memory. Therefore, when using BootP: Load an ini file with VlanMode set to 1 and SaveConfiguration set to 1. Then (after the device is active) reset the device with TFTP disabled or by using any method except for BootP. • For information on how to configure VLAN parameters, refer to ''Configuring the IP Settings'' on page 52.
SIP User's Manual 8.8.3 8. Networking Capabilities Getting Started with VLANS and Multiple IPs By default, the device operates without VLANs and multiple IPs, using a single IP address, subnet mask and default Gateway IP address.
MediaPack Series c. 4. Click the Submit button to save your changes. Configure the multiple IP parameters by completing the following steps: a. In the ‘IP Settings’ page, modify the IP parameters to correspond to the values shown in the figure below. Note that the OAM, Control, and Media Network Settings parameters appear only after you select the options ‘Multiple IP Networks’ or 'Dual IP' in the field ‘IP Networking Mode’.
SIP User's Manual 5. 8. Networking Capabilities Configure the 'IP Routing' table to define static routing rules for the OAMP and Control networks, since a default gateway isn’t supported on these networks: Open the ‘IP Routing Table’ page (refer to ''Configuring the IP Routing Table'' on page 63). a. Figure 8-6: Static Routes for OAM/Control in IP Routing Table b.
MediaPack Series Below is an example of an ini file containing VLAN and Multiple IPs parameters: ; Interface Table Configuration: [InterfaceTable] FORMAT InterfaceTable_Index = InterfaceTable_ApplicationTypes, InterfaceTable_IPv6InterfaceMode, InterfaceTable_IPAddress, InterfaceTable_PrefixLength, InterfaceTable_Gateway, InterfaceTable_VlanID, InterfaceTable_InterfaceName; InterfaceTable 0 = 0, 0, 10.31.174.50, 16, 0.0.0.0, 4, OAMP; InterfaceTable 0 = 1, 0, 10.33.174.50, 16, 10.33.0.
SIP User's Manual 8. Networking Capabilities Instead of using the ini file table parameter InterfaceTable, you can configure multiple IPs and VLANs using the individual ini file parameters, as shown below: ; VLAN Configuration VlanMode=1 VlanOamVlanId=4 VlanNativeVlanId=4 VlanControlVlanId=5 VlanMediaVlanID=6 ; Multiple IPs Configuration EnableMultipleIPs=1 LocalMediaIPAddress=10.33.174.50 LocalMediaSubnetMask=255.255.0.0 LocalMediaDefaultGW=10.33.0.1 LocalControlIPAddress=10.32.174.
MediaPack Series Reader’s Notes SIP User's Manual 378 Document #: LTRT-65411
SIP User's Manual 9 9. Supplied SIP Software Package Supplied SIP Software Package The table below lists the standard SIP software package supplied with the SIP device. Table 9-1: Supplied Software Package File Name Description Ram.cmp file MP124_SIP_xxx.cmp Image file containing the software for the MP-124/FXS device. MP118_SIP_xxx.cmp Common Image file Image file containing the software for MP-11x/FXS devices. ini files SIPgw_MP124.ini Sample ini file for MP-124/FXS device. SIPgw_fxs_MP118.
MediaPack Series Reader’s Notes SIP User's Manual 380 Document #: LTRT-65411
SIP User's Manual 10 10. Selected Technical Specifications Selected Technical Specifications The main technical specifications of the MP-11x and MP-124 devices are listed in the following subsections. Note: All specifications in this document are subject to change without prior notice. 10.1 MP-11x Specifications The table below lists the main technical specifications of the MP-11x.
MediaPack Series Function Specification Note: For country-specific coefficients, use the parameter CountryCoefficients.
SIP User's Manual 10.
MediaPack Series Function Specification Maintenance Syslog according to RFC 3164 Local RS-232 terminal Web Management via HTTP or HTTPS Telnet Type Approvals UL 60950-1, FCC part 15 Class B CE Mark EN 60950-1, EN 55022, EN 55024, EN61000-3-2, EN61000-3-3, EN55024. Safety and EMC 10.2 MP-124 Specifications The table below lists the main technical specifications of the MP-124.
SIP User's Manual 10. Selected Technical Specifications Function Specification Voice & Tone Characteristics Voice Compression G.711 PCM at 64 kbps µ-law/A-law; G.723.1 MP-MLQ at 5.3 or 6.3 kbps; G.726 at 32 kbps ADPCM; G.729 CS-ACELP 8 Kbps Annex A/B Silence Suppression G.723.1 Annex A; G.729 Annex B; PCM and ADPCM [Standard Silence Descriptor (SID) with Proprietary Voice Activity Detection (VAD) and Comfort Noise Generation (CNG)]. Packet Loss Concealment G.711 appendix 1; G.723.1; G.
MediaPack Series Function Specification Interfaces FXS Telephony Interface 24 Analog FXS phone or fax ports, loop start (RJ-11) Network Interface 10/100Base-TX RS-232 Interface RS-232 Terminal Interface (DB-9) Indicators Channel status and activity LEDs Connectors & Switches Rear Panel: 24 Analog Lines 50-pin Telco shielded connector Ethernet 10/100Base-TX, RJ-45 shielded connector RS-232 DB-9 console port AC power supply socket 100-240~0.
SIP User's Manual 11 11. Glossary Glossary Table 11-1: Glossary of Terms Term Meaning ADPCM Adaptive Differential PCM - voice compression A-law Standard companding algorithm, used in European digital communications systems to optimize the dynamic range of an analog signal for digitizing.
MediaPack Series Term Meaning MIB Management Information Base MLPP Multilevel Precedence and Preemption ms or msec Millisecond; a thousandth part of a second MWI Message Waiting Indicator NAPTR Naming Authority Pointer NAT Network Address Translation NPI Numbering Plan Indicator NTP Network Time Protocol OAMP Operations, Administration, Maintenance and Provisioning OSI Open Systems Interconnection (Industry Standard) PBX Private Branch Exchange PCM Pulse-Code Modulation PKI Publi
SIP User's Manual 11. Glossary Term Meaning TFTP Trivial File Transfer Protocol TLS Transport Layer Security TON Type of Numbering UA SIP User Agent UDP User Datagram Protocol URI (SIP URIs) SIP Uniform Resource Indicators VBD Voice-band data VLAN Virtual Local Area Network VoIP Voice over Internet Protocol VoP Voice over Packet(s) VPN Virtual Private Network µ-Law A companding algorithm, used in the digital telecommunication systems Version 5.
User's Manual Version 5.6 www.audiocodes.
