Administration

Galaxy PG1

Overview

E-Metrotel’s Galaxy PG1 is a SIP to Paging and Video intercom device for use in various industry applications. The media stream transmission adopts standard IP/RTP/RTSP protocol, with the stability and sound quality of an IP phone. It has various functions and interfaces that include intercom, broadcast, video, security and recording. It can adapt to different application environments and is simple to install, making PG1 ideal for any business.

Device Port Description

SpeakerExternal Amplified Speaker ports. The input power adapter size needs to correspond to the output speaker wattage. See Set the Speaker Settings for further details. Recommend using 18 AWG gauge speaker wire.
HeadsetExternal panel speaker or headset ports.
LEDExternal LED ports for registration, network and call status.
Function keyExternal Function key ports for connecting buttons or door bells etc.  When engaged a phone number can be speed dialed.
MicrophoneExternal Microphone ports.
Switch inputSecurity sensor ports. Example: Door sensor, infrared probe or an emergency switch.
Switch outputExternal ports for alarm lights, electric locks and other similar equipment.
Power input12V ~ 24V 2A input. The maximum output power amplifier is determined by the input voltage.
The device supports PoE. Power adapter is not included.

CameraStandard RJ45 interface for security camera.
Ethernet10M/100M ethernet WAN port. PoE supported
Registration / Network LED

Indicates the network status, call status, registration status.

  • Fast flashing: Network error or unsuccessful SIP account registration.
  • Slow flashing: An active call in progress.
  • Always on: Successful registration.
Volume Control buttons

Volume adjustments depending on state of the device.

  • Standby mode adjusts the volume of the ringtone.
  • Call mode adjusts the call volume.
  • Broadcast mode adjusts the broadcast volume.

Press and hold the volume down key to broadcast the IP address.
Press and hold the volume up key to toggle the IP address acquisition mode. (Static <-> DHCP)

Factory reset buttonPress and hold for 3 seconds to flash the device to restart and restore the factory settings.
Recording outputExternal ports for recording devices or computer mic input.
Grounding screwGrounding screw for external devices to prevent static build and shock to system.

Device Installation

Step One

Install device securely in desired location using the provided screw slots.

Step Two

Connect Speakers and other external equipment to the back of the unit as desired. It is recommended to attach a grounding cable.

Step Three

Plug the network cable to a PoE enabled switch, the device light flashes when the power connection is normal.

Wait 30 seconds after power on before using the device.

Discover IP address

There are 3 methods to find the IP address of the device.

Method One

Download the simple Network Scanner tool from iDoorPhone: https://idoorphonenetworkscanner.software.informer.com/1.0/
Run the Network Scanner tool. Press the Refresh button to search the door phone and find the IP address of the E-MetroTel Galaxy PG1 device.

Method Two

Make sure speakers are connected to the device first, then press the volume down button for 3 seconds, the device will automatically voice broadcast the device’s IP address.

Note: The default factory setting network mode is Static with IP address 192.168.1.128.

Method Three

Make sure speakers are connected to the device first, then press and hold the volume up button for 10 seconds, when a rapid beep is played, quickly press the volume up button 3 times and the beep will stop. Wait 10 seconds. The device will switch to Dynamic IP and voice broadcast its IP address.

Repeating the above steps will toggle the network mode back to Static IP address.

Configure SIP account on UCX Server

  1. Open UCX Administration
  2. From the PBX tab, select PBX Configuration
  3. From the left side column, select Extensions
  4. In the drop-down list box, select the device type Generic SIP Device and press the Submit button
  5. Enter the extension number of the SIP extension in the User Extension field
  6. You must enter something in the Display Name field.
  7. In the Device Options section, you will need to enter a password in the Secret field.
  8. If the PG1 device is connected to a UCx Cloud Server, change the nat field to Yes. Otherwise you can leave it at the default value No.
  9. Press the Submit button at the bottom of the page.
  10. When done, click on the Apply Config bar.

Configure the PG1 Device

Access the Web Interface

Enter the IP address of the PG1 device from a browser window. The default IP address of the device is 192.168.1.128.

Login using the default root login credentials:

  • Username = admin
  • Password = admin

Add SIP account

Navigate to Line -> SIP page and perform the following steps:

  • SIP Server Address field, enter IP address of UCX Server
  • SIP Server Port field, enter the port number used by UCX Server (default 5060 or 5960 for UCX Cloud)
  • Phone numberDisplay nameAuthentication Name fields, enter the SIP extension configured on the UCx Server
  • Authentication Password field, enter the secret/password of the SIP extension configured on the UCx Server
  • Select the Activate checkbox
  • Click the Apply button to save the settings

Set the Volume

Navigate to Intercom settings -> Audio page and adjust the volume levels of the following parameters:

  • Speakerphone 
  • Broadcast Output Volume
  • MIC Input Volume
  • Singal Tone Volume

Click the Apply button to save the settings.

Set the Function Key

Navigate to Function Key page and configure the following:

DSS Key 1

  • Type = Hot Key
  • Number 1 = Enter the number to dial
  • Number 2 = Enter the second number to dial if Number 1 is not available, or leave blank
  • Line = Select a working line
  • Subtype = Speed Dial

Click the Apply button to save the settings.

Set the Speaker Settings

The Speaker Settings default mode is Panel Speaker. Use this mode when connecting to a device that contains both speakers and microphone. Device cabling is connected to the Headset ports. The output power is limited to 10W.

If external loud speakers are used for broadcasting, it can be changed to External Speaker mode. Device cabling is connected to the Speaker ports.

Select the speaker audio output wattage: 10W / 20W / 30W, and they must match the corresponding power supply as indicated in the table below:

Input PowerAudio outputSpeaker Type
POE10W10W/4Ω
12V/2A DC10W10W/4Ω
18V/2A DC20W20W/4Ω
24V/2A DC30W30W/4Ω

Set the Camera

A security camera can either be connected directly to the unit or to the local network. It is recommended to use an external camera connected to the local network.

Perform the following steps to configure the device:

  • Navigate to Intercom Settings -> Video page
  • For Connect mode, select External and click Apply
  • Under IP Camera Settings, configure the following parameters:
    • User – External camera login account username
    • Password – External camera login account password
    • IP Camera brand – Select the camera’s manufacturer
    • IP – IP address of the camera
  • Click the Apply button

PG1 will automatically connect to the external camera and display the status.

Basic Operation

Answer a Call

If auto answer feature (Enable Auto Answer) is enabled, an incoming call is answered automatically. If Auto Answer Timeout is set to greater than 0, the call will ring for the set time and then auto answered.

Configuration is done under the Intercom settings -> Features page.

Make a Call

Configure function key as a Hot key with the desired target number(s). To make a call, press the function key and a call will be made to the configured number(s). 

Configuration is done under the Function Key page.

End Call

The function key can be configured to end a call when pressed. On the Function Key page, use one of the following two methods:

  1. Configure DSS key as Key Event, subtype Release
    OR
  2. Configure DSS key as Hot Key and set Use Hot Key to Hangup to Enable

Security linkage

If the switch input port connected to a door, emergency button or other sensory device receives a signal, the output port connected to alarm lights, electric locks or other equipment will automatically respond and send the alarm information. 

Input, Output and Trigger settings are configured under the Security settings page.

Video linkage

•Connect camera to the switch input port. •For camera configuration, please refer to the section on Set the Camera.

Administration

System

Information

FIELDDESCRIPTION
System InformationDisplay equipment model, hardware version, software version, uptime, last uptime, MEMInfo and system time.
NetworkShows the configuration information of WAN port, including connection mode of WAN port (Static, DHCP, PPPoE), MAC address, IP address of WAN port.
SIP AccountsShows the phone numbers and registration status of the 2 SIP LINES.

Account

There are two types of user accounts: Administrators and Users.

  • Users: Can set all configuration parameters except server parameters for SIP
  • Administrators: Can set all configuration parameters

A default adminstrator user account is provided:

  • Username: admin
  • Password: admin

Configurations

FIELDDESCRIPTION
Export ConfigurationsSave the equipment configuration to a TXT or XML file. Right click on the desired file type and select Save link as ...
Import ConfigurationsClick Select and find the saved configuration file. Then click Import to load the configuration.
Reset to factory defaultsDevice will be restored to factory default configuration. All existing configuration will be erased.

Upgrade

Click Select and find the software image file, then click Upgrade to load the new software version.

Auto Provision

Common Settings
FIELDDESCRIPTION
Current Configuration VersionShows the current config file version. If the config file to be downloaded is higher than the current version, the configuration would be upgraded.
If the endpoints confirm the configuration by the Digest method, the configuration would not be upgraded unless it differs from the current configuration.
General Configuration VersionShows the common config file version. If the config file to be downloaded and this version is the same, the auto provision would stop.
If the endpoints confirm the configuration by the Digest method, the configuration would not be upgraded unless it differs from the current configuration.
CPE Serial NumberSerial number of the equipment.
Authentication NameUsername for the FTP/HTTP/HTTPS configuration server. If this field is blank, anonymous will be used.
Authentication PasswordPassword for the FTP/HTTP/HTTPS configuration server.
Configuration File Encryption KeyEncryption key for the configuration file.
General
Configuration File
Encryption Key
Encryption key for common configuration file.
Save Auto Provision
Information
Save the auto provision username and password in the phone until the server URL changes.

DHCP Option
FIELDDESCRIPTION
Option ValueThe device supports Option 43Option 66Custom Option or it can be Disabled.
Custom Option ValueThis field is enabled only when Custom Option is selected. The DHCP option value must be from 128 to 254.

SIP Plug and Play
FIELDDESCRIPTION
Enable SIP PnPIf it is enabled, the device sends SIP SUBSCRIBE messages to the server address when it boots up. Any SIP server compatible with that message will reply with a SIP NOTIFY message containing the Auto Provisioning Server URL where the phones can request their configuration.
Server AddressPnP Server Address
Server PortPnP Server Port
Transportation ProtocolPnP Transfer protocol – UDP or TCP
Update IntervalInterval time for querying PnP server. Default is 1 hour.

Static Provisioning Server
FIELDDESCRIPTION
Server AddressSet FTP/TFTP/HTTP server IP address for auto update. The address can be an IP address or domain name with subdirectory.
Configuration File NameSpecify configuration file name. The device uses its MAC ID as the config file name if this is left blank.
Protocol TypeSpecify the protocol type FTP, TFTP or HTTP.
Update IntervalSpecify the update interval time. Default is 1 hour.
Update ModeDisable – Not to update
Update After Reboot – Update only after reboot.
Update at Time Interval – Update at every update interval.

TR069
FIELDDESCRIPTION
Enable TR069Enable/Disable TR069 configuration.
Enable TR069 Warning ToneEnable/Disable TR069 warning tone.
ACS Server TypeSelect the server type from the drop-down list.
ACS Server URLACS Server URL.
ACS UserUsername of ACS.
ACS PasswordACS password.
TLS VersionSelect the TLS version from the drop-down list.
INFORM Sending PeriodTime between transmissions of INFORM. The value is in seconds.
STUN Server AddrEnter the STUN server address.
STUN EnableEnable/Disable STUN.

FDMS

FDMS Settings
FIELDDESCRIPTION
Enable FDMSDisplay equipment model, hardware version, software version, uptime, last uptime, MEMInfo and system time.
FDMS IntervalShows the configuration information of WAN port, including connection mode of WAN port (Static, DHCP, PPPoE), MAC address, IP address of WAN port.
Doorphone Info Settings
FIELDDESCRIPTION
Community NameThe name of the community where the device is installed.
Building NumberThe name of the building where the device is installed.
Room NumberThe name of the room where the device is installed.

Tools

Syslog

Syslog is a protocol used to record log messages using a client/server mechanism. The Syslog server receives the messages from clients, and classifies them based on priority and type. Then these messages would be written into a log by rules which the administrator has configured. There are 8 levels of syslog information.

  1. Emergency: System is unusable. This is the highest debug info level.
  2. Alert: Action must be taken immediately.
  3. Critical: System is probably working incorrectly.
  4. Error: System may not work correctly.
  5. Warning: System may work correctly but needs attention.
  6. Notice: It is normal but significant condition.
  7. Information: It is normal daily messages.
  8. Debug: Debug messages normally used by system designer. This level can only be displayed via telnet.
Network Packets Capture

Capture packet streams from the device. This is normally used to troubleshoot problems.

Auto Reboot Setting

Reserved for future use.

Reboot Phone

Some configuration changes require a reboot to take effect. Click the Reboot button to reboot immediately.
Note: Be sure to save the configuration before rebooting.

Network

Basic

Network Status

Displays the network information of the device.

Settings

Select the network mode: Static IP or DHCP or PPPoE.
The fields below will change based on the mode selected.
Click the Apply button to save the new settings. If the IP address of the device has been changed, you have to relogin to the web interface with the new IP address.

FIELDDESCRIPTION
Static IPNetwork parameters must be entered manually and does not change.
DHCPNetwork parameters are provided automatically by a DHCP server.
PPPoEAccount and Password must be input manually. These are provided by your ISP.
Service Port Settings

If modifying the HTTP(S) ports, the new port must be greater than 1024.
Click the Apply button to save the new settings and reboot the device.

HTTPS Certification File

Upload or Delete the certification file.

Advanced

VPN

The device supports remote connection via VPN. It supports both Layer 2 Tunneling Protocol (L2TP) and OpenVPN protocol. (Only one protocol can be activated.)
This allows users at remote locations on the public network to make secure connections to local networks.

Once the selection and configuration is saved, the phone must be rebooted.

FIELDDESCRIPTION
VPN IP AddressShows the current VPN IP address.
VPN Mode
Enable VPNEnable/Disable VPN.
L2TPSelect Layer 2 Tunneling protocol.
OpenVPNSelect OpenVPN prootocol.
Layer 2 Tunneling Protocol (L2TP)

If L2TP mode is selected, then fill in the parameters under this section.

OpenVPN Files

If OpenVPN mode is selected, then Upload or Delete the OpenVPN cerfification files here.

Line

SIP

FIELDDESCRIPTION
LineChoose the SIP line to configure
Line StatusDisplays the current line status. To update the line status, refresh the page manually.
Phone numberThe extension number of the account.
Display nameThe display name of the account.
Authentication NameThe username of the account.
Authentication PasswordThe password or secret of the account.
ActivateActive/Deactivate the line.
SIP Proxy Server AddressEnter the IP address or FQDN of the SIP server address.
SIP Proxy Server PortEnter the SIP server port number. Default is 5060.
Backup Proxy Server AddressEnter the IP address or FQDN of the Backup SIP server address.
Backup Proxy Server PortEnter the Backup SIP server port number. Default is 5060.
Outbound proxy addressEnter the IP or FQDN address of outbound proxy server.
Outbound proxy portEnter the outbound proxy server port.
RealmEnter the SIP domain if requested by the service provider.

Codecs Settings

Set the priority and availability of the codecs by adding or remove them from the list.

Advanced Settings
FIELDDESCRIPTION
Subscribe for Voice MessageEnable the device to subscribe for voice message waiting notification. If enabled, the device will receive notification from the server if there is voice message waiting on the server.
Voice Message NumberSet the number for retrieving voice message.
Voice Message
Subscribe Period
Set the interval of voice message notification subscription.
Enable DNDEnable Do-not-disturb, any incoming call to this line will be rejected automatically.
Blocking Anonymous CallReject any incoming call that does not have a CallerID.
Use 182 Response for Call waitingSet the device to use 182 response code for call waiting.
Anonymous Call StandardSelect from the drop-down list the standard to be used for anonymous calls.
Dial Without RegisteredAllow call out by proxy without registration.
Click To TalkAllow Click to Talk.
User AgentSet the user agent, the default is Model with Software Version.
Response Single CodecIf enabled, the device will use single codec in response to an incoming call request.
Ring TypeSelect the ring tone type from the drop-down list.
Conference Type

Seelct from the drop-down list the type of call conference.

  • Local: Conference is setup on the device itself, supports a maximum of two remote parties.
  • Server: Conference is setup on the server by dialing the conference number.
Server Conference NumberConference number to dial when conference type is set to be Server.
Transfer TimeoutSet the timeout value (in seconds) for call transfers.
Enable Long ContactAllow more parameters in contact field per RFC 3840.
Enable User Inactive Hold 
Use Quote in Display NameAdd quote in display name.
Use Feature CodeWhen this setting is enabled, the features in this section will be controlled by the server instead of the device. Feature control is done by dialing the number specified in each feature code field.
Enable DNDNumber to dial to enable DND.
Enable Blocking Anonymous CallNumber to dial to enable blocking anonymous call.
DND DisabledNumber to dial to disable DND.
Disable Blocking Anonymous CallNumber to dial to disable blocking anonymous call.
Specific Server TypeSet the line to collaborate with specific server type. Use COMMON for UCx Server.
Registration ExpirationSet the SIP expiration interval.
Use VPNSet the line to use VPN restrict route.
Use STUNSet the line to use STUN for NAT traversal.
Convert URIConvert non digits and alphabetic characters to $hh hex code.
DTMF Type

Select from the drop-down list the DTMF type to be used for the line:

  • In-band
  • RFC2833
  • SIP_INFO
  • Auto
DTMF SIP INFO Mode

Select from the drop-down list the SIP INFO mode:

  • Send * / #
  • Send 10 / 11
Transportation Protocol

Select protocol to use for SIP transmission from the drop-down list:

  • UDP
  • TCP
  • TLS
Local PortSet the local port. Default is 5060.
SIP Version

Select the SIP version:

  • RFC3261
  • RFC2543
Caller ID Header

Select the Caller ID header:

  • FROM
  • PAI-FROM
  • RPID-FROM
  • PAI-RPID-FROM
  • RPID-PAI-FROM
Enable Strict ProxyEnables the use of strict routing. When the phone receives packets from the server, it will use the source IP address, not the address in via field.
Enable user=phoneSets user=phone in SIP messages.
Enable SCAEnable/Disable SCA (Shared Call Appearance )
Enable DNS SRVSet the line to use DNS SRV which will resolve the FQDN in proxy server into a service list.
Keep Alive Type

Select the NAT keep alive type:

  • Disabled
  • SIP OPTION
  • UDP
Keep Alive IntervalSet the keep alive packet transmitting interval.
Sync Clock Time 
Enable Session TimerSet the line to enable call ending by session timer refreshment. The call session will be ended if there is not new session timer event update received after the timeout period.
Session TimeoutSet the session timer timeout period.
Enable RportSet the line to add rport in SIP headers.
Enable PRACKSet the line to support PRACK SIP message.
Auto Change PortEnable/Disable Auto Change Port.
Keep AuthenticationKeep the authentication parameters from previous authentication.
Auto TCPUsing TCP protocol to guarantee usability of transport for SIP messages above 1500 bytes.
Enable Feature SyncFeature synchronization with server.
Enable GRUUSupport Globally Routable User-Agent URI (GRUU).
RTP EncryptionEnable RTP encryption such that RTP transmission will be encrypted.
RTP Encryption KeySet the pass phrase for RTP encryption.

Basic Settings

SIP settings
FIELDDESCRIPTION
Local SIP PortSet the local SIP port used to send/receive SIP messages.
Registration Faiure Retry IntervalSet the retry interval of SIP REGISTRATION when registration failed.
Enable Strickt UA MatchEnable or disable Strict UA Match.
Strict Branch 
STUN settings

A STUN (Simple Traversal of UDP through NAT) server allows a phone in a private network to know its public IP and port as well as the type of NAT being used. The device can then use this information to register itself to a SIP server so that it can make and receive calls while in a private network.

FIELDDESCRIPTION
STUN NAT TraversalShows status of STUN NAT Traversal.
Server AddressSTUN Server IP address.
Server PortSTUN Server Port – Default is 3478.
Binding PeriodSTUN blinding period – STUN packets are sent at this interval to keep the NAT mapping active.
SIP Waiting TimeWaiting time for SIP. This will vary depending on the network.
TLS Certification File

Upload or Delete the TLS certification file used for encrypted SIP transmission.

Intercom settings

Features

FIELDDESCRIPTION
Enable DNDIf enabled, all incoming calls are rejected. Outgoing calls are not affected.
Enable Intercom MuteIf enabled, incoming calls are muted during an intercom call.
Enable Auto AnswerIf enabled, incoming calls are automatically answered.
No Answer Auto HangupIf enabled, hang up automatically when there is no answer.
Voice Read IPEnable or disable voice broadcast IP address.
Enable Delay StartEnable / disable start delay.
DescriptionDescription displayed on IP scanning tool software or FDMS.
Ban OutgoingIf enabled, outgoing calls are disallowed.
Enable Intercom RingingIf enabled, intercom ring tone will be played when there is a new incoming call.
Auto Answer TimeoutSet auto answer timeout. Call is auto answered after the set time.
Auto Hangup TimeoutAutomatically hang up when there is no answer within the set time.
Voice Play LanguageSet language of the voice prompt.
Delay Start TimeSe start delay time.

Audio

FIELDDESCRIPTION
First CodecThe first codec choice: G.711A/U, G.722, G.723.1, G.726-32, G.729AB
Second CodecThe second codec choice: G.711A/U, G.722, G.723.1, G.726-32, G.729AB, None
Third CodecThe third codec choice: G.711A/U, G.722, G.723.1, G.726-32, G.729AB, None
Fourth CodecThe fourth codec choice: G.711A/U, G.722, G.723.1, G.726-32, G.729AB, None
Fifth CodecThe fifth codec choice: G.711A/U, G.722, G.723.1, G.726-32, G.729AB, None
Sixth CodecThe sixth codec choice: G.711A/U, G.722, G.723.1, G.726-32, G.729AB, None
DTMF Payload TypeThe RTP Payload type that indicates DTMF. Default is 101.
Default Ring TypeSelect ring type from the drop-down list.
G.729AB Payload LengthG.729AB Payload length – adjust from 10 msec to 60 msec.
Tone StandardSelect the country tone standard from the drop-down list.
G.722 TimestampsSelect from: 160/20 ms or 320/20ms.
G.723.1 Bit RateSelect from: 5.3kb/s or 6.3kb/s.
Speakerphone VolumeSet speaker volume level.
MIC Input VolumeSet MIC input volume level.
Broadcast Output VolumeSet broadcast output volume level.
Signal Tone VolumeSet audio signal output volume level.
Enable VADEnable or disable Voice Activity Detection (VAD). If VAD is enabled, G729 Payload length cannot be greater than 20 msec.

Speaker Settings

The Speaker Settings default mode is Panel Speaker. Use this mode when connecting to a device that contains both speakers and microphone. Device cabling is connected to the Headset ports. The output power is limited to 10W.

If external loud speakers are used for broadcasting, it can be changed to External Speaker mode. Device cabling is connected to the Speaker ports.

Select the speaker audio output wattage: 10W / 20W / 30W, and they must match the corresponding power supply as indicated in the table below:

Input PowerAudio outputSpeaker Type
POE10W10W/4Ω
12V/2A DC10W10W/4Ω
18V/2A DC20W20W/4Ω
24V/2A DC30W30W/4Ω
AEC Settings

Video

IP Camera Settings

A security camera can either be connected directly to the unit or to the local network. It is recommended to use an external camera connected to the local network.

Perform the following steps to configure the device:

  • For Connect mode, select External and click Apply
  • Under IP Camera Settings, configure the following parameters:
    • User – External camera login account username
    • Password – External camera login account password
    • IP Camera brand – Select the camera’s manufacturer
    • IP – IP address of the camera
  • Click the Apply button

PG1 will automatically connect to the external camera and display the status.

MCAST

It is easy and convenient to use multicast function to send notice to each member of the multicast via setting the multicast key on the device and sending multicast RTP stream to pre-configured multicast address. By configuring monitoring multicast address on the device, the device monitors and plays the RTP stream which sent by the multicast address.

MCAST Settings

Equipment can be set up to monitor up to 10 different multicast addresses, used to receive the multicast RTP stream sent by the multicast address.
Here are the ways to change equipment receiving multicast RTP stream processing mode in the web interface:

Priority

In the drop-down box to choose priority of ordinary calls, if the priority of the incoming streams of multicast RTP, lower precedence than the current common calls, device would automatically ignore the group RTP streams. If the priority of the incoming stream of multicast RTP is higher than the current common calls priority, device would automatically receive the group RTP streams, and keep the current common calls in maintained status. You can also choose to disable the function in the receiving threshold drop-down box, the device would automatically ignore all local network multicast RTP streams. The options are as follows:

  • 1 – 10 : To definite the priority of the common calls, 1 is the highest while 10 is the lowest
  • Disable : ignore all incoming multicast RTP streams
Enable the page priority

Page priority determines the device how to deal with the new receiving multicast RTP streams when it is in multicast session currently. When Page priority switch is enabled, the device would automatically ignore the low priority multicast RTP streams but receive top-level priority multicast RTP streams, and keep the current multicast session in maintained statu; If it is not enabled, the device would automatically ignore all receiving multicast RTP streams.

Note: when you press the multicast key for multicast session, both multicast sender and receiver would beep.

Index/Priority

Multicast is a sign of listening, but also the monitoring multicast priority. The smaller number refers to higher priority.

Name

Define your monitoring multicast name.The group name would be displayed on the screen when you answer the multicast. If you have not set, the screen would display the Host:port value.

Host:port

It is a set of addresses and ports to listen, separated by a colon.

Multicast service
  • Send: when you configure the item, pressing the corresponding key on the equipment shell, equipment would directly enter the Talking interface; the premise is to ensure no current multicast call and three-way conference, so the multicast can be established.
  • Monitor: IP port and priority are configured to monitor the device, when the call is initiated by multicast and the call is successful; the device would directly enter the Talking interface.

Action URL

URL for various actions performed by the phone. These actions are recorded and sent as xml files to the server. Sample format is http://InternalServer/FileName.xml.

Time/Date

Network Time Server Settings

Click on Apply to save the changes before proceeding to the next section.

FIELDDESCRIPTION
Time Synchronized via SNTPUse SNTP protocol for time synchronization.
Time Synchronized via DHCPUse DHCP protocol for time synchronization.
Primary Time ServerSet address of primary time server.
Secondary Time ServerSet address of secondary time server.
Time zoneSelect time zone from the drop-down list.
Resync PeriodTime interval for resynchronization wiht time server.
Date FormatSelect format from the drop-down list.
Date Separator

This field is displayed only when certain date formats are selected. Choose the separator used for the date:

  • /
  • .
  • space
Daylight Saving Time Settings

Click on Apply to save the changes.

FIELDDESCRIPTION
LocationSelect location from the drop-down list.
Note: the list of locations available in the drop-down list will change based on your time zone setting.
DST Set TypeSelect from: DisabledAutomaticManual
If Manual is selected, more fields will be displayed where you can define the rules for DST start and end.
Manual Time Settings

Set the time manually here. Note that SNTP must be disabled first.

Note: If the device is restarted, the system time will reset.

Security settings

FIELDDESCRIPTION
Input Settings
Input DetectEnable / disable input detect.
Trigger ModeLow Level Trigger (Closed Trigger) – detect the input port (low level) closed trigger.
High Level Trigger (Disconnected Trigger) – detect the input port (high level) disconnected trigger.
Alert message send to serverEnable or disable Input port send message to server
Output Settings
Output ResponseEnable / disable output response
Output LevelLow Level (NO: open) – when trigger condition is met, trigger the NO port disconnected.
High Level (NO: closed) – when trigger condition is met, trigger the NO port close.
Output DurationThe port changes the duration. Default value is 5 seconds.
Alert Trigger Setting
Alarm Ring DurationDuration of alarm ring
Input triggerWhen the input port trigger condition is met, the output port will trigger (The Port level time change, By < Output Duration > control).
You can choose to enable or disable the ringtone.
Remote DTMF TriggerReceived the terminal equipment to send the DTMF password, if correct, which triggers the corresponding output port.
You can choose to enable or disable the ringtone.
Remote SMS TriggerEnable or disable Remote SMS Trigger.
You can choose to enable or disable the ringtone.
Call State Trigger

The port output continuous time synchronization and trigger state changes, including the trigger conditions. Four models, such as: call trigger output port, will be in a call state to continue to respond.

  • Talking
  • Talking and Ringing
  • Ringing
  • Calling
Output LastBy Duration – The Port level time change, By < Output Duration > control
By State – By call state control, after the end of the call, port to return the default state.
Trigger CodeDuring the call, receive the terminal equipment to send the DTMF password, if correct, which triggers the corresponding output port. The default is 1234.
Trigger MessageIn the remote device or server to send instructions to ALERT=[instructions], if correct, which triggers the corresponding output port.
Server Settings
Server AddressConfigure remote response server address (including remote response server address and tamper alarm server address). When the input port is triggered will send a short message to the server, the message format is as follows:
Alarm Info: Description=PA2;SIP User=;Mac=00:a8:34:68:23:d1;IP=172.18.2.243;port=Input1

Function Key

Function Key Settings

DSS Keys can be configured according to the following types: Key Event, Hot Key, Multicast.

Key Event
TypeSubtypeUsage
Key EventNoneNo response
ReleaseDelete password input, cancel dialing input and end call
OKIdentification key
Hot Key

Enter the phone number in the input box. When you press the shortcut key, device will dial the preset telephone number. This button can also be used to set the IP address: you can press the shortcut key to directly make an IP call.

TypeNumberLineSubtypeUsage
Hot KeyTarget SIP account or IP addressCorresponding SIP lineSpeed DialUsing Speed Dial mode together with Enable Speed Dial Hangup can define whether this call is allowed to be disconnected when the speed dial key is pressed again.
IntercomIf the caller’s phone supports the intercom feature, the device can automatically answer intercom calls.
Multicast

Multicast function is to deliver voice streams to configured multicast address; all equipment monitored the multicast address can receive and play it. Using multicast functionality would make deliver voice one to many which are in the multicast group simply and conveniently.

TypeNumberSubtypeUsage
MulticastEnter the host IP address and port number separated by a colon.G.711ANarrowband speech coding (4Khz)
G.711U
G.722Wideband speech coding (7Khz)
G.723.1Narrowband speech coding (4Khz)
G.726-32
G.729AB
Operation mechanism

You can define the Function Key configuration with multicast address, port and used codec. The device can configure via WEB to monitor the multicast address and port. When the device make a multicast, all devices monitoring the address can receive the multicast data.

Calling configuration

If the device is in calls, or it is three-way conference, or initiated multicast communication, the device would not be able to launch a new multicast call.

Advanced Settings

FIELDDESCRIPTION
Input port Multiplexing as DSS Key2Enable / disable input port reuse for DSS key 2.
Use Function Key to AnswerEnable / disable Function Key to answer
Use Hot Key to HangupEnable / disable Hot Key to hang up
Hot Key Dial Mode Select

This field is enabled only when DSS Key is defined as a Hot Key.

Number 1 Transfer Call Number 2 Mode Select.

  • Main-Secondary :  Dial the primary extension first, if there is no answer then cancel the call and dial the secondary extension.
  • Day-Night : Check system time, if  Day time dial Number 1 otherwise dial Number 2.

User just press the speed dial key once.

Call Switched TimeThis field is enabled only when Main-Secondary mode is selected.
Set number 1 to transfer call number 2 time, default 16 seconds.
Day Start TimeThis field is enabled only when Day-Night mode is selected.
Set the start time of the day.
Day End TimeThis field is enabled only when Day-Night mode is selected.
Set the end time of the day.

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