8305 Multi-Interface IP Paging Adapter User Guide

8305 - Right Side

RESET Button

A recessed reset button. This button is used to reset the device or play the device’s IP address during setup.

RJ45 Ethernet Jack

For network connection. A cable run from the switch can be terminated to a modular jack with a connection by patch cord or terminated with an RJ45 plug.

PoE (Power over Ethernet) must be 48 V 350 mA IEEE 802.3af compliant, whether provided by the network switch or injector.

RJ45 Ethernet Jack Light

There are two lights on the Ethernet jack:

• Amber: Turns on immediately after the Ethernet cable is first connected indicating that PoE power has been successfully applied. Once the device connects to the network, the light will turn off.

• Green: Turns on when Ethernet is working after a 100Mbps link has been established. Flickers to indicate activity on the port.

SIP Light (Blue)

A steady light will appear when a SIP extension is registered. The light will blink when the device is engaged in a SIP call.

MCAST Light (Blue)

A steady light will appear when the 8305 receives multicast audio as a Receiver. The light will blink when the 8305 sends multicast audio as a Transmitter.

INPUT Light (Blue)

Not currently used on this device.

OUTPUT Indicator (Blue)

Turns on when the analog output is enabled.

AUX OUT 3.5 mm Jack

Analog line level output for compatible PC speakers or headset. Non-isolated.

8305 - Left Side

Telephone Port

Emulates an analog telephone that can go on and off hook. Has built in ring detection with auto-answer.

The telephone port is intended only for the connection to the FXS port on a legacy communication system and must not be connected to the PSTN.

Terminal Block 8 W Output (1/2)

Balanced and isolated wire pair output to connect one or many external self-amplified speakers connected in parallel with a total minimum resistance of 8 W. Intended use is for up to 100 nominal 2 kW or 1 kW self-amplified speakers.

8 W maximum output: +3dBm @ 8 W

2 kW maximum output: +1.5dB higher

Terminal Block Line Output (3/4)

Balanced and isolated wire pair. Output level defined using web interface.

Terminal Block Relay In (5/6)

Used to connect an external button or to detect a contact closure. Connection options include a normally closed switch, normally open switch, 1202 Call Button, 1203 Call Switch, 1204 Volume Control Switch, 1205 Audio Interface, or EOL resistor termination.

Terminal Block Relay Out (7/8)

By default, these terminals provide a contact closure when the 8305 IP Paging Adapter is active. Note this is a normally open relay only.

1. Use appropriate drywall anchors for #8 screws and pre-drill per the anchor manufacturer’s instructions.

2. Insert 4 anchors into the wall, and then attach the bracket to the wall anchors using #8 screws.

3. Snap the 8305 into the bracket.

Figure 3: 8305 wall mount.

• Device Name

• SIP Registration

• Call Status

• Proxy Status

• Provisioning Status

• MAC

• IP Address

• Date/Time

• Next Scheduled Event

• Multicast Mode

• Volume

• Relay Input Status

• InformaCast License

• ADMP Cloud Monitoring

SIP

SIP Domain (Proxy Server)

The SIP Server's IP address (e.g., 192.168.1.111) or domain name (e.g., myserver.com).

Ring/Alert Mode

Ring extensions do not answer incoming calls but play a customizable, pre-recorded announcement, such as a loud ringer (night bell). Announcements are customizable and can be pre-recorded.

Use this setting to add a second SIP extension for a Ring event. If Monitor "Ring" event on registered SIP extension is selected, you will see additional settings for Ring extension parameters. None is set by default.

If set, the device will detect inbound ring events on this extension and play the alerting tone (and multicast if configured) until the inbound call stops ringing. The 8305 will not answer the call on this extension.

Often, the 8305 will be a member of a hunt group or ring group to ring in conjunction with a telephone.

You may change the alert tone via Basic Settings →  Features.

Ring Extension

Enter the SIP extension for the ring parameter of the 8305.

The device will detect inbound ring events on this extension and play the alerting tone (and multicast if configured) until the inbound call stops ringing. It will not answer the call on this extension.

Page Extension

Page extensions auto-answer and open a voice path, enabling live announcements.

Enter the SIP page extension for the 8305 so the device will auto-answer any inbound call received on this extension and provide a voice paging path (and multicast if configured).

Authentication ID

The Authentication ID is a name that represents the page extension. It is also referred to as ‘Username’ for some SIP servers. This may be the same as the Ring or Page extension in some cases.

Authentication Password

This is the SIP password for the registered SIP account. Up to eight (8) characters can be used. The password can be used to authenticate SIP users.

Contact your System Administrator for the password to obtain access.

Display Name (Optional)

Enter the name you want displayed when an SIP call is made. For the display name to be shown, the PBX and phone(s) must be configured to display this message as the Caller ID.

Settings

Default Announcement Duration

An announcement can be played once or continuously until canceled. Select Play Once to play a single cycle of the chosen tone file. If Play Until Cancelled is selected, the announcement will continue to play until the "Call-to-Cancel" extension is called to clear the announcement or a defined timeout is reached.

Default Maximum Announcement Time

Select the maximum time an announcement can be played.

Announcement Selection Mode

Select Direct Extensions to register a separate extension for each announcement. Select DTMF Selectable to register a single extension that accepts DTMF input to select which announcement to play.

Answer Inbound Call

This setting indicates how Announcement calls are handled. In both cases, the Emergency Announcement is started when the appropriate extension is called and continues until the “Call-to-Cancel” extension is called.

Select Enabled to answer the inbound call and provide the option to play a Confirmation Tone before starting the alert, then automatically release the call or request a passcode before playing the announcement. Select Disabled to detect the inbound Ring signal but not answer the call.

Select Disabled to only detect the inbound Ring signal, but not answer the call.

In both instances, the announcement will play until the time limit is reached or the “Call-to-Cancel” extension is called. Enabling Answer Inbound Call can be useful when the caller cannot hear the announcement from their location. However, if the call might go to a group or multiple extension(s) (including this device), the auto-answer may intercept that call and prevent it from ringing on other devices.

Passcode Protected Announcement Extensions

Select Enabled to require the caller to enter a passcode after dialing an announcement or “Call-to-Cancel” extension. Setting a passcode helps prevent unintentional announcements.

Announcement Passcode

Enter a passcode that a caller must enter to play or cancel an announcement.

When prompted, the caller must enter the passcode followed by the # sign before the announcement will be played or canceled. The passcode prompt will be played before any other action. If the passcode is not correctly entered within 15 seconds, the call will end.

Passcode Prompt Tone

Select a tone to play when the passcode is ready to be entered.

DTMF Selection

Extension

Enter the SIP extension for the DTMF Selection parameter of the 8305.

Authentication ID

Enter the Authentication ID. It may also be called Username for some SIP servers or may be the same as the extension.

Authentication Password

Enter the SIP password provided by the system administrator for the SIP account.

Display Name (Optional)

Enter a ‘Display Name’ that will be sent when the SIP call is made. The PBX and phone(s) must be configured to display this message as the Caller ID.

Prompt Tone

Select a tone to play when the passcode is ready to be entered.

Call-to-Cancel

Call-to-Cancel Selection Mode

If using “DTMF 0”, the user should dial the main DTMF Selection extension and select ‘0’ to cancel the announcement.

Extension

Enter the SIP extension for the Call-to-Cancel Selection parameter of the 8305.

Authentication ID

Enter the Authentication ID provided by the System Administrator. It may also be called Username for some SIP servers or may be the same as the extension.

Display Name (Optional)

Enter a ‘Display Name’ that will be sent when the SIP call is made. The PBX and phone(s) must be configured to display this message as the Caller ID.

Prompt Tone

Select a tone to play when the passcode is ready to be entered.

Announcements

Announcement #

To configure an Emergency Alert extension, select Enabled for an announcement number.

Up to 10 extensions can be registered allowing up to 10 different announcements. Audio files can be easily uploaded to create custom announcements. Only one ‘Call-to-Cancel’ extension is needed.

Some SIP telephone systems may not support multiple announcements if they limit the number of extensions that can be registered on a single device.

Announcement Duration

Choose the duration of an announcement. The Default option follows the behavior configured in Default Announcement Duration.

Maximum Announcement Time

Select the maximum announcement time.

Tone/Pre-recorded Announcement

Select a file to use as a ringtone or announcement.

Confirmation Tone

Select a file to use as a confirmation tone.

General

SIP Transportation

Select a transport layer protocol to use for SIP messages from the dropdown. These options include:

• Auto: Will check the DNS NAPTR record, then try UDP/TCP.

• UDP

• TCP

• TLS: Ensures the encryption of SIP traffic. In this mode, if the SIP Server requires endpoints to be authenticated, a PEM file containing both a device certificate and a private key must be installed on the 8305. Upload a certificate via System → File Manager and rename it to 'sipclient.pem' in the 'certs' folder.

SIPS Scheme

Only visible when SIP Transportation is set to TLS. Enable to require the SIP connection from endpoint to endpoint to be secure.

Validate Server Certificate

Enable to validate the SIP server against common certificate authorities. To validate additional certificates, navigate to System → File Manager to upload a Base64 encoded X.509 certificate file in .pem, .cer, or .crt format to the certs folder.

SIP Outbound Support (RFC 5626)

Enable this option to support best networking practices according to RFC 5626. This option should be enabled if the 8305 is registered with a hosted server or TLS is used for SIP Transportation.

Outbound Proxy

Enter the IP address for an outbound proxy.

Register Period (seconds)

Enter the maximum requested period where the 8305 will re-register with the SIP server. The default setting is 3600 seconds (1 hour).

Note that if an Expires header is provided by the SIP response 200 (OK), this time will take precedence over the Register Period defined time here.

Only change if instructed to do so.

SRTP

SDP SRTP Offer

Select an option from the dropdown menu:

• Disabled: Only unencrypted calls will be accepted.

• Standard: Only encrypted calls will be accepted. This option secures the audio data  (SRTP) between parties by ensuring that it cannot be reconstructed and listened to later.

• Optional (Non-standard AVP Profile): Both unencrypted and encrypted SIP calls will be accepted, no matter if the other party supports SRTP. The SIP call’s RTP data will be unencrypted if the other party does not support SRTP.

NAT

Media NAT  

IP address for STUN server if present or IP address/credentials for a TURN server.

ICE – TURN Server

Enter the IP address or domain of the ICE server.

ICE – TURN User

Enter the username.

ICE – TURN Password

Enter the password.

STUN - Server

Enter the IP address or domain of the STUN server.

Server Redundancy

Server Redundancy Feature

Enable to configure up to two secondary backup servers.

When enabled, the 8305 will attempt to register with the primary server but switch to a secondary server when necessary. The configuration allows re-registration to the primary server upon availability or to stay with a server until unresponsive.

Backup Server #1, #2

Provided by your SIP provider or IT team.

Polling Intervals (seconds)

Select the time interval for sending monitoring packets to each server from the dropdown menu. Inactive servers are always polled and the active server may optionally be polled.

Poll Active Server

Enable to explicitly poll the current server to monitor availability. Other regular events may also handle this automatically and can be disabled to reduce network traffic.

Automatic Fallback

Enable to allow the 8305 to reconnect with a higher priority server once available, even if the backup connection is still working.

Polling Method

Select a polling method based on what your SIP provider supports.

Interoperability

Keep-Alive Method

Select a keep-alive method:

• None

• Double CRLF: The 8305 will send a packet regularly to maintain connection with the SIP Server if behind NAT.

Keep-Alive Interval

Set the interval in seconds that the CRLF message should be sent. 30 seconds is recommended.

Use Outgoing TLS port in SIP Headers

Enable to use the ephemeral port number from an outgoing SIP TLS connection instead of the listening port number in SIP Contact and Via headers. This is useful for connecting the device to some local SIP servers, like Asterisk or FreeSWITCH.

Do Not Reuse Authorization Headers

Enable so all SIP authorization information from the last successful request will not be reused in the next request.

Allow Missing Subscription-State Headers

Enable to allow SIP NOTIFY messages that do not contain a ‘Subscription-State’ header.

Multicast Mode

Always ensure that the multicast settings on all Receiver devices match those of the Transmitter.

Multicast Mode

If Transmitter (Sender) is selected, the 8305 will broadcast an IP stream when activated in addition to playing audio through the audio output. The 8305 cannot be both a multicast Transmitter and Receiver simultaneously.

Multicast Type

The 8305 may broadcast multicast paging compatible with Poly “on-premise group paging” protocol and most multicast-enabled phones that use RTP audio packets.

Select Regular (RTP) if you are only multicasting to Algo IP endpoints or multicast-enabled phones.

To multicast page announcements to Poly phones, select Poly Group Page or Poly Push-to-Talk.

Select Regular RTP + Poly Group Page or Regular RTP + Push-to-Talk to multicast page audio to Poly phones, Algo IP endpoints, and multicast-enabled phones.

Number of Zones

Select Basic Zones Only if configuring nine or fewer multicast zones. Select Basic and Expanded Zones to configure up to 50 zones. The expanded zones have the same behavior as the basic Receiver zones but are hidden by default to simplify the interface.

Poly Group Paging/Push-to-Talk

This section is used if the Multicast Type includes Poly Group Page or Poly Push-to-Talk.

Poly Zone

Enter the same Multicast IP Address and Port number configured on the Poly phones.

Poly Group Selection Mode

Select Single Group to broadcast on one pre-configured group. Multiple SIP extensions can be registered on the Transmitter device. Each extension is mapped to a unique group, allowing groups to be called directly (e.g., from speed-dial keys). See Additional Features → More Page Extensions for additional configuration settings.

If DTMF Selectable Group is selected, the group is determined by the DTMF selection between 0 – 25.

To page using DTMF Selectable Zone:

1. Dial the SIP extension of the Transmitter device

2. Dial the desired DTMF page group number on the keypad when prompted. Groups 10 and higher start with “*”.

DTMF group definitions include:

• DTMF Extension 1 for Zone 1

• DTMF Extension 2 for Zone 2

…

• DTMF Extension *10 for Zone 10

• DTMF Extension *11 for Zone 11

All DTMF codes and respective zones are available in Advanced Settings →  Advanced Multicast.

Poly Default Channel

Select the default group for the multicast stream to be sent to. If DTMF Selectable Group is chosen, this single group setting will not apply to paging since the group will be dynamically selected per call using DTMF. The Single Group setting will still apply to the ring extension and relay triggered events.

The Poly Default Channel is the default channel used for multicast actions unless an option is available for a custom channel with specific parameters.

Speaker Playback Groups

Select Speaker Playback Groups to control which specific groups can play audio from the device. This is useful if using the DTMF Selectable Group mode or additional page extensions (Additional Features →  More Page Extensions) per group to make 8305 a member of only certain zones. In this case, the Transmitter does not participate in the Zone but transmits certain traffic.

Transmitter (Sender) Zone Settings

This section is used if the Multicast Type includes Regular (RTP).

Zone Selection Mode

Select Single Zone to broadcast on one pre-configured zone. Multiple SIP extensions can be registered on the Transmitter device. Each extension is mapped to a unique zone, allowing zones to be called directly (e.g., from speed-dial keys). See Additional Features → More Page Extensions for additional configuration settings.

If DTMF Selectable Zone is selected, the zone is determined by the DTMF selection between 0 – 50. Once multicast Transmitter mode is enabled, navigate to Advanced Settings → Advanced Multicast to find the DTMF codes corresponding to each zone.

To page using DTMF Selectable Zone:

1. Dial the SIP extension of the Transmitter device

2. Dial the desired DTMF page zone number on the keypad when prompted. Zones 10 and higher start with “*”.

DTMF zone definitions include:

• DTMF Extension 9 for Priority Call

• DTMF Extension 0 or 8 for All Call

• DTMF Extension 1 for Zone 1

• DTMF Extension *10 for Zone 10

• DTMF Extension *11 for Zone 11

All DTMF codes and respective zones are available in Advanced Settings → Advanced Multicast.

Transmitter Single Zone

Select the default zone for the multicast stream to be sent to. If DTMF Selectable Zone is chosen, this single zone setting will not apply to Paging since the zone will be dynamically selected per call using DTMF. However, this single zone setting will still apply to the ring extension and relay triggered events.

The Transmitter Single Zone is the default zone used for multicast actions unless an option is available for a custom zone with specific parameters.

Speaker Playback Zones

Select Speaker Playback Zones to control which specific zones the 8305 can play audio. This is useful if using the DTMF Selectable Zone mode or additional page extensions (Additional Features →  More Page Extensions) per zone to make 8305 a member of only certain zones. In this case, the Transmitter does not participate in the Zone but transmits certain traffic.

DTMF Settings

Zone Selection Tone

Select a tone to be played to prompt a user to select a zone to multicast to.

This may be used as an interactive voice response (IVR) menu by uploading a custom audio file in the tones folder through System →  File Manager. Each zone may use a different tone. This can be configured in Advanced Settings →  Advanced Multicast.

Two-Digit Selection

When enabled, all DTMF Selectable Zones will require two digits. As a result, Basic Zones must be prefixed with 0, and Expanded Zones will no longer need to be prefixed with *.

Multicast Mode

Always ensure that the multicast settings on all Receiver devices match those of the Transmitter.

Multicast Mode

If Receiver (Listener) mode is selected, the 8305 will activate when receiving a multicast message. It will mimic the audio stream of the transmitter but use local volume settings. This can be set via Basic Settings → Features → Page Speaker Volume.

Multicast Type

Select Regular if receiving multicast from other Algo IP endpoint(s) and/or multicast-enabled phone(s) that use RTP audio packets.

Select Poly Group Page or Poly Push-to-Talk if receiving multicast paging compatible with Poly “on-premise group paging” protocol.

Number of Zones

Select Basic Zones Only if configuring nine or fewer multicast zones. Select Basic and Expanded Zones to configure up to 50 zones. The expanded zones have the same behavior as the basic Receiver zones but are hidden by default to simplify the interface.

Receiver (Listener) Zone Settings

Basic Receiver Zones

Select one or more multicast zones for the 8305 to listen to. Multicast zone priority will be based on the zone definition list order defined in Advanced Settings → Advanced Multicast.

Expanded Receiver Zones

Select additional zones (up to 50) for the device to listen to. This is only possible when Basic and Expanded Zones is selected.

Poly Group Paging/Push-to-Talk

Poly Zone

Enter the Poly Zone (IP Address and Port) that matches the configuration of the Poly phones and Channels.

Poly Receiver Channels

If using a Poly telephone as a Multicast Transmitter, a tone may be set for any of the 25 Poly Groups configured on the 8305. Poly Group Tones can be set in Advanced Settings →  Advanced Multicast.

The Poly telephone used as page audio source for the 8305 must be configured to use either the G.711 or G.722 audio codec.

Note that Poly phone(s) must be configured with the “Compatibility” setting (“ptt.compatibilityMode”) disabled for this codec setting to be applied.

Transmitter Settings

Transmitter Output Codec

Select an audio encoding format for the Transmitter device to use when sending output to the Receivers. Supported formats include:

• G.711 ulaw

• G.722

• Opus

• L16

Only G.711 and G.722 are supported when using Two-Way Paging mode.

Output Packetization Time (milliseconds)

Select the size of the audio packets the Transmitter sends to the Receivers from the dropdown menu. The default of 20 milliseconds is recommended unless a different value is specifically required for compatibility with other devices.

Multicast TTL

Only change the multicast time to live (TTL) setting if custom routing is configured on the network that specifically routes multicast packets between subnets and a longer TTL count is required. This ensures packets are not bounced back and forth in a network identity. When the TTL is reached, the router drops the packet.

RTP Control Protocol (RTCP)

RTCP Port Selection

Select how a port will be chosen to send or receive RTCP packets.

Note: If Next Higher Port is selected, ensure that the default multicast zone definitions are modified so that zones are only assigned to even-numbered ports, leaving the next higher odd-numbered ports free for RTCP packets.

Receiver Settings

Audio Sync

Available if under Basic Settings →  Multicast the Multicast Mode is set to Receiver (Listener) and Multicast Type is set to Poly Group Page or Poly Push-to-Talk. When using multicast with other third-party devices that have a delay in their audio path, the audio on the 8305 may be heard slightly earlier than on these other devices. Use this feature to add a small delay to the audio output on the 8305 to synchronize with these other devices.

Poly Receiver Tones

Poly Receiver Tones

Available if under Basic Settings →  Multicast the Multicast Mode is set to Receiver (Listener) and Multicast Type is set to Poly Group Page or Poly Push-to-Talk. A tone may be set for any of the 25 Poly Groups. If using an Algo device as a Multicast Transmitter, it is recommended to set the Receiver tones to None to avoid conflicts, as the Algo devices already multicast a tone by default.

Inbound Ring Settings

Ring settings apply to events triggered by Ring Extensions and Emergency Alerts. Emergency Alert tones are configured under Additional Features →  Emergency Alerts.

Ring/Alert Tone

Select an audio file to play when a ring event is detected on the SIP Ring Extension. Test the audio file immediately using the Play, Loop, and Stop buttons if the 8305 is connected to a speaker.

During multicast, the device will broadcast an audio stream using the Transmitter’s selected ringtone. This is the default tone that will be played if selected in the settings Multicast →  Additional Ring Extension.

Ring/Alert Volume

Set the volume for a SIP Ring event using the dropdown. This setting is for gain control and the output level depends on the levels recorded into the source audio file played from memory. This setting is only used for local tones, not multicast.

See Page Speaker Volume below for multicast settings.

Ring Limit

Typically set to no limit. Ring Limit will limit how long the speaker will ring before timing out. A new ring event must occur for the speaker to play the audio file again.

Inbound Page Settings

Page Volume

This setting is for gain control for SIP or multicast paging. The output level will depend on the streaming level. Page Speaker Volume will apply to all inbound multicast streams (for Receiver mode only) regardless of audio source or type.

Page Mode

Set calls to the SIP page extension as one-way, two-way (using an external microphone), or delayed.

In delayed mode, the speaker will record a message to be played after disconnecting. The device will buffer an announcement up to 5 minutes long.

To cancel a page while in delay mode, press “*” while recording to prevent it from being sent after hanging up.

Page Timeout

Set the maximum duration for a page. The page will end when the timeout limit has been reached. This is useful to ensure the paging system is not stuck in an active state in cases where someone accidentally forgets to hang up.

Page Tone

Select a pre-page tone to be played when a page is starting. Use only the Default or custom uploaded files. Other pre-installed tone files contain silence at the end to generate a ring "cadence" of 6 seconds. This silence will block the voice path for several seconds at the start of a page. The “Default” tone is set to page-notif.wav.

The Default Page Tone in Advanced Multicast will play the tone set here.

G.722 Support

Enable or disable the G.722 codec. G.722 enables wideband audio for optimum speech intelligibility.

Passcode Protected Page Extensions

When enabled, the caller must enter the set passcode followed by the # sign before the page can be made. Setting a passcode helps prevent unintentional pages.

Apply to All Page Extensions

Only visible when Passcode Protected Page Extensions is set to Enabled. Enable or disable a passcode for all page extensions.

Passcode

Only visible when Passcode Protected Page Extensions is set to Enabled. Passcodes can be up to 15 digits and must be numbers only.

Passcode Prompt Tone

Only visible when Passcode Protected Page Extensions is set to Enabled. Select the tone to be played to prompt the user to enter the passcode before paging.

DTMF Detection Type

Select the preferred dual-tone multi-frequency (DTMF) detection method. DTMF is a technology used with touch tone phones (the sound made when pressing a number key). The 8305 uses this for multi-zone selection, passcode, etc.

Audio

Ambient Noise Compensation

When enabled, Ambient Noise Compensation will allow the speaker level to adjust automatically in response to ambient noise levels detected at the device before the start of each call. The 8305 requires an external microphone to be connected to do this.

Automatic Gain Control (AGC)

Enable or disable AGC to normalize the audio level. Enabling ensures the speaker is always played at a consistent volume.

Analog Output Port/Level

Set the maximum voltage level available on the 'Line Out' analog audio port, telephone port, or 8Ω port. To achieve the maximum level, also set the volume control to level 10 and enable the Automatic Gain Control (AGC) to increase the level received from the far-end telephone when paging.

Audio Input Settings

Audio Input Port

Select telephone Port or Aux Out to choose where audio will be played to.

Files

Download and Install Ring Tones from the Algo Server

Tone files can be downloaded manually from the Algo website.

Cache

Rebuild Tone Cache Files

Only needed when the tone cache is out of sync. The operation might take a long time depending on the types and sizes of the tone files.

Test Tones

Listen to uploaded audio files before selecting them for your system.

Functions

Dynamic Range Compression (DRC)

Enable to compress the dynamic range of page audio to increase loudness.

Dynamic Range Compression Gain

Select the amount of compression gain from the dropdown menu. More gain increases distortion.

Jitter Buffer Range

Enter a value between 10-500 to add more buffering if necessary to correct for inconsistent delays on the network. It is recommended to use the lowest value.

Generate In-Band DTMF Tones

Enable to play DTMF tones during an SIP call to allow interoperability with DTMF-controlled multi-zone legacy communication systems.

Always Send RTP Media

Enable to send audio packets at all times, even during one-way paging mode. This option is needed when the server expects to always see audio packets.

Audio Filters

Speaker Filter

Select a frequency from the dropdown to apply a high-pass filter to the speaker output. This setting reduces audio artifacts like humming or buzzing by filtering out unwanted frequencies.

Speaker Noise Filter

Enable to filter below 145 Hz to reduce mains-induced noise like fans.

Microphone Filter

Select a frequency from the dropdown to apply a high-pass filter to the microphone input. This setting reduces audio artifacts like humming or buzzing by filtering out unwanted frequencies.

Microphone Noise Filter

Enable to filter below 145 Hz to reduce mains-induced noise like fans.

Ambient Noise Compensation

Only available if Ambient Noise Compensation is Enabled in Basic Settings → Features.

Ambient Noise Compensation No Loss

Configure the Ambient Noise Compensation algorithm to only use levels at or above the current volume. The current volume is the minimum volume when this setting is enabled.

Ambient Noise Compensation Max Volume

Based on ambient noise levels, a maximum volume can be set.

Backup/Restore Data

Download Scheduler Data File

Download a backup database file of schedules with events, times, and calendar dates. Note that this backup is independent from the rest of the configuration backup on the device. 

Restore Scheduler Data File

Upload and restore a saved Scheduler data file.

Convert JSON Scheduler Data File

If you are migrating from old firmware and would like to restore your data to a newer device, you may upload the JSON file from your old system here.

Clear All Data

Clear all the Scheduler data including saved schedules and set calendar dates.

General

Relay Input Mode

The 8305 has dry contact input terminals to connect external accessories, including Algo and third-party accessories.

Options for Relay Input Mode include:

• Disabled

• Relay Normally Open

• Relay Normally Open with Supervision (e.g. Algo 1203 Call Switch)

• Relay Normally Closed

• Relay Normally Closed with Supervision

• Mute Switch

• Mute Switch with Supervision

• Algo 1202 Call Button

• Algo 1204 Volume Control Switch (Local or Remote)

• Algo 1204 Volume Control Switch with Supervision (Local or Remote)

• Algo 1205 Audio Interface Switch

• Algo 1205 Audio Interface Switch with Supervision

• Algo 2507 Ring Detector

Notification actions can be triggered via supervision settings if the input switch is disconnected.

For more information on how to configure each of these devices with the 8305, see Algo Compatible Accessories.

Action When Input Triggered

Action

Play Tone
When the 8305 input is triggered, a tone or a pre-recorded audio file will play over the speaker or multicast. This function can be used to request support or assistance in service or retail environments, notify about an emergency at a specific location in medical or educational facilities, or sound an alarm during an intrusion.

Make Two-Way SIP Voice Call
When the 8305 input is triggered, a voice path will open for an intercom-like call via an external microphone connected to a pre-configured telephone extension. This option can be used when a call needs to be made from a public place where a telephone would not be practical to use.

Make SIP Call with Tone
When the 8305 input is triggered, a private call can be made to a pre-configured telephone extension with a pre-recorded message. For instance, a call to a supervisor’s telephone notifying about an emergency or intrusion at some location.

Tone/Pre-recorded Announcement

Available when Action is set to Play Tone or Make SIP Call with Tone.

Select a recording or tone to use. Custom audio files may be used and uploaded through System →  File Manager.

Tone Duration

Available when Action is set to Play Tone.

Extension to Dial

Available when Action is set to Make Two-Way SIP Voice Call or Make SIP Call with Tone.

A SIP account is required in Page Extension fields to make a call.

Call Mode

Available when Action is set to Make Two-Way SIP Voice Call.

Allow 2nd Button Press

Available when Action is set to Make Two-Way SIP Voice Call or Make SIP Call with Tone.

If enabled, the 2nd button press will End Call or End and Restart Call. Therefore, if an input is triggered a second time, the SIP call will be terminated and, in some cases, immediately called again.

Interval Between Tones

Available when Action is set to Make SIP Call with Tone.

Specify the time delay (seconds) between tones.

Maximum Tone Duration

Available when Action is set to Make SIP Call with Tone.

Select the maximum tone duration. The tone will be terminated once the maximum time is reached.

Action When Tamper Detected

8305 can be configured with supervision to execute one of the above three actions (Play Tone, Make Two-Way SIP Voice Call, Make SIP Call with Tone) if the device goes offline due to wiring failure or after being tampered with.

For example, a tone could sound over the speaker(s) or a private pre-recorded message could be sent to a specified telephone extension. The supervision configuration options will appear if a Relay Input Mode with supervision is selected.

See “Action When Input Triggered” above for information on additional settings.

Wiring Fault Supervision Mode

Short circuit detection will be triggered when the current draw is <4 mA. Short circuit detection will trigger when the current draw is >36 mA. The nominal source voltage on the Relay Input circuit is 13 V with a 40 mA current limit.

Action

Play Tone
When the 8305 input is triggered, a tone or a pre-recorded audio file will play over the speaker or multicast. This function can be used to request support or assistance in service or retail environments, notify about an emergency at a specific location in medical or educational facilities, or sound an alarm during an intrusion.

Make Two-Way SIP Voice Call
When the 8305 input is triggered, a voice path will open for an intercom-like call via an external microphone connected to a pre-configured telephone extension. This option can be used when a call needs to be made from a public place where a telephone would not be practical to use.

Make SIP Call with Tone
When the 8305 input is triggered, a private call can be made to a pre-configured telephone extension with a pre-recorded message. For instance, a call to a supervisor’s telephone notifying about an emergency or intrusion at some location.

Tone/Pre-recorded Announcement

Available when Action is set to Play Tone or Make SIP Call with Tone.

Select a recording or tone to use. Custom audio files may be used and uploaded through System →  File Manager.

Tone Duration

Available when Action is set to Play Tone.

Tone Multicast Settings

Use Separate Multicast

When enabled, the set tone will be played via multicast even if the 8305 is configured as a receiver. See additional options when enabled.

Multicast Mode

Use the same details as the receiver zone that is being listened to.

IP Address

Use the same details as the receiver zone that is being listened to.

Port

Use the same details as the receiver zone that is being listened to.

Outbound SIP Call Settings

Outbound Ring Limit

Available when Action is set to Make Two-Way SIP Voice Call or Make SIP Call with Tone.

Select the number of rings that will occur before the call reaches voicemail. One ring is six seconds.

Ringback Tone

Available when Action is set to Make Two-Way SIP Voice Call.

Select a ringback tone to play during an outbound SIP call while waiting for the far-end party to answer.

Maximum Call Duration

Available when Action is set to Make Two-Way SIP Voice Call.

Audio Input Settings

Audio Input Port

Select the input port your device is connected to.

Audio Input Volume

Set the desired input volume.

Output

Output Relay

Enable to close the circuit when you attempt to trigger events elsewhere.

This setting controls whether the output relay activates or not. Note that when enabled, the output relay will activate whenever the 8305 is activated (paging, alerting, etc.) This is a normally open relay only.

API Support

RESTful API

Enable a secure API for remote access and device control via HTTP. For more information, see the Algo RESTful API Guide.

Authentication Method

Speak to your IT Administrator for more information.

RESTful API Password

Speak to your IT Administrator for more information.

SCI Support

SCI

Simple Control Interface (SCI) is a separate control interface for certain applications. Its primary purpose is to support phones that may have programmable keys that can only send out HTTP GET requests.

InformaCast

InformaCast Support  

This feature requires a valid InformaCast license to be activated. Please contact sales@algosolutions.com for assistance.

Syn-Apps

The SA-Announce feature cannot be used when Multicast Transmitter mode or Poly mode is enabled. To enable SA-Announce mode, set Multicast Mode to None in Basic Settings → Multicast.

SA-Announce Support

Enable to convert unicast streams to multicast and deliver them to the target endpoints.

SA-Announce Server  

Enter the SA-Announce Server to use the Syn-Apps paging feature. Leave the field blank to use the server provided by the DHCP Option 72.

Local Management Port

Enter the local management port for the SA-Announce Server.

Microsoft

Microsoft Teams Support

Enable to provision the device via Microsoft’s servers. The device reboot will take up to 5 minutes to complete. This feature requires a compatible release from Microsoft.

ADMP Cloud Monitoring

Enable ADMP Cloud Monitoring

The Algo Device Management Platform (ADMP) simplifies the process of managing, monitoring, and maintaining Algo devices from any location. This feature requires a valid Account ID. To learn more about ADMP and how to purchase a license, visit the website.

Simple Network Management Protocol

SNMP Support

The existing setting will respond to a simple status query for automated supervision.

SNMP Community String  

Speak to your IT Administrator for more information.

SNMPv3 Security

Speak to your IT Administrator for more information.

RTP Control Protocol (RTCP)

RTCP Port Selection

Select how a port will be chosen to send or receive RTCP packets.

Note: If Next Higher Port is selected, ensure that the default multicast zone definitions are modified so that zones are only assigned to even-numbered ports, leaving the next higher odd-numbered ports free for RTCP packets.

Common

Internet Protocol

Use the dropdown to select IPv4 Only or IPv4 and IPv6. If IPv6 is also configured, it will have to be set up via DHCP or statically, similarly to the IPv4.

Supersede DNS provided by DHCP

Use this option to enter a custom DNS server address and supersede the one sent via DHCP.

DNS Servers

Add one or multiple DNS servers when Supersede DNS provided by DHCP is enabled. Separate each server by a space, comma, or semicolon.

IPv4

IPv4 Method

The 8305 can be set to a static or DHCP IP address.

DHCP is an IP standard designed to simplify the administration of IP addresses. When selected, DHCP will automatically configure IP addresses for each 8305 on the network. DHCP is selected by default.

When Static is selected, the IP address entered in the fields below will be used by the device.

IPv4 Address/Netmask

Enter the static IP address and netmask (CIDR format) for the 8305 (e.g., 192.168.1.23/24).

IPv4 Gateway

Enter the gateway address.

IPv6

IPv6 Method

The 8305 can be set to a static or DHCP IP address.

DHCP is an IP standard designed to simplify the administration of IP addresses. When selected, DHCP will automatically configure IP addresses for each 8305 on the network.

When DHCP is selected, the DHCP will automatically configure IP addresses for the 8305 on the network.

IPv6 Address/Netmask

Enter the static IP address and netmask (CIDR format) for the 8305 (e.g., 2001:123::abcd:1234/64).

IPv46Gateway

Enter the gateway address.

ICMPv6 Options

Destination Unreachable messages

Enable to restrict traffic by filtering ICMPv6 packets.

Neighbour Discovery Redirect messages

Enable to restrict traffic by filtering ICMPv6 packets.

Anycast Echo Replies

Enable to restrict traffic by filtering ICMPv6 packets.

802.1Q Virtual LAN

VLAN Mode

VLAN tagging is the networking standard that supports Virtual LANs (VLANs) on an Ethernet network. The standard defines a system of VLAN tagging for Ethernet frames and the accompanying procedures to be used by bridges and switches in handling such frames. The standard also provides provisions for a quality-of-service prioritization scheme known as IEEE 802.1p and defines the Generic Attribute Registration Protocol.

VLAN ID

Specify the VLAN that the Ethernet frame belongs to. The hexadecimal values 0x000 and 0xFFF are reserved. All other values may be used as VLAN identifiers, allowing up to 4094 VLANs.

The reserved value 0x000 indicates that the frame does not belong to any VLAN. In this case, the 802.1Q tag specifies only a priority and is referred to as a priority tag.

VLAN Priority

Set the frame priority level. Otherwise known as Priority Code Point (PCP), VLAN Priority is a 3-bit field that refers to the IEEE 802.1p priority or frame priority level. Values are from 0 (lowest) to 7 (highest).

802.1X Port-based Network Access Control

802.1x Authentication

Enable to add credentials to access LAN or WLAN that have 802.1X network access control (NAC). You can ask your IT Administrator for this information

Authentication Mode

Select the desired authentication mode.

Anonymous ID

If configured, the 8305 will send the anonymous ID to the authenticator instead of the 802.1X client username.

ID

The ID should contain a string identifying the IEEE 802.1X authenticator originating the request. Ask your IT administrator for details.

Password

Ask your IT administrator for details.

Validate Server Certificate

Enable to validate the authentication server against common authorities. To validate additional certificates, go to the System →  File Manager to upload a Base64 encoded X.509 certificate file in .pem, .cer, or .crt format to the ‘certs’ folder.

Differentiated Services

Differentiated Services provide quality of service if the DSCP protocol is supported on your network. Differentiated Services can be specified independently for SIP control packets and RTP and RTCP audio packets.

SIP (6-bit DSCP value)

Enter the DSCP value for SIP packets.

RTP (6-bit DSCP value)

Enter the DSCP value for RTP packets.

RTCP (6-bit DSCP value)

Enter the DSCP value for RTCP packets.

DNS

DNS Caching Mode

There are three mode options:

1. Disabled: No DNS queries will be cached.

2. SIP: Only the results of DNS queries for SIP requests will be cached.

3. All: The results of all DNS queries will be cached.

Admin Password

Use this section to change the admin password for logging into your 8305 web interface. It’s recommended that you change the admin password from the default to secure the device on your network.

Old Password

Enter the old admin password. The default password when you first get the device is algo.

Password

Enter a new admin password to log into the 8305 web interface. Make sure the new password is stored safely. If the password is forgotten, you must reset the device entirely with the Reset Button to restore the default password. All other settings will be reset to the original default settings as well.

For additional password security, see the setting: Force Strong Password.

Confirmation

Re-enter your new admin password.

General

Device Name (Hostname)

Add a name to identify the device in the Algo Network Device Locator Tool.

Introduction Section on Status Page

Turn On to show the introduction text on the login screen.

Show Status Section on Status Page when Logged Out

Turn On to allow others to view the status page without logging in. If turned Off, the settings and configurations on the status page will be hidden entirely unless a user is logged in to ensure only trusted users can view device information.

Display Switch Port ID on Status Page

Turn On to display the Switch Port ID on the Status Page. This option is only possible if the 8305 is connected to a switch that supports LLDP or CDP.

Web Interface Session Timeout

Set the maximum duration of inactivity to log a user out of the web interface automatically.

Play Tone at Startup

Enable to play a tone at start-up to confirm that the device has booted. This can be useful when testing or configuring a device but might not be desirable if the device is connected to an external legacy communication system and paging system.

Log Settings

Log Level  

This setting should only be used after consulting with the Algo support team.

Log Method  

Select a Log Method:

• Local: The log file is saved on the device in RAM.

• Method: Send the log file to a server repeatedly so settings are not lost if the device is rebooted.

• Both: Use both methods.

Log Server

Enter the Syslog server address provided by your IT administrator.

Management

Web Interface Protocol  

HTTPS is always enabled on the device. HTTP is enabled by default but may be disabled. To do so, select HTTPS Only mode so requests are automatically redirected to HTTPS.

Note that no security certificate exists since the device can have any address on the local network. Therefore, most browsers will provide a warning when using HTTPS.

Force Strong Password  

When Enabled, you can enforce a secure password for the 8305 web interface for additional protection. The password requirements for a strong password are:

• Must contain at least 10 characters

• Must contain at least 1 uppercase character

• Must contain at least 1 digit (0 – 9)

• Must contain at least 1 special character

Allow Secure SIP Password

When Enabled, SIP passwords are stored in the configuration file in an encrypted format to prevent viewing and recovery. If enabled, navigate to Basic Settings → SIP and fill out the field Realm. To obtain your SIP Realm information, contact your SIP Server administrator or check the SIP log file for a registration attempt. The Realms may be the same or different for all the extensions used.

All the configured Authentication Password(s) must be re-entered here as well as any other locations where SIP extensions have been configured to save the encrypted password(s).

If the Realm is changed later, all passwords must be re-entered to save the passwords with the new encryption.

Display Switch Port ID on Status Page

Turn On to display the Switch Port ID on the Status Page. This option is only possible if the 8305 is connected to a switch that supports LLDP or CDP.

Simple Network Management Protocol

SNMP Support

The existing setting will respond to a simple status query for automated supervision.

SNMP Community String  

Speak to your IT Administrator for more information.

SNMPv3 Security

Speak to your IT Administrator for more information.

API Support

RESTful API

Enable a secure API for remote access and device control via HTTP. For more information, see the Algo RESTful API Guide.

Authentication Method

Speak to your IT Administrator for more information.

RESTful API Password

Speak to your IT Administrator for more information.

SCI Support

SCI

Simple Control Interface (SCI) is a separate control interface for certain applications. Its primary purpose is to support phones that may have programmable keys that can only send out HTTP GET requests.

System Integrity

System Integrity Checking

Enable this feature to verify that installed system packages have not been tampered with by running a check. Enabling this feature may cause reboots and upgrades to take 30 seconds longer. Verification results can be found on the Status tab.

Syn-Apps

The SA-Announce feature cannot be used when Multicast Transmitter mode or Poly mode is enabled. To enable SA-Announce mode, set Multicast Mode to None in Basic Settings → Multicast.

SA-Announce Support

Enable to convert unicast streams to multicast and deliver them to the target endpoints.

SA-Announce Server  

Enter the SA-Announce Server to use the Syn-Apps paging feature. Leave the field blank to use the server provided by the DHCP Option 72.

Local Management Port

Enter the local management port for the SA-Announce Server.

InformaCast

InformaCast Support  

This feature requires a valid InformaCast license to be activated. Please contact sales@algosolutions.com for assistance.

Microsoft

Microsoft Teams Support

Enable to provision the device via Microsoft’s servers. The device reboot will take up to 5 minutes to complete. This feature requires a compatible release from Microsoft.

ADMP Cloud Monitoring

Enable ADMP Cloud Monitoring

The Algo Device Management Platform (ADMP) simplifies the process of managing, monitoring, and maintaining Algo devices from any location. This feature requires a valid Account ID. To learn more about ADMP and how to purchase a license, visit the website.

General

Timezone

Select a time zone for 8305 settings.

NTP Time Servers 1/2/3/4

The device will attempt to use Timer Server 1 and work down the list if one or more of the time servers become unresponsive.

These settings are pre-populated with public NTP servers hosted on the internet. To use these, the device requires internet connection. Alternatively, this can be customized to point the device to any other NTP server hosted or premise-based.

Supersede NTP provided by DHCP

By default, if an NTP Server address is provided via DHCP Option 42, it will be used instead of the NTP servers listed above. Enable this option to ignore DHCP Option 42.

Device Date/Time

This field shows the current time and date set on the device. If you are testing the device on a lab network that does not have access to an external NTP server, click Sync with browser to temporarily set the time on the device.

This time value will be lost at power down or overwritten if connection to the NTP server is available. Time and date are used for logging purposes and the scheduler feature.

Manually Override Time

Manual time and date are intended for testing purposes only. Time will be lost upon power down if the NTP server is reachable.

Mode

Provisioning Mode

Enabling provisioning allows installers to pre-configure 8305 on a network before installation. This is typically done for large deployments to save time and ensure consistent setups.

It is recommended that Provisioning Mode be set to Disabled if this feature is not in use. This will prevent unauthorized re-configuration of the device if DHCP is used.

Visit the Algo Provisioning Guide for more information.

Settings

Server Method

Select a Server Method.

• Auto: All three DHCP options (66, 160, 150) will be automatically checked for an active provisioning server

• DHCP Option 66 Only: Only DHCP Option 66 will be checked for a provisioning server

• DHCP Option 160 Only: Only DHCP Option 160 will be checked for a provisioning server

• DHCP Option 150 Only: Only DHCP Option 150 will be checked for a provisioning server

• Static: Only the specified static server will be checked for a provisioning server

For provisioning to work with a DHCP option, DHCP must be enabled under Advanced Settings → Network → IPv4.

Static Server

Enter the server address or domain.

Download Method

Select your preferred method for downloading provisioning files. The options are:

• TFTP (Trivial File Transfer Protocol) — See MD5 Checksum below for more details.

• FTP

• HTTP

• HTTPS — This may help prevent configuration files from being read by an unwanted third party and having sensitive data stolen.

The 8305 configuration files can be automatically downloaded from a provisioning server using DHCP Option 66. This option code (when set) supplies a TFTP boot server address to the DHCP client to boot from.

One of two files can be uploaded on the Provisioning Server (for access via TFTP, FTP, HTTP, or HTTPS):

• Generic (for all Algo 8305 Multi-Interface IP Paging Adapter) algop8305 .conf

• Specific (for a specific MAC address) algom[MAC].conf

Both protocol and path are supported for Option 66, allowing for http://myserver.com/config-path to be used.

Validate Server Certificate

Enable to verify the server. This checks if the certificate provided by the server is signed by any CAs included in the list of trusted CAs (used by the Debian infrastructure and Mozilla browsers). If a certificate signed by any of these CAs is received, that server will be trusted.

This parameter can also be enabled through provisioning:

Prov.download.cert = 1

(FTP) Auth User Name

Speak to your IT Administrator for more information.

(FTP) Auth Password

Speak to your IT Administrator for more information.

(HTTP) Auth User Name

Speak to your IT Administrator for more information.

(HTTP) Auth Password

Speak to your IT Administrator for more information.

(HTTPS) Validate Server Certificate

Speak to your IT Administrator for more information.

(HTTPS) Auth User Name

Speak to your IT Administrator for more information.

(HTTP) Auth Password

Speak to your IT Administrator for more information.

Config Download Path

Enter the path where the configuration file is located within the provisioning server (e.g., algo/config/8305 ).

Firmware Download Path

Enter the path where the firmware file is located within the provisioning server (e.g., algo/firmware/8305 ).

Partial Provisioning

Enable to allow support for “-i” incremental provisioning files. Disable for enhanced security if this is not required.

Check-sync Behavior

Select Always Reboot to set the 8305 to always reboot despite other settings.

Select Conditional Reboot to set the 8305 to check the provisioning server. Only reboot if a new config is found (unless “reboot=true” is provided as a parameter in the check-sync event).

Sync Start Time

Set a time (HH:mm:ss) for the device to perform a sync according to the Check-sync Behavior setting. Leave this blank if not needed.

Sync End Time

If set, the device will sync randomly in the window between Sync Start Time and Sync End Time. Setting an End Time earlier than the Start Time indicates an overnight period. Leave blank to lank to sync exactly at the set start time.

Sync Frequency

Select the sync frequency. Frequency can be set to Daily or Selected Days Only.

Sync Days

Select the days of the week to for syncs to occur.

Backup/Restore Configuration

Download Configuration File

Save 8305 configuration settings to a text file for backup or to set up a provisioning configuration file.

Restore Configuration File

Restore settings by uploading a backup file.

Restore Configuration to Defaults

Reset all 8305 Multi-Interface IP Paging Adapter settings to factory default values.

Backup/Restore All User Files

Download Backup Zip File

Download the 8305 configuration settings and uploaded files in File Manager (ex., certificates, licenses, and tones) to a backup ZIP file.

Restore from Backup Zip File

Restore the 8305 configuration settings and files in File Manager (ex., certificates, licenses, and tones) by uploading a backup zip file.

Restore All Settings and Files to Defaults

Reset the 8305 configuration settings. All preloaded and uploaded files, including tone files, will be deleted

Reboot

Reboot the Device

Reboots the device.

Installed Firmware

Product Firmware

Displays the current firmware on the device.

Online Upgrade

Check for Firmware Updates

Click Check to check for the latest firmware. If the firmware is up to date, Latest Firmware will state Firmware up to date. If your firmware is outdated, the new firmware availability will be listed. Internet connection is required.

Custom Upgrade

Method

Select a method for firmware upgrades to occur. This can be done From Local Files or From URL.

Signed Firmware File

Use to upgrade firmware from a local file.

To do this, download the firmware file from https://www.algosolutions.com/firmware-downloads/ then upload the file by clicking on Choose File and selecting the firmware file.

Click Upgrade at the bottom of the interface.

Once the upgrade is complete, you can confirm the firmware version is changed by looking at the top right of the web interface.

Upgrade URL

Instead of downloading the firmware file https://www.algosolutions.com/firmware-downloads/, you may add the download link here instead.

Click Upgrade at the bottom of the interface.

Once the upgrade is complete, you can confirm the firmware version is changed by looking at the top right of the web interface.

Allow Downgrade

Enable to allow product or base firmware to be downgraded to an older patch version. Enabling this option could cause future upgrade issues.

If you require downgrading, please contact support@algosolutions.com for assistance.

Figure 31: 1202 Call Button – the insert card is interchangeable.

Figure 32: 1202 Call Button wiring.

Figure 33: 1203 Call Switch

Figure 34: 1203 Call Switch wiring

Volume Control Switch Settings

Mute On Lowest Setting

Enable to mute audio when the volume control switch is turned to the lowest setting (1)

Wire Length

Set to calibrate impedance for 24 AWG.

Multicast Override

Multicast Override allows selected multicast zones to override the 1204 Volume Control settings for the selected zones.

Remote Volume Settings

Remote Mode

Configure the device to subscribe to a remote 1204 volume input or to notify remote devices of 1204 volume input.

Note that if Notify (Local 1204) or Subscribe (Remote 1204) are selected that a RESTful API must be enabled under Advanced Settings →  Admin.

Figure 38: 1204 Volume Control

Figure 39: 1204 Volume Control Switch Wiring