SIP Intercom Configuration
From Zenitel Wiki
The stations are easy to install and configure. They can be configured using a standard web browser (like Firefox or Chrome), or one can use the dedicated Intercom Management Tool IMT. This article describes configuration using the Web interface.
- 1 Getting Started
- 2 The station web interface
- 2.1 Station Main
- 2.2 SIP Configuration
- 2.2.1 SIP Settings
- 2.2.2 Audio Settings
- 2.2.3 Direct Access Key & Ringlist Settings
- 2.2.4 Relay/Output Settings
- 2.2.5 Time Settings
- 2.2.6 I/O Settings
- 2.2.7 Frontboard Mapping [Turbine only]
- 2.2.8 RTSP Settings
- 2.2.9 Scripts [Turbine only]
- 2.2.10 Audio Messages [Turbine only]
- 2.2.11 Multicast Paging
- 2.3 Station Administration
- 2.4 Advanced SIP
- 2.5 Advanced Network Settings
Main steps to get the station registered to a SIP server:
- Log on to the station
- Main Settings: Go to Station Main > Main Settings and set "Station Mode" = Use SIP, and enter the IP Settings.
- Account Settings: Select SIP Configuration > SIP Settings and enter as a minimum Display Name, Directory Number, Server Domain and Authentication User Name.
- Call Buttons (DAK keys): Select SIP Configuration > Direct Access Key Settings.
The station web interface
- Station Information page shows current status and information about the station, suuch as firmware version and IP Address
- Main Settings: Here the mode of operation is set, the network settings, and the model type.
- Account Settings: Configure relevnt parameters for the SIP account
- Call Settings: A number of Call related parameters are available
Audio Settings: A number of Audio related parameters are available
Direct Access Key & Ringlist Settings
- Direct Access Keys (DAK) are the Call Buttons or speed-dial buttons you find on most intercom stations.
- Ring List is used to configure Call Escalation (i.e. forwarding of unattended calls), and for Parallel Ringing.
To configure Call Buttons and Ring List, see Direct Access Key & Ringlist Settings (SIP)
All IP Intercom stations (execpt the desktop models) have a built-in relay which can be used for door opening, call indication lamp etc. The relay can be controlled by a DTMF digit, or by a number of different call events and station statuses.
For details about relay configuration, see Relay Settings
The user is able to configure time by either enabling NTP and specifying NTP server, or setting up time manually. On Master stations with display the configured time is shown on display.
See Time Settings for details.
The Turbine stations have 6 I/O's which can be used as Inputs or Outputs. By default all I/O's are set as Inputs.
For details about configuration of inputs and outputs, see I/O Settings.
Frontboard Mapping [Turbine only]
When making customized stations based on the Turbine Frontboard, it is possible from the station web interface to define DAK actions and LED behavior. This is supported when front board type is set to TCIS 1/2/3 and TCIV 1/2/3, and the station operates in Pulse or SIP mode.
For more details, see Button and LED configuration for customized stations (SIP)
- Enable RTSP - enables RTSP traffic by allowing/blocking traffic from/to RTSP port
- RTSP Port - defines RTSP port - by default RTSP port is set to 554
- Enable RTSP Authentication - enables mandatory usage of RTSP credentials on stream setup
- RTSP URL - RTSP URL that can be copied into RTSP player. All supported RTSP streams are listed here: audio-only, video-only and audio+video stream URL for video Turbine devices and audio-only stream URL for other Turbine devices.
- Stream Audio Source - defines when acoustic echo canceler is applied to mic+speaker voice. With this setting it is possible to choose whether only microphone signal is sent to RTSP stream or both microphone and speaker signals are sent to RTSP stream. On video Turbine stations this option can be defined separately for each stream.
- Enable video stream tunneling [TCIV only] - defines if tunneling is used in video RTSP stream. This option is using lots of resources, so it is not recommended to be used.
Scripts [Turbine only]
Scripts (Virtual I/O) is a feature for activating scripts on station events. The supported script languages are Lua and other shell scripting languages. This can be used for example to execute scripts towards other systems, e.g. access control systems, triggered by a DTMF digit. These scripts can be uploaded and configured via the menu options:
- Script Upload
- Script Configuration
- Script Events
The Scripting feature is supported on Turbine stations only.
For more details, see Virtual I/O
Audio Messages [Turbine only]
Prerecorded audio files can be uploaded to a Turbine IP Station, and the audio messages can be triggered by various events occuring on the station. The audio message files are uploaded from the station webinterface or from the VS-IMT tool.
See Audio Messaging for more details.
Multicast Paging enables IP intercom stations in SIP mode to receive VoIP audio as multicast paging from 3rd party iPBX (e.g. Asterisk). The feature is supported by both INCA stations and Turbine stations. Up to 10 different Multicast paging groups can be defined.
See Multicast Paging (SIP) for details.
Logs can be useful for debugging and troubleshooting purposes. See Logging - IP Stations on how to enable and collect logs.
The password for web access and for accessing the display setup menus can be changed, see Password (IP Stations).
Backup and Restore
From the web interface of the station it is possible to backup and restore the configuration data. See Backup and Restore for more information.
The software in the stations can be upgraded via the web interface of the station. The procedure is slighty different depending on the type of station:
Language Settings [INCA only]
Keyboard [INCA only]
On the INCA Stations there is an option to set Keyboard Type. This is typically used for the IP Substation kit or the IP Master kit when building customized stations. See Keyboard Settings for further details.
Updates (TFTP Provisioning)
IP stations may be set up to automatically poll configuration from a TFTP server. The IP address of this TFTP server can be obtained using DHCP procedures or be manually configured. See TFTP Provisioning for more information.
The tone test is designed to check whether the microphone and loudspeaker is working by playing a tone and detecting the sound pressure level. If the sound pressure level is below a chosen threshold the test fails. The result of the tone test is reported using SNMP traps, which must be turned on in the SNMP configuration.
For configuration of the Tone test, see Tone Test (IP Stations)
Webcall makes it possible to establish a call directly from the station web. See Web Call for more information.
Advanced Network Settings
A set of SNMP functions are available in the IP station. SNMP (Simple Network Management Protocol) is a protocol for centralizing the management of devices in IP networks.
See SNMP in IP Stations for futher information.
IEEE 802.1X is an IEEE Standard for port-based Network Access Control (PNAC). It provides an authentication mechanism to devices wishing to attach to a LAN, either establishing a point-to-point connection or preventing it if authentication fails. For details on configuration, see IEEE 802.1X.
All IP Stations have an embedded firewall. See Firewall for details.
VLAN Tagging [INCA only]
VLAN Tagging is the practice of inserting a VLAN ID into a packet header in order to identify which VLAN (Virtual Local Area Network) the packet belongs to. More specifically, switches use the VLAN ID to determine which port(s), or interface(s), to send a broadcast packet to.
For details on configuration, see VLAN Tagging (IEEE 802.1Q).