Dialogic® 1000 and 2000 Media Gateway Series User’s Guide November 2012 64-0346-12
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Contents Revision History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15 About This Publication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21 Purpose . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Intended Audience . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Contents 3.4 3.5 3.6 3.7 3.8 3.9 3.10 3.11 3.12 3.13 3.14 3.15 3.16 6 3.3.2 Server Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61 3.3.3 TCP/UDP Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63 3.3.4 TLS Group . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64 3.3.5 Timing Group . . . . . . . . . . . . . . . . . .
Contents 3.17 3.18 3.19 3.20 4 151 152 152 152 153 153 153 154 158 161 161 161 162 163 173 177 178 178 179 179 179 180 180 180 181 181 181 Call Progress Tones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183 4.1 4.2 5 3.16.5 CPID Length . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3.16.6 CPID Padding String. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Contents 5.4 6 Media Gateway Parsers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 247 6.1 6.2 7 7.3 7.4 7.5 Summary Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265 Alarm Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266 Call Log Status Information . . . . . . . . . . . . . . . . . . . . . . .
Contents 9.7 9.6.2 Trace Utility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9.6.3 Trace Commands . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9.6.4 Examples of Trace Commands and Displays . . . . . . . . . . . . . . . . . . . . . . . . . . . Diagnostic Commands. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9.7.1 Devstat Command . . . . . .
Contents 10 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Figures 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 Typical IP Gateway Phone Emulating Topology - PBX Connection . . . . . . . . . . . . . . . . . . . . . 27 Typical IP Gateway Phone Emulating Topology - PSTN Connection . . . . . . . . . . . . . . . . . . . . 27 IP Gateway Using Serial Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28 Multiple IP Gateways Using Serial Link . . . .
Contents 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 12 Default Analog CPID Configuration in the .ini File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 249 Sample Analog Type II CPID Configuration Data in the .adt File . . . . . . . . . . . . . . . . . . . . . . . 250 Default Mitel Digital CPID Configuration Data (cpid.htm) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Tables 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 Coder/Decoder Parameters. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86 Syntax for Number Matching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137 Default Number Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138 Default Number Type. . . . . . . . . . . . . . . . . .
Contents 14 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Revision History This revision history summarizes the changes made in each published version of this document. Document No. Publication Date Description of Revisions 64-0346-12 November 2012 Updated to support Version 6.0 SU9 Software. Chapter 1, “Overview” added new DMG2060DTISQV34 model listed under DMG2000 section in Product Description. Chapter 3, “Parameter Reference” added new parameter in Audio Group for Unhold on RE-INVITE with No SDP.
Revision History Document No. Publication Date 64-0346-10 November 2010 Description of Revisions Updated to support Version 6.0 SU7 Software. Chapter 3, “Parameter Reference” added new Privacy Headers Group section with new parameters for Enable Privacy Headers and Privacy Header Method. Chapter 3, “Parameter Reference” added new parameter in DSP Settings Parameters for VoIP Packet Loss Concealment (DMG2000 Only). 64-0346-09 July 2010 Updated to support Version 6.0 SU6 Software.
Revision History Document No. Publication Date Description of Revisions 64-0346-06 July 2009 Updated to support Version 6.0 SU4 Software. Chapter 3, “Parameter Reference” updated parameter to Default Value = 90 for TCP Inactivity Timer in TCP/UDP Group. Chapter 3, “Parameter Reference” added new VoIP Network Group Parameters to support VoIP Network Group feature. Chapter 3, “Parameter Reference” added new VoIP Authentication Parameters to support VoIP Authentication feature.
Revision History Document No. Publication Date 64-0346-03 January 2009 Description of Revisions Updated to support Version 6.0 SU2 Software. Chapter 3, “Parameter Reference” added new parameter in User-Agent Group for Reliable Provisional Responses. Chapter 3, “Parameter Reference” updated parameter to Default Value = Off for Signaling Digit Relay Mode in Audio Group.
Revision History Document No. Publication Date Description of Revisions 64-0346-01 March 2008 Updated to support Version 6.0 Software. Global Updates: Added and revised screen shots for the enhanced Web interface. Added new DMG2060DTISQ and DMG2120DTISQ models which include support for survivability. Removed some parameters that are obsoleted in Version 6.0 Software. Chapter 3, “Parameter Reference” added new parameter in Server Group for DNS Server Address 2.
Revision History 20 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
About This Publication The following topics provide information about this guide: • Purpose • Intended Audience • How to Use This Publication • Related Information Purpose This document provides information about installing, configuring, operating, and maintaining the Dialogic® Media Gateway.
About This Publication • Chapter 6, “Media Gateway Parsers” describes the Media Gateway in-band Type I (on-hook) and Type II (off-hook) integration parsers for analog units, and the display parser for digital units. These parsers allow the user to define the meaning of either the in-band/on-hook integration strings or display strings received from the telephony network. • Chapter 7, “Data Security” provides information about configuring security on the Media Gateway for HTTP, call control, and voice.
About This Publication Related Information For additional information related to the Dialogic® 1000 Media Gateway (DMG1000) and Dialogic® 2000 Media Gateway (DMG2000) products, see the following: • Dialogic® 1000 and 2000 Media Gateway Series Getting Started Guide for information about installing, cabling, and initializing the product prior to performing configuration and operation tasks.
About This Publication 24 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Overview 1 The following information provides an overview of the Dialogic® Media Gateway: • Product Description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25 • Call Routing in Phone Emulating Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30 • Voice over IP Address Translation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31 • Security . . . . . . . . . . . . . . . . . . . . . . . .
Overview The DMG2000 is available in the following models: • DMG2030DTIQ - Supports phone emulation mode for a single T1 or E1 interface. • DMG2060DTIQ - Supports phone emulation mode for two T1 or E1 interfaces. • DMG2120DTIQ - Supports phone emulation mode for four T1 or E1 interfaces. • DMG2060DTISQ - Supports phone emulation mode with survivability for two T1 or E1 interfaces. • DMG2060DTISQV34 - Supports phone emulation mode with survivability for two T1 or E1 interfaces.
Overview Figure 1. Typical IP Gateway Phone Emulating Topology - PBX Connection PSTN VoIP Terminal Device Digital Telephone Media Gateway (All Phone Emulating Models) Digital PBX 1-8 Digital Phone Lines or 1-4 T1/E1 Trunks Digital Telephone VoIP Terminal Device LAN Figure 2.
Overview provide full call party information across the station set interfaces while others provide little or no call party information across the station set interface. The PBX switches that provide little or no call party information typically will provide full call party information across a separate serial interface connection. For this reason, the Media Gateway supports a serial link interface to the switch or PSTN, and supports several serial protocols.
Overview The switch only provides a single serial link connection. At a site where there are multiple Media Gateway units, only one of the units can be physically connected to the switch or PSTN serial link. This unit is the serial protocol Master Media Gateway while the remaining units are considered serial protocol slaves. Figure 4 shows how multiple Media Gateways connect to a switch or PSTN that uses a serial link to provide call party information.
Overview Both the serial interface and the selection of which serial protocol to use are configurable using the Web interface. Refer to Chapter 3, “Parameter Reference” for information about configuring the serial interface and serial protocols. 1.2 Call Routing in Phone Emulating Mode The Media Gateway routes calls from the Switch network to a VoIP destination on the IP network. Conversely, it routes calls from the IP network through a Switch port to a destination telephone number on the Switch network.
Overview 1.2.1.1 IP to Switch Calls There are a few circumstances where an inbound IP call may not be routed to the Switch network. They are the following: • No available Switch Ports – There are no available Media Gateway Switch ports on which to carry the call. • Invalid or No Switch destination address specified – The IP call information does not contain a valid E.164 destination address for the Switch network.
Overview address of the Media Gateway as the desired gateway, or they may rely on the VoIP Address Translator to address the Media Gateway using the gateway prefix configured on the Media Gateway. 1.4 Security The Media Gateway supports two kinds of security: • Access Security • Data Security 1.4.1 Access Security Access to the Media Gateway is secured by requiring a user name and password to login to the gateway.
Overview Figure 5. DMG1000 Web Interface Figure 6.
Overview The Media Gateway’s Web interface is divided into four sections, Status, Configuration, Diagnostics, and System. The Status Web pages provide run-time information and statistics about the operation of the unit. The Diagnostics Web pages provide the system administrator with an interface to perform diagnostics tasks. The Configuration Web pages provide the system administrator with an interface to configure the Media Gateway.
Overview • DSP Settings - Selecting DSP Settings brings up the DSP Settings Web page which allows you to configure DSP Settings parameters. For detailed information about the DSP Settings parameters, see Section 3.19, “DSP Settings Parameters”, on page 162. The System Web pages provide the system administrator with an interface to change password, upgrade system, and restart system. • Password - Selecting Password brings up the Password Web page. This page allows you to change your password.
Overview 36 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Media Gateway Configuration 2 Information about configuring, and upgrading the Dialogic® 1000 Media Gateway (DMG1000) and Dialogic® 2000 Media Gateway (DMG2000) is contained in the following sections: • Setting the IP Address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37 • Basic Configuration Via the Serial Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38 • Changing the Password . . . . . . . . . . . . . . . . . . . .
Media Gateway Configuration 8. Once the system completes its initialization (after approximately one minute), browse to the new IP address. The Web browser can now access the unit at the new IP address. 9. If it was necessary to change the IP address of your Windows® workstation in the Initial Log On procedure (described in the Getting Started Guide), you should now change it back to the original IP address and access the Media Gateway using the Media Gateway’s newly configured IP address. 10.
Media Gateway Configuration • For DMG1000 models, select Telephony Switch Type (See Telephony Switch Type parameter information in Section 3.10, “TDM Digital Parameters”, on page 135.) Note: The Telephony Switch Type parameter does not apply to the Models DMG1008MTLDNIW, DMG1008LSW, DMG1004LSW, DMG2030DTIQ, DMG2060DTIQ, and DMG2120DTIQ. • For DMG2000 models, select: – Line Mode (Section 3.8.1.1, “Line Mode”, on page 109) – CAS Protocol (Section 3.8.1.2, “Signaling Mode”, on page 109 and Section 3.8.2.
Media Gateway Configuration 3. When the System Login Web page appears, enter admin in the User Name box and your password in the Password box, then click on the Log On button. Note: The user name and password are case sensitive. 4. Select the appropriate configuration Web page from the Configuration menu. For example, to define the PCM Coding as aLaw, select the System Web page from the menu. When the System Web page appears, select aLaw from the drop down box as the PCM Coding parameter value. 5.
Media Gateway Configuration the configuration has been changed and now you wish to return to the previous configuration, you can import the previous configuration data back to the Media Gateway. • Multiple unit implementation - At a site where there are multiple Media Gateway units, much of the configuration will be the same between units (e.g. Audio Parameters, SIP Parameters, etc.). An installer could configure the first unit, export the *.ini file, edit the unit specific parameters in the *.
Media Gateway Configuration d. Click the Only CPID Parsing Rules button in the Export Files box. The File Download dialog box then appears. You may click the Open button to view/edit the configuration file in Notepad or click the Save button to download and save the file to a directory. The default name for the file is cpid.adt. Clicking the Save button causes the Save As dialog box to appear. e. Click the Export Routing Table Schema button in the Export Files box.
Media Gateway Configuration 4. When the System Login Web page appears, enter the user name and current password in the boxes provided and click on the Log On button. Note: The user name and password are case sensitive. 5. Once the login has been accepted, the Media Gateway Summary Web page will appear. Select the Upgrade Web page from the Configuration menu on the left side of the page. 6. Enter the path and filename of the upgrade file or click on the Browse... button to select the upgrade file. 7.
Media Gateway Configuration 44 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Parameter Reference 3 This section lists each Dialogic® 1000 Media Gateway (DMG1000) and Dialogic® 2000 Media Gateway (DMG2000) configuration parameter that may be changed using the Web browser Configuration menu. Also listed are non-menu (hidden) parameters which are not accessible from the Configuration menu. Changing non-menu parameters significantly modifies the operation of the Media Gateway.
Parameter Reference 3.1 IP Settings The IP Settings include the following groups: • IP Settings, LAN1 • IP Settings, LAN2 (DMG2000) 3.1.1 IP Settings, LAN1 The IP Settings, LAN1 group includes the following parameters: • Client IP Address • Client Subnet Mask • Default Network Gateway Address • BOOTP Enabled • SNTP Server IP Address 3.1.1.1 Client IP Address Description: Sets the IP address of the Media Gateway. Allowed Values: Any valid IP address in dotted decimal notation. Default Value = 10.12.13.
Parameter Reference 3.1.1.3 Default Network Gateway Address Description: Sets the IP address of the default network gateway router. Allowed Values: Any valid IP address in dotted decimal notation. Default Value = blank INI File Parameter Name = ipRouterAddr Note: 3.1.1.4 Unit requires a restart if this parameter value is changed. BOOTP Enabled Description: When enabled, this parameter causes the Media Gateway to issue a BOOTP request on startup.
Parameter Reference 3.1.2 IP Settings, LAN2 (DMG2000) The IP Settings, LAN2 group includes the following parameters: • Client IP Address • Client Subnet Mask Note: 3.1.2.1 Currently, LAN2 is only supported in Version 5.1 SU1 Software or later. Client IP Address Description: Sets the IP address of the Media Gateway second Ethernet port. This Ethernet port may be used as the Maintenance port depending on the “IP Management Interface” setting. A value of 0.0.0.0 disables the LAN2 Ethernet port.
Parameter Reference 3.1.3 IP Advanced Parameters (DMG2000) The IP advanced parameters allow an advanced network topology to be configured using the LAN interfaces. The administrator can configure which LAN interfaces provide access to the IP management protocols. The administrator can also configure custom routing rules that replace or supplement the functionality of the default network gateway address. Management Interface The management IP protocols are accessible on all LAN interfaces by default.
Parameter Reference Figure 7. Example of a Network Topology Two routes are required in this topology so that: 1. LAN1 can access LAN1 Subnet B 2. LAN2 can access LAN2 Subnet B The IP Router 1 may be configured as the default network gateway. However, the DMG2000 must also be configured to use IP Router 2 as a network gateway. A custom route must be created so that LAN2 can access LAN2 Subnet B. In addition, the default network gateway may be optionally replaced by another custom route.
Parameter Reference An IP route rule is a custom routing rule on a specific LAN interface. It allows access to a specific subnet that matches the destination address/mask via the specified gateway address. This rule supplements or replaces the functionality of the default network gateway address by providing access to multiple subnets across all LAN interfaces. The IP route rules are available on the DMG2000 only. Ethernet Interface Description: Specifies the LAN interface used for the route.
Parameter Reference Gateway Address Description: IP address of the gateway router that receives all IP packets from the LAN that match this route. The gateway IP address must be in the same subnet as the route's LAN interface. Allowed Values: Any valid IP address in dotted decimal notation. Default Value = blank INI File Parameter Name = ipRouteGw Note: 3.2 Unit requires a restart if this parameter value is changed.
Parameter Reference E-Mail Minimum Alarm Severity Description: Defines the minimum alarm severity level that generates an e-mail notification. Allowed Values: • Error • Warning • Info Default Value = Info INI File Parameter Name = emailAlarmSeverity Destination E-Mail List Description: Semi-colon delimited list of e-mail addresses that will receive e-mail alarms. Allowed Values: Semi-colon delimited list of e-mail addresses.
Parameter Reference 3.2.2 SysLog Group The SysLog subgroup includes the following parameters: • SysLog Server IP Address • Alarms to SysLog Enabled • SysLog Minimum Alarm Severity • Diagnostics Trace to SysLog Enabled SysLog Server IP Address Description: IP address of the SysLog (RFC3164) server (notifications sent to server's UDP port 514).
Parameter Reference Diagnostics Trace to SysLog Enabled Description: Defines whether diagnostic trace messages are sent to SysLog server. Allowed Values: • Yes = Diagnostics trace messages are sent as user-level facility/debug messages. • No = Diagnostics trace messages are not sent as user-level facility/debug messages. Default Value = No INI File Parameter Name = syslogTraceEnabled 3.2.
Parameter Reference SNMP Minimum Alarm Severity Description: Defines the minimum alarm severity level to generate an SNMP trap. Allowed Values: • Error • Warning • Info Default Value = Info INI File Parameter Name = snmpAlarmSeverity SNMP Trap IP List Description: Semi-colon delimited list of the IP addresses of SNMP Managers who are to receive SNMP traps generated by the IP gateway. Allowed Values: Semi-colon delimited list of IP addresses. Default Value = 255.255.255.
Parameter Reference SNMP System Contact Description: Specifies the SNMP System administration contact name. Allowed Values: Any string with length between 0 - 63 characters Default Value = (no default value) INI File Parameter Name = snmpSysContact Note: Unit requires a restart if this parameter value is changed. SNMP System Location Description: Specifies the SNMP System location.
Parameter Reference HTTPs Server Enabled Description: Defines whether the HTTPs Server is enabled or disabled. Disable the HTTP Server to disable non-secure Web access. Allowed Values: • Yes = HTTPs Server is enabled. • No = HTTPs Server is disabled. Default Value = No INI File Parameter Name = webHttpsEnabled Note: 3.2.5 Unit requires a restart if this parameter value is changed.
Parameter Reference Maintenance Port Enabled Description: Defines whether the Maintenance port is enabled. Disable to prevent any Maintenance messages from being transmitted by the port. If disabled, the Maintenance port will show only boot messages before the port is disabled. Note: If the Web interface and Maintenance port are both turned off, the only way to re-enable the Maintenance port is to re-enable the port through booter as follows: 1. Cycle power on unit. 2.
Parameter Reference 3.3.1 User-Agent Group The User-Agent subgroup includes the following parameters: • Host and Domain Name • User-Agent Header Value • Call as Domain Name? • Invite Expiration • Reliable Provisional Responses 3.3.1.1 Host and Domain Name Description: The host and domain name of the Media Gateway. This name is used when the unit registers with the SIP Registration Server. Allowed Values: Domain Name string Default Value = pbxgw.default.
Parameter Reference 3.3.1.4 Invite Expiration Description: Specifies the amount of time in seconds that an INVITE request sent by the Media Gateway is valid and can be accepted by the SIP endpoint. After the defined time in seconds, the INVITE request expires and is no longer valid. Allowed Values: 1 - 60000 seconds Default Value = 60000 seconds INI File Parameter Name = sipExpInvSec 3.3.1.
Parameter Reference Notes: 1. If the DNS Server IP Address parameter is configured but the Primary Proxy Server Address parameter is not, the Media Gateway will use the DNS Server to resolve IP address information. 2. If both the DNS Server IP Address parameter and the Primary Proxy Server Address parameter are configured, the Media Gateway will use the Proxy Server for all requests. 3.
Parameter Reference 3.3.3 TCP/UDP Group The TCP/UDP subgroup includes the following parameters: • UDP/TCP Transport Enabled • TCP/UDP Server Port • TCP Inactivity Timer 3.3.3.1 UDP/TCP Transport Enabled Description: Enables or disables the UDP/TCP transports. Allowed Values: • Yes = UDP and TCP transports are enabled. • No = UDP and TCP transports are disabled. Default Value = Yes INI File Parameter Name = sipUdpTcpEnabled Note: 3.3.3.2 Unit requires a restart if this parameter value is changed.
Parameter Reference 3.3.4 TLS Group The TLS subgroup includes the following parameters: • TLS Transport Enabled • TLS Server Port • SSL TLS Protocol • Mutual TLS Authentication Required • TLS Inactivity Timer • Verify TLS Peer Certificate Date • Verify TLS Peer Certificate Trust • Verify TLS Peer Certificate Purpose 3.3.4.1 TLS Transport Enabled Description: Determines whether the TLS transport is enabled or disabled. Allowed Values: • Yes = TLS transport is enabled. • No = TLS transport is disabled.
Parameter Reference 3.3.4.2 TLS Server Port Description: The Transport Layer Security (TLS) Port of the Media Gateway on which SIP TLS messages are sent/received. Allowed Values: 1024 - 65000 Default Value = 5061 INI File Parameter Name = sipTlsServerPort Note: 3.3.4.3 Unit requires a restart if this parameter value is changed. SSL TLS Protocol Description: Specifies the default SSL record type to be used on TLS connections.
Parameter Reference 3.3.4.5 TLS Inactivity Timer Description: Number of seconds after which an idle Transport Layer Security (TLS) connection will be closed. Allowed Values: 10 - 60000 seconds Default Value = 30 seconds INI File Parameter Name = sipTlsInactivitySec 3.3.4.6 Verify TLS Peer Certificate Date Description: Determines whether the TLS peer's certificate date is validated. This requires that the Media Gateway has already received the network time via SNTP.
Parameter Reference 3.3.4.8 Verify TLS Peer Certificate Purpose Description: Determines if the purpose of the TLS Peer Certificate should be verified. Allowed Values: • Yes = Validate • No = Do not validate Default Value = Yes INI File Parameter Name = sipTlsCertVerifyPurpose 3.3.5 Timing Group The Timing subgroup includes the following parameters: • T1 Time • T2 Time • T4 Time • T1 Multiplier 3.3.5.1 T1 Time Description: The T1 Time specifies the SIP request retransmit timeout in milliseconds.
Parameter Reference 3.3.5.2 T2 Time Description: The T2 Time specifies the maximum retransmit time of SIP request messages (except for INVITE) in milliseconds. The retransmitting of the request and the doubling of the timeout continues until the timeout reaches T2 Time milliseconds. Once the timeout reaches T2 Time, the request is retransmitted at T2 Time intervals until a response is received or until the message has been retransmitted “NumRetriesRequest” times - at which point the request expires.
Parameter Reference 3.3.6 Monitoring Group Note: The Monitoring subgroup parameters only apply to a Media Gateway operating in the Phone Emulating Mode. The Monitoring subgroup includes the following parameters: • Monitor Call Connections • Call Monitor Interval • VoIP Host Monitor Interval • Proactive DNS Monitoring Monitor Call Connections Description: When the Monitor Call Connections parameter is enabled, the Media Gateway will monitor the connection state of active IP calls.
Parameter Reference VoIP Host Monitor Interval Description: Interval in which VoIP hosts are monitored to determine on-line status. Valid only if there are VoIP Host Groups in the Routing Table that have Fault Tolerance enabled. Allowed Values: 10 - 3600 Default Value = 30 INI File Parameter Name = gwMonitorVoipHostsIntSec Note: Unit requires a restart if this parameter value is changed.
Parameter Reference IPv4 TOS Byte Bits: 0-2 = Precedence (RFC 1122) 3-6 = Type of Service (RFC 1349/1455) 7 = Must be zero PrecedenceBits (0-2): 111xxxxx (224) = Network Control 110xxxxx (192) = Internetwork Control 101xxxxx (160) = CRITIC/ECP 100xxxxx (128) = Flash Override 011xxxxx (96) = Flash 010xxxxx (64) = Immediate 001xxxxx (32 = Priority 000xxxxx (0) = Routine Type of Service Bits (3-6): xxx0000x (0) = Normal xxx1000x (16) = Minimize delay xxx0100x (8) = Maximize throughput xxx0010x (4) = Maximize
Parameter Reference 3.3.8 Privacy Headers Group Note: The Privacy Headers parameters only apply to a DMG2000 that is configured for SIP (on the VoIP side) and ISDN (on the TDM side). Privacy Headers are used in SIP messages to specify to the receiving side how to present ‘Calling Party ID’ data to a user. When privacy is enabled, the normal ‘From’ fields in a SIP message are masked and the actual ‘Calling Party ID’ is placed in ‘privacy headers’.
Parameter Reference Allowed Values: • P-Asserted-Identification = When a SIP ‘P-Asserted-Identity’ header is received its data is translated into ISDN privacy fields. When ISDN privacy fields are received they are converted into a SIP ‘P-Asserted-Identity’ header. If a SIP ‘Remote-Party-ID’ header is received it is ignored. • Remote-Party-ID = When a SIP ‘Remote-Party-ID’ header is received its data is translated into ISDN privacy fields.
Parameter Reference Figure 8. Example Call Flow for SIP-SIP Calls ISDN-ISDN Calls The UUI information element is passed unchanged between calling and called parties. The gateway support a single UUI information element in the 'SETUP', 'ALERTING' and 'CONNECT' messages. Figure 9 is an example call flow for ISDN-ISDN calls: Figure 9.
Parameter Reference SIP-ISDN and ISDN-SIP Calls For calls between SIP and ISDN, the UUI is translated between formats. SIP UUI headers are translated to ISDN UUI information elements and ISDN UUI information elements are translated to SIP UUI headers. Translation occurs only for specific message pairs. Translation occurs between SIP 'INVITE' and ISDN 'SETUP', between SIP '180 Ringing' and ISDN 'ALERTING', and between SIP '200 OK' and ISDN 'CONNECT'.
Parameter Reference Transfers using SIP REFER When the DMG2000 receives a SIP REFER that contains UUI, the UUI is passed to the resulting SIP INVITE or ISDN SETUP. Figure 11 is an example call flow for transfers using SIP REFER: Figure 11.
Parameter Reference The User-User Information subgroup includes the following parameters: • User-User Enabled • Translation Method User-User Enabled Description: Specifies how the DMG2000 processes SIP UUI headers and ISDN UUI information elements are processed by the DMG 2000. Allowed Values: • Yes = The SIP UUI headers and ISDN UUI information elements are processed. • No = The SIP UUI headers and ISDN UUI information elements are ignored.
Parameter Reference 3.3.10 ISUP Using Multipart MIME Group The ISUP Using Multipart MIME parameters control the propagation of ISDN information elements to SIP messages. The information elements are, typically, proprietary to a telephony vendor and do not map to any defined SIP header. The DMG2000 uses multipart MIME bodies to encapsulate the information element’s data. Various header fields allow the SIP endpoint to decode the MIME body.
Parameter Reference Lines 5 and 6 specify that multipart MIME is appended to the INVITE message and that it uses MIME version 1.0. Line 6 defines the boundary between MIME bodies; in our example the boundary is the string ?(boundary)*. Line 17 starts the first MIME body. This body sends SDP (session description protocol) to the SIP endpoint. We are not concerned with this. Line 29 starts the second MIME body. This body sends the ISDN information element’s data.
Parameter Reference 3.4 VoIP Network Group Parameters The VoIP Network Group feature on the gateway provides a means for the gateway to communicate to more than one type of VoIP Network. Specifically, the user can configure up to four unique VoIP Networks - each having their own specific VoIP attributes.
Parameter Reference Figure 12.
Parameter Reference 3.4.1 Network Group The Network subgroup includes the following parameters: • Network Group Label 3.4.1.1 Network Group Label Description: Label of the currently selected VoIP network group. Allowed Values: A string with length between 0 - 40 characters Default Value = Default 3.4.2 Transport Group The Transport subgroup includes the following parameters: • Transport Protocol • SIPS URI Scheme 3.4.2.
Parameter Reference 3.4.3 URI Parameters Group The URI Parameters subgroup includes the following parameters: • User Phone Parameter • Local Phone Context • Remote Phone Context 3.4.3.1 User Phone Parameter Description: Determines if the “user=phone” attribute is included in the SIP URI. Allowed Values: • Yes • No Default Value = Yes 3.4.3.2 Local Phone Context Description: Defines the string that will be used in the “phone-context” attribute included in “From” header of the SIP URI.
Parameter Reference 3.4.4 Proxy Group The Proxy subgroup includes the following parameters: • Primary Proxy Server Address • Primary Proxy Server Port • Backup Proxy Server Address • Backup Proxy Server Port • Proxy Query Interval 3.4.4.1 Primary Proxy Server Address Description: The IP Address of the SIP Proxy Server through which the Media Gateway may send/receive requests. If blank, the Media Gateway will not use a Proxy Server.
Parameter Reference 3.4.4.5 Proxy Query Interval Description: Interval in seconds at which the Primary Proxy Server is queried. If the Primary Proxy Server does not respond to the query, the Media Gateway will switch to the Backup Proxy Server. Once the Primary Proxy Server responds to the query, the Media Gateway will switch back to the Primary Proxy Server. Note: This parameter is only valid if the Backup Proxy Server Address parameter is configured.
Parameter Reference 3.4.5.3 Registration User Description: Specifies the address-of-record (AOR) to be registered. If it does not contain ‘@domain’ then the gateway IP address is appended. Allowed Values: A string with length between 0 - 64 characters Default Value = (no default value) 3.4.5.4 Registration Expiration Description: Specifies the amount of time in seconds that the registration with the SIP Proxy Server is valid. Allowed Values: 1 - 60000 seconds Default Value = 3600 seconds 3.4.
Parameter Reference INI File Parameter Name = dspFrameSizeG711, dspFramesPerPktG723k, dspFramesPerPktG729 3.4.6.2 Audio Compression Description: Sets the audio coder/decoder to be used by the Media Gateway. Allowed Values: • G.711u/G.711a = G.711 uLaw preferred, G.711 aLaw secondary • G.711u = G.711 uLaw only • G.711a = G.711 aLaw only • G.723.1 = G.723.1 only • G.729AB = G.729AB only Default Value = G.711u/G.
Parameter Reference 3.4.
Parameter Reference MKI on Transmit Stream Description: Specifies if Master Key Index (MKI) is supported on transmit stream. Allowed Values: • Yes = Transmit stream supports MKI • No = Transmit stream does not support MKI Default Value = Yes INI File Parameter Name = srtpTxMkiEnable Note: Currently, the Media Gateway uses only one master key. Key Derivation Enable Description: Specifies if the Secure RTP (SRTP) Key is changed during a voice session.
Parameter Reference Cipher Mode Description: Specifies the cipher used to encrypt voice packets. Note: Plain text is essentially no encryption and should be used only for testing purposes. For no encryption, you should specify RTP ONLY in the SRTP Preference parameter. Allowed Values: • Plain_Text = Use Plain_Text as the cipher (No encryption - for testing purposes only) Note: Plain_Text is not supported in Version 5.1 SU2 Software.
Parameter Reference Unauthenticated SRTP Enable Description: This parameter can be used to force Realtime Transport Protocol (RTP) packets to be received without authentication even if an authentication algorithm has been negotiated between parties. Allowed Values: • Yes = RTP VOICE packets will NOT be authenticated. • No = RTP VOICE packets will be authenticated per the negotiated algorithm.
Parameter Reference 3.5 VoIP Media Parameters The VoIP Media parameters include the following subgroups: • Early Media Group • Audio Group • Fax Group • RTP Group 3.5.1 Early Media Group The Early Media subgroup includes the following parameters: • RFC 3960 Early Media Support • Send Early 183 Progress Response • Early Media Requires Reliable Provisional Responses RFC 3960 Early Media Support Description: Specifies the Early Media mode to be supported by the gateway.
Parameter Reference Early Media Requires Reliable Provisional Responses Description: Specifies that Early-Media is only available if reliable provisional responses (PRACK) are supported or required on the call. Allowed Values: • Yes • No Default Value = No INI File Parameter Name = sipEarlyMediaReqRelProvRsp 3.5.
Parameter Reference Voice Activity Detection Description: Enables the use of voice activity detection (VAD) to reduce the amount of audio data traffic. If VAD is enabled, the unit will stop the transmission of RTP audio data when no voice activity is detected on the telephony port. Transmission is continued when voice activity is detected. This drastically reduces the amount of audio traffic on the network. Allowed Values: • On = Voice activity detection is enabled.
Parameter Reference Packet Time for Inbound VoIP Description: Specifies a packet time to use when there is no packet time specified by the SDP of Inbound VoIP calls. The actual packet time used may be different depending on negotiated packet time, codec, or system limitations. Allowed Values: • 10 • 20 • 30 • 40 • 50 • 60 Codec Type Valid Packet Times (ms) G.711 10, 20, 30, 40*, 50*, 60* G.723 30, 60 G.729 10, 20, 30, 40, 50, 60 G.722 20 * = available on DMG2000 only.
Parameter Reference Allowed Values (Phone Emulating): • None = The Media Gateway will not use RTP packets to transmit/receive tone information. • RFC2833 = The (DTMF) tone information is sent between the Media Gateway and VoIP endpoints via RTP packets as defined by RFC 2833. Note: A payload type of 101 is used in the implementation of the RFC2833 method. • RFC2833-LowLatency = Same as RFC2833, but with 20 milliseconds DTMF detection delay removed.
Parameter Reference Unhold on RE-INVITE with No SDP Description: If set to No and the call is on hold, when receiving a RE-INVITE with no SDP, the hold is not removed. If set to Yes and the call is on hold, when receiving a RE-INVITE with no SDP, the hold is removed. Allowed Values: • Yes • No Default Value = No INI File Parameter Name = sipReInviteNoSdpIsUnhold 3.5.
Parameter Reference Allowed Values: IP address in dotted decimal notation Default Value = (no default value) INI File Parameter Name = gwFaxServerAddress Fax Server Network Group Description: Defines the Network Group to be used for inbound TDM fax calls when Fax Server Address is non-null.
Parameter Reference V.34 Fax Speed Limit Description: The gateway V.34 fax modem supports rate configurability. Note: The V.34 fax modem rate setting is currently only supported on the DMG2030DTIQ and DMG2060DTISQV34. Allowed Values: {33600, 31200, 28800, 26400, 24000, 21600, 19200, 16800, 14400, 12000, 9600, 7200, 4800, 2400, None} Default Value = 33600 INI File Parameter Name = gwV34SpeedLimit 3.5.
Parameter Reference RTP End Port Description: Specifies the last UDP IP port used for Realtime Transport Protocol (RTP) traffic. This value must be greater than the Start Port for RTP. It is recommended that this value be at least 1000 more than Start Port for RTP. Allowed Values: 1024 to 65000 Default Value = 50000 INI File Parameter Name = gwRTPEndPort Note: Unit requires a restart if this parameter value is changed.
Parameter Reference RTP QoS Byte Description: The RTP QoS Type parameter defines a decimal value that represents QoS bit flags. This parameter is used in each RTP data packet transmitted from the Media Gateway with a QoS byte code. Routers use this byte code to assign priority levels to packets. The QoS byte may be interpreted as either IPv4 TOS or DiffServ.
Parameter Reference 3.6 VoIP Authentication Parameters Information about VoIP Authentication and how it is supported by the Dialogic® Media Gateway is described in the following sections: • VoIP Authentication Overview • Inbound VoIP (Server) • Outbound VoIP (Client) 3.6.
Parameter Reference Algorithm - Sets the algorithm of the gateway. MD5 is the default hashing algorithm for authentication as specified by RFC 2617. MD5-sess is the algorithm that employs MD5 hashing with the use of a session key. Methods to Challenge - Determines which received SIP methods will be challenged if the Inbound Authentication Enabled is set to Yes. • Invite • Register • Notify • Info • Bye • Refer • Options 3.6.1.3 Users Figure 14 is a sample screen shot of the Users Web page: Figure 14.
Parameter Reference 3.6.2.1 Outbound VoIP Configuration Figure 15 is a sample screen shot of the Outbound VoIP Configuration Web page: Figure 15. Outbound VoIP Configuration Web Page Outbound Authorization Enabled - Enables or disables the ability of the gateway to respond to Authorization challenges. If Yes, the gateway is capable of resending a SIP method with Authorization information as a result of a received challenge from a SIP peer. 3.6.2.
Parameter Reference 3.
Parameter Reference Maximum Call Party Delay (Phone Emulating Only) Description: Specifies the maximum number of milliseconds that the Media Gateway will wait for display information (calling party information) on an inbound PBX call before the call is routed to the IP destination.
Parameter Reference Turn MWI On FAC (Phone Emulating Only) Description: Specifies the Features Access Code (FAC) to dial to turn on the Message Waiting Indicator (MWI) of a PBX extension. Allowed Values: Any string of less than 11 characters Default Value = (no default value) INI File Parameter Name = telMwiOnFAC Turn MWI Off FAC (Phone Emulating Only) Description: Specifies the Features Access Code (FAC) to dial to turn off the Message Waiting Indicator (MWI) of a PBX extension.
Parameter Reference Hunt Group Extension (Phone Emulating Only) Description: If the telephony ports are configured in a Hunt Group on the switch, then enter the Hunt Group extension number in this field. The gateway will use this field to ignore the Hunt Group extension when generating the call party information. This ensures that the Hunt Group extension number is not mistaken for a calling or called party number.
Parameter Reference 3.8 TDM T1/E1 Parameters The T1/E1 parameters are used to configure the T1 and E1 port(s) on DMG2000 only and include the following groups: • T1/E1 Mode Group • T1 CAS Protocol Group (T1 CAS Signaling Mode) • T1 ISDN Protocol Group (ISDN Signaling Mode) • E1 ISDN Protocol Group (ISDN Signaling Mode) 3.8.1 T1/E1 Mode Group The parameters in the T1/E1 Mode group include the following: • Line Mode • Signaling Mode • Telephony Port Interface Side • Clock Control 3.8.1.
Parameter Reference 3.8.1.3 Telephony Port Interface Side Description: Identifies the side of the connection. Allowed Values: • Terminal = Port is Terminal Equipment (TE) • Network = Port is Network Termination (NT) Default Value = Terminal INI File Parameter Name = telPortInterfaceSide 3.8.1.4 Clock Control Description: Selects if a port receives clock from external equipment or drives clock to external equipment.
Parameter Reference • Consult Call Disconnected Drop Code • Consult Call Error Drop Code • MWI Confirmation Tone • Use Same Port for MWI Clear/Set • Initial Wait for Inband CPID • Inband CPID Complete Timeout • Inband Type I CID to First Ring Timeout 3.8.2.1 T1 CAS Protocol Description: If T1 CAS is selected as the Signaling Mode, specifies the T1 CAS protocol to be used. Allowed Values: • Loop Start = Loop Start protocol will be used. • Ground Start = Ground Start protocol will be used.
Parameter Reference Default Value = Yes INI File Parameter Name = telDialToneExpAfterFlashHook 3.8.2.4 Delay After Flash-Hook Description: Specifies the duration (in milliseconds) to delay after a Flash Hook if a dial tone confirmation is not expected. Note: This parameter can only be changed in the configuration file. It is not accessible through the Web interface. Allowed Values: 100 to 10000 milliseconds Default Value = 2000 milliseconds INI File Parameter Name = telFlashHookNoDialToneDelayMs 3.8.
Parameter Reference INI File Parameter Name = msRingCycleTime 3.8.2.8 Enable Glare Detection Description: When making an outbound call on the TDM network, it specifies whether the gateway will treat the failure to detect Dialtone as a Glare Condition. A Glare condition is when an incoming call arrives at the same time a port goes off hook to dial. Allowed Values: • Yes = The failure to detect dialtone will be treated as a Glare. • No = The outbound call will proceed.
Parameter Reference 3.8.2.11 Consult Call Proceeding Drop Code Description: Defines the Feature Code to dial in order to drop a Consult call that is in the Proceeding state (dialed but not connected) and to reconnect to the original call. Allowed Values: A string from 0 to 10 digits in length using any of the following characters: 0-9, *, #, ! Default Value = !! Note: Each ! is a hook flash. For !, the Media Gateway will perform a single hook-flash operation.
Parameter Reference 3.8.2.14 Consult Call Disconnected Drop Code Description: Defines the Feature Code to dial in order to drop a Consult call that is in the Disconnected state and to reconnect to the original call. Allowed Values: A string from 0 to 10 digits in length using any of the following characters: 0-9, *, #, ! Default Value = !! Note: Each ! is a hook flash. For !, the Media Gateway will perform a single hook-flash operation.
Parameter Reference 3.8.2.17 Use Same Port for MWI Clear/Set Description: Specifies if the PBX requires that the telephony station that set an MWI be used to clear the MWI. Note: Some PBX types require that the port that set an MWI be the same port that clears the MWI. Allowed Values: • Yes = Use the same port to set and clear the MWI. • No = Not necessary to use the same port to clear an MWI that was used to set the MWI. Default Value = Yes INI File Parameter Name = telMwiSamePort 3.8.2.
Parameter Reference 3.8.2.20 Inband Type I CID to First Ring Timeout Description: Specifies the number of milliseconds that are allowed from the end of the Type I CPID on the first ring. If this time is exceeded, the CPID is cleared. Note: This parameter is only needed when CPID arrives at the Gateway before the first ring. This parameter can be ignored if the CPID arrives between the first and second ring.
Parameter Reference 3.8.3.2 Framing Description: Specifies the type of T1 framing that will be used by the line. Allowed Values: • SF = D4 Superframe format (12 consecutive T1 frames) is used. • ESF = Extended Superframe format (24 consecutive T1 frames) is used. Default Value = ESF INI File Parameter Name = t1Framing 3.8.3.3 Selects Transmit Pulse Waveform Description: Selects the method to be used for generating the Transmit Waveform Shape.
Parameter Reference 3.8.3.5 ISDN Protocol Variant Description: Specifies the type of T1 ISDN protocol variant to be used. Allowed Values: • None = None (Standard) • Alcatel = Support Alcatel Extensions • Ericsson = Support Ericsson Extensions • Avaya_IP_Office = Support Avaya IP Office Extensions • Nortel_DMS-100 = Support Nortel DMS-100 Extensions Default Value = None INI File Parameter Name = t1IsdnProtocolVariant 3.8.3.
Parameter Reference 3.8.3.7 Network Specific Facilities (NSF) Description: If NI-2 protocol is used and any value other than 'None' is selected, then the specified service type is included as an NSF Information Element in the outgoing ISDN packet.
Parameter Reference 3.8.3.9 Enable Failover Description: Specifies the Failover mode used when the gateway is powered down. Note: This parameter is only applicable for DMG2060DTISQ and DMG2120DTISQ models. Allowed Values: • Yes = Failover enabled (span 1 to span 2) and (span 3 to span 4). • No = Failover disabled (spans not interconnected). Default Value = No INI File Parameter Name = t1e1Failover 3.8.
Parameter Reference 3.8.4.2 Framing Description: Specifies the type of E1 framing that will be used by the line. Allowed Values: • CRC_MF = Multiframe format with CRC is used. • FR = Basic frame format is used. • MF = Multiframe format is used. Default Value =CRC_FMF INI File Parameter Name = e1Framing Note: 3.8.4.3 Unit requires a restart if this parameter value is changed. Selects Transmit Pulse Waveform Description: Selects the method to be used for generating the Transmit Waveform Shape.
Parameter Reference 3.8.4.5 ISDN Protocol Variant Description: Specifies the type of E1 ISDN protocol variant to be used. Allowed Values: • None = None (Standard) • Alcatel = Support Alcatel Extensions • Ericsson = Support Ericsson Extensions • Avaya IP Office = Support Avaya IP Office Extensions Default Value = None INI File Parameter Name = e1IsdnProtocolVariant 3.8.4.6 Contiguous B-Channel Description: Specifies B-Channel Selection Mode Used By Switch. Only applicable to E1 ISDN protocols.
Parameter Reference 3.8.4.8 Outbound TDM Calling Party Source Description: The source of the calling party that will be provided to the TDM interface for outbound calls and MWI requests. Note: When a DMG2000 is connected to a Mitel PBX and is configured for E1 QSIG, this parameter must be set to one of the following options in conjunction with the Static TDM Calling Party parameter: • Outbound TDM Calling Party Source set to Static and Static TDM Calling Party set to a number (e.g. 1234).
Parameter Reference 3.8.4.10 ISDN Answer Supervision Enable Description: If ‘Yes’, and an outbound call receives a Progress Indicator with a 1 or 8 (non-ISDN, inband), then the gateway will use in-band answer-supervision to determine when the call has been answered. If ‘No’, then a received CONNECT message will be used to determine when the call has been answered. Notes: 1. Progress Indicator 1-Call is not end-to-end ISDN or may be in-band information. 2.
Parameter Reference 3.9.1 Timing Group The Analog parameters in the Timing group include: • Flash Hook • Loop Current Off Debounce • Incoming Rings Before Answer • Ringing Timeout 3.9.1.1 Flash Hook Description: Specifies the duration in milliseconds that the DMG1000 will remain on-hook during a hook flash operation. Allowed Values: 50 to 4000 milliseconds Default Value = 500 milliseconds INI File Parameter Name = telFlashMs 3.9.1.
Parameter Reference 3.9.1.4 Ringing Timeout Description: Specifies the number of milliseconds of non-ringing that will signal that an incoming call has gone away. Allowed Values: 100 to 10000 milliseconds Default Value = 6000 milliseconds INI File Parameter Name = telRingOffMs 3.9.
Parameter Reference 3.9.2.2 Consult Call Dialtone Drop Code Description: Defines the Feature Code to dial in order to drop a Consult call that is in the Dialtone state and to reconnect to the original call. Allowed Values: A string from 0 to 10 digits in length using any of the following characters: 0-9, *, #, ! Default Value = !! Note: Each ! is a hook flash. For !, the Media Gateway will perform a single hook-flash operation. For !!, the Media Gateway will perform a double hook-flash operation.
Parameter Reference 3.9.2.5 Consult Call Connected Drop Code Description: Defines the Feature Code to dial in order to drop a Consult call that is in the Connected state and to reconnect to the original call. Allowed Values: A string from 0 to 10 digits in length using any of the following characters: 0-9, *, #, ! Default Value = !! Note: Each ! is a hook flash. For !, the Media Gateway will perform a single hook-flash operation. For !!, the Media Gateway will perform a double hook-flash operation.
Parameter Reference 3.9.3 Message Waiting Control Group The Analog parameters in the Message Waiting Control group include: • MWI Confirmation Tone • Use Same Port for MWI Clear/Set 3.9.3.1 MWI Confirmation Tone Description: Specifies whether or not the PBX sends a confirmation tone to signal the successful completion of a Message Waiting Indication (MWI) request. Allowed Values: • Yes = PBX will send a confirmation tone. • No = PBX will not send a confirmation tone.
Parameter Reference 3.9.4 CPID Settings Group The Analog parameters in the CPID Settings group include: • Initial Wait for Inband CPID • Inband CPID Complete Timeout • CID to First Ring Timeout • Analog Interface Type • Central Office (Type I) Caller ID Type • Central Office (Type I) Caller ID Alert Type • Central Office (Type I) FSK Caller ID Expiration • Central Office (Type I) FSK Caller ID Timeout • Auto-Answer Inbound TDM Calls (Type II CPID) 3.9.4.
Parameter Reference 3.9.4.3 CID to First Ring Timeout Description: Specifies the number of milliseconds that are allowed from the end of the Type I CPID on the first ring. If this time is exceeded, the CPID is cleared. Note: This parameter is only needed when CPID arrives at the Gateway before the first ring. This parameter can be ignored if the CPID arrives between the first and second ring.
Parameter Reference 3.9.4.6 Central Office (Type I) Caller ID Alert Type Description: Specifies the kind of Type I caller ID alert that is used on the trunk(s) connected to the DMG1000, either pause in ring cycle, ring burst, polarity reversal, or none. This configuration option is only enabled when the Central Office (Type I) Caller ID Type is set to Bellcore FSK. Notes: 1.
Parameter Reference 3.9.4.9 Auto-Answer Inbound TDM Calls (Type II CPID) Description: Type II CPID requires that the inbound TDM call be answered by the gateway before the switch will send the inband CPID to the gateway. If this parameter is enabled, then the gateway will answer the call and gather the inband CPID from the switch prior to sending a call-request to the VoIP endpoint. This allows the gateway to send the CPID information as part of the initial callrequest to the VoIP endpoint.
Parameter Reference 3.10 TDM Digital Parameters The TDM Digital subgroup includes the following parameter: • Telephony Switch Type Telephony Switch Type Description: Specifies the type of PBX telephony switch to which the DMG1000 is connected. Note: This parameter does not apply to the Models DMG1008MTLDNIW, DMG1008LSW, DMG1004LSW, DMG2030DTIQ, DMG2060DTIQ, and DMG2120DTIQ. Note: The list of allowed values that will be displayed depends on your model type.
Parameter Reference 3.11 TDM Port Enable Parameters The TDM Port Enable parameters allow you to individually configure the call control capabilities of each port connected to the Media Gateway. These parameters also allow you to enable or disable individual ports. The TDM Port Enable group includes the following parameters: • Port # • Telephony Port Enabled 3.11.0.1 Port # Description: This is a read-only parameter that defines the Media Gateway port numbers. Note: 3.11.0.
Parameter Reference Figure 17. ISDN Call Type Rules Web Page Select Use this column to select a rule and change its priority. Click and hold the left mouse button and drag the row up or down. Rule Label Specifies name of the call type rule. Calling Number Match Each CPID Matching rule must use the following syntax: Table 2. Syntax for Number Matching Token Description * Matches all. 0123456789 Identifies a specific digit. [digit_string-digit_string] Specifies a range of digit strings.
Parameter Reference • Private • Reserved Table 3. Default Number Plan NI-2, DMS-100, 5ESS Length = 7 ISDN_E.164 Length = 10 ISDN_E.164 All other lengths Unknown QSIG, ETSI All number lengths Unknown Calling Type of Number Type of Number for the calling party information. The user must select matching number type according to the selected number plan. If the value is set to default, the gateway will set an appropriate value depending on the calling number.
Parameter Reference Called Number Match Each CPID Matching rule must use the following syntax: Table 5. Syntax for Number Matching Token Description * Matches all. 0123456789 Identifies a specific digit. [digit_string-digit_string] Specifies a range of digit strings. x Matches any single digit. . Matches any number of ending digits. Called Numbering Plan Numbering Plan for the called party information.
Parameter Reference Called Type of Number Type of Number for the called party information. The user must select matching number type according to the selected number plan. If the value is set to default, the gateway will set an appropriate value depending on the called number. Allowed Values: • Default • Unknown • International • National • Network • Subscriber • Abbreviated • Reserved • Level2Regional • Level1Regional • PISNSpecific • Level0Regional Table 7.
Parameter Reference 3.13 TDM Cause Codes Group The TDM Cause Codes group provides tables for mapping ISDN Cause Codes and SIP Responses. 3.13.1 ISDN Cause Code Mapping The Web interface contains two tables that define mapping between ISDN Cause Codes and SIP Responses. • ISDN Cause Codes to SIP Response • SIP Responses to ISDN Cause Codes 3.13.1.1 ISDN Cause Codes to SIP Response The ISDN to SIP Mapping table maps ISDN Cause Codes to generated SIP Responses.
Parameter Reference Table 8 lists the supported SIP Responses. Table 8.
Parameter Reference Table 8. ISDN to SIP Mapping SIP Responses 3.13.1.2 505 Version Not Supported 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere SIP Responses to ISDN Cause Codes The SIP to ISDN Mapping table maps SIP Responses to ISDN Cause Codes. This table takes effect under the following scenarios: disconnect request and release request. Figure 19 is a screen shot of the SIP to ISDN Mapping Web page: Figure 19.
Parameter Reference Table 9 lists the supported ISDN Cause Codes. Table 9.
Parameter Reference Table 9. SIP to ISDN Mapping ISDN Cause Codes 3.
Parameter Reference The text box on this page can be easily modified by either pasting parsing rules into it from a text file or directly modifying the contents of the control. See Chapter 6, “Media Gateway Parsers” for further information on the creation and use of parsing rules. INI File Parameter Name = ;-CPID RULES 3.15 Serial Ports Parameters The following parameters are in the Serial ports group: • Serial Port, COM1 Group • Serial Port, COM2 Group (DMG2000 Only) 3.15.0.
Parameter Reference Serial Port Parity Description: Specifies the parity of the Media Gateway Serial port. Allowed Values: • None = Parity will not be used. • Even = Even parity will be used. • Odd = Odd parity will be used. Default Value = None INI File Parameter Name = sysSerialParity Serial Port Data Bits Description: Specifies the number of data bits used by the Media Gateway Serial port.
Parameter Reference 3.15.0.2 Serial Port, COM2 Group (DMG2000 Only) The Serial port, COM2 group parameters are used to configure the DMG2000 diagnostics/administration Serial port (COM2 connector). The system parameters in the Serial port group include: • Serial Port Baud Rate • Serial Port Parity • Serial Port Data Bits • Serial Port Stop Bits Serial Port Baud Rate Description: Specifies the baud rate of the DMG2000 Serial port.
Parameter Reference Serial Port Data Bits Description: Specifies the number of data bits used by the DMG2000 Serial port. Allowed Values: • 7 data bits • 8 data bits Default Value = 8 data bits INI File Parameter Name = sysSerialDataBitsCom2 Serial Port Stop Bits Description: Specifies the number of stop bits used by the DMG2000 Serial port. Allowed Values: • 1 stop bit • 2 stop bits Default Value = 1 stop bit INI File Parameter Name = sysSerialStopBitsCom2 3.
Parameter Reference 3.16.1 Serial Mode (Master/Slave) Description: Specifies if the Media Gateway is using the Serial protocol and, if so, whether the unit is the Serial Protocol Master or Slave. The Media Gateway must be configured as the Master if it is physically connected to the Serial link of the PBX. In a system with multiple Media Gateways, only one of the units can be connected to the Serial link of the PBX.
Parameter Reference 3.16.3 MCI Message Extension Length Description: Specifies the extension length used in MCI messages. Messages with six-digit extensions or with eight-digit extensions. Allowed Values: • Six-Digits • Eight-Digits Default Value = Six-Digits INI File Parameter Name = telSerMciMsg 3.16.4 MCI Message Type Description: Specifies the type of MCI messages. Messages can be Type B (default) or Type A.
Parameter Reference 3.16.6 CPID Padding String Description: Specifies the pad string to strip from the CPID fields (calling and called parties) in the incoming Serial packet data. Also used to pad extensions in MWI requests sent to the PBX from the Media Gateway. This pad string must match the pad string configured by the PBX. This parameter is only required when the Serial Protocol parameter is set to SMDI and the Media Gateway is the Serial Protocol Master.
Parameter Reference 3.16.9 MWI Response Timeout Description: Specifies the time in milliseconds the Media Gateway Serial protocol Master will wait for a failed response from the PBX before sending a success message to a Serial protocol Slave in response to a message waiting indicator (MWI) request from the slave. This parameter is only required when this Media Gateway is configured as the Serial Protocol Master.
Parameter Reference 3.17 Tone Detection Parameters The Tone Detection parameters define the characteristics (frequencies, durations, and deviations) of the tones that the Media Gateway detects during call progress analysis. These call progress tones can be viewed and edited manually from the Web interface if the characteristics are known, or the Media Gateway can be directed to analyze and learn the characteristics of specific call progress tones.
Parameter Reference 3.17.0.1 ID Description: Automatically assigned IDs for reference. These IDs cannot be edited. 3.17.0.2 Tone Event Description: Specifies the tone event for this tone definition. Allowed Values: • None • Dialtone • Ringback • Busy • Congestion • Disconnect • Error • SIT • Pager • Modem • Fax • FaxCNG Default Value = None INI File Parameter Name = cpToneEvent 3.17.0.3 Tone Name Description: Specifies the text name associated with this tone definition.
Parameter Reference 3.17.0.4 Cadence Type Description: Specifies the cadence type for this tone definition.
Parameter Reference 3.17.0.7 Tone Frequency Deviation Description: Specifies the deviation for the tone frequencies for this tone definition. Allowed Values: 0-200 Hz Default Value = 0 INI File Parameter Name = cpToneFreqDeviation 3.17.0.8 Tone Cadence Time Description: Specifies the cadence times for this tone definition. A tone can be described by up to three (3) cadence times. The interpretation of each time parameter is dependent on the selection of Cadence Type.
Parameter Reference 3.17.1 Tone Generation Configuration Parameters 3.17.1.1 Call Progress Tone Generation Event Description: The Tone event or type. The tone will be interpreted as this type of call progress tone generation. Allowed Values: The configurable tone events are Dialtone, Busy, Ringback, Congestion, Error, Disconnect, Intercept, Reorder, and Special. INI File Parameter Name = cpGenToneEvent Note: 3.17.1.2 This is a *.ini file parameter only.
Parameter Reference 3.17.1.5 Call Progress Tone Generation Frequency 2 (Hz) Description: Frequency 2 contained within the tone. Allowed Values: Number between 0 – 3000. A value of 0 means to ignore the entry. INI File Parameter Name = cpGenToneFreqHz2 Note: 3.17.1.6 This is a *.ini file parameter only. Call Progress Tone Generation Frequency 3 (Hz) Description: Frequency 3 contained within the tone. Allowed Values: Number between 0 – 3000. A value of 0 means to ignore the entry.
Parameter Reference 3.17.1.10 Call Progress Tone Generation Amplitude 3 (dBm) Description: Amplitude for frequency 3. Allowed Values: Number between -80 - 3. INI File Parameter Name = cpGenToneAmpDbm3 Note: 3.17.1.11 This is a *.ini file parameter only. Call Progress Tone Generation Amplitude 4 (dBm) Description: Amplitude for frequency 4. Allowed Values: Number between -80 - 3. INI File Parameter Name = cpGenToneAmpDbm4 Note: 3.17.1.12 This is a *.ini file parameter only.
Parameter Reference 3.17.2 Editing the INI File Directly The call progress tone parameters can be edited directly in the INI file if desired. In older versions of the INI file, 16 tone definitions were listed regardless of the number of actual tone definitions in use. In these older INI files unused tone definitions would have the cpToneEvent parameter set to none. In newer versions of the INI file only the tone definitions in use are listed.
Parameter Reference 3.18.1.2 HTTPS Certificate Type Description: Determines whether the Media Gateway uses a self-assigned or CA signed certificate for HTTP over TLS (HTTPS). Allowed Values: • Self Signed = Use the self-signed TLS certificate • CA Signed = Use the CA-signed TLS certificate Default Value = Self Signed INI File Parameter Name = secSipTlsUseSelfSignedCert Note: 3.19 Unit requires a restart if this parameter value is changed.
Parameter Reference 3.19.
Parameter Reference 3.19.1.1 TDM to IP Gain Adjustment (DMG1000 Only) Description: Adjusts the gain of the audio signal in the Telephony-to-IP direction. Allowed Values: -14 dB to +14 dB Default Value = 0 dB INI File Parameter Name = dspPbxToIPGain 3.19.1.2 IP to TDM Gain Adjustment (DMG1000 Only) Description: Adjusts the gain of the audio signal in the IP-to-Telephony direction. Allowed Values: -14 dB to +14 dB Default Value = 0 dB INI File Parameter Name = dspIPtoPbxGain 3.19.1.
Parameter Reference 3.19.1.5 Voice Activity Noise Floor Description: Defines the noise floor for the voice activity detector (VAD). Signal levels below the value selected will be treated as silence by the VAD. Allowed Values: -80 dB to -10 dB Default Value = -40 dB (DMG1000) -32 dB (DMG2000) INI File Parameter Name = vadNoiseFloor Note: 3.19.1.6 Unit requires a restart if this parameter value is changed.
Parameter Reference 3.19.1.9 Call Progress Filter Debounce Description: Specifies the debounce time in milliseconds for the Call Progress Tone Detector. This value defines the minimum time that a call progress signal must be present before the detector is triggered. Allowed Values: 100 to 32768 milliseconds Default Value = 100 milliseconds INI File Parameter Name = cpFilterDebounce Note: 3.19.1.10 Unit requires a restart if this parameter value is changed.
Parameter Reference 3.19.1.12 Call Progress Filter High Cutoff (DMG1000 Only) Description: Specifies the high cut-off frequency for the Call Progress Tone Detector. Call progress signals above this frequency will be ignored. Allowed Values: 500 Hz to 2000 Hz Default Value = 650 Hz INI File Parameter Name = cpFilterHighCutoff Note: 3.19.1.13 Unit requires a restart if this parameter value is changed.
Parameter Reference 3.19.1.15 Jitter-Buffer Minimum Delay (DMG2000 Only) Description: Specifies the minimum jitter-buffer delay. The smaller the delay, the smaller the jitter-buffer. It is recommended that this value be at least the duration of one RTP packet interval. If the minimum, maximum, and initial jitter-buffer delays are set to the same value, the jitter-buffer will operate in the non-adaptive mode.
Parameter Reference 3.19.1.19 Jitter-Buffer Deletion Threshold (DMG2000 Only) Description: When the jitter-buffer grows past this value, frames exceeding the threshold are deleted immediately. Allowed Values: 0 to 500 milliseconds Default Value = 500 milliseconds INI File Parameter Name = dspJbDeletionThresholdMs 3.19.1.20 Jitter-Buffer Frame Deletion Mode (DMG2000 Only) Description: Determines how frames are deleted when the jitter-buffer adapts downward.
Parameter Reference 3.19.1.22 VoIP Packet Loss Concealment (DMG2000 Only) Description: Enables packet loss concealment (PLC). The gateway tries to account for missed media packets. It may attempt generating voice samples based on prior packets for short audio drops. When used in conjunction with IVR systems, PLC may cause unwanted artifacts at the end of voice prompts, in which case PLC should be turned Off. Allowed Values: • On • Off Default Value = On INI File Parameter Name = dspPktLossConclmt 3.19.
Parameter Reference 3.19.1.25 IP to TDM AGC Target Level (DMG2000 Only) Description: Defines the IP to TDM AGC target level for a range of -50 to 0 dBm. Allowed Values: -50 to 0 dBm Default Value = -14 dBm INI File Parameter Name = dspIpToPbxAgcLevel 3.19.1.26 IP to TDM AGC Max Gain (DMG2000 Only) Description: Defines the IP to TDM AGC maximum gain within the range of 0 to 15 dB. Allowed Values: 0 to 15 dB Default Value = 12 dB INI File Parameter Name = dspIpToPbxAgcMaxGain 3.19.1.
Parameter Reference 3.19.1.29 TDM to IP AGC Slew Rate (DMG2000 Only) Description: Determines the speed at which the AGC adapts in the TDM to IP direction. Allowed Values: • Fast = More emphasis on AGC gain adjustments • Medium = Mid point • Slow = More emphasis on Audio Quality Default Value = Medium INI File Parameter Name = dspPbxToIpAgcSlewRate 3.19.1.30 TDM to IP AGC Target Level (DMG2000 Only) Description: Defines the TDM to IP AGC target level for a range of -50 to 0 dBm.
Parameter Reference 3.19.2 T.38 Fax Advanced Settings The T.38 Fax Advanced Settings parameters include: • Allow T.38 ECM Faxes (DMG2000 Only) • Transmit Small T4 ECM T.38 Packets (DMG2000 Only) • Enable T.38 Spoofing (DMG2000 Only) • TSI Removal (DMG2000 Only) • CSI Removal (DMG2000 Only) • NSF Removal (DMG2000 Only) • DIS Removal (DMG2000 Only) • T.38 Packet Loss Concealment Method (DMG2000 Only) • T.38 UDPTL Redundancy Count for Fax Page Data (DMG2000 Only) • T.38 UDPTL Redundancy Count for T.
Parameter Reference 3.19.2.3 Enable T.38 Spoofing (DMG2000 Only) Description: Enables/disables spoofing in T.38. Allowed Values: • On = Enable T.38 spoofing. • Off = Disable T.38 spoofing. Default Value = On INI File Parameter Name = dspT38SpoofingEnable 3.19.2.4 TSI Removal (DMG2000 Only) Description: Determines if the Transmitting Subscriber Identification (TSI) is removed. Allowed Values: • On - Remove TSI from TDM to reduce packet delay. • Off - Include TSI in packet sent to packet network.
Parameter Reference 3.19.2.6 NSF Removal (DMG2000 Only) Description: Determines if Non-Standard Facilities (NSF) is removed. Allowed Values: • On - Remove NSF from TDM to reduce packet delay. • Off - Include NSF in packet sent to packet network. Default Value = Off INI File Parameter Name = dspT38NsfRemovalEnable 3.19.2.7 DIS Removal (DMG2000 Only) Description: Determines if the Digital Identification signal (DIS) is removed. Allowed Values: • On - Remove DIS from TDM to reduce packet delay.
Parameter Reference 3.19.2.10 T.38 UDPTL Redundancy Count for T.30 Messages (DMG2000 Only) Description: Specifies the number of redundant IFP frames transferred with each UDPTL packet. Allowed Values: 0 to 3 Default Value = 2 INI File Parameter Name = dspUdptlT30Redundancy 3.19.2.11 Fax Transmit Level (dBm) (DMG2000 Only) Description: Set the fax transmit level in T.38 fax relay mode. Allowed Values: -15 to 0 Default Value = -10 INI File Parameter Name = dspT38FaxTransmitLevel 3.19.2.
Parameter Reference 3.19.3 Positive Answer Machine Detection The Positive Answer Machine Detection parameters include: • Maximum Live Answer Time • Minimum Live Answer Time • Maximum Silence after Voice has been Detected • Maximum Time to Wait for Voice 3.19.3.1 Maximum Live Answer Time Description: Sets the time limit at which point a continuous voice answer will be considered an answer machine.
Parameter Reference Allowed Values: 1000 to 20000 milliseconds Default Value = 4000 milliseconds INI File Parameter Name = pamdFailSafeTimeout 3.20 Non-Menu (Hidden) Parameters The following parameters do not appear in the Web interface and can only be changed by editing the configuration file directly. These parameters are listed alphabetically.
Parameter Reference 3.20.2 Incompatible Message STATUS Description: If 'Yes' then a STATUS message (with cause #98) will be sent if an incompatible message (for the current call state) is received. If 'No' then a STATUS message will not be sent. Applicable only when ISDN Protocol is QSIG or ETSI. Note: This parameter can only be changed in the configuration file. It is not accessible through the Web interface.
Parameter Reference 3.20.5 ISDN Overlap Receive Minimum Digits Description: Specifies the minimum number of called-party digits that must be received on an inbound call before the inbound call is processed. If the value is 0, then the call is processed immediately. Applicable only when the switch is using overlap sending. Note: This parameter can only be changed in the configuration file. It is not accessible through the Web interface.
Parameter Reference 3.20.8 Routing Proxy Group Name Description: Specifies the routing proxy group name. Note: This parameter can only be changed in the configuration file. It is not accessible through the Web interface. Allowed Values: 0 to 39 characters Default Value = blank INI File Parameter Name = gwRoutingProxyGroupName 3.20.9 Routing Table Only On Proxy Down Description: If a VoIP proxy server is available, then all inbound TDM calls and messages will be routed to the VoIP proxy server.
Parameter Reference 182 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Call Progress Tones 4 Call progress tone parameters define the characteristics (frequencies, durations, and deviations) of the tones that the Dialogic® Media Gateway detects during call progress analysis. The following sections discuss how to view, edit, learn, and validate call progress tones using a Web interface: • Viewing and Editing Call Progress Tones. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183 • Learning and Validating Call Progress Tones . . . . . . . . . . . . . . . .
Call Progress Tones You can also add and delete call progress tones from this Web page. Use the following buttons on the Manual Tones Web page to manipulate call progress tones: Apply Changes button Saves changes to the configuration once the parameters have been edited. Add button Adds a row to the tone table, allowing a new tone definition to be added. Reset to Default Values button Deletes all existing tones and restores factory default tones.
Call Progress Tones Figure 21. Learn Tone Web Page The Learn Tone user interface has the following components of interest: Acquire Tone field A check indicates that the corresponding tone event is to be learned or validated. Tone Event field Indicates one of four types: dialtone, busy, error, or ringback. Destination Address field Specifies the phone number or extension that when dialed will generate the call progress tone associated with the tone event.
Call Progress Tones • Ringback – Call a valid extension not in use. Caution: Make sure that this extension is not forwarded on “no answer” and that it will ring until answered. Learn button Specifies that the Media Gateway will automatically learn the tone. Used when the frequency and timing information of a call progress tone are unknown. For more information on using the Learn feature, see Section 4.2.2, “Learning the Characteristics of Unknown Call Progress Tones”, on page 186.
Call Progress Tones Progress messages are described as follows: Learning The Gateway has gone off hook and dialed the destination address. It is collecting raw data on the call progress tone. Analyzing The Gateway is analyzing the raw data to extract the tone characteristics. Comparing The Gateway is comparing the new tones to existing tones and other new tones for possible conflicts.
Call Progress Tones Failed Validation unexpected CP tone The Gateway dialed the destination address and was able to extract tone characteristics, but when the Gateway dialed the destination address to validate the newly learned tone, an unexpected call progress (CP) tone was detected. The detected CP tone is listed in the error message. This detected CP tone did not match the newly learned tone or any expected conflicts.
Call Progress Tones bandwidth is specified as all of the frequencies from the specified frequency minus the frequency deviation to the specified frequency plus the frequency deviation. For example, if the frequency is specified as 400 Hz and the frequency deviation is specified as 50 Hz then the bandwidth is 350 Hz to 450 Hz. The frequencies of tone definitions are considered to match if their bandwidths overlap.
Call Progress Tones The validation process is useful when there was an error in learning new tones and some tone characteristics have been manually adjusted to remove conflicts. The Validate Only button can be used to validate the changes. Follow these steps to validate the timing and frequency characteristics of call progress tones. 4.2.6 1. For each call progress tone that is to be validated, determine how that tone can be generated by the system. 2.
Call Progress Tones Validation Errors An error occurred when trying to validate the tones listed in this section. These tones cannot be added to the configuration due to the error encountered. Possible reasons for error include: Failed Validation unexpected CP tone When the Gateway dialed the destination address to validate the tone, an unexpected call progress (CP) tone was detected. The detected CP tone is listed in the error message.
Call Progress Tones 192 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Routing Table 5 Information about Routing Table and how it is supported by the Dialogic® Media Gateway is described in the following sections: • Routing Table Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193 • Router Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195 • Offline Testing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Routing Table Figure 22 below shows how this information enters the Routing Table control. Figure 22. VoIP to TDM calls Calling Number: 101 Calling Number Dial Plan Called Number: 8675309 VoIP Address: 172.16.3.131 Called Number Physical TDM channel The Routing Table configuration will determine what calling and called numbers are sent to the TDM device, as well as the physical TDM destination. 5.1.
Routing Table 5.2 Router Configuration The Router Configuration manages the configuration of the Routing Table for the gateway. The Routing Table defines how all telephony requests are routed through the gateway between networks. It also defines the call-party information manipulation rules for passing call-party information between networks. TDM Trunk Groups and VoIP Host Groups should be created first, as they are referenced by both the Inbound TDM and Inbound VoIP routing rules.
Routing Table Figure 24 below shows call routing flow for Routing Table. Figure 24. Routing Table Call Routing Flow Incoming Call Get next rule in table Do incoming call caracteristics match the rule? no yes yes Apply CPID manipulation Route Call Are there any more rules in the table? no Do Not route call If no match is found, the call is not routed by the Routing Table. 5.2.2 Inbound TDM Rules Table The Inbound TDM Rules is the main table used to route calls originating from the TDM interface.
Routing Table 5.2.2.1 Inbound TDM Rules Use this table to define the routing rules for all telephony requests that are inbound from the TDM network. Rules are evaluated from the top down until a matching rule is found. Figure 25 is a screen shot of the Inbound TDM Rules configuration Web page: Figure 25. Inbound TDM Rules Configuration Web Page Select to configure the routing rules for requests that are inbound from the TDM network. Select - Click anywhere in this column to select an entry.
Routing Table 5.2.2.2 Inbound TDM Request Matching Defines the call-party information expressions that are used to match an inbound telephony request from the TDM network to this routing rule. The Inbound TDM Request Matching is performed when the inbound TDM request matches the Request Type and Trunk Group of one of the defined rules. CPID Matching Part of the criteria for matching the characteristics of an incoming call to a rule in the Routing Table is number matching.
Routing Table Figure 26 is a screen shot of the CPID Matching configuration Web page: Figure 26. CPID Matching Configuration Web Page Calling Party Number - The formula as specified in the section above that is used to match the calling number of the incoming call. Calling Party Name- The formula as specified in the section above that is used to match the calling name of the incoming call.
Routing Table Figure 27 is a screen shot of the Call Type Property Matching configuration Web page: Figure 27. Call Type Property Matching Configuration Web Page Calling Party Numbering Plan - Numbering Plan match for Calling Number (ISDN only). – – – – – – – – Any Unknown ISDN_E.164 Data_X.121 Telex_F.69 Standard Private Reserved Calling Party Type of Number - Type of Number match for Calling Number (ISDN only).
Routing Table Called Party Numbering Plan - Numbering Plan match for Called Number (ISDN only). – – – – – – – – Any Unknown ISDN_E.164 Data_X.121 Telex_F.69 Standard Private Reserved Called Party Type of Number - Type of Number match for Called Number (ISDN only).
Routing Table Redirecting Party Type of Number - Type of Number match for Redirect Number (ISDN only). – – – – – – – – – – – – Any Unknown International National Network Subscriber Abbreviated Reserved Level2Regional Level1Regional PISNSpecific Level0Regional Redirecting Party Presentation Indicator - Presentation Indicator match for Redirect Number (ISDN only). – – – – – 5.2.2.
Routing Table Figure 28 is a screen shot of the Device Selection configuration Web page: Figure 28. Device Selection Configuration Web Page Outbound Destination - Specifies the network to which the request is to be routed: – VoIP = Route to the VoIP network. – TDM = Route to the TDM network. – Blocked = Request is not routed. The inbound request will be immediately rejected. Host Group - Name of VoIP Host Group to which the request is to be routed.
Routing Table Table 11 shows the syntax used for CPID Manipulation. Table 11.
Routing Table Name Manipulation Rule Syntax Description Example Example Result S Source (calling) number S Amanda D Destination (called) number D Sydney R Redirection number R Dave S# Source (calling) number S# 7168675309 D# Destination (called) number D# 5551212 R# Redirection number R# 4815162342 S@ Source (calling) name S@ Amanda D@ Destination (called) name D@ Sydney R@ Redirection name R@ Dave “” Takes what’s in quotes as literal “353” 353 + Concatenate “8
Routing Table Figure 29 is a screen shot of the CPID Manipulation configuration Web page: Figure 29. CPID Manipulation Configuration Web Page Calling Party Number - The formula as specified in the section above that determines the calling number of the outgoing call. Calling Party Name- The formula as specified in the section above that determines the calling name of the outgoing call.
Routing Table 5.2.3 Inbound VoIP Rules Table The Inbound VoIP Rules is the main table used to route calls originating from the VoIP interface. When a call is received, the incoming call characteristics are compared against the values entered in the first row of the table. If a match is found, an action is taken. If no match is found, the next row in the table is checked. Rows will be tested until a match is found, or there are no more entries in the table.
Routing Table Rule Label - Specifies label for this entry. This label is used only to help identify the rule to the administrator. It is not used by any other tables in the Routing Table configuration. Request Type - Specifies the type of telephony request to which the rule applies: – Any = All telephony request types. – Call = Calls only. – Message = Message-waiting notifications only. Originating VoIP Host Address - Specifies the VoIP hosts from which requests are accepted for use by the rule.
Routing Table Table 13 shows the syntax used for CPID Matching. Table 13. Syntax Used for CPID Matching Number Matching Token Description * Matches any number string when entered alone. 0123456789 Identifies a specific digit. [digit_string-digit_string] Specifies a range of digit strings. Identifies any digit string that is included in the range. The range delimiters must both contain the same number of digits. A digit string must be no more than 9 digits in length.
Routing Table Called Party Name - The formula as specified in the section above that is used to match the called name of the incoming call. Redirecting Party Number - The formula as specified in the section above that is used to match the redirect number of the incoming call. Redirect Party Name - The formula as specified in the section above that is used to match the redirect name of the incoming call. 5.2.3.3 Outbound Routes Defines how the request is to be routed by the gateway.
Routing Table CPID Manipulation CPID Manipulation defines rules for how Calling Number, Calling Name, Called Number, Called Name, Redirect Number, and Redirect Name appear in the destination call. Typically, the calling and called numbers from the incoming call are simply passed through to the outgoing call. However, there are cases where it is desirable to alter either or both of these numbers. A “CPID Manipulation” rule is applied to all calls routed via the Routing Table.
Routing Table Name Manipulation Rule Syntax Description Example Example Result S Source (calling) number S Amanda D Destination (called) number D Sydney R Redirection number R Dave S# Source (calling) number S# 7168675309 D# Destination (called) number D# 5551212 R# Redirection number R# 4815162342 S@ Source (calling) name S@ Amanda D@ Destination (called) name D@ Sydney R@ Redirection name R@ Dave “” Takes what’s in quotes as literal “353” 353 + Concatenate “8
Routing Table Called Party Number - The formula as specified in the section above that determines the called number of the outgoing call. Called Party Name - The formula as specified in the section above that determines the called name of the outgoing call. Redirecting Party Number - The formula as specified in the section above that determines the redirect number of the outgoing call.
Routing Table 5.2.4.1 TDM Port Types The gateway supports multiple types of TDM ports. The number of physical ports and channels supported vary. Table 15 shows the possibilities: Table 15. TDM Port Types Maximum number of supported ports Gateway Type Number of supported channels per port Analog 8 1 Digital (PBX) 8 1 T1 - CAS 4 24 T1 - ISDN 4 23 E1 4 30 Figure 36 is a sample screen shot of the TDM Trunk Groups configuration Web page: Figure 36.
Routing Table Selection Mode - Specifies the method of channel selection when selecting channels from the trunk group for outbound requests: Linear • For gateways that have multiple channels per port, the highest free channel on the highest port is used for the next call (Descending) or the lowest free channel on the lowest port is used for the next call (Ascending).
Routing Table 5.2.5 VoIP Host Groups Table The configuration of VoIP Host Groups are used for routing requests to the VoIP network. VoIP Host Groups are used to specify outbound routes to the VoIP network. Figure 37 is a screen shot of the VoIP Host Groups configuration Web page: Figure 37. VoIP Host Groups Configuration Web Page Name - Defines the name of the host group. This name is referenced by routing rules that use the VoIP host group.
Routing Table 5.2.5.1 Host List Use this table to Add hosts to this group and Delete hosts from this group. Figure 38 is a screen shot of the Host List configuration Web page: Figure 38. Host List Configuration Web Page 5.3 Offline Testing An offline facility exists to test the Routing Table without receiving actual calls. To use this feature, navigate to the Tests-->Router link via the Media Gateway’s user interface. Two tables exist in this page.
Routing Table Figure 39.
Routing Table Figure 40.
Routing Table 5.4 Call Routing Examples Each example presents the requirements (i.e. how you want the calls routed) and the steps required to setup the routing table to meet the requirements. The examples are presented in increasing complexity and build on each other. The explanations presented in one example are not repeated in subsequent ones.
Routing Table Example #1 – Basic TDM to/from VoIP System Configuration: Central Office T1 PBX T1 Gateway Media Server WAN ... Requirements: 1. All inbound TDM calls must be routed to the media server (IP address 172.13.4.3). 2. All inbound VoIP calls must be routed out the TDM. Any available TDM channel can be used. Steps: 1. 2. Determine the required TDM Trunk Groups and the VoIP Host Groups. • From requirement #1 we have a TDM trunk group of “all TDM requests” and a VoIP host group of “172.
Routing Table 4. Set up Inbound TDM Rules. Inbound TDM Rules Enable X Rule Label Request Type All Inbound TDM Any Trunk Group All TDM Requests Inbound TDM Request Matching CPID Matching Calling Number * Called Number * Redirect Number * Calling Name * Called Name * Redirect Name * Outbound Routes Device Selection Outbound Destination Host Group VoIP All to 172.13.4.
Routing Table 5. Set up Inbound VoIP Rules.
Routing Table 6. Validate Routing Table. The Routing Table can be validated using the Tests -> Router web page. • Navigate to Tests -> Router and enter data as per the following table. • Inbound TDM Request Type Port Channel Calling Number Calling Name Called Number Called Name Redirect Number Redirect Name • Click on “Simulate Route”.
Routing Table Input: Inbound Route o Inbound TDM Request Type Host IP Port Calling Number Calling Name Called Number Called Name Redirect Number Redirect Name • Inbound VoIP Call 172.13.4.
Routing Table Example #2 – Basic Call with Load Balancing System Configuration: Requirements: 1. All inbound TDM calls must be routed to one of the three Media Servers using Load Balancing. Media Server 1 is 172.16.3.1, Media Server 2 is 172.16.3.2 and Media Server 3 is 172.16.3.3. 2. All inbound VoIP calls must be routed out the TDM. Any available TDM channel can be used. Steps: 1. 2. Determine the required TDM Trunk Groups and the VoIP Host Groups.
Routing Table 4. Set up Inbound TDM Rules.
Routing Table 5. Set up Inbound VoIP Rules.
Routing Table Example #3 – Basic Call with CPID manipulation System Configuration: Central Office T1 PBX T1 Gateway Media Server WAN ... Requirements: Routing Requirements: 1. All inbound TDM calls must be routed to the media server (IP address 172.13.4.3). 2. All inbound VoIP calls must be routed out the TDM. Any available TDM channel can be used. CPID manipulation Requirements: Inbound VoIP calls: • The VoIP calls use E.164 format. We must translate from E.
Routing Table 4. Set up Inbound TDM Rules. There is a rule for every CPID manipulation. Inbound TDM Rule #1: CPID Manipulation: CPID of the form YYYxxxxxxx must be changed to +YYYxxxxxxx and CPID of the form ZZxxxxxxxx must be changed to ZZxxxxxxxx both map to the same rule. That is, prefix the number with a plus sign.
Routing Table 5. Set up Inbound VoIP Rules. There is a rule for every CPID manipulation plus an optional catch all. Inbound VoIP Rule #1: CPID Manipulation: CPID of the form +1YYYxxxxxxx must be changed to 91YYYxxxxxxx. Inbound VoIP Rules Enable Rule Label Request Type Country Code One (1) X Any Originating VoIP Host Address * Inbound VoIP Request Matching CPID Matching Calling Number +1. Called Number +1.
Routing Table Inbound VoIP Rule #2: CPID Manipulation: CPID of the form +ZZxxxxxxxx must be changed to 9011ZZxxxxxxxx. Inbound VoIP Rules Enable Rule Label Request Type Originating VoIP Host Address X Country Code One (1) Any * X Other Country Codes Any * Inbound VoIP Request Matching CPID Matching Calling Number * Called Number +.
Routing Table Inbound VoIP Rule #3: An optional Catch All rule. This rule does not correspond to any required CPID manipulation. It is a catch all rule that is invoked if all the previous rules fail. This rule is optional but if you do not use a catch all rule, any inbound VoIP calls that do not match a rule are dropped.
Routing Table 6. Validate the Routing Table. • For Inbound TDM CPID of the form 1YYYxxxxxxx is changed to +1YYYxxxxxxx. Input: • Inbound TDM Request Type Port Channel Calling Number Calling Name Called Number Called Name Redirect Number Redirect Name Inbound Route o Inbound VoIP Call 1 1 5553000 9145552345 Output: Results Reason Inbound Rule Outbound Group VoIP Bridged 172.13.4.
Routing Table • For Inbound VoIP CPID of the form +1YYYxxxxxxx is changed to 91YYYxxxxxxx. Input: Inbound Route o Inbound TDM Request Type Host IP Port Calling Number Calling Name Called Number Called Name Redirect Number Redirect Name • Inbound VoIP Call 172.16.5.
Routing Table Example #4 – Basic Call with MWI System Configuration: Requirements: 1. All inbound TDM requests must be routed to the media server (IP address 10.1.1.1). 2. All inbound VoIP calls must be routed out the TDM. Only TDM ports 1-6 can be used for calls. 3. All inbound VoIP MWIs (Message Waiting Indications) must be routed out the TDM. Only TDM ports 7 and 8 can be used for MWIs. Steps: 1. 2. Determine the required TDM Trunk Groups and the VoIP Host Groups.
Routing Table 4. Set up Inbound TDM Rules. Inbound TDM Rule #1: For Inbound TDM Requests (requirement #1). Inbound TDM Rules Enable X Rule Label Request Type All TDM Requests Any Trunk Group All TDM Requests Inbound TDM Request Matching CPID Matching Calling Number * Called Number * Redirect Number * Calling Name * Called Name * Redirect Name * Outbound Routes Device Selection Outbound Destination Host Group VoIP All to 10.1.1.
Routing Table 5. Set up Inbound VoIP Rules. Inbound VoIP Rule #1: For inbound VoIP calls (requirement #2).
Routing Table Inbound VoIP Rule #2: For inbound VoIP MWI requests (requirement #3).
Routing Table Example #5 – Basic Calls with Proxy This example is the same as Example #1 except a Proxy is used. Physical System Configuration verses Logical System Configuration: The proxy is connected to the same network as the Media Server. However, logically the proxy sits between the Gateway and the Media Server. That is, all messages from the Gateway go to the proxy and the proxy retransmits the messages to the Media Server.
Routing Table Example #6 – Toll Bypass Toll Bypass allows a company to make interoffice calls using the company’s WAN and without using the public telephone network – reducing the cost of phone calls. System Configuration: The sample system configuration shows three offices. We named them New York City (NYC), Los Angeles (LA) and Seattle to make the example more real-world. Each office has a number of phones connected to an internal PBX.
Routing Table Routing Requirements: New York City Gateway: 1. All inbound VoIP calls must be routed out the TDM span connected to the PBX. 2. All inbound TDM calls from the span connected to the CO must be routed out the TDM span connected to the PBX. 3. All inbound TDM calls from the span connected to the PBX and with a destination of 2xxx must be routed out the VoIP to the Los Angeles gateway (172.16.5.3). 4.
Routing Table 3. Set up VoIP Host Groups. Name Load Balanced Los Angeles Gateway 4. false Fault Tolerant Host Summary false 172.16.5.3 Set up Inbound TDM Rules. Inbound TDM Rule #1: All inbound TDM calls from the span connected to the CO must be routed out the TDM span connected to the PBX (routing requirement #2).
Routing Table Inbound TDM Rule #2: All inbound TDM calls from the span connected to the PBX and with a destination of 2xxx must be routed out the VoIP to the Los Angeles gateway (172.16.5.3) (requirement #3). Inbound TDM Rules Enable X X Rule Label Request Type Calls From CO Calls From PBX To LA Call Call Trunk Group Span to CO Span to PBX Inbound TDM Request Matching CPID Matching Calling Number * Called Number 2.
Routing Table Inbound TDM Rule #3: All other inbound TDM calls from the span connected to the PBX must be routed out the TDM span connected to the CO (requirement #4).
Routing Table 5. Set up Inbound VoIP Rules. Inbound VoIP Rule #1: All inbound VoIP calls must be routed out the TDM span connected to the PBX (requirement #1).
Media Gateway Parsers 6 This section describes the Dialogic® Media Gateway in-band Type I (on-hook) and Type II (offhook) integration parsers for analog and T1 CAS integrations, and the display parsers for digital integrations. These parsers allow the user to define the meaning of either the in-band/on-hook integration strings or display strings received from the telephony network. For this discussion, the term parser will be used when referencing all three variants listed above.
Media Gateway Parsers Figure 41. Default Analog CPID Configuration Data on Analog Web Page 6.2 Parsing Configuration Syntax The configuration data syntax is used to describe the type I or II CPID (analog or T1 CAS) or display (digital) strings that may be received from the telephone network. The type of information that can be received using analog DTMF integrations is a subset of that which can be gleaned from digital displays.
Media Gateway Parsers Figure 42. Default Analog CPID Configuration in the .ini File ;Version Information: ;MAC:00-a0-e6-06-05-07 ;IP:10.0.1.41 ;DSP Firmware: |9.1w/Fax|FRI MAY 02 17:59:51 2003| ;DSP Firmware (ROM): 9.1 w/Fax|FRI MAY 02 17:59:51 2003 ;Main Board Boot (ROM): |4.
Media Gateway Parsers Figure 43. Sample Analog Type II CPID Configuration Data in the .
Media Gateway Parsers Figure 44. Default Mitel Digital CPID Configuration Data (cpid.htm) The parser attempts to match an input integration or display string to a rule defined by the configuration data. If a match is made, the parser uses the call-party, reason code, and origin specifiers of the matching rule. In this manner, the parser can extract the call party information from the correct locations of the integration or display strings. 6.2.
Media Gateway Parsers The above provides a translation of the switch-specific reason code “FWD” to the application string “fwd-all”. If the token “FWD” is found in the reason-code section of a display, the reason code presented to the application will be “fwd-all”. This provides a means of normalizing different switch-specific reason codes into strings that can be recognized by the application. The following is an origin example.
Media Gateway Parsers Table 16. Parser Regular Expressions (Continued) Characters and Meta-Characters \nnn (min,max) Purpose Analog Parsing Applicable Octal control character. No Braces. Follows other chars. If non-null, length must be within min/max. Yes Rule Syntax A call-class rule starts with the tag “rule”. All characters following the tag define the rule. Following the rule are specifiers that define the location in the integration or display string of the call party information.
Media Gateway Parsers Rule Order The parser attempts to match an input integration or display string with a rule contained in the configuration data. The rules are compared to the specified string from the top rule to the bottom rule. Because of this, rules that contain the most specific information should be listed in the configuration file first. In this way, exact matches can occur before a more generic 'catch-all' rule is reached.
Data Security 7 Information about data security and how it is supported by the Dialogic® Media Gateway is described in the following sections: • Data Security Overview. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255 • Secure HTTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255 • SIP Call Control Security using TLS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Data Security • When accessing the Media Gateway, use https:// instead of the non-secure http:// followed by the Media Gateway's URL. This section includes the following information about HTTP security: • HTTPS Certificate Configuration • HTTPS Example 7.2.1 HTTPS Certificate Configuration An HTTPS certificate can be either self-signed or certificate authority (CA) signed. A self-signed certificate can be generated by the Media Gateway.
Data Security 7.2.2 HTTPS Example An example of how HTTPS is used with a self-signed certificate is described below. In the example, the Media Gateway has an IP address of 172.16.3.10 and uses a self-signed certificate. 1. Start the Media Gateway. 2. Start Internet Explorer (or any Web browser that supports HTTPS). 3. In the Web browser Address box, enter http://172.16.3.10. 4. At the login screen, enter a User name and Password, and click OK to login to the Media Gateway. 5.
Data Security 7.3.1 TLS Certificate Configuration A TLS certificate can be self-signed or certificate authority (CA) signed. A self-signed certificate can be generated by the Media Gateway. CA signed certificates must be requested by the Media Gateway and signed by a CA. When using a self-signed certificate: • The Media Gateway generates a certificate which will be installed on VoIP devices that will communicate with the Media Gateway via TLS.
Data Security • TLS Cipher List - The Cipher list is not a configurable parameter. The Media Gateway supports 6 ciphers in a list but cannot be changed. Valid OpenSSL ciphers can be found at: http://www.openssl.org/docs/apps/ciphers.html A default cipher list must be specified for TLS to work. The Media Gateway uses the default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH. • SSL TLS Protocol - This parameter specifies the SSL record type to be used with the TLS connections and can be set to use SSLv3 and/or TLSv1.
Data Security • Verify TLS Peer Certificate trust: No Example 2: Assume a company has five sites: one in Indiana, one in Illinois, one in California, one in New York, and one in Washington. Each of these sites uses a local telephone company. People at different sites frequently call each other. They have decided to use VoIP service with one Media Gateway at each site. They also use one media server. A CA signed certificate, which will expire in 1 year, is used. There is no convenient SNTP server.
Data Security for secure audio, and the gateway will accept requests for both secure and non-secure audio. If RTP_Only is specified, the gateway will only request non-secure audio and will only accept requests for non-secure audio. Note: SRTP_Preferred is not supported in Version 5.1 SU2 Software. • Master Key Index (MKI) on Transmit Stream - With SRTP, audio data for a conversation is encrypted using a key that is called the session key.
Data Security The Media Gateway should be configured as follows: • SRTP Preference: SRTP Only • Cipher Mode: AES Counter Mode • Authentication Type: SHA1 • Authentication Tag Length: SHA1 32 bits or 80 bits • Master Key Index (MKI) on Transmit Stream: No • Key Derivation Enable: No • Key Derivation Rate (KDR): Not applicable Example 2: In this example, the Media Gateway is talking to an IP phone that does not support security: • SRTP Preference: RTP Only • Master Key Index (MKI) on Transmit Stream: Not ap
Data Security Figure 45. Storing Self-Signed Certificate by Certificate Import Wizard 11. If the Certificate store is not Trusted Root Certification Authorities, use the Browse button to locate it. 12. Click Next. 13. Click Finish. 14. Click Yes. 15. Click the OK button. 16. Close the windows and Internet Explorer.
Data Security 264 Dialogic® 1000 and 2000 Media Gateway Series User’s Guide
Unit Status 8 Status information about the Dialogic® 1000 Media Gateway (DMG1000) and Dialogic® 2000 Media Gateway (DMG2000) units can be obtained through the Web interface. The DMG1000 Status and DMG2000 Status Web pages allow an administrator to view the number of calls processed by the unit as well as a log of all calls (with call party information). The Media Gateway supports SNMP Version 1. The Media Gateway-specific SNMP MIB can be downloaded from the unit using the Web interface. The MIB file dmg.
Unit Status 8.2 Alarm Information Selecting Alarm from the Status menu displays the following types of alarm information: • Time- Timestamp of alarm entry. • ID - Identifier of alarm entry. • Severity - Severity Level of alarm entry. • Description - Text Description of alarm entry. • Port - Port Number of alarm entry. • Text Message - Text message of alarm entry. Clicking on the Clear Alarms button will clear all alarm information. 8.
Unit Status 8.4 Telephony Status Information Selecting Telephony from the Status menu displays information about the state of each PBX port that connects to the Media Gateway. For each port, the following status information is provided: • State (DMG1000): – In Service – No Link – Red Alarm – Yellow Alarm – Pend D-Chan – n/a • State (DMG2000): – In Service – No Link – Red Alarm – Yellow Alarm – Pend D-Chan – n/a Note: 8.
Unit Status 8.6 Version Information Selecting Version from the Status menu displays version information about the Media Gateway software and hardware. Note: The firmware version information displayed will vary, depending on the Media Gateway model.
Diagnostics 9 This chapter describes how to perform diagnostics tasks on the Dialogic® 1000 Media Gateway (DMG1000) and Dialogic® 2000 Media Gateway (DMG2000) units and includes the following sections: • VoIP Interface Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269 • TDM Interface Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 272 • TDM Self Verification Test. . . . . . . . . . .
Diagnostics 9.1.2 VoIP Interface Test Operation To operate the VoIP interface diagnostic tool, the user must first navigate to the Tests-->VoIP link via the Media Gateway’s user interface. In the VoIP Interface Test Configuration table: 1. Choose the diagnostic test to perform by selecting one of the options in the Test Selection field. 2. Then the user must fill in the Destination VoIP Address field of the VoIP endpoint that the diagnostic test will use when attempting to establish a connection. 3.
Diagnostics Source Name The alphanumeric name representing the source party that is delivered to the destination endpoint during the diagnostic test. This is an optional field that is not required to perform the diagnostic test. Source Number The number representing the source party that is delivered to the destination endpoint during the diagnostic test. This is an optional field that is not required to perform the diagnostic test.
Diagnostics If an error occurs some other suggestions are: • Verify gateway VoIP interface is in-service and configured properly. • Verify destination VoIP endpoint is valid and in-service. • Run the test again with the Trace Capture and Network Capture diagnostic utilities started to log the detailed call flow of the VoIP interface diagnostic test. 9.1.2.3 Call Log The test results from the VoIP interface diagnostic tool are also stored in the Media Gateway call log.
Diagnostics The Media Gateway diagnostic utilities are accessible via the Web interface by selecting the Diagnostics link on the left side menu of any Web page. 9.2.2 TDM Interface Test Operation To operate the TDM interface diagnostic tool, the user must first navigate to the Tests-->TDM link via the Media Gateway’s user interface. In the TDM Interface Test Configuration table: 1. Choose the diagnostic test to perform by selecting one of the options in the Test Selection field. 2.
Diagnostics 9.2.2.1 Configuration Options The following configuration items are available to be modified by the user: Test Selection Choose the type of diagnostic test to perform. Available options are Initiate Call and Send Message Waiting Status. Interface Selection Select the specific interface to use in the diagnostic test. If a specific interface is not chosen the test will select the first interface available. Channel Selection Select the specific interface to use in the diagnostic test.
Diagnostics When the test is completed the results will show if the diagnostic test was able to establish a TDM connection with the specified Destination Number. Test Passed: Normal TDM connection was successful Test Failed: CallEndReason TDM connection failed The CallEndReason description is the error response received either from the specified destination TDM endpoint or the TDM interface of the gateway.
Diagnostics 9.3 TDM Self Verification Test TDM Self Verification is discussed in the following topics: • TDM Self Verification Test Overview • TDM Self Verification Test Operation 9.3.1 TDM Self Verification Test Overview The TDM self verification diagnostic tool is a Web page user application that can be used to provide verification of PBX compatibility and also to confirm the Media Gateway configuration values relating to the PBX interface are valid. 9.3.1.
Diagnostics 9.3.1.3 Capabilities The ability to route calls and send message waiting status updates through a particular PBX interface and channel is determined by the capabilities, enabled state, and if the interface and channel is in-service. If the interface and channel is not in-service or is not enabled, or it does not support the capability for the requested action, then the interface and channel will not be used for routing through the gateway and will not be included in a diagnostic test.
Diagnostics Figure 52.
Diagnostics 9.3.2.1 Configuration Options The following configuration items are available to be modified by the user: Interface Extension Numbers Enter the extension number of the interfaces to include in the selected diagnostic test. Interface extension numbers are only valid for Initiate Call / Answer Call, and Transfer Call diagnostic tests. Interface extension numbers that are left blank will not be included in the diagnostic test.
Diagnostics Figure 53. Call Flow for Initiate Call / Answer Call Initiate Call / Answer Call and Transfer Call The call flow of the Initiate Call / Answer Call and Transfer Call diagnostic test tries to verify not only the operation of originating an outbound call and answering an inbound call, but transferring the call as well. These tests also include verification of outbound call progress, inbound CPID, send/receive DTMF and disconnect supervision.
Diagnostics Figure 54. Call Flow for Initiate Call / Answer Call and Transfer Call Send Message Waiting Status The call flow of the Send Message Waiting Status diagnostic test tries to verify the message waiting notification set and clear update operation.
Diagnostics Figure 55. Call Flow for Send Message Waiting Status 9.3.2.3 Test Status When a diagnostic test is executed the test results are shown in the TDM Self Verification Test Status table on the Web page. Test Result Symbol Description - Test skipped because not configured or not available (no capability, not enabled, or not in-service). n Test was pending but did not execute because a failure occurred.
Diagnostics 9.3.2.4 Test Results After the diagnostic test(s) are completed or canceled the test results can be opened or saved to a text file (.txt) for documented proof of the verification process. To open the test results in another Web browser window click the pbxtest.txt hyperlink. Figure 57. TDM Self Verification Test Results 9.3.2.
Diagnostics • Check the Call Log for the call end reason of the connections used in the diagnostic tests. Some of the most common test result failures and causes are: Test Failure Reasons Originate an outbound PBX call • The extension numbers for the PBX interface are not correct. • The PBX interface and channel are not configured properly*. • Dial tone is not detected or being provided. Answer an outbound PBX call • The extension numbers for the PBX interface are not correct.
Diagnostics The call log is accessible through the Web by selecting the Call Log link on the left side menu of any Web page. Figure 58. TDM Self Verification Call Log Web Page 9.4 Diagnostic Logging The following is an overview describing the diagnostic logging capabilities of the Media Gateway and includes the following sections: • Overview • Trace Capture • Network Capture • TDM Capture 9.4.
Diagnostics • TDM Capture: The TDM log is a copy of the voice data transferred on the T1/E1 line. It is available only on the DMG2000. To view a log you must transfer the log to a PC. When transferred to a PC, each log is formatted into an industry standard to facilitate viewing and sharing of data. The format for each log is: • Trace Capture: Uses an ASCII text format (.TXT) • Network Capture: Uses an Ethereal format. Ethereal is a freeware program downloadable from the Internet.
Diagnostics Capture control page is shown in Figure 60. While the DMG2000 control page has more trace keys, the ideas presented in this section apply equally to both the DMG1000 and the DMG2000. Figure 60. Trace Capture Control Page - DMG2000 Figure 60 shows the default trace keys (i.e. immediately after power-up). Notice the Error and Trace masks are not shown as they are always enabled.
Diagnostics To use the Trace Capture, first check the appropriate trace masks. Click Apply Masks and then click Start Logging. Logging starts and the display changes to show the state of the log. An example of a running log is shown in Figure 61. Figure 61. Example of a Running Log - Trace Capture When the Trace Capture is running, the fill level of the log is displayed. The log uses a circular buffer, so when it reaches capacity, new data overwrites the old.
Diagnostics • The DMG1000 log can store up to 1.5 Mbytes, while the DMG2000 log can store up to 32 Mbytes. • Trace Capture is automatically enabled at power-on-with Error, Debug and Init trace masks for all trace keys. 9.4.3 Network Capture The Network Capture creates an in-memory log of Media Gateway IP traffic. Since there can be a great deal of traffic, the Network Capture control page allows filtering of IP packets.
Diagnostics to disable individual interfaces. The second level of filtering is on the protocol level. When an interface is enabled (Capture), you have the ability to filter (not log) different protocols. A check mark in the box means the Gateway will not log packets of the corresponding protocol. To use the Network Capture, first select the interfaces to capture, check the protocols you do not want to log, click Apply Filters and then click Start Capturing.
Diagnostics • After selecting protocol filters, be sure to click Apply Filters. • Size of the log is determined at startup. DMG1000 ~1.1 Mbytes, DMG2000 ~24 Mbytes. • Downloading the Network Capture clears the log and starts another. • Stopping the log and restarting it does not clear the log, new data is just appended. 9.4.4 TDM Capture The TDM Capture creates an in-memory log of voice traffic on the DMG2000's T1 or E1 line.
Diagnostics Figure 67. Example of Running Log - TDM Capture While the TDM Capture is running, the fill level of the log is displayed. The log is a circular buffer, so when it reaches capacity, new data overwrites the old. Once the log reaches capacity, the fill level does not change. The log can be transferred to a PC clicking on Download. Doing this brings up a File Download dialog box as shown in Figure 68. Select Save to save the file to the PC.
Diagnostics • Downloading the TDM Capture clears the log and starts another. • Stopping the log and restarting clears the log. Caution: This operation is different from the other logs. 9.
Diagnostics 9.5.2 Connecting to Terminal Interface Via LAN Connector Perform the following to initiate a Telnet session at the workstation connected to the Media Gateway via the LAN interface: 1. From the command prompt, type telnet followed by a space and the IP address of the Media Gateway. Then press Enter. 2. The telnet client connected message will appear. Press Enter. The following prompt will appear: PIMG> 3. At the prompt, type pwd and then press Enter. 4.
Diagnostics Telephony port 2 events: 002:59:314 [Tel-2 002:59:550 [Tel-2 002:59:552 [Tel-2 ] Event ] Event ] Event Lamp 60:CallApp0:0 OFF->FLASH 0:0:48| 251 | Cpid (251->) (Direct) Time-stamping of the trace output can be enabled or disabled by the user. The telephony switch protocol can be traced, but the output is encoded for security. 9.6.1 Trace Format The following is a description of the format of a trace display. The example display is broken down into its components.
Diagnostics Table 19. Supported Trace Keys Trace Keys Description Prot Code Error Warn Init Alarm ● ● ● ● ● Event System OS-Independent layer, System initialization Tel Digital Telephony Interface (for protocol, hook, lamp, display, CPID, in phone-emulation and phonedriving etc.) ●(8) ● ● ● ● ● ●(6) TelDrv Low layer ISDN trace coming from the Telesoft driver. Includes the RAW ISDN layer 3 packets.
Diagnostics Table 19.
Diagnostics Table 20. Supported Trace Types Trace Types Trace Description Prot Protocol Code Software/Code Error Errors (1) Warn Warnings Init Software/Device initialization (1) Alarm Alarms (1) Event Hookswitch, Lamps, Tones, Displays, Button Presses Stat Statistics (1) System starts with these enabled on all trace keys. 9.6.
Diagnostics Pbn VoIP Gw Gcc Adept teldrv CallLog Log Error Error Error Error Error Error Error Error Init Init Init Init Init Init Init Init Debug Debug Debug Debug Debug Debug Debug Debug Channel Trace State ('*' enabled, '-' disabled) -----------------------------------------------* * * * * * * * 9.6.3.2 trace default The trace default command sets the default tracing of all trace keys, and enables VoIP and telephony event tracing. PIMG>trace default 9.6.3.
Diagnostics 9.6.3.5 trace reset This command resets specified (or all) trace keys to their defaults (from system start). 9.6.3.6 trace all off This command disables all traces, except Error and Debug, on all trace keys. Error and Debug cannot be disabled. 9.6.3.7 trace port <1-based port number|all> This command enables or disables port-related traces.
Diagnostics 002:59:314 002:59:550 002:59:552 003:02:300 003:02:372 003:02:490 003:02:570 [Tel-2 [Tel-2 [Tel-2 [Tel-1 [Tel-2 [Tel-1 [Tel-2 ] ] ] ] ] ] ] Event Event Event Event Event Event Event Lamp 60:CallApp0:0 OFF->FLASH 0:0:48 | 251 Cpid (251->) (Direct) Lamp 60:CallApp0:0 FLASH->OFF Lamp 60:CallApp0:0 FLASH->OFF 0:0:48 | MAY 13 7:44 P 0:0:48 | MAY 13 7:44 P | | | Example 2 Enable tracing of SIP messages: PIMG>trace voip prot on Ok PIMG>exit Good-Bye.
Diagnostics [VoIP ] Prot [VoIP ] Prot ---->SIP/2.0 100 Trying! [VoIP ] Prot Via:SIP/2.0/UDP 10.0.1.110:5060! [VoIP ] Prot Call-ID:01B2270D9C81400000000006@pbxgw.default.com! [VoIP ] Prot CSeq:1 INVITE! [VoIP ] Prot From:"250";user=phone;tag=1A1F32463135364100000ABE! [VoIP ] Prot To:sip:101@10.0.1.215;tag=3246313536412A29002455E5! [VoIP ] Prot Contact:! [VoIP ] Prot Server:DMG1000 1.
Diagnostics [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1
Diagnostics [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 [DspCpi-1 ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] ] Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Stat Error Stats: invalid_header_count: to_micro_overflow_count: lost_enh_packet_count: no_core_packet_count: pkt_lost_by_network: rc4key_update
Diagnostics 9.7 Diagnostic Commands This section describes a number of diagnostic commands and includes: • Devstat Command • Restart Command • Ping Command • Ver Command • Alarm List Command 9.7.1 Devstat Command The Devstat command displays the current high-level settings of the unit and the unit's main status and telephony port status. The telephony port state can be 'In Service', 'Out of Service', or 'Active'. 'Active' means that the port is off-hook.
Diagnostics PIMG>ping 10.0.1.1 10.0.1.1 is alive PIMG>ping 10.0.1.1 5 PING 10.0.1.1: 56 data bytes 64 bytes from 10.0.1.1: icmp_seq=0. time=2. ms 64 bytes from 10.0.1.1: icmp_seq=1. time=0. ms 64 bytes from 10.0.1.1: icmp_seq=2. time=0. ms 64 bytes from 10.0.1.1: icmp_seq=3. time=0. ms 64 bytes from 10.0.1.1: icmp_seq=4. time=0. ms ----10.0.1.1 PING Statistics---5 packets transmitted, 5 packets received, 0% packet loss round-trip (ms) min/avg/max = 0/0/2 PIMG>ping 10.10.10.10 no answer from 10.10.10.10 9.
Index A alarm information 266 alarms command 306 B basic configuration via the serial port 38 C call class rules 252 reason tokens 254 regular expressions 252 rule order 254 rule syntax 253 call log status information 266 call progress tones characteristics 184 editing 183 editing INI file directly 161 learn tone issues 188 Learn Tone Progress web page 186 Learn Tone Results web page 187 learn tone solutions 188 Learn Tone web page 184 learning 184 learning, conflicting tones 188 learning, existing tones
configuration syntax 248 configuration via the serial port 38 connecting to terminal interface via Diagnostics connector 293 connecting to terminal interface via LAN connector 294 control of log files 286 D debug trace capture 286 Description 107, 108, 119, 123 devstat command 305 diagnostic logging 285 diagnostic logging overview 285 diagnostics control of log files 286 debug trace capture 286 diagnostic logging 285 diagnostic logging overview 285 network capture 289 PBX/PSTN Interface Test 272 TDM captur
IP Media Gateway changing the password 39 configuration procedure 39 parsers 247 product description 25 setting the IP address 37 upgrading the software 42 Web interface 32 IP parameters LAN2 Group 48 IP Web page 34, 37 L LAN2 Group parameters 48 Learn Tone Progress web page 186 Learn Tone Results web page 187 Learn Tone web page 184 M Manual Tones web page 183 MIB-II status information 267 N network capture 289 O online Help 35 operating modes phone emulating 26 overview diagnostic logging 285 Dialogi
P parameters Acceptable Media 94 Allow T.
DNS NAPTR and SRV Record Enable 178 DNS Server Address 61 DNS Server Address 2 62 DNS Translation of Phone Numbers 62 Early Media Requires Reliable Provisional Responses 93 E-Mail Alarms Enabled 52, 53, 54, 55, 57, 58 E-Mail Server IP Address 53 Enable Failover - E1 125 Enable Failover - T1 121 Enable Glare Detection 113 Enable Privacy Headers 72 Enable T.38 Spoofing 174 Enable V.
Jitter-Buffer Maximum Delay 168 Jitter-Buffer Minimum Delay 168 Key Derivation Enable 89 Key Derivation Rate 89 Line Coding - E1 121 Line Echo Cancellation 164 Line Echo Cancellation NLP 164 Line Encoding - T1 117 Line Mode 109 Local Phone Context 83 Loop Current Off Debounce 126 Maintenance Port Enabled 59 Maximum Call Party Delay 106 Maximum Live Answer Time 177 Maximum Silence after Voice has been Detected 177 Maximum Time to Wait for Voice 177 Minimum Call Party Delay 105 Minimum Live Answer Time 177 MK
Send Early 183 Progress Response 92 Serial CPID Expiration 153 Serial Interface Protocol 150 Serial Mode (Master/Slave) 150 Serial Port Baud Rate 146, 148 Serial Port Data Bits 147, 149 Serial Port Parity 147, 148 Serial Port Stop Bits 147, 149 Server Port 63, 65 Signaling Digit Relay Mode 93 Signaling Mode 109 SIPS URI Scheme 82 SNMP Community Name 56 SNMP System Contact 57 SNMP System Location 57 SNMP System Name 56 SNMP Trap IP List 56 SNMP Traps Enabled 55, 56 SNTP Server IP Address 47 Source E-Mail Add
Use Same Port for MWI Clear/Set - Analog 130 Use Same Port for MWI Clear/Set - T1 116 User Phone Parameter 83 User-Agent Header Value 60 User-User Enabled 77 V.
Serial Protocol Web page 34 setting the IP address 37 software upgrading 42 status alarm information 266 call log information 266 MIB-II information 267 summary information 265 telephony information 267 version information 268 summary information 265 Summary Web page 38 System parameters Serial Port, COM1 Group 146 Serial Port, COM2 Group 148 T TDM capture 291 telephony status information 267 terminal commands alarms command 306 devstat command 305 ping command 305 restart command 305 ver command 306 termi
trace default command 299 trace format 295 trace function 294 trace key 295 trace -l command 300 trace mechanism trace format 295 trace function 294 trace utility 295 trace port command 300 trace reset command 300 trace show command 298 trace time command 300 trace type 297 trace utility 295 trace key 295 trace type 297 U un-routable calls IP to PBX calls 31 PBX to IP calls 31 Upgrade Web page 35, 43 upgrading the software 42 V Validate Tone Progress web page 190 Validate Tone Results web page 190 ver com