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Stentofon Pulse: IP Based Intercom System
Stentofon Pulse: IP Based Intercom System
Document Scope
This document provides a step-by-step procedure for the configuration of
the Pulse IP-based intercom system for up to 16 intercom stations and 10
SIP terminals.
Publication Log
Rev. Date Author Status
1.0 1.12.2012 HKL Published
Related Documentation
For further information, refer to the following documentation:
Turbine IP Station
Directory Number: 14
Name: Elevator
IP Address: 169.254.1.104
or DHCP
Turbine IP Station
Directory Number: 13
Name: Entrance
IP Address: 169.254.1.103
or DHCP
2.1.1 Cabling
●● The cable between the PoE network switch and the station must be
Category 5 cable (Cat 5) or better
●● Maximum cable distance between the PoE switch and the station is
100 meters
●● All intercom stations in a Pulse installation must be located on the
same LAN (IP subnet)
●● Connect the LAN port of the Pulse Server station to a PoE network
switch.
LL Do not connect any other stations at this point
PoE port
RJ45
24V EXT
0V EXT
RELAY
LS
AUX
LAN
6
P3 1
1 P4 6
STATION A C
KEY1
KEY2
KEY3
LED
LED
Access the Pulse Server station by logging into the web interface using
a standard web browser on your PC:
1. Click Login
2. Enter the default User name: admin
3. Enter the default password: alphaadmin
The Station Information page will now be displayed, showing the default
IP master station configuration.
Station Mode
Since this is the Pulse Server station, it will function as the intercom
server or exchange:
IP Settings
The Pulse Server station will auto-discover all the other stations on the
LAN.
●● Click Play
-- You should now hear a tone from the station you selected.
All stations that have been configured should be displayed in the table.
Stations that are up and running will have the status Registered.
The Call and Audio Settings menu include the following parameters:
Autoanswer
Unchecked: Ringing Mode - the station will ring when it is called.
Checked: Open Mode - the call will be automatically connected when
the station is called. Optionally, a time period could be set for ringing
after which the station goes into open mode.
Auto-answer is typically enabled (checked) for substations and not for
master stations.
Disconnect By Button
When Enabled the call button will function as a cancel button during
conversation. This parameter is applicable to substations with a call
button.
When the relay is activated, this parameter determines how long the
relay should stay ON.
Speaker Volume
This parameter sets the speaker volume level (0-7) for the station.
Noise Reduction
This parameter enables active noise cancellation level (0-7). For
stations that are located in noisy environments, it is recommended that
the noise cancellation level be set to 3 or 4.
Call To
This parameter sets the directory number to call to when the key is
pressed.
Select Line
Select Line can be used to park an on-going call, and to make a new
call. To receive more than 1 call, Select Line should be assigned to at
least one DAK.
This parameter assigns the key to an external phone line. This requires
that the Pulse system is connected to an external telephone system
via a SIP gateway. You can have multiple lines on an IP master station.
The active line is indicated by a green LED. You can change the line
by pressing the associated DAK. Lines that have an active call are
indicated by a red LED.
The DAK keys can be configured via the Pulse Server:
●● Use a web browser and log into the Pulse Server station (e.g. IP
address: 169.254.1.99)
●● Select Server Management > Server Configuration > Direct
Access Key Settings
●● Select the station on which you want to program the Call Button
-- The Call Button is Input Button 1
●● Under Value enter the directory number to call when the button is
pressed.
Profile nickname
The name of the profile. Any text can be used.
Operation
A group call is activated by dialing the appropriate code (e.g. 84). A
ding-dong signal is heard in all member stations.
Nickname
This is the display text shown in the initiating station when the group call
is activated. Can be any text.
Directory number
The number to dial to activate the group call. If changed, a reboot of the
Pulse Server is required for the changes to take effect.
Priority
Priorities can be set for one group call at a time. A group call with higher
priority will override a lower priority group call.
EMERGENCY: Volume and handset override. The Group Call audio is
mixed with conversation audio
HIGH: The Group Call audio is mixed with conversation audio
NORMAL: The Group Call audio is not played if there is an ongoing
conversation
LOW: The Group Call audio is not played if there is an ongoing
conversation
Stations in group
Select the stations that should receive Group Call audio. You can select
all stations in one operation be checking the Add all stations box.
●● Remember to click Save before you make changes to the next group
call.
Troubleshooting
Q: When making a group call, the ding-dong chime is heard in the
stations, but there is no audio.
A: The problem is that the multicast audio does not reach the stations,
which may be due to two things:
To configure a ringlist:
There are variants of ringlists that can be set such as ringing time
intervals, loopback, and ringing together with previous stations.
●● Enter the license key in the New License field and click Insert
License
The Licensing list should now show all the licenses that are available.
●● Under Third Party SIP Terminals enter the Directory Number, Name,
Profile, and Password for the IP telephone.
●● Click Add followed by Save
When a valid Pulse Gateway license has been installed, the Gateways
parameter will be available.
3.8.5 Troubleshooting
When dialling the gateway access number (e.g. 0), the station display
may show one of the following error messages:
Forbidden
The station is restricted to call via the gateway due to Station Profiles
settings. In the Pulse Server station:
Disconnected. No number 0
●● The SIP Gateway is not registered to the Pulse Server station. The
SIP Gateway registration can be verified via the Pulse Server:
-- Select Server Management > Server Monitoring
-- The SIP Gateway should be displayed and have status
Registered. If it has status Not Registered, check the setup of the
SIP gateway.
●● The gateway is an AudioCodes MP114/118 and the telephone line is
not connected to the gateway
DOC NO.
A100K11311 support@stentofon.com
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