™ CPE & Access Analog Gateways SIP MediaPack™ MP-124 & MP-11x User’s Manual Version 5.
SIP User's Manual Contents Table of Contents 1 Overview ............................................................................................................19 1.1 1.2 1.3 Gateway Description .............................................................................................. 19 SIP Overview ......................................................................................................... 20 MediaPack Features ......................................................................
MediaPack 5 Web Management..............................................................................................49 5.1 5.2 Computer Requirements ........................................................................................ 49 Protection and Security Mechanisms..................................................................... 49 5.2.1 5.2.2 5.2.3 5.3 Accessing the Embedded Web Server .................................................................. 51 5.3.1 5.4 Main Menu Bar ...
SIP User's Manual 5.6.2 5.6.3 5.6.4 5.6.5 5.6.6 5.7 5.7.2 5.7.3 5.7.4 Gateway Statistics ................................................................................................ 187 5.7.1.1 IP Connectivity ...................................................................................... 187 5.7.1.2 Call Counters......................................................................................... 189 5.7.1.3 Call Routing Status...........................................................
MediaPack 7 Using BootP / DHCP........................................................................................211 7.1 7.2 7.3 BootP/DHCP Server Parameters ......................................................................... 211 Using DHCP......................................................................................................... 212 Using BootP ......................................................................................................... 213 7.3.1 7.3.
SIP User's Manual Contents 8.12.4 Remote IP Extension between FXO and FXS...................................................... 242 8.12.4.1 Dialing from Remote Extension............................................................. 243 8.12.4.2 Dialing from other PBX line, or from PSTN ........................................... 244 8.12.4.3 FXS MediaPack Configuration (using the Embedded Web Server) ..... 244 8.12.4.4 FXO MediaPack Configuration (using the Embedded Web Server).....
MediaPack 12 Security ............................................................................................................279 12.1 IPSec and IKE...................................................................................................... 279 12.1.1 IKE ........................................................................................................................ 280 12.1.2 IPSec .......................................................................................................
SIP User's Manual Contents 14.8.3 Trusted Managers................................................................................................. 317 14.8.3.1 Configuration of Trusted Managers via ini File ..................................... 318 14.8.3.2 Configuration of Trusted Managers via SNMP ..................................... 318 14.8.4 SNMP Ports .......................................................................................................... 319 14.8.
MediaPack C.9 Log Window ......................................................................................................... 352 C.10 Setting the Preferences........................................................................................ 353 C.10.1 BootP Preferences................................................................................................ 353 C.10.2 TFTP Preferences ................................................................................................ 354 C.
SIP User's Manual Contents List of Figures Figure 1-1: Typical MediaPack VoIP Application ...................................................................................20 Figure 2-1: MP-118 Front Panel Connectors .........................................................................................25 Figure 2-2: MP-118 Rear Panel Connectors ..........................................................................................26 Figure 2-3: MP-124 Front Panel....................................
MediaPack Figure 5-38: NFS ini File Example ...................................................................................................... 144 Figure 5-39: VLAN Settings Screen .................................................................................................... 147 Figure 5-40: Voice Settings Screen..................................................................................................... 152 Figure 5-41: Fax / Modem / CID Settings Screen ...................................
SIP User's Manual Contents Figure 11-1: Metering Tone Relay Architecture .................................................................................. 277 Figure 11-2: Proprietary INFO Message for Relaying Metering Tones............................................... 277 Figure 12-1: IPSec Encryption ............................................................................................................ 279 Figure 12-2: Example of an IKE Table .....................................................
MediaPack List of Tables Table 1-1: Supported Product Configurations ........................................................................................19 Table 2-1: Definition of MP-11x Front Panel LED Indicators (continues on pages 26 to 26) ................26 Table 2-2: MP-11x Rear Panel Component Descriptions ......................................................................26 Table 2-3: Front Panel Buttons on the MP-124..........................................................................
SIP User's Manual Contents Table 5-43: Media Settings, Fax/Modem/CID Parameters (continues on pages 154 to 157) ............ 154 Table 5-44: Media Settings, RTP / RTCP Parameters (continues on pages 157 to 159)................... 158 Table 5-45: Media Settings, Hook-Flash Settings Parameters ........................................................... 160 Table 5-46: Media Settings, General Media Settings Parameters ......................................................
MediaPack Table D-1: Packet Types Defined in RFC 3551 .................................................................................. 361 Table D-2: Defined Payload Types ..................................................................................................... 361 Table D-3: Default RTP/RTCP/T.38 Port Allocation............................................................................ 362 Table F-1: acBoardFatalError Alarm Trap................................................................
SIP User's Manual Notices 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 Note 1: MediaPack refers to the MP-124, MP-118, MP-114, and MP-112 VoIP gateways. Note 2: MP-11x refers to the MP-118, MP-114, and MP-112 VoIP gateways. Note: Where ‘network’ appears in this manual, it means Local Area Network (LAN), Wide Area Network (WAN), etc. accessed via the gateway’s Ethernet interface.
SIP User's Manual 1 1.
MediaPack The layout diagram (Figure 1-1), illustrates a typical MediaPack VoIP application. Figure 1-1: Typical MediaPack VoIP Application 1.2 SIP Overview SIP (Session Initialization Protocol) is an application-layer control (signaling) protocol used on the MediaPack for creating, modifying, and terminating sessions with one or more participants. These sessions can include Internet telephone calls, media announcements and conferences.
SIP User's Manual 1.3 1. Overview MediaPack Features This section provides a high-level overview of some of the many MediaPack supported features. 1.3.1 1.3.2 General Features Superior, high quality Voice, Data and fax over IP networks. Toll quality voice compression. Enhanced capabilities including MWI, long haul, metering, CID and out door protection. Proven integration with leading PBXs, IP-PBXs, Softswitches and SIP servers. Spans a range of 2 to 24 FXS/FXO analog ports.
MediaPack 1.3.3 1.3.4 MP-124 Hardware Features 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 and rear panels that provide information on the operating status of the media gateway and the network interface. Reset button on the front panel that restarts the MP-124 gateway, and is also used to restore the MP-124 parameters to their factory default values.
SIP User's Manual 1. Overview The REGISTER message is sent to the Registrar’s IP address (if configured) or to the Proxy’s IP address. The message is sent per gateway or per gateway endpoint according to the AuthenticationMode parameter. Usually the FXS gateways are registered per gateway port, while FXO gateways send a single registration message, where Username is used instead of phone number in From/To headers. The registration request is resent according to the parameter ‘RegistrartionTimeDivider’.
MediaPack SIP URL: sip:”phone number”@IP address (such as 1225556@10.1.2.4, where “122556” is the phone number of the source or destination) or sip:”phone_number”@”domain name”, such as 122556@myproxy.com. Note that the SIP URI host name can be configured differently per called number. Supports RFC 4040, RTP payload format for a 64 kbit/s transparent data. Can negotiate coder from a list of given coders. Supports negotiation of dynamic payload types.
SIP User's Manual 2 2. MediaPack Physical Description MediaPack Physical Description This section provides detailed information on the hardware, the location and functionality of the LEDs, buttons and connectors on the front and rear panels of the MP-11x (refer to Section 2.1 below) and MP-124 (refer to Section 2.2 on page 27) gateways. For detailed information on installing the MediaPack, refer to Chapter 0 on page 29. 2.1 MP-11x Physical Description 2.1.
MediaPack Table 2-1: Definition of MP-11x Front Panel LED Indicators (continues on pages 26 to 26) LED Type Color Channels Status Telephone Interface State Green Blinking The phone is ringing (incoming call, before answering). Fast Blinking Line malfunction Off On Uplink Fail Ready Power 2.1.
SIP User's Manual 2. MediaPack Physical Description 2.2 MP-124 Physical Description 2.2.1 MP-124 Front Panel Figure 2-3 illustrates the front layout of the MP-124. Table 2-3 describes the Reset button located on the front panel. Table 2-4 lists and describes the front panel LEDs.
MediaPack 2.2.2 MP-124 Rear Panel The figure below illustrates the rear panel of the MP-124. For descriptions of the rear panel components, refer to Table 2-5. For the functionality of rear panel Ethernet LEDs, refer to Table 2-6. Figure 2-4: MP-124 (FXS) Rear Panel Connectors Table 2-5: MP-124 Rear Panel Component Descriptions Item # Label Component Description Protective earthing screw (mandatory for all installations). Accepts a 6-32 UNC screw.
SIP User's Manual 3 3. Installing the MediaPack Installing the MediaPack This section provides information on the installation procedure for the MP-11x (refer to Section 3.1 below) and the MP-124 (refer to Section 3.2 on page 36). For information on how to start using the gateway, refer to Chapter 4 on page 41. Caution Electrical Shock The equipment must only be installed or serviced by qualified service personnel 3.1 Installing the MP-11x ¾ To install the MP-11x, take these 4 steps: 1.
MediaPack 3.1.3 19-inch Rack Installation Package (Optional) An additional option is available for installing the MP-11x in a 19-inch rack. The 19-inch rack installation package contains a single shelf (shown in Figure 3-1) and eight shelf-todevice screws. Note: The 19-inch rack shelf is not supplied in the standard package kit, but can be ordered separately: Bulk Pack package (MCMK00015) containing 10 rack mounting shelves for MP-11x.
SIP User's Manual 3. Installing the MediaPack Table 3-1: View of the MP-11x Base Item # Functionality 1 Square slot used to attach anti-slide bumpers (for desktop mounting) 2 Oval notch used to attach the MP-11x to a wall 3 Screw opening used to attach the MP-11x to a 19-inch shelf rack 3.1.4.1 Mounting the MP-11x on a Desktop Attach the four (supplied) anti-slide bumpers to the base of the MP-11x (refer to item #1 in Figure 3-2) and place it on the desktop in the position you require. 3.1.4.
MediaPack 3.1.4.3 Installing the MP-11x in a 19-inch Rack The MP-11x can be installed in a standard 19-inch rack by placing it on an AudioCodes' 19-inch rack-mounting shelf that is pre-installed in the rack. The shelf can hold up to two MP-11x gateways. This shelf can be ordered separately from AudioCodes. Note: The 19-inch rack shelf is not supplied in the standard package kit, but can be ordered separately (Bulk Pack package MCMK00015 with 10 rack mounting shelves for MP-11x).
SIP User's Manual 3.1.5 3. Installing the MediaPack Cabling the MP-11x Cable your MP-11x according to each section of Table 3-3. For detailed information on the MP-11x rear panel connectors, refer to Table 2-2 on page 26. Table 3-3: MP-11x Cables and Cabling Procedure Cable Cabling Procedure RJ-45 Ethernet cable Connect the Ethernet connection on the MP-11x directly to the network using a crossover RJ-45 Ethernet cable. For connector pinouts refer to Figure 3-4 below.
MediaPack 3.1.5.1 Connecting the MP-11x RS-232 Port to Your PC Using a standard RS-232 straight cable (not a cross-over cable) with DB-9 connectors, connect the MP-11x RS-232 port (using a DB-9 to PS/2 adaptor) to either COM1 or COM2 RS-232 communication port on your PC. The pinouts of the PS/2 connector is shown below in Figure 3-6. A PS/2 to DB-9 adaptor is not included with the MP-11x package. For the PS/2 to DB-9 pinouts, refer to Figure 3-7 below.
SIP User's Manual 3. Installing the MediaPack ¾ To cable the combined MP-11x FXS/FXO Lifeline, take these 2 steps: 1. 2. Connect a fax machine, modem, or phone to each of the FXS ports. Connect an analog PSTN line to each of the FXO ports. Note: Version 5.0 The use of the Lifeline on network failure can be disabled using the ‘LifeLineType’ ini file parameter (described in Table 5-55 on page 182).
MediaPack 3.2 Installing the MP-124 ¾ To install the MP-124, take these 4 steps: 1. Unpack the MP-124 (refer to Section 3.2.1 below). 2. Check the package contents (refer to Section 3.2.2 below). 3. Mount the MP-124 (refer to Section 3.2.3 on page 36). 4. Cable the MP-124 (refer to Section 3.2.4 on page 38). After connecting the MP-124 to the power source, the Ready and LAN LEDs on the front panel turn to green (after a self-testing period of about 1 minute).
SIP User's Manual 3. Installing the MediaPack Rack Mount Safety Instructions (UL) When installing the chassis in a rack, be sure to implement the following Safety instructions recommended by Underwriters Laboratories: • • • • • 3.2.3.2 Elevated Operating Ambient - If installed in a closed or multi-unit rack assembly, the operating ambient temperature of the rack environment may be greater than room ambient.
MediaPack 3.2.4 Cabling the MP-124 Cable your MP-124 according to each section of Table 3-4. For detailed information on the MP-124 rear panel connectors, refer to Section 2.2.2 on page 28. Table 3-4: MP-124 Cables and Cabling Procedure Cable Cabling Procedure Protective earthing strap Connect an earthed strap to the chassis protective earthing screw (6-32 UNC screw) and fasten it securely according to the safety standards.
SIP User's Manual 3. Installing the MediaPack Figure 3-13: MP-124 in a 19-inch Rack with MDF Adaptor Table 3-5: Pin Allocation in the 50-pin Telco Connector Phone Channel Connector Pins Phone Channel Connector Pins 1 2 3 4 5 6 7 8 9 10 11 12 1/26 2/27 3/28 4/29 5/30 6/31 7/32 8/33 9/34 10/35 11/36 12/37 13 14 15 16 17 18 19 20 21 22 23 24 13/38 14/39 15/40 16/41 17/42 18/43 19/44 20/45 21/46 22/47 23/48 24/49 Version 5.
MediaPack 3.2.4.1 Connecting the MP-124 RS-232 Port to Your PC Using a standard RS-232 straight cable (not a cross-over cable) with DB-9 connectors, connect the MP-124 RS-232 port to either COM1 or COM2 RS-232 communication port on your PC. The required connector pinouts and gender are shown below in Figure 3-14. For information on establishing a serial communications link with the MP-124, refer to Section 10.2 on page 262.
SIP User's Manual 4 4. Getting Started Getting Started The MediaPack is supplied with default networking parameters (show in Table 4-1 below) and with an application software already resident in its flash memory (with factory default parameters). Before you begin configuring the gateway, change its default IP address to correspond with your network environment (refer to Section 4.2) and learn about the configuration methods available on the MediaPack (refer to Section 4.1 below).
MediaPack 4.2.1 Assigning an IP Address Using HTTP ¾ To assign an IP address using HTTP, take these 8 steps: 1. Disconnect the MediaPack from the network and reconnect it to your PC using one of the following two methods: • Use a standard Ethernet cable to connect the network interface on your PC to a port on a network hub / switch. Use a second standard Ethernet cable to connect the MediaPack to another port on the same network hub / switch. • 2. 3. 4. 5.
SIP User's Manual 4.2.3 4. Getting Started ¾ To assign an IP address using BootP, take these 3 steps: 1. 2. 3. Open the BootP application (supplied with the MediaPack software package). Add client configuration for the MediaPack, refer to Section C.11.1 on page 355. Use the reset button to physically reset the gateway causing it to use BootP; the MediaPack changes its network parameters to the values provided by the BootP.
MediaPack Table 4-2: Configuration Parameters Available via the Voice Menu (continues on pages 43 to 44) Item Number at Menu Prompt 4.2.4 Description 12 MGCP call agent port number (N/A) 99 Voice menu password (initially 12345). Note: The voice menu password can also be changed using the parameter ‘VoiceMenuPassword’ (refer to Table 5-50 on page 174). Assigning an IP Address Using the CLI First access the CLI using a standard Telnet application or using a serial communication software (e.g.
SIP User's Manual 4.2.4.2 4. Getting Started Assign an IP Address ¾ To assign an IP address via the CLI, take these 4 steps: 1. 2. At the prompt type ‘conf’ and press enter; the configuration folder is accessed. To check the current network parameters, at the prompt, type GCP IP, and then press Enter; the current network settings are displayed. Change the network settings by typing: SCP IP [ip_address] [subnet_mask] [default_gateway] (e.g., ‘SCP IP 10.13.77.7 255.255.0.0 10.13.0.
MediaPack 4.3 Configure the MediaPack Basic Parameters To configure the MediaPack basic parameters use the Embedded Web Server’s ‘Quick Setup’ screen (shown in Figure 4-1 below). Refer to Section 5.3 on page 51 for information on accessing the ‘Quick Setup’ screen. Figure 4-1: Quick Setup Screen ¾ To configure basic SIP parameters, take these 9 steps: 1. If the MediaPack is connected to a router with Network Address Translation (NAT) enabled, perform the following procedure.
SIP User's Manual 5. 4. Getting Started Configure ‘Enable Registration’ to ‘Yes’ or ‘No’: • ‘No’ = the MediaPack does not register to a Proxy server/Registrar (default). • ‘Yes’ = the MediaPack registers to a Proxy server/Registrar at power up and every ‘Registration Time’ seconds; The MediaPack sends a REGISTER request according to the ‘Authentication Mode’ parameter. For detailed information on the parameters ‘Registration Time’ and ‘Authentication Mode’, refer to Table 5-4 on page 66. 6.
MediaPack Reader's Notes SIP User's Manual 48 Document #: LTRT-65408
SIP User's Manual 5 5. Web Management Web Management The Embedded Web Server is used both for gateway configuration, including loading of configuration files, and for run-time monitoring. The Embedded Web Server can be accessed from a standard Web browser, such as Microsoft™ Internet Explorer, Netscape™ Navigator, etc. Specifically, users can employ this facility to set up the gateway configuration parameters.
MediaPack Table 5-1 lists the available access levels and their privileges.
SIP User's Manual 5.2.2 5. Web Management Limiting the Embedded Web Server to Read-Only Mode Users can limit access to the Embedded Web Server to read-only mode by changing the ini file parameter ‘DisableWebConfig’ to 1. In this mode all Web screens, regardless of the access level used, are read-only and cannot be modified. In addition, the following screens cannot be accessed: ‘Quick Setup’, ‘Web User Accounts’, ’Reset‘, ‘Save Configuration‘ and all of the file-loading screens. Notes: 5.2.
MediaPack 5.3.1 Using Internet Explorer to Access the Embedded Web Server Internet explorer’s security settings may block access to the gateway’s Web browser if they’re configured incorrectly. In this case, the following message is displayed: Unauthorized Correct authorization is required for this area. Either your browser does not perform authorization or your authorization has failed. RomPager server. ¾ To troubleshoot blocked access to Internet Explorer™, take these 3 steps 1.
SIP User's Manual 5. Web Management The Web Interface screen features the following components: 5.4.1 Title bar: contains three configurable elements: corporate logo, a background image and the product’s name. For information on how to modify these elements, refer to Section 10.5 on page 267. Product name: the gateway model name. Main menu bar: always appears on the left of every screen to quickly access parameters, submenus, submenu options, functions and operations.
MediaPack 5.4.3 Entering Phone Numbers in Various Tables Phone numbers entered into various tables on the gateway, such as the Tel to IP routing table, must be entered without any formatting characters. For example, if you wish to enter the phone number 555-1212, it must be entered as 5551212 without the hyphen (-). If the hyphen is entered, the entry does not work. The hyphen character is used in number entry only, as part of a range definition.
SIP User's Manual 3. 5. Web Management In the searched result list, click the required parameter to open the screen in which the parameter appears. The relevant parameter is highlighted in green in the screen for easy viewing. Figure 5-4: Searched Parameter Highlighted in Screen Note: Version 5.0 If the searched parameter is not located, the "No Matches Found For This String" message is displayed.
MediaPack 5.5 Protocol Management Use this menu to configure the gateway’s SIP parameters and tables. Note: 5.5.1 Those parameters contained within square brackets are the names used to configure the parameters via the ini file. Protocol Definition Parameters Use this submenu to configure the gateway’s specific SIP protocol parameters. 5.5.1.1 General Parameters Use this screen to configure general SIP parameters.
SIP User's Manual 5. Web Management ¾ To configure the general parameters under Protocol Definition, take these 4 steps: 1. Open the ‘General Parameters’ screen (Protocol Management menu > Protocol Definition submenu > General Parameters option); the ‘General Parameters’ screen is displayed. Figure 5-5: Protocol Definition, General Parameters Screen Version 5.
MediaPack 2. 3. 4. Configure the general parameters under Protocol Definition according to Table 5-3. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description PRACK Mode [PRACKMode] PRACK mechanism mode for 1xx reliable responses: Disable [0]. Supported [1] (default). [2].
SIP User's Manual 5. Web Management Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description Session Expires Method [SessionExpiresMethod] Defines the SIP method used for session-timer updates. Re-INVITE [0] = Use Re-INVITE messages for session-timer updates (default). UPDATE [1] = Use UPDATE messages. Note 1: The gateway can receive session-timer refreshes using both methods.
MediaPack Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description SIP UDP Local Port [LocalSIPPort] Local UDP port used to receive SIP messages. The valid range is 1 to 65534. The default value is 5060. SIP TCP Local Port [TCPLocalSIPPort] Local TCP port used to receive SIP messages. The default value is 5060. SIP TLS Local Port [TLSLocalSIPPort] Local TLS port used to receive SIP messages. The default value is 5061.
SIP User's Manual 5. Web Management Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description Enable History-Info Header Enables usage of the History-Info header. [EnableHistoryInfo] Valid options include: [0] = Disable (default) [1] = Enable 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 Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description Use Display Name as Source Number [UseDisplayNameAsSourc eNumber] Applicable to IPÆTel calls. No [0] = The IP Source Number is used as the Tel Source Number and the IP Display Name is used as the Tel Display Name (if IP Display Name is received). If no Display Name is received from IP, the Tel Display Name remains empty (default).
SIP User's Manual 5. Web Management Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description Enable GRUU [EnableGRUU] Determines whether or not the GRUU mechanism is used. Valid options include: Disable [0] (default) Enable [1] The gateway obtains a GRUU by generating a normal REGISTER request. This request contains a Supported header field with the value “gruu”. The gateway includes a “+sip.
MediaPack Table 5-3: Protocol Definition, General Parameters (continues on pages 58 to 64) Parameter Description Enable Semi-Attended Transfer [EnableSemiAttendedTran sfer] Determines the gateway behavior when Transfer is initiated while still in Alerting state. Valid options include: Disable [0] = Send REFER with Replaces (default). Enable [1] = Send CANCEL, and after a 487 response is received, send REFER without Replaces.
SIP User's Manual 5.5.1.2 5. Web Management Proxy & Registration Parameters Use this screen to configure parameters that are associated with Proxy and Registration. ¾ To configure the Proxy & Registration parameters, take these 4 steps: 1. Open the ‘Proxy & Registration’ parameters screen (Protocol Management menu > Protocol Definition submenu > Proxy & Registration option); the ‘Proxy & Registration’ parameters screen is displayed. Figure 5-6: Proxy & Registration Parameters Screen Version 5.
MediaPack 2. 3. 4. Configure the Proxy & Registration parameters according to Table 5-4. 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. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205.
SIP User's Manual 5. Web Management Table 5-4: Proxy & Registration Parameters (continues on pages 66 to 71) Parameter Description First Redundant Proxy IP IP addresses of the first redundant Proxy you are using. Address Enter the IP address as FQDN or in dotted format notation (for example [ProxyIP] 192.10.1.255). You can also specify the selected port in the format: :. Note 1: This parameter is available only if you select ‘Use Proxy' in the ‘Enable Proxy’ field.
MediaPack Table 5-4: Proxy & Registration Parameters (continues on pages 66 to 71) Parameter Description Proxy Load Balancing Method [ProxyLoadBalancingMe thod] Enables the usage of the Proxy Load Balancing mechanism. Valid options include: Disable [0] = Load Balancing is disabled (default). Round Robin [1] = Round Robin algorithm. Random Weights [2] = Random Weights. When Round Robin (1) algorithm is used, a list of all possible Proxy IP addresses is compiled.
SIP User's Manual 5. Web Management Table 5-4: Proxy & Registration Parameters (continues on pages 66 to 71) Parameter Description Proxy DNS Query Type [ProxyDNSQueryType] Enables the use of DNS Naming Authority Pointer (NAPTR) and Service Record (SRV) queries to discover Proxy servers. Valid options include: [0] = A-Record (default) [1] = SRV [2] = NAPTR If set to A-Record, no NAPTR or SRV queries are performed.
MediaPack Table 5-4: Proxy & Registration Parameters (continues on pages 66 to 71) Parameter Description Re-registration Timing (%) Defines the re-registration timing (in percentage). The timing is a percentage of the [RegistrationTimeDivide re-register timing set by the Registration server. r] The valid range is 50 to 100. The default value is 50.
SIP User's Manual 5. Web Management Table 5-4: Proxy & Registration Parameters (continues on pages 66 to 71) Parameter Description Use Routing Table for Use the internal Tel to IP routing table to obtain the URI Host name and (optionally) Host Names and Profiles an IP profile (per call), even if Proxy server is used. [AlwaysUseRouteTable] No [0] = Don’t use (default). Yes [1] = Use. Note: This domain name is used, instead of Proxy name or Proxy IP address, in the INVITE SIP URI.
MediaPack 5.5.1.3 Coders From the Coders screen you can configure the first to fifth preferred coders (and their attributes) for the gateway. The first coder is the highest priority coder and is used by the gateway whenever possible. If the far end gateway cannot use the coder assigned as the first coder, the gateway attempts to use the next coder and so forth. ¾ To configure the gateway’s coders, take these 9 steps: 1.
SIP User's Manual 5. Web Management Table 5-5: Supported Coders and their Attributes Coder Name Packetization Time Rate Payload Type Silence Suppression G.711 A-law [g711Alaw64k] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 Always 8 Disable [0] Enable [1] G.711 µ-law [g711Ulaw64k] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 Always 0 Disable [0] Enable [1] G.
MediaPack 5.5.1.4 DTMF & Dialing Parameters Use this screen to configure parameters that are associated with DTMF and dialing. ¾ To configure the dialing parameters, take these 4 steps: 1. Open the ‘DTMF & Dialing’ screen (Protocol Management menu > Protocol Definition submenu > DTMF & Dialing option); the ‘DTMF & Dialing’ screen is displayed. Figure 5-8: DTMF & Dialing Screen 2. 3. 4. Configure the DTMF & Dialing parameters according to Table 5-7. Click the Submit button to save your changes.
SIP User's Manual 5. Web Management Table 5-7: DTMF & Dialing Parameters (continues on pages 74 to 76) Parameter Description Declare RFC 2833 in SDP [RxDTMFOption] Defines the supported Receive DTMF negotiation method. No [0] = Don’t declare RFC 2833 telephony-event parameter in SDP Yes [3] = Declare RFC 2833 telephony-event parameter in SDP (default) The MediaPack is designed to always be receptive to RFC 2833 DTMF relay packets.
MediaPack Table 5-7: DTMF & Dialing Parameters (continues on pages 74 to 76) Parameter Description Digit Mapping Rules [DigitMapping] Digit map pattern. If the digit string (dialed number) has matched one of the patterns in the digit map, the gateway stops collecting digits and starts to establish a call with the collected number The digit map pattern contains up to 52 options separated by a vertical bar (|) and enclosed in parenthesis.
SIP User's Manual 5. Web Management ¾ To configure the general parameters under Advanced Parameters, take these 4 steps: 1. Open the ‘General Parameters’ screen (Protocol Management menu > Advanced Parameters submenu > General Parameters option); the ‘General Parameters’ screen is displayed. Figure 5-9: Advanced Parameters, General Parameters Screen Version 5.
MediaPack 2. 3. 4. Configure the general parameters under ‘Advanced Parameters’ according to Table 5-8. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Table 5-8: Advanced Parameters, General Parameters (continues on pages 78 to 82) Parameter Description Signaling DiffServ [ControlIPDiffServ] Obsolete parameter, use PremiumServiceClassControlDiffServ instead.
SIP User's Manual 5. Web Management Table 5-8: Advanced Parameters, General Parameters (continues on pages 78 to 82) Parameter Description Enable DID Wink [EnableDIDWink] Disable [0] = DID is disabled (default). Enable [1] = Enable DID. If enabled, the MediaPack can be used for connection to EIA/TIA-464B DID Loop Start lines. Both FXO (detection) and FXS (generation) are supported. An FXO gateway dials DTMF digits after a Wink signal is detected (instead of a Dial tone).
MediaPack Table 5-8: Advanced Parameters, General Parameters (continues on pages 78 to 82) Parameter Description Disconnect Call on Silence Detection [EnableSilenceDisconnect] Yes [1] = The MediaPack disconnect calls in which silence occurs in both (call) directions for more than 120 seconds. No [0] = Call is not disconnected when silence is detected (default). The silence duration can be set by the ‘FarEndDisconnectSilencePeriod’ parameter (default 120).
SIP User's Manual 5. Web Management Table 5-8: Advanced Parameters, General Parameters (continues on pages 78 to 82) Parameter Description Enable Busy Out [EnableBusyOut] No [0] = ‘Busy out’ feature is not used (default). Yes [1] = ‘Busy out’ feature is enabled. When Busy out is enabled, the MediaPack gateway performs a specific behavior (e.g., plays a reorder tone when the phone is offhooked) due to one of the following reasons: Physically disconnected from the network (i.e.
MediaPack Table 5-8: Advanced Parameters, General Parameters (continues on pages 78 to 82) Parameter Description Enable Calls Cut Through [CutThrough] Enables users to receive incoming IP calls while the port is in an offhooked state. = Disabled (default). Disable [0] = Enabled. Enable [1] If enabled, FXS gateways answer the call and ‘cut through’ the voice channel, if there is no other active call on that port, even if the port is in offhooked state.
SIP User's Manual 5.5.2.2 5. Web Management Supplementary Services Use this screen to configure parameters that are associated with supplementary services. For detailed information on the supplementary services, refer to Section 8.1 on page 215. ¾ To configure the supplementary services’ parameters, take these 4 steps: 1. Open the ‘Supplementary Services’ screen (Protocol Management menu > Advanced Parameters submenu > Supplementary Services option); the ‘Supplementary Services’ screen is displayed.
MediaPack 2. 3. 4. Configure the supplementary services parameters according to Table 5-9. Click the Submit button to save your changes, or click the Subscribe for MWI or UnSubscribe for MWI buttons to save your changes and to subscribe / unsubscribe to the MWI server. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205.
SIP User's Manual 5. Web Management Table 5-9: Supplementary Services Parameters (continues on pages 84 to 86) Parameter Description Time Between Call Waiting Indications Difference (in seconds) between call waiting indications (FXS only) for call waiting. [TimeBetweenWaitingIndi The default value is 10 seconds. cations] Time before Waiting Defines the interval (in seconds) before a call waiting indication is played to the Indication port that is currently in a call (FXS only).
MediaPack Table 5-9: Supplementary Services Parameters (continues on pages 84 to 86) Parameter Description Disable [0] = Disable MWI subscription (default). Subscribe to MWI [EnableMWISubscription] Enable [1] = Enable subscription to MWI (to MWIServerIP address). Note: Use the parameter ‘SubscriptionMode’ (described in Table 5-35 on page 132) to determine whether the gateway subscribes separately per endpoint of for the entire gateway. MWI Server IP Address [MWIServerIP] MWI server IP address.
SIP User's Manual 5. Web Management ¾ To configure the Metering Tones, take these 6 steps: 1. Open the ‘Metering Tones’ screen (Protocol Management menu > Advanced Parameters submenu > Metering Tones option); the ‘Metering Tones’ screen is displayed. Figure 5-11: Metering Tones Parameters Screen 2. 3. 4. 5. 6. From the ‘Generate Metering Tones’ drop-down list, select the method used to configure the metering tones that are generated to the Tel side (refer to Table 5-10).
MediaPack 5.5.2.3.1 Charge Codes Table The Charge Codes table is used to configure the metering tones (and their time interval) that the FXS gateway generates to the Tel side. To associate a charge code to an outgoing Tel to IP call, use the Tel to IP Routing table. ¾ To configure the Charge Codes table, take these 6 steps: 1. Access the ‘Metering Tones’ screen (Protocol Management menu > Advanced Parameters submenu > Metering Tones option); the ‘Metering Tones’ screen is displayed (Figure 5-11).
SIP User's Manual 5.5.2.4 5. Web Management Keypad Features The Keypad Features screen (applicable only to FXS gateways) enables you to activate / deactivate the following features directly from the connected telephone’s keypad: Hotline (refer to Section 5.5.9.2 on page 120). Caller ID Restriction (refer to Section 5.5.9.3 on page 121). Call Forward (refer to Section 5.5.9.4 on page 122). ¾ To configure the keypad features, take these 4 steps: 1.
MediaPack Table 5-12: Keypad Features Parameters Parameter Description Forward Note that the forward type and number can be viewed in the Call Forward Table (refer to Section 5.5.9.5 on page 124) Unconditional [KeyCFUnCond] Keypad sequence that activates the immediate 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 5.5.3 5. Web Management Configuring the Number Manipulation Tables The VoIP gateway provides four Number Manipulation tables for incoming and outgoing calls. These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly.
MediaPack 4. 5. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Table 5-13: Number Manipulation Parameters (continues on pages 92 to 93) Parameter Description Dest. Prefix Each entry in the Destination Prefix fields represents a destination telephone number prefix. An asterisk (*) represents any number. Source Prefix Each entry in the Source Prefix fields represents a source telephone number prefix.
SIP User's Manual 5. Web Management Table 5-13: Number Manipulation Parameters (continues on pages 92 to 93) Parameter Description Presentation Select ‘Allowed’ to send Caller ID information when a call is made using these destination / source prefixes. Select ‘Restricted’ if you want to restrict Caller ID information for these prefixes. When set to ‘Not Configured’, the privacy is determined according to the Caller ID table (refer to Section 5.5.9.3 on page 121).
MediaPack Table 5-14: Number Manipulation ini File Parameters (continues on pages 93 to 95) Parameter Description NumberMapIP2Tel Manipulate the destination number for IP to Tel calls. NumberMapIP2Tel = a,b,c,d,e,f,g,h,i a = Destination number prefix. b = Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed.
SIP User's Manual 5. Web Management Table 5-14: Number Manipulation ini File Parameters (continues on pages 93 to 95) Parameter Description SourceNumberMapIP2Tel Manipulate the source number for IP to Tel calls. NumberMapIP2Tel = a,b,c,d,e,f,g,h,i a = Source number prefix b = Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix.
MediaPack 5.5.4 Mapping NPI/TON to Phone-Context The Phone-Context table is used to configure the mapping of NPI and TON to the PhoneContext SIP parameter. When a call is received from the ISDN, the NPI and TON are compared against the table and the Phone-Context value is used in the outgoing SIP INVITE message. The same mapping occurs when an INVITE with a Phone-Context attribute is received.
SIP User's Manual 5. Web Management 4. Configure the Phone Context table according to Table 5-15. 5. Click the Submit button to save your changes. 6. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Notes: • Several rows with the same NPI-TON or Phone-Context are allowed. In such a scenario, a Tel-to-IP call uses the first match. • Phone-Context '+' is a unique case as it doesn't appear in the RequestURI as a Phone-Context parameter.
MediaPack 5.5.5 Configuring the Routing Tables Use this submenu to configure the gateway’s IPÆTel and TelÆIP routing tables and their associated parameters. 5.5.5.1 General Parameters Use this screen to configure the gateway’s IPÆTel and TelÆIP routing parameters. ¾ To configure the general parameters under Routing Tables, take these 4 steps: 1. Open the ‘General Parameters’ screen (Protocol Management menu > Routing Tables submenu > General option); the ‘General Parameters’ screen is displayed.
SIP User's Manual 5. Web Management Table 5-16: Routing Tables, General Parameters (continues on pages 98 to 99) Parameter Description IP to Tel Remove Routing Table Prefix [RemovePrefix] No [0] = Don't remove prefix (default) Yes [1] = Remove the prefix (defined in the IP to Hunt Group Routing table) from a telephone number for an IPÆTel call, before forwarding it to Tel.
MediaPack 5.5.5.2 Tel to IP Routing Table The Tel to IP Routing Table is used to route incoming Tel calls to IP addresses. This routing table associates a called / calling telephone number’s prefixes with a destination IP address or with an FQDN (Fully Qualified Domain Name).
SIP User's Manual Tip: 5. Web Management Tel to IP routing can be performed either before or after applying the number manipulation rules. To control when number manipulation is done, set the Tel to IP Routing Mode parameter (described in Table 5-17). ¾ To configure the Tel to IP Routing table, take these 6 steps: 1. Open the ‘Tel to IP Routing’ screen (Protocol Management menu > Routing Tables submenu > Tel to IP Routing option); the ‘Tel to IP Routing’ screen is displayed (shown in Figure 5-17).
MediaPack Table 5-17: Tel to IP Routing Table (continues on pages 101 to 102) Parameter Description Destination IP Address In each of the IP Address fields, enter the IP address (and optionally port number) that is assigned to these prefixes. Domain names, such as domain.com, can be used instead of IP addresses. For example: : To discard outgoing IP calls, enter 0.0.0.0 in this field.
SIP User's Manual 5. Web Management To use hunt groups you must also do the following. You must assign a hunt group ID to the VoIP gateway channels on the Endpoint Phone Number screen. For information on how to assign a hunt group ID to a channel, refer to Section 5.5.7 on page 115. You can configure the Hunt Group Settings table to determine the method in which new calls are assigned to channels within the hunt groups (a different method for each hunt group can be configured).
MediaPack Table 5-18: IP to Hunt Group Routing Table (continues on pages 103 to 104) Parameter Description Source IP Address Each entry in the Source IP Address fields represents the source IP address of an IPÆTel call (obtained from the Contact header in the INVITE message). Note: The source IP address can include the ‘x’ wildcard to represent single digits. For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.
SIP User's Manual 5.5.5.4 5. Web Management Internal DNS Table The internal DNS table, similar to a DNS resolution, translates hostnames into IP addresses. This table is used when hostname translation is required (e.g., ‘Tel to IP Routing’ table). Two different IP addresses can be assigned to the same hostname. If the hostname isn’t found in this table, the gateway communicates with an external DNS server.
MediaPack 5.5.5.5 Internal SRV Table The Internal SRV table is used for resolving host names to DNS A-Records. Three different A-Records can be assigned to a hostname. Each A-Record contains the host name, priority, weight, and port. If the Internal SRV table is configured, the gateway first tries to resolve a domain name using this table. If the domain name isn’t found, the gateway performs an SRV resolution using an external DNS server.
SIP User's Manual 5.5.5.6 5. Web Management Reasons for Alternative Routing The Reasons for Alternative Routing screen includes two tables (TelÆIP and IPÆTel). Each table enables you to define up to 4 different release reasons. If a call is released as a result of one of these reasons, the gateway tries to find an alternative route to that call. The release reason for IPÆTel calls is provided in Q.931 notation. The release reason for TelÆIP calls is provided in SIP 4xx, 5xx and 6xx response codes.
MediaPack 4. 5. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Table 5-21: Reasons for Alternative Routing ini File Parameter Parameter Name in ini File Parameter Format AltRouteCauseTel2IP AltRouteCauseTel2IP = For example: AltRouteCauseTel2IP = 408 AltRouteCauseTel2IP = 486 (Response timeout). (User is busy). Note: This parameter can appear up to 4 times.
SIP User's Manual 5. Web Management ¾ To configure the coder group settings, take these 11 steps: 1. Open the ‘Coder Group Settings’ screen (Protocol Management menu > Profile Definitions submenu > Coder Group Settings option); the ‘Coder Group Settings’ screen is displayed. Figure 5-22: Coder Group Settings Screen From the 'Coder Group ID' drop-down list, select the coder group you want to edit (up to four coder groups can be configured). 3.
MediaPack Table 5-22: ini File Coder Group Parameter Parameter Description CoderName_ID Defines groups of coders that can be associated with IP or Tel profiles (up to five coders in each group). Enter coder groups in the following format: CoderName_=,,,,. Note 1: This parameter (CoderName_ID) can appear up to 20 times (five coders in four coder groups). Note 2: The coder name is case-sensitive.
SIP User's Manual 5.5.6.2 5. Web Management Tel Profile Settings Use the Tel Profile Settings screen to define up to four different Tel Profiles. These Profiles are used in the ‘Endpoint Phone Number’ table to associate different Profiles to gateway’s endpoints, thereby applying different behavior to different MediaPack ports. ¾ To configure the Tel Profile settings, take these 9 steps: 1.
MediaPack 4. 5. 6. 7. 8. 9. From the ‘Profile Preference’ drop-down list, select the preference (1-20) of the current Profile. The preference option is used to determine the priority of the Profile. Where ‘20’ is the highest preference value. If both IP and Tel profiles apply to the same call, the coders and other common parameters (noted by an asterisk in the description of the parameter TelProfile_ID) of the preferred Profile are applied to that call.
SIP User's Manual 5.5.6.3 5. Web Management IP Profile Settings Use the IP Profile Settings screen to define up to four different IP Profiles. These Profiles are used in the Tel to IP and IP to Hunt Group Routing tables to associate different Profiles to routing rules. IP Profiles can also be used when working with Proxy server (set ‘AlwaysUseRouteTable’ to 1). ¾ To configure the IP Profile settings, take these 9 steps: 1.
MediaPack 4. 5. 6. 7. 8. 9. From the ‘Profile Preference’ drop-down list, select the preference (1-20) of the current Profile. The preference option is used to determine the priority of the Profile. Where ‘20’ is the highest preference value. If both IP and Tel profiles apply to the same call, the coders and other common parameters (noted by an asterisk in the description of the parameter IPProfile_ID) of the preferred Profile are applied to that call.
SIP User's Manual 5.5.7 5. Web Management Configuring the Endpoint Phone Numbers From the 'Endpoint Phone Number Table' screen you can enable and assign telephone numbers, hunt groups (optional) and profiles to the VoIP gateway ports. ¾ To configure the Endpoint Phone Number table, take these 4 steps: 1. Open the ‘Endpoint Phone Number Table’ screen (Protocol Management menu > Endpoint Phone Numbers); the ‘Endpoint Phone Number Table’ screen is displayed.
MediaPack Table 5-25: Endpoint Phone Number Table (continues on pages 115 to 116) Parameter Description Hunt Group ID In each of the Hunt Group ID fields, enter the hunt group ID (1-99) assigned to the channel(s). The same hunt group ID can be used for more than one channel and in more than one field. The hunt group ID is an optional field that is used to define a group of common behavior channels that are used for routing IP to Tel calls.
SIP User's Manual 5.5.8 5. Web Management Configuring the Hunt Group Settings The Hunt Group Settings Table is used to determine the method in which new calls are assigned to channels within each hunt group. If such a rule doesn’t exist (for a specific hunt group), the global rule, defined by the Channel Select Mode parameter (Protocol Definition > General Parameters), applies. ¾ To configure the Hunt Group Settings table, take these 8 steps: 1.
MediaPack Table 5-26: Channel Select Modes Mode Description By Dest Phone Number Select the gateway port according to the called number (refer to the note below). Cyclic Ascending Select the next available channel in ascending cycle order. Always select the next higher channel number in the hunt group. When the gateway reaches the highest channel number in the hunt group, it selects the lowest channel number in the hunt group and then starts ascending again.
SIP User's Manual 5.5.9 5. Web Management Configuring the Endpoint Settings The Endpoint Settings screens enable you to configure port-specific parameters. 5.5.9.1 Authentication The Authentication Table (normally used with FXS gateways) defines a username and password combination for authentication for each MediaPack port. The ‘Authentication Mode’ parameter (described in Table 5-4) determines if authentication is performed per port or for the entire gateway.
MediaPack 5.5.9.2 Automatic Dialing Use the Automatic Dialing Table to define telephone numbers that are automatically dialed when a specific port is used. ¾ To configure the Automatic Dialing table, take these 6 steps: 1. Open the ‘Automatic Dialing’ screen (Protocol Management menu > Endpoint Settings submenu > Automatic Dialing option); the ‘Automatic Dialing’ screen is displayed. Figure 5-28: Automatic Dialing Table Screen 2. 3.
SIP User's Manual 5. Web Management Table 5-28: Automatic Dialing ini File Parameter Parameter Name in ini File Parameter Format TargetOfChannelX TargetOfChannel = , For example: TargetOfChannel0 = 1001,1 TargetOfChannel3 = 911,2 Note 1: The numbering of channels starts with 0. Note 2: Define this parameter for each gateway port you want to use for Automatic Dialing. Note 3: This parameter can appear up to 8 times for 8-port gateways and up to 24 times for MP-124 gateways. 5.5.9.
MediaPack 3. From the ‘Presentation’ drop-down list, select: • ‘Allowed’ [0] to send the string in the Caller ID/Name field when a (TelÆIP) call is made using this VoIP gateway port. • ‘Restricted’ [1] if you don’t want to send this string. Notes: 4. 5. 6. • When the ‘Presentation’ field is set to ‘Restricted’, the caller identity is passed to the remote side using only the P-Asserted-Identity and P-Preferred-Identity headers (AssertedIdMode).
SIP User's Manual 5. Web Management ¾ To configure the Generate Caller ID to Tel Table, take these 5 steps: 1. Open the ‘Generate Caller ID to Tel’ screen (Protocol Management menu > Endpoint Settings > Generate Caller ID to Tel option); the ‘Generate Caller ID to Tel’ screen is displayed. Figure 5-30: MediaPack FXS Generate Caller ID to Tel Screen 2. In the ‘Caller ID’ field, select one of the following: • Enable: Enables Caller ID generation (FXS) or detection (FXO) for the specific port.
MediaPack 5.5.9.5 Call Forward The VoIP gateway allows you to forward incoming IPÆTel calls (using 302 response) based on the VoIP gateway port to which the call is routed (applicable only to FXS gateways). The Call Forwarding Table is applicable only if the Call Forward feature is enabled. To enable Call Forward set ‘Enable Call Forward’ to ‘Enable’ in the ‘Supplementary Services’ screen, or ‘EnableForward=1’ in the ini file (refer to Table 5-9).
SIP User's Manual 5. Web Management Table 5-31: Call Forward Table (continues on pages 124 to 125) Parameter Description Time for No Reply Forward If you have set the Forward Type for this port to no reply, enter the number of seconds the VoIP gateway waits before forwarding the call to the phone number specified.
MediaPack 5.5.10 Configuring RADIUS Accounting Parameters The RADIUS Parameters screen is used for configuring the Remote Authentication Dial In User Service (RADIUS) accounting parameters. ¾ To configure the FXO parameters, take these 4 steps: 1. Open the ‘RADIUS Parameters' screen (Protocol Management menu > RADIUS Parameters); the ‘RADIUS Parameters' screen is displayed. Figure 5-32: RADIUS Parameters Screen 2. 3. 4. Configure the RADIUS accounting parameters according to Table 5-33.
SIP User's Manual 5. Web Management 5.5.11 Configuring the FXO Parameters Use this screen to configure the gateway’s specific FXO parameters. ¾ To configure the FXO parameters, take these 4 steps: 1. Open the ‘FXO Settings’ screen (Protocol Management menu > FXO Settings > FXO Settings option); the ‘FXO Settings’ screen is displayed. Figure 5-33: FXO Settings Screen 2. 3. 4. Configure the FXO parameters according to Table 5-33. Click the Submit button to save your changes.
MediaPack Table 5-33: FXO Parameters (continues on pages 127 to 130) Parameter Description Waiting For Dial Tone [IsWaitForDialTone] No [0] = Don’t wait for dial tone. Yes [1] = Wait for dial tone (default). Used for IPÆMediaPack/FXO gateways, when ‘One Stage Dialing’ is enabled. If ‘wait for dial tone’ is enabled, the FXO gateway dials the phone number (to the PSTN/PBX line) only after it detects a dial tone. Note 1: The correct dial tone parameters should be configured in the Call Progress Tones file.
SIP User's Manual 5. Web Management Table 5-33: FXO Parameters (continues on pages 127 to 130) Parameter Description Send Metering Message to IP [SendMetering2IP] No [0] = Disabled (default). Yes [1] = FXO gateways send a metering tone INFO message to IP on detection of 12/16 kHz metering pulse. FXS gateways generate the 12/16 kHz metering tone on reception of a metering message. Note 1: Suitable (12 kHz or 16 kHz) coeff must be used for both FXS and FXO gateways.
MediaPack 5.5.12 Configuring Voice Mail (VM) Parameters Use this screen to configure the VM parameters. The VM application applies only to FXO gateways. For detailed information on VM, refer to the CPE Configuration Guide for Voice Mail. ¾ To configure the VM parameters, take these 4 steps: 1. Open the ‘Voice Mail’ screen (Protocol Management menu > FXO Settings > Voice Mail option); the ‘Voice Mail’ screen is displayed. Figure 5-34: Voice Mail Screen 2. 3. 4.
SIP User's Manual 5. Web Management Table 5-34: Voice Mail Parameters ((continues on pages 130 to 131) Parameter Line Transfer Mode [LineTransferMode] Description Determines the transfer method used by the gateway. Disable [0] = IP (default). Blind Transfer [1] = PBX blind transfer. In this mode, after receiving a REFER message from the IP side, the FXO: sends a hook-flash to the PBX, dials the digits (that are received in the Refer-To header), and then immediately drops the line (on-hook).
MediaPack 5.5.13 Protocol Management ini File Parameters Table 5-35 describes the SIP Protocol Management parameters that can only be configured via the ini file. Table 5-35: Protocol Management, ini File Parameters (continues on pages 132 to 137) ini File Parameter Name Valid Range and Description EnablePtime 0 = Remove the ptime header from SDP. 1 = Include the ptime header in SDP (default).
SIP User's Manual 5. Web Management Table 5-35: Protocol Management, ini File Parameters (continues on pages 132 to 137) ini File Parameter Name DisableAutoDTMFMute Valid Range and Description Enables / disables the automatic mute of DTMF digits when out-of-band DTMF transmission is used. 0 = Auto mute is used (default). 1 = No automatic mute of in-band DTMF.
MediaPack Table 5-35: Protocol Management, ini File Parameters (continues on pages 132 to 137) ini File Parameter Name CurrentDisconnectDurat ion Valid Range and Description Duration of the current disconnect pulse (in msec). The default is 900 msec, The range is 200 to 1500 msec. Applicable for both FXS and FXO gateways. Note: The FXO gateways’ detection range is +/-200 msec of the parameter’s value + 100. For example if CurrentDisconnectDuration = 200, the detection range is 100 to 500 msec.
SIP User's Manual 5. Web Management Table 5-35: Protocol Management, ini File Parameters (continues on pages 132 to 137) ini File Parameter Name Valid Range and Description 3WayConferenceMode Defines the mode of operation when the 3-Way Conference feature is used.
MediaPack Table 5-35: Protocol Management, ini File Parameters (continues on pages 132 to 137) ini File Parameter Name EnableRport Valid Range and Description Enables / disables the usage of the ‘rport’ parameter in the Via header. [0] = Enabled. [1] = Disabled (default). The gateway adds an ‘rport’ parameter to the Via header field 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.
SIP User's Manual 5.6 5. Web Management Advanced Configuration Use this menu to set the gateway’s advanced configuration parameters. Note: 5.6.1 Those parameters contained within square brackets are the names used to configure the parameters via the ini file. Configuring the Network Settings From the Network Settings you can: Configure the IP Settings (refer to Section 5.6.1.1). Configure the Application Settings (refer to Section 5.6.1.2 on page 141).
MediaPack 5.6.1.1 Configuring the IP Settings ¾ To configure the IP Settings parameters, take these 4 steps: 1. Open the ‘IP Settings’ screen (Advanced Configuration menu > Network Settings > IP Settings option); the ‘IP Settings’ screen is displayed. Figure 5-35: IP Settings Screen 2. 3. 4. Configure the IP Settings according to Table 5-36. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205.
SIP User's Manual 5. Web Management Table 5-36: Network Settings, IP Settings Parameters (continues on pages 138 to 140) Parameter Description Subnet Mask Subnet mask of the gateway. Enter the subnet mask in dotted format notation, for example 255.255.0.0 Note 1: A warning message is displayed (after pressing the button ‘Submit’) if the entered value is incorrect.
MediaPack Table 5-36: Network Settings, IP Settings Parameters (continues on pages 138 to 140) Parameter Description DHCP Settings Enable DHCP [DHCPEnable] Disable [0] = Disable DHCP support on the gateway (default). Enable [1] = Enable DHCP support on the gateway. After the gateway is powered up, it attempts to communicate with a BootP server. If a BootP server is not responding and if DHCP is enabled, then the gateway attempts to get its IP address and other network parameters from the DHCP server.
SIP User's Manual 5.6.1.2 5. Web Management Configuring the Application Settings ¾ To configure the Application Settings parameters, take these 4 steps: 1. Open the ‘Application Settings’ screen (Advanced Configuration menu > Network Settings > Application Settings option); the ‘Application Settings’ screen is displayed. Figure 5-36: Application Settings Screen 2. 3. 4. Version 5.0 Configure the Application Settings according to Table 5-37. Click the Submit button to save your changes.
MediaPack Table 5-37: Network Settings, Application Settings Parameters Parameter Description NTP Settings For detailed information on NTP, refer to Section 9.7 on page 253. NTP Server IP Address [NTPServerIP] IP address (in dotted format notation) of the NTP server. The default IP address is 0.0.0.0 (the internal NTP client is disabled). NTP UTC Offset [NTPServerUTCOffset] Defines the UTC (Universal Time Coordinate) offset (in seconds) from the NTP server. The default offset is 0.
SIP User's Manual 5.6.1.3 5. Web Management Configuring the NFS Settings Network File System (NFS) enables the MediaPack to access a remote server’s shared files and directories and to handle them as if they’re located locally. A file system, the NFS is independent of machine types, OSs, and network architectures. Up to five different NFS file systems can be configured. NFS is utilized by the MediaPack to load the cmp, ini and configuration files via the Automatic Update mechanism (refer to Section 10.
MediaPack ¾ To delete a remote NFS file system, take these 3 steps: 1. 2. 3. Select the 'Edit' radio button for the row to be deleted. Click the Delete Line button; the row is deleted. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. ¾ To modify an existing remote NFS file system, take these 4 steps: 1. 2. 3. Select the 'Edit' radio button for the row to be modified. Change the values on the selected row according to your requirements.
SIP User's Manual 5.6.1.4 5. Web Management Configuring the IP Routing Table The IP routing table is used by the gateway to determine IP routing rules. It can be used, for example, to define static routing rules for the OAM and Control networks since a default gateway isn’t supported for these networks (refer to Section 9.9.1 on page 254). Before sending an IP packet, the gateway searches this table for an entry that matches the requested destination host / network.
MediaPack Table 5-39: IP Routing Table Column Description (continues on pages 145 to 146) Column Name [ini File Parameter Name] Description The address of the host / network you want to reach is determined by an AND operation that is applied on the fields ‘Destination IP Address’ and ‘Destination Mask’. For example: To reach the network 10.8.x.x, enter 10.8.0.0 in the field ‘Destination IP Address’ and 255.255.0.0 in the field ‘Destination Mask’.
SIP User's Manual 5.6.1.5 5. Web Management Configuring the VLAN Settings For detailed information on the MediaPack VLAN implementation, refer to Section 9.8 on page 253. ¾ To configure the VLAN Settings parameters, take these 4 steps: 1. Open the ‘VLAN Settings’ screen (Advanced Configuration menu > Network Settings > VLAN Settings option); the ‘VLAN Settings’ screen is displayed. Figure 5-39: VLAN Settings Screen 2. 3. 4. Version 5.0 Configure the VLAN Settings according to Table 5-40.
MediaPack Table 5-40: Network Settings, VLAN Settings Parameters Parameter Description VLAN Mode [VlanMode] Sets the VLAN functionality. Disable [0] (default). Enable [1]. PassThrough [2] = N/A. Note: This parameter cannot be changed on-the-fly and requires a gateway reset. IP Settings Native VLAN ID [VlanNativeVlanID] Sets the native VLAN identifier (PVID, Port VLAN ID). The valid range is 1 to 4094. The default value is 1.
SIP User's Manual 5.6.1.6 5. Web Management Network Settings ini File Parameters Table 5-41 describes the Network parameters that can only be configured via the ini file. Table 5-41: Network Settings, ini File Parameters (continues on pages 149 to 151) ini File Parameter Name Valid Range and Description EnablePPPoE Enables the PPPoE (Point-to-Point Protocol over Ethernet) feature. [0] = Disable (default) [1]= Enable PPPoEUserName User Name for PAP or Host Name for CHAP authentication.
MediaPack Table 5-41: Network Settings, ini File Parameters (continues on pages 149 to 151) ini File Parameter Name Valid Range and Description DisableNAT Enables / disables the Network Address Translation (NAT) mechanism. [0] = Enabled. [1]= Disabled (default). Note: The compare operation that is performed on the IP address is enabled by default and is controlled by the parameter ‘EnableIPAddrTranslation’.
SIP User's Manual 5. Web Management Table 5-41: Network Settings, ini File Parameters (continues on pages 149 to 151) ini File Parameter Name Valid Range and Description NoOpInterval Defines the time interval in which RTP or T.38 No-Op packets are sent in the case of silence (no RTP / T.38 traffic) when No-Op packet transmission is enabled. The valid range is 20 to 65,000 msec. The default is 10,000. Note: To enable No-Op packet transmission, use the NoOperationSendingMode parameter.
MediaPack 5.6.2 Configuring the Media Settings From the Media Settings page you can define: Voice Settings (refer to Section 5.6.2.1 below). Fax / Modem / CID Settings (refer to Section 5.6.2.2 on page 154). RTP/RTCP Settings (refer to Section 5.6.2.3 on page 157). Hook-Flash Settings (refer to Section 5.6.2.4 on page 160). General Media Settings (refer to Section 5.6.2.5 on page 161). These parameters are applied to all MediaPack channels. Notes: 5.6.2.
SIP User's Manual 2. 3. 4. 5. Web Management Configure the Voice Settings according to Table 5-42. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Table 5-42: Media Settings, Voice Settings Parameters Parameter Description Voice Volume [VoiceVolume] Voice gain control in dB. This parameter sets the level for the transmitted (IPÆTel) signal. The valid range is -32 to 31 dB. The default value is 0 dB.
MediaPack 5.6.2.2 Configuring the Fax / Modem / CID Settings ¾ To configure the Fax / Modem / CID Settings parameters, take these 4 steps: 1. Open the ‘Fax / Modem / CID Settings’ screen (Advanced Configuration menu > Media Settings > Fax / Modem / CID Settings option); the ‘Fax / Modem / CID Settings’ screen is displayed. Figure 5-41: Fax / Modem / CID Settings Screen 2. 3. 4. Configure the Fax / Modem / CID Settings according to Table 5-43. Click the Submit button to save your changes.
SIP User's Manual 5. Web Management Table 5-43: Media Settings, Fax/Modem/CID Parameters (continues on pages 154 to 157) Parameter Description Caller ID Type [CallerIDType] Defines one of the following standards for detection (FXO) and generation (FXS) of Caller ID and detection (FXO) of MWI (when specified) signals. Bellcore [0] (Caller ID and MWI) (default). [1] (Caller ID and MWI) ETSI NTT [2].
MediaPack Table 5-43: Media Settings, Fax/Modem/CID Parameters (continues on pages 154 to 157) Parameter Description Fax Relay Enhanced Redundancy Depth [FaxRelayEnhancedRedundanc yDepth] Number of times that control packets are retransmitted when using the T.38 standard. The valid range is 0 to 4. The default value is 2. Fax Relay ECM Enable [FaxRelayECMEnable] Disable [0] = Error Correction Mode (ECM) mode is not used during fax relay. Enable [1] = ECM mode is used during fax relay (default).
SIP User's Manual 5.6.2.3 5. Web Management Configuring the RTP / RTCP Settings ¾ To configure the RTP / RTCP Settings parameters, take these 4 steps: 1. Open the ‘RTP / RTCP Settings’ screen (Advanced Configuration menu > Media Settings > RTP / RTCP Settings option); the ‘RTP / RTCP Settings’ screen is displayed. Figure 5-42: RTP / RTCP Settings Screen 2. 3. 4. Version 5.0 Configure the RTP / RTCP Settings according to Table 5-44. Click the Submit button to save your changes.
MediaPack Table 5-44: Media Settings, RTP / RTCP Parameters (continues on pages 158 to 159) Parameter Description Dynamic Jitter Buffer Minimum Delay [DJBufMinDelay] Minimum delay for the Dynamic Jitter Buffer. The valid range is 0 to 150 milliseconds. The default delay is 10 milliseconds. Note: For more information on the Jitter Buffer, refer to Section 8.6 on page 230. Dynamic Jitter Buffer Optimization Factor [DJBufOptFactor] Dynamic Jitter Buffer frame error / delay optimization factor.
SIP User's Manual 5. Web Management Table 5-44: Media Settings, RTP / RTCP Parameters (continues on pages 158 to 159) Parameter Description RTP Base UDP Port [BaseUDPPort] Lower boundary of UDP port used for RTP, RTCP (Real-Time Control Protocol) (RTP port + 1) and T.38 (RTP port + 2). The upper boundary is the Base UDP Port + 10 * (number of gateway’s channels). The range of possible UDP ports is 6,000 to 64,000. The default base UDP port is 6000.
MediaPack 5.6.2.4 Configuring the Hook-Flash Settings ¾ To configure the Hook-Flash Settings parameters, take these 4 steps: 1. Open the ‘Hook-Flash Settings’ screen (Advanced Configuration menu > Media Settings > Hook-Flash Settings option); the ‘Hook-Flash Settings’ screen is displayed. Figure 5-43: Hook-Flash Settings Screen 2. 3. 4. Configure the Hook-Flash Settings according to Table 5-45. Click the Submit button to save your changes.
SIP User's Manual 5.6.2.5 5. Web Management Configuring the General Media Settings ¾ To configure the General Media Settings parameters, take these 4 steps: 1. Open the ‘General Media Settings’ screen (Advanced Configuration menu > Media Settings > General Media Settings option); the ‘General Media Settings’ screen is displayed. Figure 5-44: General Media Settings Screen 2. 3. 4. Configure the General Media Settings according to Table 5-46. Click the Submit button to save your changes.
MediaPack 5.6.2.6 Media Settings ini File Parameters Table 5-47 describes the Media Settings parameters that can only be configured via the ini file. Table 5-47: Media Settings, ini File Parameters (continues on pages 162 to 163) ini File Parameter Name Valid Range and Description RTPSIDCoeffNum Determines the number of spectral coefficients added to an SID packet being sent according to RFC 3389. Valid only if ‘EnableStandardSIDPayloadType’ is set to 1.
SIP User's Manual 5.
MediaPack Table 5-47: Media Settings, ini File Parameters (continues on pages 162 to 163) ini File Parameter Name Valid Range and Description StunServerDomainName Defines the domain name for the STUN server's address (used for retrieving all STUN servers with an SRV query). The STUN client can perform the required SRV query to resolve this domain name to an IP address and port, sort the server list, and use the servers according to the sorted list.
SIP User's Manual 5.6.3 5. Web Management Restoring and Backing up the Gateway Configuration The Configuration File screen enables you to restore (load a new ini file to the gateway) or to back up (make a copy of the VoIP gateway ini file and store it in a directory on your computer) the current configuration the gateway is using. Back up your configuration if you want to protect your VoIP gateway programming.
MediaPack 5.6.4 Regional Settings The ‘Regional Settings’ screen enables you to set and view the gateway’s internal date and time and to load to the gateway the following configuration files: Call Progress Tones, coefficient and Voice Prompts (currently not applicable to MediaPack gateways). For detailed information on the configuration files, refer to Chapter 6 on page 209. ¾ To configure the date and time of the MediaPack, take these 3 steps: 1.
SIP User's Manual 5. Web Management ¾ To load a configuration file to the gateway, take these 8 steps: 1. Open the ‘Regional Settings’ screen (Advanced Configuration menu > Regional Settings); the ‘Regional Settings’ screen is displayed (shown in Figure 5-46). Click the Browse button adjacent to the file you want to load. Navigate to the folder that contains the file you want to load. Click the file and click the Open button; the name and path of the file appear in the field beside the Browse button.
MediaPack 5.6.5 Security Settings From the Security Settings you can: 5.6.5.1 Configure the Web User Accounts (refer to Section 5.6.5.1 below). Configure the Web & Telnet Access List (refer to Section 5.6.5.2 on page 170). Configure the Firewall Settings (refer to Section 5.6.5.3 on page 171). Configure the Certificates (refer to Section 5.6.5.4 on page 172). Configure the General Security Settings (refer to Section 5.6.5.5 on page 173).
SIP User's Manual 5. Web Management ¾ To change the Web User Accounts attributes, take these 4 steps: 1. Open the ‘Web User Accounts’ screen (Advanced Configuration menu > Security Settings > Web User Accounts option); the ‘Web User Accounts’ screen is displayed. Figure 5-47: Web User Accounts Screen (for Users with ‘Security Administrator’ Privileges) 2. 3. 4.
MediaPack 5.6.5.2 Configuring the Web and Telnet Access List Use this screen to define up to ten IP addresses that are permitted to access the gateway’s Web and Telnet interfaces. Access from an undefined IP address is denied. This security feature is inactive (the gateway can be accessed from any IP address) when the table is empty. ¾ To manage the Web & Telnet access list, take these 4 steps: 1.
SIP User's Manual 5.6.5.3 5. Web Management Configuring the Firewall Settings The MediaPack accommodates an internal Firewall, allowing the security administrator to define network traffic filtering rules. For detailed information on the internal Firewall, refer to Section 12.5 on page 297. ¾ To create a new access rule, take these 6 steps: 1. Open the ‘Firewall Settings’ screen (Advanced Configuration menu > Security Settings > Firewall Settings option); the ‘Firewall Settings’ screen is displayed.
MediaPack ¾ To delete a rule, take these 3 steps: 1. 2. 3. Select the radio button of the entry you want to activate. Click the Delete Rule button; the rule is deleted. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Table 5-49: Internal Firewall Fields Parameter Description Is Rule Active A read-only field that indicates whether the rule is active or not. Note: After reset all rules are active.
SIP User's Manual 5.6.5.5 5. Web Management Configuring the General Security Settings ¾ To configure the General Security Settings parameters, take these 4 steps: 1. Open the ‘General Security Settings’ screen (Advanced Configuration menu > Security Settings > General Security Settings option); the ‘General Security Settings’ screen is displayed. Figure 5-50: General Security Settings Screen 2. 3. 4. Version 5.0 Configure the General Security Settings according to Table 5-50 below.
MediaPack Table 5-50: Security Settings, General Security Settings Parameters (continues on pages 174 to 175) Parameter Description Secured Web Connection [HTTPSOnly] Determines the protocol types used to access the Embedded Web Server. HTTP and HTTPS [0] (default). HTTPS only [1] (unencrypted HTTP packets are blocked). HTTP Authentication Mode [WebAuthMode] Determines the authentication mode for the Embedded Web Server. Basic [0] = Basic authentication (clear text) is used (default).
SIP User's Manual 5. Web Management Table 5-50: Security Settings, General Security Settings Parameters (continues on pages 174 to 175) Parameter Description Local RADIUS Password Cache Mode [RadiusLocalCacheMode] Defines the gateway’s mode of operation regarding the timer (configured by the parameter RadiusLocalCacheTimeout) that determines the validity of the username and password (verified by the RADIUS server).
MediaPack 5.6.5.7 Configuring the IKE Table Use the IKE Table screen to configure the IKE parameters. For detailed information on IPSec and IKE, refer to Section 12.1 on page 279. 5.6.6 Configuring the Management Settings ¾ To configure the Management Settings parameters, take these 4 steps: 1. Open the ‘Management Settings’ screen (Advanced Configuration menu > Management Settings); the ‘Management Settings’ screen is displayed. Figure 5-51: Management Settings Screen 2. 3. 4.
SIP User's Manual 5. Web Management Table 5-51: Management Settings Parameters (continues on pages 177 to 178) Parameter Description Syslog Settings Syslog Server IP address [SyslogServerIP] IP address (in dotted format 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 VoIP gateway. For information on the Syslog, refer to Section 13.2 on page 301.
MediaPack Table 5-51: Management Settings Parameters (continues on pages 177 to 178) Parameter Description Access to Restricted Domains [ActivityListToLog = ARD] Access to Restricted Domains. The following screens are restricted: (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.
SIP User's Manual 5. Web Management If you clear a checkbox and click Submit, all settings in the same row revert to their defaults. Note: Table 5-52: SNMP Managers Table Parameters Parameter Description Checkbox [SNMPManagerIsUsed_x] Up to five parameters, each determines the validity of the parameters (IP address and port number) of the corresponding SNMP Manager used to receive SNMP traps.
MediaPack 5.6.6.2 Configuring the SNMP Community Strings Use the SNMP Community Strings table to configure up to five read-only and up to five read / write SNMP community strings, and to configure the community string that is used for sending traps. For detailed information on SNMP community strings, refer to Section 14.8.1 on page 313. ¾ To configure the SNMP Community Strings, take these 5 steps: 1.
SIP User's Manual 5.6.6.3 5. Web Management Configuring SNMP V3 Use the SNMP V3 Table to configure authentication and privacy for up to 10 SNMP V3 users. For detailed information on SNMP community strings, refer to Section 14.8.1 on page 313. ¾ To configure the SNMP V3 Users, take these 5 steps: 1. Access the ‘Management Settings’ screen (Advanced Configuration menu > Management Settings); the ‘Management Settings’ screen is displayed (Figure 5-51).
MediaPack 5.6.6.4 Advanced Configuration ini File Parameters Table 5-55 describes the board parameters that can only be configured via the ini file.
SIP User's Manual 5. Web Management Table 5-55: Board, ini File Parameters (continues on pages 182 to 184) ini File Parameter Name Valid Range and Description RADIUSRetransmission Determines the number of RADIUS retransmission retries for the same request. The valid range is 1 to 10. The default value is 3. RADIUSTo Determines the time interval (measured in seconds) the gateway waits for a response before a RADIUS retransmission is issued. The valid range is 1 to 30. The default value is 10.
MediaPack Table 5-55: Board, ini File Parameters (continues on pages 182 to 184) ini File Parameter Name BootPSelectiveEnable Valid Range and Description Enables the Selective BootP mechanism. 1 = Enabled. 0 = Disabled (default). The Selective BootP mechanism (available from Boot version 1.92) enables the gateway’s integral BootP client to filter unsolicited BootP/DHCP replies (accepts only BootP replies that contain the text ‘AUDC’ in the vendor specific information field).
SIP User's Manual 5. Web Management Table 5-56: Automatic Updates Parameters (continues on pages 184 to 185) ini File Parameter Name Description IniFileURL Specifies the name of the ini file and the location of the server (IP address or FQDN) from which the gateway loads the ini file. The ini file can be loaded using: HTTP, HTTPS, FTP, FTPS or NFS. For example: http://192.168.0.1/filename http://192.8.77.
MediaPack 5.6.6.6 SNMP ini File Parameters Table 5-57 describes the SNMP parameters that can only be configured via the ini file. Table 5-57: SNMP ini File Parameters ini File Parameter Name Description SNMPPort The device’s local UDP port used for SNMP Get/Set commands. The range is 100 to 3999. The default port is 161. SNMPTrustedMGR_x Up to five IP addresses of remote trusted SNMP managers from which the SNMP agent accepts and processes get and set requests.
SIP User's Manual 5.7 5. Web Management Status & Diagnostics Use this menu to view and monitor the gateway’s channels, Syslog messages, hardware / software product information, and to assess the gateway’s statistics and IP connectivity information. 5.7.1 Gateway Statistics Use the screens under Gateway Statistics to monitor real-time activity such as IP Connectivity information, call details and call statistics, including the number of call attempts, failed calls, fax calls, etc. Notes: 5.7.1.
MediaPack ¾ To view the IP connectivity information, take these 2 steps: 1. 2. Set ‘AltRoutingTel2IPEnable’ to 1 or 2. Open the ‘IP Connectivity’ screen (Status & Diagnostics menu > Gateway Statistics submenu > IP Connectivity); the ‘IP Connectivity’ screen is displayed (Figure 5-55). Figure 5-55: IP Connectivity Screen Table 5-58: IP Connectivity Parameters Column Name Description IP Address IP address defined in the destination IP address field in the Tel to IP Routing table.
SIP User's Manual 5.7.1.2 5. Web Management Call Counters The Call Counters screens provide you with statistic information on incoming (IPÆTel) and outgoing (TelÆIP) calls. The statistic 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 CDR message is sent). The release reason can be viewed in the Termination Reason field in the CDR message. For detailed information on each counter, refer to Table 5-59 on page 189.
MediaPack Table 5-59: Call Counters Description (continues on pages 189 to 190) Counter Number of Failed Calls due to a Busy Line Number of Failed Calls due to No Answer Description This counter indicates the number of calls that failed as a result of a busy line. It is incremented as a result of the following release reason: GWAPP_USER_BUSY (17) This counter indicates the number of calls that weren’t answered.
SIP User's Manual 5.7.1.3 5. Web Management Call Routing Status The Call Routing Status screen provides you with information on the current routing method used by the gateway. This information includes the IP address and FQDN (if used) of the Proxy server the gateway currently operates with. Figure 5-57: Call Routing Status Screen Table 5-60: Call Routing Status Parameters Parameter Current Call-Routing Method Description Proxy = Proxy server is used to route calls.
MediaPack 5.7.2 Activating the Internal Syslog Viewer The Message Log screen displays Syslog debug messages sent by the gateway. Note that it is not recommended to keep a ‘Message Log’ session open for a prolonged period (refer to the Note below). For prolong debugging use an external Syslog server, refer to Section 13.2 on page 301. Refer to the Debug Level parameter ‘GwDebugLevel’ (described in Table 5-8) to determine the Syslog logging level. ¾ To activate the Message Log, take these 4 steps: 1.
SIP User's Manual 5.7.3 5. Web Management Device Information The Device Information screen displays specific hardware and software product information. This information can help you to expedite any troubleshooting process. Capture the screen and email it to ‘our’ Technical Support personnel to ensure quick diagnosis and effective corrective action. From this screen you can also view and remove any loaded files used by the MediaPack (stored in the RAM).
MediaPack 5.7.4 Viewing the Ethernet Port Information The Ethernet Port Information screen provides read-only information on the Ethernet connection used by the MediaPack. The Ethernet Port Information parameters are displayed in Table 5-61. For detailed information on the Ethernet interface configuration, refer to Section 9.1 on page 247.
SIP User's Manual 5.8 5. Web Management Monitoring the MediaPack Channels (Home Page) The 'Channel Status' screen provides real-time monitoring on the current channels status. In addition, this screen allows you to assign a brief description or name to each port as well as releasing a channel. The Web interface provides the Home icon 5.8.1 for quick-and-easy access to this screen.
MediaPack ¾ To monitor the details of a specific channel, take these 3 steps: 1. Click the numbered port icon of the specific channel whose detailed status you need to check/monitor; a shortcut menu appears. From the shortcut menu, choose Port Settings; the channel-specific Channel Status screen appears, shown in Figure 5-62. Click the submenu links to check/view a specific channel’s parameter settings. 2. 3. Figure 5-62: Channel Status Details Screen 5.8.
SIP User's Manual 5.9 5. Web Management Software Update The ‘Software Update’ menu enables users to upgrade the MediaPack software by loading a new cmp file along with the ini and a suite of auxiliary files, or to update the existing auxiliary files. The ‘Software Update’ menu comprises two submenus: Software Upgrade Wizard (refer to Section 5.9.1 below). Load Auxiliary Files (refer to Section 5.9.2 on page 202). Note: 5.9.
MediaPack ¾ To use the Software Upgrade Wizard, take these 11 steps: 1. 2. Stop all traffic on the MediaPack (refer to the note above). Open the ‘Software Upgrade Wizard’ (Software Update menu > Software Upgrade Wizard); the ‘Start Software Upgrade’ screen appears. Figure 5-63: Start Software Upgrade Screen Note: 3. At this point, the process can be canceled with no consequence to the MediaPack (click the Cancel button).
SIP User's Manual 5. Web Management Figure 5-64: Load a cmp File Screen 4. Click the Browse button, navigate to the cmp file and click the button Send File; the cmp file is loaded to the MediaPack and you’re notified as to a successful loading (refer to Figure 5-65). Figure 5-65: cmp File Successfully Loaded into the MediaPack Notification 5. 6. Note that the four action buttons (Cancel, Reset, Back, and Next) are now activated (following cmp file loading).
MediaPack Figure 5-66: Load an ini File Screen 7. In the ‘Load an ini File’ screen, you can now choose to either: • Click Browse and navigate to the ini file; the check box ‘Use existing configuration’, by default checked, becomes unchecked. Click Send File; the ini file is loaded to the MediaPack and you’re notified as to a successful loading. • Ignore the Browse button (its field remains undefined and the check box ‘Use existing configuration’ remains checked by default). • 8.
SIP User's Manual 5. Web Management Follow the same procedure you followed when loading the ini file (refer to Step 7). The same procedure applies to the ‘Load a coefficient file’ screen. 10. In the ‘Finish’ screen (refer to Figure 5-68), the Next button is disabled. Complete the upgrade process by clicking Reset or Cancel. 9. • Click Reset, the MediaPack ‘burns’ the newly loaded files to flash memory. The ‘Burning files to flash memory’ screen appears. Wait for the ‘burn’ to finish.
MediaPack 5.9.2 Auxiliary Files The ‘Auxiliary Files’ screen enables you to load to the gateway the following files: Call Progress Tones, coefficient, Prerecorded Tones (PRT) and User Information. The Voice Prompts file is currently not applicable to the MediaPack. For detailed information on these files, refer to Section 6 on page 209. For information on deleting these files from the MediaPack, refer to Section 5.7.3 on page 193. Table 5-63 presents a brief description of each auxiliary file.
SIP User's Manual 5. Web Management Figure 5-70: Auxiliary Files Screen 5.9.2.1 Loading the Auxiliary Files via the ini File ¾ To load the auxiliary files via the ini file, take these 3 steps: In the ini file, define the auxiliary files to be loaded to the MediaPack. You can also define in the ini file whether the loaded files should be stored in the non-volatile memory so that the TFTP process is not required every time the MediaPack boots up. 2.
MediaPack 5.10 Maintenance The Maintenance menu is used for the following operations: Locking and unlocking the gateway (refer to Section 5.10.1 on page 204) Saving the gateway's configuration (refer to Section 5.10.2 on page 205) Resetting the gateway (refer to Section 5.10.3 on page 206) 5.10.1 Locking and Unlocking the Gateway The Lock and Unlock options allow you to lock the gateway so that it does not accept any new incoming calls.
SIP User's Manual 5. Web Management 4. Click the LOCK button. If 'Graceful Option' is set to 'Yes', the lock is delayed and a screen displaying the number of remaining calls and time is displayed. Otherwise, the lock process begins immediately. The Current Admin State displays the current state: LOCKED or UNLOCKED. ¾ To unlock the gateway, take these 2 steps: 1. Access the 'Maintenance Actions' screen as described above in the previous procedure. Click the UNLOCK button.
MediaPack 5.10.3 Resetting the MediaPack The 'Maintenance Actions' screen enables you to remotely reset the gateway. Before you reset the gateway, you can choose the following options: Save the gateway's current configuration to the flash memory (non-volatile). Perform a graceful shutdown. Reset starts only after a user-defined time expires or no more active traffic exists (the earliest thereof). ¾ To reset the gateway, take these 5 steps: 1.
SIP User's Manual 5.11 5. Web Management 4. In the 'Shutdown Timeout' field (relevant only if the 'Graceful Option' in the previous step is set to 'Yes'), enter the time after which the gateway resets. Note that if no traffic exists and the time has not expired, the gateway resets. 8. Click the Reset button. If 'Graceful Option' is set to 'Yes', the reset is delayed and a screen displaying the number of remaining calls and time is displayed.
MediaPack Reader's Notes SIP User's Manual 208 Document #: LTRT-65408
SIP User's Manual 6 6. ini File Configuration of the MediaPack ini File Configuration of the MediaPack As an alternative to configuring the VoIP gateway using the Web Interface (refer to Chapter 5 on page 49), it can be configured by loading the ini file containing Customer-configured parameters. The ini file is loaded via the BootP/TFTP utility (refer to Appendix C on page 349) or via any standard TFTP server. It can also be loaded through the Web Interface (refer to Section 5.6.3 on page 165).
MediaPack 6.3 The ini File Structure The ini file can contain any number of parameters. The parameters are divided into groups by their functionality. The general form of the ini file is shown in Figure 6-1 below. Figure 6-1: ini File Structure [Sub Section Name] Parameter_Name = Parameter_Value Parameter_Name = Parameter_Value ; REMARK [Sub Section Name] 6.3.1 6.3.2 The ini File Structure Rules The ini file name mustn’t include hyphens or spaces, use underscore instead.
SIP User's Manual 7 7. Using BootP / DHCP Using BootP / DHCP The MediaPack uses the Bootstrap Protocol (BootP) and the Dynamic Host Configuration Protocol (DHCP) to obtain its networking parameters and configuration automatically after it is reset. BootP and DHCP are also used to provide the IP address of a TFTP server on the network, and files (cmp and ini) to be loaded into memory. DHCP is a communication protocol that automatically assigns IP addresses from a central point.
MediaPack 7.2 Using DHCP When the gateway is configured to use DHCP (DHCPEnable = 1), it attempts to contact the local DHCP server to obtain the networking parameters (IP address, subnet mask, default gateway, primary/secondary DNS server and two SIP server addresses). These network parameters have a ‘time limit’. After the time limit expires, the gateway must ‘renew’ its lease from the DHCP server.
SIP User's Manual 7. Using BootP / DHCP 7.3 Using BootP 7.3.1 Upgrading the MediaPack When upgrading the MediaPack (loading new software onto the gateway) using the BootP/TFTP configuration utility: From version 4.4 to version 4.4 or to any higher version, the device retains its configuration (ini file). However, the auxiliary files (CPT, logo, etc.) may be erased. From version 4.6 to version 4.
MediaPack 7.3.2 Vendor Specific Information Field The MediaPack uses the vendor specific information field in the BootP request to provide device-related initial startup information. The BootP/TFTP configuration utility displays this information in the ‘Client Info’ column (refer to Figure C-1). This option is not available on DHCP servers. Note: The Vendor Specific Information field is disabled by default.
SIP User's Manual 8. Telephony Capabilities 8 Telephony Capabilities 8.1 Working with Supplementary Services The MediaPack SIP FXS and FXO gateways support the following supplementary services: Call Hold / Retrieve; refer to Section 8.1.1 on page 215. Consultation / Alternate; refer to Section 8.1.2 on page 216. Transfer (Refer + Replaces); refer to Section 8.1.3 on page 216. Call Forward (3xx Redirect Responses); refer to Section 8.1.4 on page 217.
MediaPack 8.1.2 8.1.3 Consultation / Alternate The Consultation feature is relevant only for the holding party (applicable only to the MediaPack/FXS gateway). After holding a call (by pressing hook-flash), the holding party hears dial tone and can now initiate a new call, which is called a consultation call. While hearing a dial tone or when dialing to the new destination (before dialing is complete) the user can retrieve the held call by pressing hook-flash.
SIP User's Manual 8.1.4 8. Telephony Capabilities Call Forward Five forms of call forward are supported: Immediate: Any incoming call is forwarded immediately and unconditionally. Busy: Incoming call is forwarded if the endpoint is busy. No Reply: The incoming call is forwarded if it isn't answered for a specified time. On Busy or No Reply: Forward incoming calls when the port is busy or when calls are not answered after a specified time.
MediaPack 8.1.6 Message Waiting Indication Support for Message Waiting Indication (MWI) according to IETF , including SUBSCRIBE (to MWI server). MediaPack/FXS gateways can accept an MWI NOTIFY message that indicates waiting messages or that the MWI is cleared. Users are informed of these messages by a stutter dial tone. The stutter and confirmation tones are defined in the CPT file (refer to Section 15.1 on page 325).
SIP User's Manual 3. 8. Telephony Capabilities Using NOTIFY messages according to : In this mode DTMF digits are carried to the remote side using NOTIFY messages. To enable this mode set: • ‘RxDTMFOption = 0’ (Declare RFC 2833 in SDP = No) ‘TxDTMFOption = 2’ (1st to 5th DTMF Option = NOTIFY) Note that in this mode DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0 (DTMF Mute)).
MediaPack 8.3 Fax & Modem Transport Modes 8.3.1 Fax/Modem Settings Users may choose to use one of the following transport methods for fax and for each modem type (V.22/V.23/Bell/V.32/V.34): Fax relay: demodulation / modulation Bypass: using a high bit rate coder to pass the signal Transparent: passing the signal in the current voice coder When the fax relay mode is enabled, distinction between fax and modem is not immediately possible at the beginning of a session.
SIP User's Manual 8.3.4 8. Telephony Capabilities Supporting V.34 Faxes V.34 faxes don’t comply with the T.38 relay standard. We therefore provide the optional modes described under Sections 8.3.4.1 and 8.3.4.2: Note that the CNG detector is disabled (CNGDetectorMode=0) in all the following examples. 8.3.4.1 Using Bypass Mechanism for V.34 Fax Transmission In this proprietary scenario, the media gateway uses a high bit-rate coder to transmit V.34 faxes, enabling the full utilization of its speed.
MediaPack 8.3.5 Supporting V.152 Implementation The MediaPack gateway supports the ITU-T recommendation V.152 (Procedures for Supporting Voice-Band Data over IP Networks). Voice-band data (VBD) is the transport of modem, facsimile, and text telephony signals over a voice channel of a packet network with a codec appropriate for such signals. For V.152 capability, the gateway supports T.38 as well as VBD codecs (i.e., G.711 A-law and G.711 µ-law).
SIP User's Manual 8.4 8. Telephony Capabilities FXO Operating Modes This section provides a description of the FXO operating modes and gateway configurations for Tel-to-IP and IP-to-Tel calls. 8.4.1 IP-to-Telephone Calls The FXO gateway provides the following FXO operating modes for IP-to-Tel calls: 8.4.1.
MediaPack One -stage dialing incorporates the following FXO functionality: Waiting for Dial Tone The Waiting for Dial Tone feature enables the gateway to dial the digits to the Tel side only after detecting a dial tone from the PBX line. The ini file parameter IsWaitForDialTone is used to configure this operation. Time to Wait Before Dialing The Time to Wait Before Waiting feature defines the time (in msec) between seizing the FXO line and starting to dial the digits.
SIP User's Manual 8. Telephony Capabilities Two-stage dialing implements the Dialing Time feature. Dialing Time allows you to define the time that each digit can be separately dialed. By default, the overall dialing time per digit is 200 msec. The longer the telephone number, the greater the dialing time will be. The relevant parameters for configuring Dialing Time include the following: 8.4.1.
MediaPack Interruption of RTP stream Relevant parameters: BrokenConnectionEventTimeout and DisconnectOnBrokenConnection. Note: Protocol-based termination of the call from the IP side Note: 8.4.1.4 This method operates correctly only if silence suppression is not used. The implemented disconnect method must be supported by the CO or PBX. DID Wink The gateway's FXO ports support Direct Inward Dialing (DID).
SIP User's Manual 8.4.2.1 8. Telephony Capabilities Automatic Dialing Automatic dialing is defined using the ini file parameter TargetOfChannelX (where 'X' is the channel number) or the embedded Web server's 'Automatic Dialing' screen (refer to Section 5.5.9.2 on page 120). The SIP call flow diagram below illustrates Automatic Dialing. Figure 8-3: Call Flow for Automatic Dialing Version 5.
MediaPack 8.4.2.2 Collecting Digits Mode When automatic dialing is not defined, the gateway collects the digits. The SIP call flow diagram below illustrates the Collecting Digits Mode. Figure 8-4: Call Flow for Collecting Digits Mode 8.4.2.3 Ring Detection Timeout The ini file parameters IsWaitForDialTone and WaitForDialTone apply to Ring Detection Timeout.
SIP User's Manual 8.4.2.4 8. Telephony Capabilities FXO Supplementary Services Hold / Transfer toward the Tel side The ini file parameter LineTransferMode must be set to 0 (default). If the FXO receives a hook-flash from the IP side (using out-of-band or RFC 2833), the gateway sends the hook-flash to the Tel side by one of the following: • Performing a hook flash (i.e.
MediaPack 8.6 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. This is called jitter (delay variation), and degrades the perceived voice quality. To minimize this problem, the gateway uses a jitter buffer.
SIP User's Manual 8.7 8. Telephony Capabilities Configuring the Gateway’s Alternative Routing (based on Connectivity and QoS) The Alternative Routing feature enables reliable routing of Tel to IP calls when a Proxy isn’t used. The MediaPack gateway periodically checks the availability of connectivity and suitable Quality of Service (QoS) before routing. If the expected quality cannot be achieved, an alternative IP route for the prefix (phone number) is selected. 8.7.
MediaPack 8.8 Mapping PSTN Release Cause to SIP Response The MediaPack FXO gateway is used to interoperate 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 IPÆTel calls. The converse is also true: For TelÆIP calls, the SIP 4xx or 5xx responses are mapped to tones played to the PSTN/PBX.
SIP User's Manual 8.
MediaPack Table 8-2: Supported RADIUS Attributes (continues on pages 233 to 235) Attribute Number Attribute Name VSA No.
SIP User's Manual 8. Telephony Capabilities 8.10.1 RADIUS Server Messages In Figure 8-5 below, non-standard parameters are preceded with brackets. Figure 8-5: Accounting Example Accounting-Request (361) user-name = 111 acct-session-id = 1 nas-ip-address = 212.179.22.
MediaPack 8.11 Proxy or Registrar Registration Example The REGISTER message is sent to the Registrar’s IP address (if configured) or to the Proxy’s IP address. The message is sent per gateway or per gateway endpoint according to the ‘AuthenticationMode’ parameter. Usually the FXS gateways are registered per gateway port, while FXO gateways send a single registration message, where Username is used instead of phone number in From/To headers.
SIP User's Manual 8.12 8. Telephony Capabilities Configuration Examples 8.12.1 Establishing a Call between Two Gateways After you’ve installed and set up the MediaPack, you can ensure that it functions as expected by establishing a call between it and another gateway. This section exemplifies how to configure two 8-port MediaPack FXS SIP gateways to establish a call. After configuration, you can make calls between telephones connected to a single MediaPack gateway or between the two MediaPack gateways.
MediaPack 8.12.2 SIP Call Flow The following Call Flow describes SIP messages exchanged between two MediaPack gateways during simple call. Telephone ‘6000’ dials ‘2000’, sending INVITE message to Gateway 10.8.201.161. Figure 8-6: SIP Call Flow 10.8.201.158 10.8.201.161 INVITE F1 Ringing F2 200 OK F3 ACK F4 BYE F5 200 OK F6 F1 10.8.201.158 ==> 10.8.201.161 INVITE INVITE sip:6000@10.8.201.161;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.158;branch=z9hG4bKacolwbzYF From:
SIP User's Manual 8. Telephony Capabilities F2 10.8.201.161 ==> 10.8.201.158 180 RINGING SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.8.201.158;branch=z9hG4bKacolwbzYF From: ;tag=1c3535 To: ;tag=1c29715 Call-ID: 2123353775377NrpL-2000--6000@10.8.201.158 Server: Audiocodes-Sip-Gateway/MP-118 FXS/v.4.20.299.410 CSeq: 20214 INVITE Supported: 100rel,em Content-Length: 0 Note: Phone ‘2000’ answers the call, and sends 200 OK message to gateway 10.8.201.158. F3 10.8.
MediaPack F5 10.8.201.161 ==> 10.8.201.158 BYE BYE sip:2000@10.8.201.158;user=phone;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.8.201.161;branch=z9hG4bKacLBzZgmA From: ;tag=1c29715 To: ;tag=1c3535 Call-ID: 2123353775377NrpL-2000--6000@10.8.201.158 User-Agent: Audiocodes-Sip-Gateway/MP-118 FXS/v.4.20.299.410 CSeq: 34541 BYE Supported: 100rel,em Content-Length: 0 F6 10.8.201.158 ==> 10.8.201.161 200 OK SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.201.
SIP User's Manual 8. Telephony Capabilities On receiving this request the Registrar/Proxy returns 401 Unauthorized response. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.1.200 From: ;tag=1c17940 To: Call-ID: 634293194@10.1.1.200 Cseq: 1 REGISTER Date: Mon, 30 Jul 2001 15:33:54 GMT Server: Columbia-SIP-Server/1.17 Content-Length: 0 WWW-Authenticate: Digest realm="audiocodes.
MediaPack At this time a new REGISTER request is issued with the response: REGISTER sip:10.2.2.222 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200 From: ;tag=1c23940 To: Call-ID: 654982194@10.1.1.200 Server: Audiocodes-Sip-Gateway/MP-118 FXS/v.4.20.299.410 CSeq: 1 REGISTER Contact: sip:122@10.1.1.200: Expires:3600 Authorization: Digest, username: 122, realm="audiocodes.com”, nonce="11432d6bce58ddf02e3b5e1c77c010d2", uri=”10.2.2.
SIP User's Manual 8. Telephony Capabilities Connect the FXO MediaPack ports directly to the PBX lines as shown in the diagram below: Figure 8-7: MediaPack FXS & FXO Remote IP Extension 8.12.4.1 Dialing from Remote Extension (Phone connected to FXS) ¾ To configure the call, take these 6 steps: 1. Lift the handset to hear the dial tone coming from PBX, as if the phone was connected directly to PBX.
MediaPack 8.12.4.2 Dialing from other PBX line, or from PSTN ¾ To configure the call, take these 5 steps: 1. Dial the PBX subscriber number the same way as if the user’s phone was connected directly to PBX. When the PBX rings the FXO MediaPack, the ring signal is immediately ‘sent' to the phone connected to the FXS MediaPack. Once the phone’s handset, connected to the FXS MediaPack is raised, the FXO MediaPack seizes the PBX line and the voice path is established between the phone and the PBX line.
SIP User's Manual 3. 8. Telephony Capabilities In the ‘Tel to IP Routing’ screen, enter 20 in the ‘Destination Phone Prefix’ field, and the IP address of the FXO MediaPack gateway (10.1.10.2) in the field ‘IP Address’. Note: In remote extensions, for the transfer to function, hold must be disabled on the FXS (i.e., Enable Hold = 0). 8.12.4.4 FXO MediaPack Configuration (using the Embedded Web Server) ¾ To configure the FXO MediaPack, take these 4 steps: 1.
MediaPack Reader's Notes SIP User's Manual 246 Document #: LTRT-65408
SIP User's Manual 9. Networking Capabilities 9 Networking Capabilities 9.1 Ethernet Interface Configuration Using the parameter ‘EthernetPhyConfiguration‘, users can control the Ethernet connection mode. Either the manual modes (10 Base-T Half-Duplex, 10 Base-T Full-Duplex, 100 Base-TX Half-Duplex, 100 Base-TX Full-Duplex) or Auto-Negotiate mode can be used.
MediaPack The way SIP is designed creates a problem for VoIP traffic to pass through NAT. SIP uses IP addresses and port numbers in its message body. 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. A gateway (located behind a NAT) that initiates a signaling path will have problems in receiving incoming signaling responses (they will be blocked by the NAT).
SIP User's Manual 9.2.2 9. Networking Capabilities First Incoming Packet Mechanism If the remote gateway resides behind a NAT device, it’s possible that the MediaPack can activate the RTP/RTCP/T.38 streams to an invalid IP address / UDP port. To avoid such cases, the MediaPack automatically compares the source address of the incoming RTP/RTCP/T.38 stream with the IP address and UDP port of the remote gateway.
MediaPack 9.4 Point-to-Point Protocol over Ethernet (PPPoE) PPPoE is a method of sending the Point-to-Point Protocol over Ethernet network. 9.4.1 Point-to-Point Protocol (PPP) Overview Point-to-Point Protocol (PPP) provides a method of transmitting data over serial point-topoint links. The protocol defines establishing, configuring and testing the data link connection and the network protocol.
SIP User's Manual 9.4.2 9. Networking Capabilities PPPoE Overview PPPoE is a method of sending the Point-to-Point Protocol over Ethernet network. PPPoE provides the ability to connect a network of hosts over a simple bridging access device to a remote Access Concentrator. Access control, billing and type of service can be done on a per-user, rather than a per-site, basis.
MediaPack When working in a PPPoE environment, the gateway negotiates for its IP address (as described above). However, if the user desires to disable the PPPoE client, the gateway can be configured to use default values for IP address, subnet mask and default gateway. This can be done using ini file parameters PPPoERecoverIPAddress, PPPoERecoverSubnetMask and PPPoERecoverDfgwAddress.
SIP User's Manual 9.7 9. Networking Capabilities Simple Network Time Protocol Support Simple Network Time Protocol (SNTP) client functionality generates requests and reacts to the resulting responses using the NTP version 3 protocol definitions (according to RFC 1305). Through these requests and responses, the NTP client is able to synchronize the system time to a time source within the network, thereby eliminating any potential issues should the local system clock 'drift' during operation.
MediaPack 9.9 VLANS and Multiple IPs 9.9.1 Multiple IPs Media, Control, and Management (OAM) traffic in the gateway can be assigned one of the following IP addressing schemes: Single IP address for all traffic (i.e., Media, Control, and OAM). Separate IP address for each traffic type. For separate IP addresses, the different traffic types are separated into three dedicated networks.
SIP User's Manual 9. Networking Capabilities For the mapping of an application to its class-of-service and traffic type, refer to Table 9-1 below. Media traffic type is assigned ‘Premium media’ class of service, Management traffic type is assigned ‘Bronze’ class of service, and Control traffic type is assigned ‘Premium control’ class of service.
MediaPack 9.9.2.1 Operation Outgoing packets (from the gateway to the switch): All outgoing packets are tagged, each according to its interface (control, media or OAM). If the gateway’s native ID is identical to one of the other IDs (usually to the OAM ID), this ID (e.g., OAM) is set to zero on outgoing packets (VlanSendNonTaggedOnNative = 0). This method is called Priority Tagging (p tag without Q tag).
SIP User's Manual 9.9.3.1 9. Networking Capabilities Integrating Using the Embedded Web Server ¾ To integrate the MediaPack into a VLAN and multiple IPs network using the Embedded Web Server, take these 7 steps: 1. 2. Access the Embedded Web Server (Section 5.3 on page 51). Use the Software Upgrade Wizard (Section 5.9.1 on page 197) to load and burn the firmware version to the MediaPack (VLANs and multiple IPs support is available only when the firmware is burned to flash).
MediaPack Figure 9-3: Example of the IP Settings Screen 5. • Click the Submit button to save your changes. Configure the IP Routing table by completing the following steps (the IP Routing table is required to define static routing rules for the OAM and Control networks since a default gateway isn’t supported for these networks): • Open the ‘IP Routing Table’ screen (Advanced Configuration menu > Network Settings > IP Routing Table option); the ‘IP Routing Table’ screen is displayed.
SIP User's Manual 9.9.3.2 9. Networking Capabilities Integrating Using the ini File ¾ To integrate the MediaPack into a VLAN and multiple IPs network using the ini file, take these 3 steps: 1.
MediaPack Reader's Notes SIP User's Manual 260 Document #: LTRT-65408
SIP User's Manual 10. Advanced System Capabilities 10 Advanced System Capabilities 10.1 Restoring Networking Parameters to their Initial State You can use the ‘Reset’ button to restore the MediaPack networking parameters to their factory default values (described in Table 4-1) and to reset the username and password. Note that the MediaPack returns to the software version burned in flash. This process also restores the MediaPack parameters to their factory settings.
MediaPack 10.2 Establishing a Serial Communications Link with the MediaPack Use serial communication software (e.g., HyperTerminalTM) to establish a serial communications link with the MediaPack via the RS-232 connection. You can use this link to change the networking parameters (Section 4.2.4 on page 44) and to receive error / notification messages. ¾ To establish a serial communications link with the MediaPack via the RS-232 port, take these 2 steps: 1.
SIP User's Manual 10.3 10. Advanced System Capabilities Automatic Update Mechanism The MediaPack is capable of automatically updating its cmp, ini and configuration files. These files can be stored on any standard Web, FTP or NFS server/s and can be loaded periodically to the gateway via HTTP, HTTPS, FTP or NFS. This mechanism can be used even for Customer Premise(s) Equipment (CPE) devices that are installed behind NAT and firewalls. The Automatic Update mechanism is applied separately to each file.
MediaPack The following example illustrates how to utilize Automatic Updates for deploying devices with minimum manual configuration. ¾ To utilize Automatic Updates for deploying the MediaPack with minimum manual configuration, take these 5 steps: 1. Set up a Web server (in the following example it is http://www.corp.com/) where all configuration files are to be stored. To each device, pre-configure the following parameter (DHCP / DNS are assumed): IniFileURL = 'http://www.corp.com/master_configuration.
SIP User's Manual 10.4 10. Advanced System Capabilities Startup Process The startup process (illustrated in Figure 10-3 on page 266) begins when the gateway is reset (physically or from the Web / SNMP) and ends when the operational software is running. In the startup process, the network parameters, software and configuration files are obtained. After the gateway powers up or after it is physically reset, it broadcasts a BootRequest message to the network.
MediaPack Figure 10-3: MediaPack Startup Process Reset from the Web Interface or SNMP Physical Reset BootP x times No Response DHCP x times No Response Update network BootP Response parameters from DHCP Response BootP/DHCP reply BootP/DHCP reply contains firmware file name? No Yes Download firmware via TFTP BootP/DHCP reply contains ini file name? BootP/DHCP reply contains ini file name? No Preconfigured firmware URL? Yes Yes Download firmware via TFTP Yes No Device Reset No Preconfig
SIP User's Manual 10.5 10. Advanced System Capabilities Using Parameter Tables The MediaPack uses parameter tables to group related parameters of specific entities and manage them together. These tables, similar to regular parameters, can be configured via the ini file, Embedded Web Server, SNMP, etc. Tables are composed of lines and columns. Columns represent parameters’ types. Lines represent specific entities. The instances in each line are called line attributes.
MediaPack 10.5.2 Table Permissions Each column has a 'permission' attribute that is applied to all instances in the column. This permission determines if and when a field can be modified. Several permissions can be applied to each column. The following permissions are available: Read: The value of the field can be read. Write: The value of the field can be modified.
SIP User's Manual 10. Advanced System Capabilities 10.5.5 Using the ini File to Configure Parameter Tables You can use the ini file to add / modify parameter tables. When using tables, Read-Only parameters are not loaded, as they cause an error when trying to reload the loaded file. Therefore, Read-Only parameters mustn’t be included in tables in the ini file. Consequently, tables are loaded with all parameters having at least one of the following permissions: Write, Create or Maintenance Write.
MediaPack Refer to the following notes: Indices (in both the Format and the Data lines) must appear in the same order determined by the specific table's documentation. The Index field must never be omitted. The Format line can include a sub-set of the configurable fields in a table. In this case, all other fields are assigned with the pre-defined default values for each configured line. The order of the fields in the Format line isn’t significant (as opposed to the Indexfields).
SIP User's Manual 10.6 10. Advanced System Capabilities Customizing the MediaPack Web Interface Customers incorporating the MediaPack into their portfolios can customize the Web Interface to suit their specific corporate logo and product naming conventions. Customers can customize the Web Interface’s title bar (AudioCodes’ title bar is shown in Figure 10-5; a customized title bar is shown in Figure 10-7).
MediaPack 10.6.1.1 Replacing the Main Corporate Logo with an Image File Note: Use a gif, jpg or jpeg file for the logo image. It is important that the image file has a fixed height of 59 pixels (the width can be configured up to a maximum of 339 pixels). The size of the image files (logo and background) is limited each to 64 kbytes. ¾ To replace the default logo with your own corporate image via the Web Interface, take these 7 steps: 1. 2. Access the MediaPack Embedded Web Server (refer to Section 5.
SIP User's Manual 10. Advanced System Capabilities ¾ To replace the default logo with your own corporate image via the ini file, take these 2 steps: 1. Place your corporate logo image file in the same folder as where the device’s ini file is located (i.e., the same location defined in the BootP/TFTP configuration utility). For detailed information on the BootP/TFTP, refer to Appendix C on page 349. Add/modify the two ini file parameters in Table 10-3 according to the procedure described in Section 6.
MediaPack 10.6.2 Replacing the Background Image File The background image file is duplicated across the width of the screen. The number of times the image is duplicated depends on the width of the background image and screen resolution. When choosing your background image, keep this in mind. Use a gif, jpg or jpeg file for the background image. It is important that the image file has a fixed height of 59 pixels. The size of the image files (logo and background) is limited each to 64 kbytes.
SIP User's Manual 10. Advanced System Capabilities 10.6.3 Customizing the Product Name The Product Name text string can be modified according to OEMs specific requirements. To replace AudioCodes’ default product name with a text string via the Web Interface, modify the two ini file parameters in Table 10-6 according to the procedure described in Section 10.6.4 on page 276.
MediaPack 10.6.4 Modifying ini File Parameters via the Web AdminPage ¾ To modify ini file parameters via the AdminPage, take these 6 steps: 1. 2. Access the MediaPack Embedded Web Server (refer to Section 5.3 on page 51). In the URL field, append the suffix ‘AdminPage’ (note that it’s case-sensitive) to the IP address, e.g., http://10.1.229.17/AdminPage. Click the INI Parameters option, the INI Parameters screen is displayed (shown in Figure 10-8). 3. Figure 10-8: INI Parameters Screen 4. 5. 6.
SIP User's Manual 11 11. Special Applications - Metering Tones Relay Special Applications - Metering Tones Relay The MediaPack FXS and FXO gateways can be used to relay standard 12 or 16 kHz metering tones over the IP network as illustrated in Figure 11-1 below. Figure 11-1: Metering Tone Relay Architecture After a call is established between the FXS and FXO gateways, the PSTN generates 12 or 16 kHz metering tones towards the FXO gateway.
MediaPack Reader's Notes SIP User's Manual 278 Document #: LTRT-65408
SIP User's Manual 12 12. Security Security This section describes the security mechanisms and protocols implemented on the MediaPack. The following list specifies the available security protocols and their objectives: 12.1 IPSec and IKE protocols are part of the IETF standards for establishing a secured IP connection between two applications. IPSec and IKE are used in conjunction to provide security for control and management protocols but not for media (refer to Section 12.1 below).
MediaPack 12.1.1 IKE IKE is used to obtain the Security Associations (SA) between peers (the gateway and the application it’s trying to contact). The SA contains the encryption keys and profile used by the IPSec to encrypt the IP stream. The IKE table lists the IKE peers with which the gateway performs the IKE negotiation (up to 20 peers are available). The IKE negotiation is separated into two phases: main mode and quick mode.
SIP User's Manual 12. Security IPSec specifications include the following: Transport mode only Encapsulation Security Payload (ESP) only Support for Cipher Block Chaining (CBC) Supported IPSec SA encryption algorithms - DES, 3DES, and AES Hash types for IPSec SA are SHA1 and MD5 12.1.3 Configuring the IPSec and IKE To enable IPSec and IKE on the gateway set the ini file parameter ‘EnableIPSec’ to 1. 12.1.3.
MediaPack Table 12-1: IKE Table Configuration Parameters (continues on pages 281 on 282) Parameter Name Description Authentication Method Determines the authentication method for IKE. [IkePolicyAuthenticationMeth The valid authentication method values include: od] 0 = Pre-shared Key (default) 1 = RSA Signiture Note 1: For pre-shared key based authentication, peers participating in an IKE exchange must have a prior (out-of-band) knowledge of the common key (see IKEPolicySharedKey parameter).
SIP User's Manual 12.
MediaPack In the ‘Policy Index’ drop-down list, select the peer you want to edit (up to 20 peers can be configured). 4. Configure the IKE parameters according to Table 12-1 on page 281. 5. Click the button Create; a row is create in the IKE table 6. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. To delete a peer from the IKE table, select it in the ‘Policy Index’ drop-down list, click the button Delete and click OK in the prompt. 3. 12.1.3.
SIP User's Manual 12. Security Table 12-3: SPD Table Configuration Parameters (continues on pages 284 to 285) Parameter Name Description First to Fourth Proposal Encryption Type [IPSecPolicyProposalEncrypt ion_X] Determines the encryption type used in the quick mode negotiation for up to four proposals. X stands for the proposal number (0 to 3).
MediaPack In the SPD example, all packets designated to IP address 10.11.2.21 that originates from the OAM interface (regardless to their destination and source ports) and whose protocol is UDP are encrypted, the SPD also defines an SA lifetime of 900 seconds and two security proposals: DES/SHA1 and 3DES/SHA1. ¾ To configure the SPD table using the Embedded Web Server, take these 6 steps: 1. 2. Access the Embedded Web Server (refer to Section 5.3 on page 51).
SIP User's Manual 12. Security 12.1.3.3 IPSec and IKE Configuration Table’s Confidentiality Since the pre-shared key parameter of the IKE table must remain undisclosed, measures are taken by the ini file, Embedded Web Server and SNMP agent to maintain this parameter’s confidentiality. On the Embedded Web Server a list of asterisks is displayed instead of the pre-shared key. On SNMP, the pre-shared key parameter is a write-only parameter and cannot be read.
MediaPack 12.2 SSL/TLS SSL, also known as TLS, is the method used to secure the MediaPack SIP Signaling connections, Embedded Web Server and Telnet server. The SSL protocol provides confidentiality, integrity and authenticity between two communicating applications over TCP/IP. Specifications for the SSL/TLS implementation: Supports transports: SSL 2.0, SSL 3.0, TLS 1.0 Supports ciphers: DES, RC4 compatible Authentication: X.509 certificates; CRLs are not supported 12.2.
SIP User's Manual 2. 3. 12. Security If you are using Internet Explorer, click View Certificate and then Install Certificate. The browser also warns you if the host name used in the URL is not identical to the one listed in the certificate. To solve this, add the IP address and host name (ACL_nnnnnn where nnnnnn is the serial number of the MediaPack) to your hosts file, located at /etc/hosts on UNIX or C:\Windows\System32\Drivers\ETC\hosts on Windows; then use the host name in the URL (e.g.
MediaPack 12.2.4 Server Certificate Replacement The MediaPack is supplied with a working SSL configuration consisting of a unique selfsigned server certificate. If an organizational Public Key Infrastructure (PKI) is used, you may wish to replace this certificate with one provided by your security administrator. ¾ To replace the MediaPack self-signed certificate, take these 9 steps: 1. Your network administrator should allocate a unique DNS name for the MediaPack (e.g., dns_name.corp.customer.com).
SIP User's Manual 3. 4. 5. 12. Security In the Subject Name field, enter the DNS name and click Generate CSR. A textual certificate signing request, that contains the SSL device identifier, is displayed. Copy this text and send it to your security provider; the security provider (also known as Certification Authority or CA) signs this request and send you a server certificate for the device. Save the certificate in a file (e.g., cert.txt).
MediaPack 12.2.5 Client Certificates By default, Web servers using SSL provide one-way authentication. The client is certain that the information provided by the Web server is authentic. When an organizational PKI is used, two-way authentication may be desired: both client and server should be authenticated using X.509 certificates. This is achieved by installing a client certificate on the managing PC, and loading the same certificate (in base64-encoded X.
SIP User's Manual 12.3 12. Security SRTP The gateway supports Secured RTP (SRTP) according to RFC 3711. SRTP is used to encrypt RTP and RTCP transport since it is best-suited for protecting VoIP traffic. SRTP requires a Key Exchange mechanism that is performed according to . The Key Exchange is executed by adding a ‘Crypto’ attribute to the SDP. This attribute is used (by both sides) to declare the various supported cipher suites and to attach the encryption key to use.
MediaPack 12.4 RADIUS Login Authentication Users can enhance the security and capabilities of logging to the gateway’s Web and Telnet embedded servers by using a Remote Authentication Dial-In User Service (RADIUS) to store numerous usernames, passwords and access level attributes (Web only), allowing multiple user management on a centralized platform.
SIP User's Manual 2. 12. Security If access levels are required, set up a VSA dictionary for the RADIUS server and select an attribute ID that represents each user's access level. The following example shows a dictionary file for FreeRADIUS that defines the attribute ‘ACL-Auth-Level’ with ID=35.
MediaPack 6. Under section ‘RADIUS Authentication Settings’, in the field ‘Device Behavior Upon RADIUS Timeout’, select the gateway’s operation if a response isn’t received from the RADIUS server after the 5 seconds timeout expires: • 7. 8. Deny Access – the gateway denies access to the Web and Telnet embedded servers. • Verify Access Locally – the gateway checks the local username and password.
SIP User's Manual 12.5 12. Security ¾ To configure RADIUS support on the gateway using the ini file: Add the following parameters to the ini file. For information on modifying the ini file, refer to Section 6.2 on page 209.
MediaPack Figure 12-15 shows an example of an access list definition via ini file: Figure 12-15: Example of an Access List Definition via ini File [ ACCESSLIST ] FORMAT AccessList_Index = AccessList_Source_IP, AccessList_Net_Mask, AccessList_Start_Port, AccessList_End_Port, AccessList_Protocol, AccessList_Packet_Size, AccessList_Byte_Rate, AccessList_Byte_Burst, AccessList_Allow_Type; AccessList 10 AccessList 15 AccessList 20 AccessList 22 [ \ACCESSLIST = = = = ] mgmt.customer.com, 255.255.255.
SIP User's Manual 12.6 12. Security Network Port Usage The following table lists the default TCP/UDP network port numbers used by the MediaPack. Where relevant, the table lists the ini file parameters that control the port usage and provide source IP address filtering capabilities. Table 12-5: Default TCP/UDP Network Port Numbers Port Number Peer Port Application Notes 2 2 Debugging interface Always ignored 23 - Telnet Disabled by default (TelnetServerEnable).
MediaPack 12.7 Recommended Practices To improve network security, the following guidelines are recommended when configuring the MediaPack: 12.8 Set the password of the primary web user account (refer to 5.6.5.1 on page 168) to a unique, hard-to-hack string. Do not use the same password for several devices as a single compromise may lead to others. Keep this password safe at all times and change it frequently. If possible, use a RADIUS server for authentication.
SIP User's Manual 13 13. Diagnostics Diagnostics Several diagnostic tools are provided, enabling you to identify correct functioning of the MediaPack, or an error condition with a probable cause and a solution or workaround. 13.1 Front and rear-panel LEDs on the MediaPack. The MP-11x front-panel LEDs are described in Table 2-1 on page 26. The MP-124 front-panel LEDs are described in Table 2-4 on page 27, the rear-panel LEDs are described in Table 2-6 on page 28.
MediaPack Coefficients checksum. Message waiting indication status. Ring state. Reversal polarity state. Line current (only on port 0). Line voltage between TIP and RING (only on port 0). 3.3 V reading (only on port 0). Ring voltage (only on port 0). Long line current (only on port 0). The following line tests are available on FXO gateways: Hardware revision number. Hook state. Reversal polarity state.
SIP User's Manual 13.3 13. Diagnostics Syslog Support Syslog protocol is an event notification protocol that enables a machine to send event notification messages across IP networks to event message collectors -also known as Syslog servers. Syslog protocol is defined in the IETF RFC 3164 standard. Since each process, application and operating system was written independently, there is little uniformity to Syslog messages.
MediaPack 13.3.2 Operation The Syslog client, embedded in the MediaPack, sends error reports/events generated by the MediaPack unit application to a Syslog server using IP/UDP protocol. ¾ To activate the Syslog client on the MediaPack, take these 5 steps: 1. 2. Set the parameter ‘EnableSyslog’ to 1 (refer to Table 5-51 on page 177). Use the parameter ‘SyslogServerIP’ to define the IP address of the Syslog server you use (refer to Table 5-51 on page 177).
SIP User's Manual 14 14. SNMP-Based Management SNMP-Based Management Simple Network Management Protocol (SNMP) is a standard-based network control protocol used to manage elements in a network. The SNMP Manager (usually implemented by a Network Manager (NM) or an Element Manager (EM)) connects to an SNMP Agent (embedded on a remote Network Element (NE)) to perform network element Operation, Administration and Maintenance (OAM).
MediaPack Set Request: The SNMP standard provides a method of effecting an action associated with a device (via the ‘set’ request) to accomplish activities such as disabling interfaces, disconnecting users, clearing registers, etc. This provides a way of configuring and controlling network devices via SNMP.
SIP User's Manual 14. SNMP-Based Management Typically, when a MIB is compiled into the system, the manager creates new folders or directories that correspond to the objects. These folders or directories can typically be viewed with a MIB Browser, which is a traditional SNMP management tool incorporated into virtually all Network Management Systems.
MediaPack 14.3 Cold Start Trap MediaPack technology supports a cold start trap to indicate that the device is starting. This allows the manager to synchronize its view of the device's active alarms. Two different traps are sent at start-up: 14.4 The standard coldStart trap - iso(1).org(3).dod(6).internet(1). snmpV2(6). snmpModules(3). snmpMIB(1). snmpMIBObjects(1). snmpTraps(5). coldStart(1) - sent at system initialization.
SIP User's Manual 14. SNMP-Based Management The MIBs include: • acPMMedia: for media (voice) related monitoring (e.g., RTP, DSP’s). • acPMControl: for Control-Protocol related monitoring (e.g., connections, commands). • acPMAnalog: for analog channels in offhook state. • acPMSystem: for general (system related) monitoring. The log trap, acPerformanceMonitoringThresholdCrossing (non-alarm), is sent out every time the threshold of a Performance Monitored object is crossed.
MediaPack RTCP-XR: This MIB (RFC) implements the following partial support: • The rtcpXrCallQualityTable is fully supported. • In the rtcpXrHistoryTable, support of the RCQ objects is provided only with no more than 3 intervals, 15 minutes long each. • Supports the rtcpXrVoipThresholdViolation trap. In addition to the standard MIBs, the complete product series contains proprietary MIBs: AC-TYPES MIB: lists the known types defined by the complete product series.
SIP User's Manual Note 1: Note 2: 14. SNMP-Based Management The following are special notes pertaining to MIBs: • A detailed explanation of each parameter can be viewed in an SNMP browser in the ‘MIB Description’ field. • Not all groups in the MIB are functional. Refer to version release notes. • Certain parameters are non-functional. Their MIB status is marked 'obsolete'.
MediaPack 14.7 Traps Note: As of this version all traps are sent from the SNMP port (default 161). This is part of the NAT traversal solution. Full proprietary trap definitions and trap Varbinds are found in the acBoard and acAlarm MIBs. The following proprietary traps are supported. For detailed information on these traps, refer to Appendix F on page 375: Table 14-1: Proprietary Traps Description Trap Description acBoardFatalError Sent whenever a fatal device error occurs.
SIP User's Manual 14. SNMP-Based Management In addition to the listed traps, the device also supports the following standard traps: 14.8 coldStart authenticationFailure linkDown linkup entConfigChange SNMP Interface Details This section describes details of the SNMP interface that is required when developing an Element Manager (EM) for any of the TrunkPack-VoP Series products, or to manage a device with a MIB browser.
MediaPack 14.8.1.3 Configuration of Community Strings via SNMP To configure read-only and read-write community strings, the EM must use the SNMPCOMMUNITY-MIB. To configure the trap community string, the EM must also use the snmpVacmMIB and the snmpTargetMIB. ¾ To add a read-only community string (v2user), take this step: Add a new row to the srCommunityTable with CommunityName v2user and GroupName ReadGroup. ¾ To delete the read-only community string (v2user), take these 2 steps: 1. 2.
SIP User's Manual 14. SNMP-Based Management 14.8.2 SNMP v3 USM Users You can define up to 10 User-based Security Model (USM) users (USM users are referred to as “v3 users”). Each v3 user can be associated with an authentication type (none, MD5, or SHA-1) and a privacy type (none, DES, 3DES, or AES).
MediaPack 14.8.2.1 Configuring SNMP v3 Users via the ini File Use the SNMPUsers ini table to add, modify, and delete SNMPv3 users. For a description of the SNMPUsers table ini file parameters, refer to Section 5.6.6.3 on page 181. Note: The SNMPUsers ini table is a hidden parameter. Therefore, when you perform a “Get ini File” operation using the Web interface, the table will not be included in the generated file.
SIP User's Manual 14. SNMP-Based Management ¾ To delete the read-only, noAuthNoPriv SNMPv3 user (v3user), take these 3 steps: 1. If v3 user is associated with a trap destination, follow the procedure for associating a different user to that trap destination. (See below.) Delete the vacmSecurityToGroupTable row for SecurityName v3user, GroupName ReadGroup1, and SecurityModel usm. Delete the row in the usmUserTable for v3user. 2. 3.
MediaPack 14.8.3.1 Configuration of Trusted Managers via ini File To set the Trusted Mangers table from start-up, write the following in the ini file: SNMPTRUSTEDMGR_X = D.D.D.D where X is any integer between 0 and 4 (0 sets the first table entry, 1 sets the second, and so on), and D is an integer between 0 and 255. 14.8.3.2 Configuration of Trusted Managers via SNMP To configure Trusted Managers, the EM must use the SNMP-COMMUNITY-MIB, the snmpTargetMIB and the TGT-ADDRESS-MASK-MIB.
SIP User's Manual 14. SNMP-Based Management The following procedure assumes that there is at least one configured read-write community. There is currently only one Trusted Manager. The taglist for columns for all rows in the srCommunityTable are currently set to MGR. This procedure must be performed from the final Trusted Manager. ¾ To delete the final Trusted Manager, take these 2 steps: Set the value of the TransportLabel field on each row in the srCommunityTable to the empty string. 2.
MediaPack 14.8.5.2 Configuring Trap Managers via the ini File In the MediaPack ini file, the parameters below can be set to enable or disable the sending of SNMP traps. Multiple trap destinations can be supported on the device by setting multiple trap destinations in the ini file. SNMPManagerTrapSendingEnable_: indicates whether or not traps are to be sent to the specified SNMP trap manager. A value of ‘1’ means that it is enabled, while a value of ‘0’ means disabled.
SIP User's Manual 14. SNMP-Based Management 14.8.5.3 Configuring Trap Managers via SNMP The standard snmpTargetMIB interface is available for configuring trap managers. Note: The acBoard MIB is planned to become obsolete. The only relevant section in this MIB is the trap sub tree acTrap. ¾ To add an SNMPv2 trap destination, take the following step: Add a row to the snmpTargetAddrTable with these values: Name=trapN, TagList=AC_TRAP, Params=v2cparams, where N is an unused number between 0 and 4.
MediaPack 14.9 SNMP Manager Backward Compatibility With support for the Multi Manager Trapping feature, the older acSNMPManagerIP MIB object, synchronized with the first index in the snmpManagers MIB table, is also supported. This is translated in two features: SET/GET to either of the two MIB objects is identical. i.e., as far as the SET/GET are concerned OID 1.3.6.1.4.1.5003.9.10.1.1.2.7 is identical to OID 1.3.6.1.4.1.5003.9.10.1.1.2.21.1.1.3.
SIP User's Manual 14. SNMP-Based Management 14.11 SNMP Administrative State Control 14.11.1 Node Maintenance Node maintenance for the MediaPack is provided by an SNMP interface. The acBoardMIB provides two parameters for graceful and forced shutdowns of the MediaPack: acgwAdminState acgwAdminStateLockControl The acgwAdminState is used either to request (set) a shutdown (0), undo shutdown (2), or to view (get) the gateway condition (0 = locked; 1 = shutting down; 2 = unlocked).
MediaPack 14.12 AudioCodes’ Element Management System Using AudioCodes’ Element Management System (EMS) is recommended to Customers requiring large deployments (multiple media gateways in globally distributed enterprise offices, for example), that need to be managed by central personnel. The EMS is not included in the device’s supplied package. Contact AudioCodes for detailed information on AudioCodes’ EMS and on AudioCodes’ EVN - Enterprise VoIP Network – solution for large VoIP deployments.
SIP User's Manual 15 15. Configuration Files Configuration Files This section describes the configuration dat files that are loaded (in addition to the ini file) to the gateway. The configuration files are: Call Progress Tones file (refer to Section 15.1 on page 325). Prerecorded Tones file (refer to Section 15.2 on page 330). FXS Coefficient file (refer to Section 15.3 on page 331). User Information file (refer to Section 15.4 on page 332).
MediaPack Users 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, users 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 MediaPack reports dial tone detection if either of the two tones is detected.
SIP User's Manual 15. Configuration Files • Third Signal Off Time [10 msec]: ‘Signal Off’ period (in 10 msec units) for the third cadence ON-OFF cycle. Can be omitted if there isn’t a third cadence. • Forth Signal On Time [10 msec]: ‘Signal On’ period (in 10 msec units) for the fourth cadence ON-OFF cycle. Can be omitted if there isn’t a fourth cadence. • Forth Signal Off Time [10 msec]: ‘Signal Off’ period (in 10 msec units) for the fourth cadence ON-OFF cycle.
MediaPack 15.1.2 Format of the Distinctive Ringing Section in the ini File Distinctive Ringing is only applicable to MediaPack/FXS gateways. Using the distinctive ringing section of this configuration file, the user can create up to 16 distinctive ringing patterns. Each ringing pattern configures the ringing tone frequency and up to 4 ringing cadences. The same ringing frequency is used for all the ringing pattern cadences.
SIP User's Manual 15. Configuration Files 15.1.2.
MediaPack 15.2 Prerecorded Tones (PRT) File The Call Progress Tones mechanism has several limitations, such as a limited number of predefined tones and a limited number of frequency integrations in one tone. To work around these limitations and provide tone generation capability that is more flexible, the PRT file can be used. If a specific prerecorded tone exists in the PRT file, it takes precedence over the same tone that exists in the CPT file and is played instead of it.
SIP User's Manual 15.3 15. Configuration Files The Coefficient Configuration File The Coeff_FXS.dat file is used to provide best termination and transmission quality adaptation for different line types for FXS gateways. This adaptation is performed by modifying the telephony interface characteristics (such as DC and AC impedance, feeding current and ringing voltage). The coeff.
MediaPack 15.4 User Information File The User Information file maps PBX extensions (connected to the MediaPack gateway) to global IP phone numbers (alphanumerical). In this context, a global IP number serves as a routing identifier for calls in the ‘IP World’. The PBX extension uses this mapping to emulate the behavior of an IP phone. Note that the mapping mechanism is disabled by default and must be activated using the parameter ‘EnableUserInfoUsage’ (described in Section 5.5.2.1).
SIP User's Manual 16. Selected Technical Specifications 16 Selected Technical Specifications 16.1 MP-11x Specifications Table 16-1: MP-11x Functional Specifications (continues on pages 333 to 335) Channel Capacity Available Ports MP-112 2 ports* MP-114 4 ports MP-118 8 ports * The MP-112 differs from the MP-114 and MP-118. Its configuration excludes the RS-232 connector, the Lifeline option and outdoor protection.
MediaPack Table 16-1: MP-11x Functional Specifications (continues on pages 333 to 335) Voice & Tone Characteristics Voice Compression Silence Suppression G.711 PCM at 64 kbps µ-law/A-law msec) G.723.1 MP-MLQ at 5.3 or 6.3 kbps G.726 at 32 kbps ADPCM msec) G.729 CS-ACELP 8 Kbps Annex A / B (10, 20, 30, 40, 50, 60, 80, 100, 120 (30, 60, 90 msec) (10, 20, 30, 40, 50, 60, 80, 100, 120 (10, 20, 30, 40, 50, 60 msec) G.723.1 Annex A G.
SIP User's Manual 16. Selected Technical Specifications Table 16-1: MP-11x Functional Specifications (continues on pages 333 to 335) Connectors & Switches Rear Panel 8 Analog Lines (MP-118) 8 RJ-11 connectors 4 Analog Lines (MP-114) 4 RJ-11 connectors 2 Analog Lines (MP-112) 2 RJ-11 connectors AC power supply socket 100-240~0.3A max. Ethernet 10/100 Base-TX, RJ-45 RS-232 Console PS/2 port Reset Button Resets the MP-11x Physical Dimensions (HxWxD) 42 x 172 x 220 mm Weight 0.5 kg (Approx.
MediaPack 16.2 MP-124 Specifications Table 16-2: MP-124 Functional Specifications (continues on pages 336 to 338) Channel Capacity Available Ports MP-124 24 ports FXS Functionality FXS Capabilities Short or Long Haul (Automatic Detection): REN2: Up to 15.5 km (50,800 feet) using 24 AWG line. REN3: Up to 9 km (30,000 feet) using 24 AWG line. REN5: Up to 5.5 km (18,000 feet) using 24 AWG line.
SIP User's Manual 16. Selected Technical Specifications Table 16-2: MP-124 Functional Specifications (continues on pages 336 to 338) Call Progress Tone 32 tones: Detection and Generation single tone, dual tones or AM tones, programmable frequency & amplitude; 64 frequencies in the range 300 to 1980 Hz, 1 to 4 cadences per tone, up to 4 sets of ON/OFF periods.
MediaPack Table 16-2: MP-124 Functional Specifications (continues on pages 336 to 338) Type Approvals Safety and EMC UL 60950-1, FCC part 15 Class B CE Mark EN 60950-1, EN 55022, EN 55024, EN61000-3-2, EN61000-3-3, EN55024.
SIP User's Manual A A. MediaPack SIP Software Kit MediaPack SIP Software Kit Table A-1 describes the standard supplied software kit for MediaPack FXS/FXO SIP gateways. The supplied documentation includes this User’s Manual, the MP-11x & MP-124 MGCP-H.323-SIP Fast Track Guide, and the MP-11x & MP-124 SIP Release Notes. Table A-1: MediaPack SIP Supplied Software Kit File Name Description Ram.cmp files MP124_SIP_xxx.cmp Image file containing the software for the MP-124/FXS gateway. MP118_SIP_xxx.
MediaPack Reader’s Notes SIP User's Manual 340 Document #: LTRT-65408
SIP User's Manual B B. SIP Compliance Tables SIP Compliance Tables The MediaPack gateways comply with RFC 3261, as shown in the following sections. B.
MediaPack B.
SIP User's Manual B. SIP Compliance Tables Table B-3: SIP Headers (continues on pages 342 to 343) Header Field Supported Response- Key Yes Retry- After Yes Route Yes Rseq Yes Session-Expires Yes Server Yes SIP-If-Match Yes Subject Yes Supported Yes Timestamp Yes To Yes Unsupported Yes User- Agent Yes Via Yes Voicemail Yes Warning Yes WWW- Authenticate Yes B.
MediaPack B.5 SIP Responses The following SIP responses are supported by the gateway: B.5.
SIP User's Manual B.5.3 B. SIP Compliance Tables 3xx Response – Redirection Responses Table B-7: 3xx SIP Responses 3xx Response Supported Comments 300 Multiple Choice Yes The gateway responds with an ACK and resends the request to first in the contact list, new address. 301 Moved Permanently Yes The gateway responds with an ACK and resends the request to new address.
MediaPack Table B-8: 4xx SIP Responses (continues on pages 345 to 346) 4xx Response Supported Comments 409 Conflict Yes The gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call. 410 Gone Yes The gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.
SIP User's Manual B.5.5 B. SIP Compliance Tables 5xx Response – Server Failure Responses Table B-9: 5xx SIP Responses 5xx Response 500 Internal Server Error 501 Not Implemented 502 Bad gateway 503 Service Unavailable 504 Gateway Timeout 505 Version Not Supported B.5.6 Comments On reception of any of these Responses, the GW releases the call, sending appropriate release cause to PSTN side. The GW generates 5xx response according to PSTN release cause coming from PSTN.
MediaPack Reader's Notes SIP User's Manual 348 Document #: LTRT-65408
SIP User's Manual C C. BootP/TFTP Configuration Utility BootP/TFTP Configuration Utility The BootP/TFTP utility enables you to easily configure and provision our boards and media gateways. Similar to third-party BootP/TFTP utilities (which are also supported) but with added functionality; our BootP/TFTP utility can be installed on Windows™ 98 or Windows™ NT/2000/XP. The BootP/TFTP utility enables remote reset of the device to trigger the initialization procedure (BootP and TFTP).
MediaPack C.4 C.5 C.
SIP User's Manual C.7 C. BootP/TFTP Configuration Utility BootP/TFTP Application User Interface Figure C-1 shows the main application screen for the BootP/TFTP utility. Figure C-1: Main Screen Log Window C.8 Function Buttons on the Main Screen Pause: Click this button to pause the BootP Tool so that no replies are sent to BootP requests. Click the button again to restart the BootP Tool so that it responds to all BootP requests. The Pause button provides a depressed graphic when the feature is active.
MediaPack Figure C-2: Reset Screen When a gateway resets, it first sends a BootRequest. Therefore, Reset can be used to force a BootP session with a gateway without needing to power cycle the gateway. As with any BootP session, the computer running the BootP Tool must be located on the same subnet as the controlled VoIP gateway. C.9 Log Window The log window (refer to Figure C-1 on the previous page) records all BootP request and BootP reply transactions, as well as TFTP transactions.
SIP User's Manual C. BootP/TFTP Configuration Utility Use right-click on a line in the Log Window to open a pop-up window with the following options: Reset: Selecting this option results in a reset command being sent to the client VoIP gateway. The program searches its database for the MAC address indicated in the line. If the client is found in that database, the program adds the client MAC address to the Address Resolution Protocol (ARP) table for the computer.
MediaPack When ARP Manipulation is enabled on this screen, the BootP Tool creates an ARP cache entry on your computer when it receives a BootP BootRequest from the VoIP gateway. Your computer uses this information to send messages to the VoIP gateway without using ARP again. This is particularly useful when the gateway does not yet have an IP address and, therefore, cannot respond to an ARP. Because this feature creates an entry in the computer ARP cache, Administrator Privileges are required.
SIP User's Manual C. BootP/TFTP Configuration Utility C.11 Configuring the BootP Clients The Clients window, shown in Figure C-4, is used to set up the parameters for each specific VoIP gateway. Figure C-4: Client Configuration Screen C.11.1 Adding Clients Adding a client creates an entry in the BootP Tool for a specific gateway. ¾ To add a client to the list without using a template, take these 3 steps: Click Add New Client ; a client with blank parameters is displayed. 2.
MediaPack ¾ To add a client to the list using a template, take these 5 steps: 1. Click Add New Client 2. 3. 4. 5. ; a client with blank parameters is displayed. In the field 'Template', located on the right side of the 'Client Configuration Window', click on the down arrow to the right of the entry field and select the template that you want to use. The values provided by the template are automatically entered into the parameter fields on the right side of the 'Client Configuration Window'.
SIP User's Manual C. BootP/TFTP Configuration Utility C.11.4 Testing the Client There should only be one BootP utility supporting any particular client MAC active on the network at any time. ¾ To check if other BootP utilities support this client, take these 4 steps: 1. Select the client that you wish to test by clicking the client name in the main area of the Client Configuration Window. 2. 3. . Click the Test Selected Client button Examine the Log Window on the Main Application Screen.
MediaPack Boot File: This field specifies the file name for the software (cmp) file that is loaded by the TFTP utility to the VoIP gateway after the VoIP gateway receives the BootReply message. The actual software file is located in the TFTP utility directory that is specified in the BootP Preferences window. The software file can be followed by command line switches. For information on available command line switches, refer to Section C.11.6 on page 359.
SIP User's Manual C. BootP/TFTP Configuration Utility C.11.6 Using Command Line Switches You can add command line switches in the field Boot File. ¾ To use a Command Line Switch, take these 4 steps: In the field Boot File, leave the file name defined in the field as it is (e.g., ramxxx.cmp). 2. Place your cursor after cmp. 3. Press the space bar. 4. Type in the switch you require. Example: ‘ramxxx.cmp –fb’ to burn flash memory. ‘ramxxx.
MediaPack C.12 Managing Client Templates Templates can be used to simplify configuration of clients when most of the parameters are the same. Figure C-5: Templates Screen ¾ To create a new template, take these 4 steps: 1. 2. 3. 4. Click the Add New Template button . Fill in the default parameter values in the parameter fields. Click Apply to save this new template. Click OK when you are finished adding templates. ¾ To edit an existing template, take these 4 steps: 1. 2. 3. 4.
SIP User's Manual D D. RTP/RTCP Payload Types and Port Allocation RTP/RTCP Payload Types and Port Allocation RTP Payload Types are defined in RFC 3550 and RFC 3551. We have added new payload types to enable advanced use of other coder types. These types are reportedly not used by other applications. D.1 Packet Types Defined in RFC 3551 Table D-1: Packet Types Defined in RFC 3551 D.2 Payload Type Description Basic Packet Rate [msec] 0 2 4 8 18 200 G.711 µ-Law G.726-32 G.723 (6.3/5.3 kbps) G.
MediaPack D.3 Default RTP/RTCP/T.38 Port Allocation The following table shows the default RTP/RTCP/T.38 port allocation. Table D-3: Default RTP/RTCP/T.38 Port Allocation Channel Number RTP Port RTCP Port T.
SIP User's Manual E E. Accessory Programs and Tools Accessory Programs and Tools The accessory applications and tools shipped with the device provide you with friendly interfaces that enhance device usability and smooth your transition to the new VoIP infrastructure. The following applications are available: E.1 TrunkPack Downloadable Conversion Utility (refer to Section E.1 below). Call Progress Tones Wizard (refer to Section E.1.3 on page 366).
MediaPack E.1.1 Converting a CPT ini File to a Binary dat File For detailed information on creating a CPT ini file, refer to Section 15.1 on page 325. ¾ To convert a CPT ini file to a binary dat file, take these 10 steps: 1. Execute the TrunkPack Downloadable Conversion Utility, DConvert.exe (supplied with the software package); the utility’s main screen opens (shown in Figure E-1). Click the Process Call Progress Tones File(s) button; the ‘Call Progress Tones’ screen, shown in Figure E-2, opens. 2.
SIP User's Manual E.1.2 E. Accessory Programs and Tools Encoding / Decoding an ini File For detailed information on secured ini file, refer to Section 6.1 on page 209. ¾ To encode an ini file, take these 6 steps: 1. Execute the TrunkPack Downloadable Conversion Utility, DConvert.exe (supplied with the software package); the utility’s main screen opens (shown in Figure E-1). Click the Process Encoded/Decoded ini file(s) button; the ‘Encode/Decode ini File(s)’ screen, shown in Figure E-3, opens. 2.
MediaPack E.1.3 Creating a Loadable Prerecorded Tones File For detailed information on the PRT file, refer to Section 15.2 on page 330. Note: The maximum size of a PRT file that can be loaded to the gateway is 100 KB. ¾ To create a loadable PRT dat file from your raw data files, take these 7 steps: 1. Prepare the prerecorded tones (raw data PCM or L8) files you want to combine into a single dat file using standard recording utilities. Execute the TrunkPack Downloadable Conversion utility, DConvert.
SIP User's Manual 4. E. Accessory Programs and Tools To add the prerecorded tone files (you created in Step 1) to the ‘Prerecorded Tones’ screen follow one of these procedures: • Select the files and drag them to the ‘Prerecorded Tones’ screen. • 5. Click the Add File(s) button; the ‘Select Files’ screen opens. Select the required Prerecorded Tone files and click the Add>> button. Close the ‘Select Files’ screen.
MediaPack E.2.2 Installation The CPTWizard can be installed on any Windows 2000 or Windows XP based PC. Windows-compliant networking and audio peripherals are required for full functionality. To install the CPTWizard, copy the files from the supplied installation kit to any folder on your PC. No further setup is required (approximately 5 MB of hard disk space are required). E.2.3 Initial Settings ¾ To start the CPTWizard, take these 5 steps: 1. Execute the CPTWizard.
SIP User's Manual E.2.4 E. Accessory Programs and Tools Recording Screen – Automatic Mode After the connection to the MediaPack/FXO gateway is established, the recording screen is displayed. Figure E-7: Recording Screen –Automatic Mode ¾ To start recording in automatic mode, take these 3 steps: 1. Click the Start Automatic Configuration button; the wizard starts the following Call Progress Tones detection sequence (the operation takes approximately 60 seconds to complete): Version 5.
MediaPack 2. The wizard then analyzes the recorded Call Progress Tones and displays a message specifying the tones that were detected (by the gateway) and analyzed (by the wizard) correctly. At the end of a successful detection operation, the detected Call Progress Tones are displayed in the Tones Analyzed pane (refer to Figure E-8). Figure E-8: Recording Screen after Automatic Detection 3.
SIP User's Manual E.2.5 E. Accessory Programs and Tools Recording Screen – Manual Mode In manual mode you can record and analyze tones, included in the Call Progress Tones ini file, in addition to those tones analyzed when in automatic mode. ¾ To start recording in manual mode, take these 6 steps: 1. Click the Manual tab at the top of the recording screen, the manual recording screen is displayed. Figure E-9: Recording Screen - Manual Mode 2. 3.
MediaPack E.2.6 The Call Progress Tones ini File After the Call Progress Tones detection is complete, a text file named call_progress_tones.ini is created in the same directory as the directory in which the CPTWizard.exe is located. This file contains: Information about each tone that was recorded and analyzed by the wizard. This information includes frequencies and cadence (on/off) times, and is required for using this file with the TrunkPack Downloadable Conversion utility.
SIP User's Manual E. Accessory Programs and Tools Information related to matches of all tones recorded with the CPTWizard’s internal database. The database is scanned to find one or more PBX definitions that match all recorded tones (i.e., dial tone, busy tone, ringing tone, reorder tone and any other manually-recorded tone – all match the definitions of the PBX). If a match is found, the entire PBX definition is reported (as comments) in the ini file using the same format.
MediaPack E.2.7 Adding a Reorder Tone to the CPT File The following procedure describes how to add a Reorder tone that a PBX generates to indicate a disconnected call, to the CPT file. ¾ To add a Reorder tone to the CPT file, take these 11 steps: Make a call (using G.711) between the MP FXO, which is connected to the PBX, and a remote entity in the IP network. 2. Capture the call using a "network sniffer" such as Wireshark. 3.
SIP User's Manual F F. SNMP Traps SNMP Traps This section provides information on proprietary SNMP traps currently supported by the gateway. There is a separation between traps that are alarms and traps that are not (logs). Currently all have the same structure made up of the same 11 varbinds (Variable Binding) (1.3.6.1.4.1.5003.9.10.1.21.1). The source varbind is composed of a string that details the component from which the trap is being sent (forwarded by the hierarchy in which it resides).
MediaPack Table F-2: acBoardTemperatureAlarm Alarm Trap Alarm: acBoardTemperatureAlarm OID: 1.3.6.1.4.1.5003.9.10.1.21.2.0.
SIP User's Manual F. SNMP Traps Table F-4: acOperationalStateChange Alarm Trap Alarm: acOperationalStateChange OID: 1.3.6.1.4.1.5003.9.10.1.21.2.0.15 Default Severity Major Event Type: processingErrorAlarm Probable Cause: outOfService (71) Alarm Text: Network element operational state change alarm. Operational state is disabled. Note: This alarm is raised if the operational state of the node goes to disabled. The alarm is cleared when the operational state of the node goes to enabled.
MediaPack Table F-6: acBoardCallResourcesAlarm Alarm Trap acBoardCallResourcesAlarm Alarm: OID: 1.3.6.1.4.1.5003.9.10.1.21.2.0.8 Default Severity: Major Event Type: processingErrorAlarm Probable Cause: softwareError (46) Alarm Text: Call resources alarm Status Changes: Number of free channels exceeds the predefined RAI high threshold. Condition: Alarm status: Major Note: To enable this alarm the RAI mechanism must be activated (EnableRAI = 1).
SIP User's Manual F.1.2 F. SNMP Traps Component: AlarmManager#0 The source varbind text for all the alarms Board#/AlarmManager#0 where n is the slot number. under this component is Table F-9: acActiveAlarmTableOverflow Alarm Trap Alarm: acActiveAlarmTableOverflow OID: 1.3.6.1.4.15003.9.10.1.21.2.0.
MediaPack F.1.4 Log Traps (Notifications) This section details traps that are not alarms. These traps are sent with the severity varbind value of ‘indeterminate’. These traps don’t ‘clear’, they don’t appear in the alarm history or active tables. One log trap that does send clear is acPerformanceMonitoringThresholdCrossing. Table F-11: acKeepAlive Log Trap acKeepAlive Trap: OID: 1.3.6.1.4.1.5003.9.10.1.21.2.0.
SIP User's Manual F.1.5 F. SNMP Traps Other Traps The following are provided as SNMP traps and are not alarms. Table F-14: coldStart Trap Trap Name: coldStart OID: 1.3.6.1.6.3.1.1.5.1 MIB: SNMPv2-MIB Note: This is a trap from the standard SNMP MIB. Table F-15: authenticationFailure Trap Trap Name: authenticationFailure OID: 1.3.6.1.6.3.1.1.5.5 MIB: SNMPv2-MIB Table F-16: acBoardEvBoardStarted Trap Trap Name: acBoardEvBoardStarted OID: 1.3.6.1.4.1.5003.9.10.1.21.2.0.
MediaPack Reader's Notes SIP User's Manual 382 Document #: LTRT-65408
SIP User's Manual G G. Installation and Configuration of Apache HTTP Server Installation and Configuration of Apache HTTP Server This appendix describes the installation and configuration of Apache’s HTTP server with Perl script environment (required for recording). G.1 Windows 2000/XP Operation Systems Note: For detailed installation information, refer to http://perl.apache.org/docs/2.0/os/win32/config.html. Additional required software: an uploading script (put.
MediaPack 7. Open the Apache2/conf/perl.conf file for editing and add the line “Script PUT /perl/put.cgi” after the last line in the following section (note that if the following section is omitted or different in the file, insert it into the file or change it there accordingly): Alias /perl/ "C:/Apache2/perl/ SetHandler perl-script PerlResponseHandler ModPerl::Registry Options +ExecCGI PerlOptions +ParseHeaders Locate the file put.
SIP User's Manual c. d. e. G. Installation and Configuration of Apache HTTP Server Set the MaxKeepAliveRequests parameter to 0 (enables an unlimited number of requests during a persistent connection – required for multiple consecutive HTTP POST requests for uploading the file). Set MaxClients to 250. Change the mod_perl module lines to:
MediaPack Reader's Notes SIP User's Manual 386 Document #: LTRT-65408
SIP User's Manual H H. Regulatory Information Regulatory Information Declaration of Conformity 73/23/EEC (including amendments) Application of Council Directives: 89/336/EEC (including amendments) 1999/5/EC Annex-II of the Directive EN55022: 1998 + A1: 2000 + A2: 2003 Standards to which Conformity is Declared: EN55024:1998 + A1: 2001 + A2: 2003 EN61000-3-2: 2000 + A2: 2005 EN61000-3-3: 1995 + A1: 2001 EN60950-1: 2001 Manufacturer’s Name: AudioCodes Ltd.
MediaPack Hungarian Alulírott, [AudioCodes Ltd] nyilatkozom, hogy a [MP-11x/FXS & FXO Series & MP-124] megfelel a vonatkozó alapvetõ követelményeknek és az 89/336/EEC, 73/23/EEC; 1999/5/ES irányelv egyéb elõírásainak Icelandic æki þetta er í samræmi við tilskipun Evrópusambandsins 89/336/EEC, 73/23/EEC; 1999/5/ES Italian Con la presente [AudioCodes Ltd] dichiara che questo [MP-11x/FXS & FXO Series & MP-124] è conforme ai requisiti essenziali ed alle altre disposizioni pertinenti stabilite dalla diretti
SIP User's Manual H. Regulatory Information Industry Canada Notice This equipment meets the applicable Industry Canada Terminal Equipment technical specifications. This is confirmed by the registration numbers. The abbreviation, IC, before the registration number signifies that registration was performed based on a declaration of conformity indicating that Industry Canada technical specifications were met. It does not imply that Industry Canada approved the equipment.
™ CPE & Access Analog Gateways SIP MediaPack™ MP-124 & MP-11x User’s Manual Version 5.0 www.audiocodes.
