GPT iSDX/Realitis (EMEA) Confidential 13
Configuration Note 8501 - Ver. H (08/01)
Unified Messenger™
Server
GPT iSDX/Siemens Realitis (EMEA)
With In-band integration, one pathway between the PBX and the Octel system transmits both call information and voice communications
1.0 Method of integration
With Inband integration, one pathway between the PBX and the Unified Messengerä server transmits both call information and voice communications. The pathway is provided by 2-wire analog single-line circuits that connect to Brooktrout VPS4 card in the Unified Messenger server. Each Brooktrout port simulates 2-wire analog lines. Calls to a Unified Messenger port are preceded by the called party information from the PBX in DTMF format. The Unified Messenger server then answers and plays the appropriate greeting.
Unified Messenger Server ordering information
2.0 unified messenger server requirements
· Brooktrout VPS4 card (4 port card)
· Brooktrout Vantage PCI Line Cards (8 port card)
· Software Release 4.0 or higher.
PBX requirements
3.0 PBX hardware requirements
· One analog extension circuit and line cord per Unified Messenger port
3.1 PBX software requirements
· Minimum software: 3.5.301
· The Integrated Voice Mail (IVM) package
· Software patches required
NOTE: Refer to Appendix A to verify that required patches are installed.
· A special EPROM may be required to delay answer by the Unified Messenger Server and ensure that called party ID is correctly sent when an operator transfer a caller to the Unified Messenger Server.
Supported integration features
4.0 supported features
· System Forward to personal greeting
- busy
- ring-no-answer
- all calls
· Multiple return-to-operator
· Message Waiting
· Multiple forward to personal greeting
· Direct Call (iSDX/Realistis Release 5.1.101 & above, See Section 5.3)
Configuring the iSDX/Siemens Realitis to integrate
5.0 COnfiguring the PBX to integrate
The Integrated Voice Messaging (IVM) feature must be installed and enabled by the PBX vendor. Once this is done, each of the extensions connected to the Unified Messenger Server are set as VDIR (the extension attribute). The command to perform this operation for each extension is REO nnnn VDIR, where nnnn is the extension number. Make sure you disable self-test from these extensions, as it will cause integration problems if enabled. Use the CNT command to disable self-test.
The iSDX/Realistis produces a conversion table, which enables the user's mailbox number to be different from the extension number. The mailbox numbers are allocated to the extension numbers using the "Revise IVM Mailbox Number" (RVM) command on the iSDX/Realistis connected to the user. If the mailbox manager number is the same as the extension number then the LVM table must be left blank and patches 1810X, and 1811X must be fitted.
Digit Conversion: Direct voice mail access can be achieved by ISDT's, operators, and rotary phones as the iSDX/Realistis will generate the necessary DTMF tones. This also applies across a DPNSS network.
If the PBX software is below 3.7.001 then patch 2094X must be fitted to enable DTMF conversion to feature phones on forwarding.
The Integration information is passed in-band across the 2-wire extension lines. The message format detailed below can be configured by the PBX and sent across the line using DTMF tones.
5.1 Programming forward to personal greeting
Programming forward to personal greeting
Each extension can be programmed by the user to divert to the pilot number of the Unified Messenger Server. The following call-diversion types are available on the iSDX/Realistis. (Broken dial tone is heard when forwarding features are invoked.)
§ Immediate or All Calls, also known as "call forward", will forward all calls.
-Activation Code #9<Unified Messenger Server Pilot Number.
-Deactivation Code ##9
§ Busy Diversion will divert calls to a specified extension when the user is on the phone. This can be specified by the user for all calls or by the system manager for external calls, internal calls, or all calls. A mailbox user will typically have internal calls forwarded on Busy to the Unified Messenger Server and may also forward external calls if a secretary is not available.
-Activation Code #0<Unified Messenger Server Pilot Number.
-Deactivation Code ##0
§ Ring-no-Answer will divert a call if the user does not answer his phone within a specified time (usually 15 seconds). This can be specified by the user for all calls or by the system manager for external calls, internal calls, or all calls. A mailbox user will typically have internal calls forwarded on RNA to the Unified Messenger Server and may also forward external calls if a secretary is not available.
-Activate RNA: #0*<Unified Messenger Server Pilot Number>
-Cancel RNA: ##0*
The feature codes listed above are generally standard but may vary from switch to switch.
If extensions are defined as LUAL users in the LIVU table, then no diversion is required to be set by the user, as automatic redirection to the Unified Messenger Server will occur after "all" other diversion is resolved.
5.2 called party id information
Called party ID information
When a call is diverted to the Unified Messenger Server, an available connecting line is seized and the following fields sent.
§ PAUSE 1. An optional, configurable pause to allow the system greeting to be played. Not needed for the Unified Messenger Server; should be set to 0.
§ CODE. A sequence of tones to put the VMS in record mode. Not needed by Unified Messenger Server: should be left blank. (Null)
§ PAUSE 2 An optional, configurable pause to allow the system greeting to be played. Not needed by the Unified Messenger Server: should be set to 0.
§ DELIMITER. This defines the start of the mailbox number. Not needed by the Unified Messenger Server; should be left blank.
§ MAILBOX. The destination mailbox number to which the voice message was sent.
§ PAUSE 3. An optional, configurable pause to allow mailbox greeting to be played. Not needed by the Unified Messenger Server: should be set to 0.
§ STATUS. This defines the status of the call. Not seen by the Unified Messenger Server; should be left blank. (Null)
Therefore, the only information sent across the line is the mailbox number.
The caller will be disconnected during any digit sending from the iSDX/Realistis to the Unified Messenger Server and therefore will not hear the tones being dialed. Only after the digit sending has been completed will the connection be made two-way.
5.3 direct call in release 5.1.101
Beginning at Software Release 5.1.101 the new enhanced inband integration has been introduced.
This supports Direct Call.
To activate this enhanced integration a Unique Code needs to be entered on the PBX to enable the software.
Duplicate Codes can be added for the Unified Messenger Server with no call progression. ( ) = Default
5.4 integration information
Integration information
The integration information is passed in-band across the 2-wire extension lines. The message format detailed below can be configured by the PBX and sent across the line using DTMF tones.
iSDX/siemens software parameters
§ SPCDV Defines the tone sequence to be sent to the UM as indication of clear down. 177770 (= ###8). Note to activate this SPICD must be set to 1
iSDX/Realistis software parameters
§ SPCDE. The Tone sequence code to put the IVM into record mode (1 to 4 digits) (Unified Messenger Server default = null).
§ SPDLM. The mailbox delimiter number (1 to 4 digits) (Unified Messenger Server default = null).
§SPSTS. The tone sequence to be sent to the IVM to define the type of call (1 to 4 digits).
§ SPTKR. Number of first stage tinkle rings before going on to the second stage.
§ TMBIV. Length of time in seconds before a call is automatically redirected to the IVM after giving a special busy tone.
§ TMPS1. Pause to allow the system greeting to be played to the user before sending the mailbox number (multiples of 200 ms).
(Unified Messenger Server default = 0)
§ TMPS2. Pause to allow the system greeting to be played to the user after sending the code SPCDE (multiples of 200 ms).
(Unified Messenger Server default = 0)
§ TMPS3. Pause to play a specific user greeting (multiples of 200 ms).
(Unified Messenger Server default = 0)
§ TMTK1. Interval between the rings during the first stage of tinkle (in minutes).
§ TMTK2. Interval between rings during the second stage of tinkle (in minutes).
§ SPILE, IVM hunt group pilot number
§ SPIDG. Only used on ISDX/Realistis remote to IVM on DPNSS network. Otherwise 0.
§ SPIRC, used to determine under what conditions redirection to IVM will take place set to 170000 to allow RNAE,RNAI,BE and BI
§ TMTK2. Interval between rings during the second stage of tinkle (in minutes). Not used on UM
§ SPIDV. Set to 0 allows non-IVM users to set diversion to a IVM port
§ SPVMD. Set to 0 disallows non-IVM users to divert to a IVM port
§ SPVMT Interdigit pause 0 = 200mS set to 1 = 100mS
5.6 Centralized VOICE MAIL Operation
The iSDX/Realistis IVM software allows voice mail features across a network of switches connected with DPNSS
The iSDX/Realistis IVM software allows voice mail features across a network of switches connected with DPNSS. Only the switch connected to the Unified Messenger Server needs the IVM package loaded with the IVM software. Other networked switches software must be at 3.5.301 and higher to allow transparent integration features. However, lower revisions of software can deliver callers to the Unified Messenger Server if patch 2129X is fitted at software below 3.7.001 or SPVMD is set to 1 at software 3.7.001 and higher. If the extension number is being forwarded as the mailbox number, then the SPEID code ( the PBX network address) will be added to the digit string prior to the mailbox number. This will need to be considered in mailbox length.
The GPT SL-1 \ iSLX \ BteX\ Realistis can also form part of a GPT iSDX/Realistis network and have access to Voice Mail features.
Note: If a call comes into a node and has a destination extension on a second node, which is busy, DPNSS is clever enough to see that extension on the second node is busy thus never transfers the call to the second node.
5.6 Message waiting
Ensure the MWI code is set as follows:
###4xxxx*1# for MWI on Note: xxxx = extension number
###4xxxx*0# for MWI off
6.0 Configuring the Unified messenger server
Note: The following screens reflect the latest version of the Unified Messenger. Please refer to the appropriate manual according to your system’s release for older versions of Unified Messenger.
Configuring the Unified Messenger Server
Configuring the Unified Messenger platform for proper PBX integration requires configuring several menus accessed within the Voice Mail System Configuration application.
§ Access the Voice Mail System Configuration application from the Unified Messenger program group.
Note: Maximize the Voice Mail System Configuration application window. Also, expand all fields so that all applicable options are visible.
§ Access the Telephone User Interface. Select General. Within this screen, set the number of digits in a mailbox. This number should match the number of digits of extension numbers on the customer’s PBX. At this point, you can also choose to enable/disable the Automated Attendant and Call Back Notification features. All other fields and tabs are configurable according to your customer needs.
§ Access the Message Waiting Indicator tab and set the following values:
§ Enable Message Waiting Indicator (MWI) = Enable by checking the box
MWI server = Select the MWI server with the MWI software installed
Limit requests = Leave Blank
§ Return to the Voice Mail Domain and select PBX’s. Access Edit, and select Add New PBX.
§ Within the Add New PBX dialog box, select iSDX.
§ Return to the Voice Mail Domain, and then select PBXs – iSDX.
§ Access the General tab and set the following values:
§ Go Off Hook when Port Disabled = Enable by checking the box
Input Gain = 1
Record Cue during Silence = Leave Blank
Pause before Digits (ms) = 2500
Pause Interval for Comma in Dial String (ms) = 2000
DTMF Inter-Digit Delay during Dialing (ms) = 80
DTMF Length during Dialing (ms) = 80
DTMF Inter-Digit Delay during Detection = 50
DTMF Length during Detection (ms) = 50
DTMF Inter-Digit Delay during Play = 50
DTMF Length during Play (ms) = 50
§ Access the Call Transfer tab and set the following values:
§ Transfer Prefix Code = &,XN (Unsupervised transfer)
Transfer Release Code when Busy = &,*1
Transfer Release Code when No Answer = &,*1
Transfer Release Code when Reject = &,*1
Flash Time Interval (ms) = 100
Enable Call Progress = X
Start Delay for Call Progress (ms) = 1000
*Note If earth break recall is being used “&” should be replaced with “R”, i.e. R,XN for transfer prefix code.
§ Access the Hangup Detection tab and set the following values:
§ Maximum Continuous Tone before Hanging Up (ms) = 6000
Hangup String = ###8
Hangup String Timeout (ms) = 0
Minimum Duration For Drop in Loop Current (ms)= 300
Maximum Silence before Hanging Up (ms) = 6000
§ Select Add New Tone. Enter, in the Tone Name field, Dial as the
new tone to be added. Select OK
Select Tone Identifier. Enter 100
Select Tone Length (ms). Enter 3008. Select OK.
§ Return to the Voice Mail System Configuration window, and select Voice Servers and access Telephony Interface (Analog).
§ Select the Analog tab to configure the selected port(s) on your Voice Server as follows:
- Playback Volume = 2 (Default)
- Number of Ports = Enter the number of ports in your system
- Enable DTMF Progress Tones = Leave Blank
- Enable the port(s) by checking the Box field next to the Port field
- Extension = Enter the proper extension number assigned to each port
- Incoming Ring Count = 1
- Primary ID = Leave Blank
- Secondary ID = Leave Blank
§ Return to the Voice Servers section and access PBX Integration.
§ Access the General tab. Select Inband as the Integration Type.
§ Access the Inband tab and set the following values: