8186 IP Horn Speaker - User Guide

Terminal Block Relay In

Connection options are a normally closed switch, normally open switch, 1202 Call Button, 1203 Call Switch, 1204 Volume Control Switch or EOL resistor termination. The web interface of the 8186 can be used to configure the action a connected device will trigger.

Terminal Block Relay Out

By default, these terminals provide a normally open contact closure when the 8186 is active.

Terminal Block Reset

The reset terminal can be used to reset the 8186 IP Horn Speaker during power up. To return all the settings to the factory default, reboot or power cycle the 8186. Wait until the LED flashes, then connect the reset terminals and hold until the 8186 LED begins a double flash pattern. Release the reset connection and allow the unit to complete its boot process.

Do not short the reset terminals until the LED begins flashing.

A reset will set all configuration options to factory default including the password.

Once booting has completed, shorting the reset terminals will cause the device to speak its IP address.

• Device Name

• SIP Registration

• Call Status

• Proxy Status

• Provisioning Status

• MAC

• IPv4

• IPv6

• Date/Time

• Multicast Mode

• Volume

• Relay Input Status

• InformaCast License

• ADMP Cloud Monitoring

One Extension (DTMF) for Emergency Alerts

Multiple Extensions (Direct Dialing) for Emergency Alerts

Pros

• Only one extension or UC license needs to be registered, saving money

• Calls will be auto-answered and a confirmation tone can be played to provide feedback to the user

Cons

• Users must memorize DTMF keys individually, which can be challenging to recall

Pros

• Can set extensions as speed dials which requires less user training

• Option to use auto-answer or not.
  
  With auto-answer: A confirmation tone can be played to provide feedback to the user
  
  Without auto-answer: Can detect if the extension is part of a ring group and prevent interference with other calls or configurations such as loud ringing or other alerting

Cons

• Requires use of multiple extensions which increases overall cost

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 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 8186 will not answer the call on this extension.

Often, the 8186 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 8186.

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 8186 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 associated with 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 used to authenticate SIP users.

Contact your System Administrator for the password.

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.

DTMF (One Extension)

Multiple Extensions

Pros

• Only one extension or UC license needs to be registered, saving money

• Calls can be auto-answered and a confirmation tone can be played to provide feedback to the user

Cons

• Users must memorize DTMF keys individually, which can be challenging to recall

Pros

• Can set extensions as speed dials which requires less user training

• Option to use auto-answer or not.
  
  With auto-answer: A confirmation tone can be played to provide feedback to the user
  
  Without auto-answer: Can detect if the extension is part of a ring group and prevent interference with other calls or configurations such as loud ringing or other alerting

Cons

• Requires use of multiple extensions which increases overall cost

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 will play for.

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

Enable to answer inbound calls with the option to play a confirmation tone before starting an alert.

If disabled, the inbound ring signal will be detected, but the device will not answer the call. In some cases, keeping this setting disabled may be useful if the extension is part of a ring group. This will prevent interference with other calls or configurations such as loud ringing or other alerting.

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 and there will be no change to the current announcement state.

Passcode Prompt Tone

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

DTMF Selection

This extension will be used to activate and optionally cancel emergency alerts when Announcement Selection Mode is set to DTMF Selectable. Use the configurations below to register a single extension that will accept DTMF input to play selected announcements.

Extension

Enter the SIP extension for the DTMF Selection parameter.

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. Using DTMF 0 allows emergency alerts to work with only a single SIP registration rather than requiring multiple accounts.

Extension

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

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.

Confirmation Tone

Select a tone to play to confirm that an alert has been canceled.

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.

Alternatively, DTMF Selectable Mode can be used if the SIP telephone system limits 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 the announcement.

Confirmation Tone

Select a file to use as a confirmation tone.

Multicast Zone

Set the RTP multicast zone where announcements will be played.

Multicast Poly Group

Set the Poly Group where announcements will be played.

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 device. 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 device is registered with a hosted server or TLS is used for SIP Transportation.

Only enable this option if the SIP server supports RFC 5626.

Outbound Proxy

Enter the IP address for an outbound proxy.

Register Period (seconds)

Enter the maximum requested period where the device 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.

Rate Limit SIP Registration

This option should be used in cases where many SIP extensions are registered (ex. one for each zone).

Select a rate limit to stagger registration requests and prevent overloading the server by sending them all at the same time.

Wait for Successful Unregister

Enable to wait for the device to successfully unregister from the server. Enabling may cause a slight delay during configuration changes and reboots

SRTP

SDP SRTP Offer

Select an option from the dropdown menu:

Disabled

Standard: Encrypts RTP voice data to secure audio RTP packets (SRTP). SIP calls will be rejected if the other party does not support SRTP. This option secures the audio data between parties by ensuring that it’s not left out for third parties to reconstruct and listen to.

Optional (Non-standard AVP Profile): The SIP call’s RTP data will be unencrypted if the other party does not support SRTP.

SDP SRTP Offer Crypto Suite

The encryption and authentication algorithms used for voice data.

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 device 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 device 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.

Zoom Phone Local Survivability

Local Survivability

Enable to re-register with local ZPLS Node if connection to Zoom fails. This allows sites to maintain a subset of Zoom Phone features even if connectivity to the Zoom Phone cloud is lost.

Survivability Proxy

The IP address or domain name of the local ZPLS node.

Interoperability

Keep-Alive Method

Select a keep-alive method:

None

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

Keep-Alive Interval (seconds)

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

Multicast Mode

If Transmitter (Sender) is selected, the 8186 will broadcast an IP stream when activated in addition to playing audio. The 8186 cannot be both a multicast Transmitter and Receiver simultaneously.

Multicast Type

The 8186 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 8186 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

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

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.

Speaker Playback Zones

Select Speaker Playback Zones to control which specific zones the 8186 can play audio for. This is useful if using the DTMF Selectable Zone mode or additional page extensions (Additional Features → More Page Extensions) per zone to make the 8186 a member of only certain zones. In this case, the transmitter does not participate in the zone but can still send audio to speakers in different zones.

DTMF Settings

This section is used if the Zone Selection Mode is set to DTMF Selectable Zone.

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 8186 will activate when receiving a multicast audio stream. 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 8186 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 8186. Poly Group Tones can be set in Advanced Settings → Advanced Multicast.

The Poly telephone used as page audio source for the 8186 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 indefinitely. 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 8186 may be heard slightly earlier than on these other devices. Use this feature to add a small delay on the 8186 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 using the Play, Loop, and Stop buttons.

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 the volume settings used for all audio received over multicast.

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, or delayed.

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

To cancel a page while in delayed 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 or puts the call on hold by mistake.

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 8186 uses this for multi-zone selection, passcode, etc.

Audio Processing

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 volume is adjusted automatically via the speaker's microphone.

Automatic Gain Control (AGC)

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

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 via the device.

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.

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.

Microphone

Global Microphone Mute

Enable to disable the microphone entirely.

Microphone Volume

Select a volume for the microphone.

Display

Power Levels

Enable to display audio power levels in the volume control pulldown menus. This will show the maximum power in Watts when a full scale input signal is used (e.g. 1kHz sine wave, with an RMS level of -3dBFS). This allows for an easy level comparison with legacy analog products or other Algo products that support PoE+ or satellite speakers.

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.

General

Relay Input Mode

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 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 8186, see Algo Compatible Accessories.

Action When Input Triggered

Action

Play Tone
When the 8186 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 8186 input is triggered, a voice path will open for an intercom-like call. 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 8186 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

The 8186 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 accessory device connected to the relay input goes offline due to a 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

Open 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 8186 input is triggered, a tone or a pre-recorded audio file will play over the speaker or multicast. For example, an audio file with the announcement: "Wiring fault detected on the emergency button of the Algo 8186 in the warehouse."

Make Two-Way SIP Voice Call
When the 8186 input is triggered, a voice path will open for an intercom-like call.

Make SIP Call with Tone
When the 8186 input is triggered, a call can be made using a pre-recorded audio file describing the failure.

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 8186 is configured as a receiver. To do this, a different multicast channel must be used to transmit audio. The separate multicast address must use a different port number from any of the zones that are already used as listening zones.

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.

Volume Control Switch Settings

Available when the Algo 1204 Volume Control Switch is selected as the relay input device. The settings below allow for variable volume control in the speaker location. For example, turning the speaker volume down in a classroom down during an exam.

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.

Remote Volume Settings

Available when the Algo 1204 Volume Control Switch is selected as the relay input device. This feature allows one 1204 to change the volume on multiple speakers. For example, changing the volume of multiple speakers in a school theatre.

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, the RESTful API must be enabled under Advanced Settings → Admin.

IP Address

Only used if Remote Mode is set to Subscribe (Remote 1204). The IP address of the Algo IP endpoint with a connected 1204.

Remote Device RESTful API Password

The RESTful API password used between the two (or more) Algo devices that are sharing a single 1204. The password must be the same across all devices.

Output

Output Light

Enable or disable the blue light on the speaker entirely (keep the light off even when the speaker is active).

Heartbeat Light

Enable to flash the blue light every 30 seconds to indicate that the speaker is powered and running.

Output Relay

This setting controls whether the output relay activates or not. Note that when enabled, the output relay will activate whenever the 8186 is activated (paging, alerting, etc.) This is a normally open relay only.

Output Triggering Events

Select an event to trigger the output relay.

Common

Internet Protocol

Use the dropdown to select IPv4 Only or IPv4 and IPv6.

Supersede DNS provided by DHCP

This setting will not appear if the selected Internet Protocol is set to Static. When enabled, this configuration allows DNS settings to be manually configured, replacing ones that may have been provided via DHCP.

IPv4

IPv4 Method

The device 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 device on the network. DHCP is selected by default.

When Static is selected, the device will use the IP address entered in the fields below.

IPv4 Address/Netmask

Enter the static IP address and netmask (CIDR format) for the device (e.g., 192.168.1.23/24 where “/24” is equivalent to a netmask of 255.255.255.0).

IPv4 Gateway

Enter the gateway address.

IPv6

IPv6 Method

The device 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 device on the network.

When Static is selected, the device will use the IP address entered in the fields below.

IPv6 Address/Netmask

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

IPv6 Gateway

Enter the gateway address.

ICMPv6 Options

Destination Unreachable Messages

Enable to restrict traffic by filtering ICMPv6 packets.

Neighbor Discovery Redirect Messages

Enable to restrict traffic by filtering ICMPv6 packets.

Anycast Echo Replies

Enable to restrict traffic by filtering ICMPv6 packets.

Enable Rate Limiting Outbound Messages

Enable to limit the device to respond to other network devices at the specified rate below and prevent it from receiving multiple requests at the same time.

Rate Limit (packets per second)

Specify the packets per second allowed for Rate Limiting Outbound Messages.

802.1Q Virtual LAN

If the device is using VLAN, you will need to be on the same VLAN to access the web interface, unless routing has been configured between VLANs.

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 device 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

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

TLS

Allow Weak TLS Ciphers

Enables compatibility with legacy systems that may not support the most current encryptions standards

Admin Password

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 device 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 device 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

The device can play a beep tone at startup.

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 in RAM on the device.

Network: Send the log file to an external SysLog server so settings are not lost if the device is rebooted, or for ease of central access.

Both: Use both methods.

Log Server

Enter the Syslog server address provided by your IT administrator.

Select Additional Log Events

To be used by support@algosolutions.com if necessary.

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 device 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 Passwords

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 Realm field. 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.

Simple Network Management Protocol

SNMP Support

Disabled by default. 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

Disabled by default. 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

Disabled by default. 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 and allow them to initiate events remotely on an Algo device.

SCI Password

Enter your SCI password.

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.

Intrado Revolution (formerly Syn-Apps)

Revolution Support

Enable the device to register with an Intrado Revolution Server and receive audio events from the system.

Revolution Server

Enter the Revolution Server to use the Revolution 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 Revolution Server.

InformaCast IP Speaker

InformaCast IP Speaker Support

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

Configuration Mode

Auto: The device will attempt to configure Informacast using DNS SRV, SLP, and/or via DHCP and TFTP

Manual: The device will allow the configuration file location to be manually configured.

Direct: The device will register to the list of static server addresses directly, bypassing the Configuration File Server

Configuration Retry Interval

Set the amount of time to wait before attempting to obtain configuration information after failure.

SIP Support

Enter the SIP credentials provided by InformaCast during configuration

Maximum Broadcast Duration

The maximum length of broadcast.

InformaCast Scenarios API

InformaCast Scenario API Support

Enable the device to start an InformaCast Scenario via relay input. This feature can work without an InformaCast license, as only the output device requires a license.

Microsoft

Microsoft Teams Support

Enable to allow the device to register with a Microsoft Teams account. The device reboot will take up to 5 minutes to complete, as the device will communicate several times with the Microsoft server. This feature requires a compatible release from Microsoft.

After enabling this setting, please return to the status page to sign into your Teams accounts. This feature requires a compatible release from Microsoft.

For more details, please see the Microsoft Teams Configuration Guide.

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 ADMP webpage.

Account ID

Enter the account ID listed on the Settings page of your ADMP account.

Allow Configuration File Sync

Enable ADMP to query and display settings stored on the device.

Heartbeat Interval

Select how often ADMP should check the status of your device.

Time Settings

Time Zone

Use the dropdown to select the time zone required for your device.

NTP Time Server

The interface 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.

NTP Symmetric Key Authentication

To enable, create a new folder in the tab System → File Manager and create a folder named ntp. Upload the symmetric key file and rename the file to ntp.keys.

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.

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 the device 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.

Settings

Server Method

Set to Auto by default. 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 device 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 8186) algop8186.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.

Config Download Path

Enter the path where the configuration file is located in the provisioning server (e.g., algo/config/8186).

Firmware Download Path

Enter the path where the configuration file is located in the provisioning server (e.g., algo/config/8186).

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 device to always reboot despite other settings.

Select Conditional Reboot to set the device and 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 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 for syncs to occur.

Zero Touch Provisioning

ZTP is enabled by default but is disabled when any changes are made to the device configuration. This button can also be used to disable ZTP if no changes have yet been made to the device configuration.

Backup/Restore Configuration

Download Configuration File

Save 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 device settings to factory default values.

Backup/Restore All User Files

Download Backup Zip File

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

Restore from Backup Zip File

Restore the device 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 device 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 to be downgraded to an older version. Enabling this option could cause future upgrade issues.

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