Intercom Call Configuration
  • 11 Dec 2023
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Intercom Call Configuration

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Article Summary

IP Call & IP Call Configuration

An IP call is a direct call between two intercom devices using their IP addresses, without a server or a PBX. IP calls work when the devices are on the same network.

To do this configuration on web Phone > Call Feature > Direct IP interface.
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Parameters Set-up:

  • Direct IP port: set up the IP direct call port, 5060 is the default port.

SIP Call &SIP Call Configuration

Session Initiation Protocol(SIP) is a signaling transmission protocol used for initiating, maintaining, and terminating calls. 

A SIP call uses SIP to send and receive data between SIP devices, and can use the internet or a local network to offer high-quality and secure communication. Initiating a SIP call requires a SIP account, a SIP address for each device, and configuring SIP settings on the devices.

SIP Account Registration

Each device needs a SIP account to make and receive SIP calls. 

Akuvox intercom devices support the configuration of two SIP accounts, which can be registered under two independent servers.

Configure SIP Account Configuration

To perform the SIP account setting on the Web Account > Basic > SIP Account Interface. Register Name, User Name, and Password are provided by the SIP account administrator.
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Parameter Set-up:

  • Status: check to see if the SIP account is registered or not.
  • Account: select the exact account (Account 1&2) to be configured.
  • Display Name: configure the name, for example, the device’s name to be shown on the device being called to.
  • Display Label: configure the device label to be shown on the device screen.

SIP Server Configuration

SIP servers enable devices to establish and manage call sessions with other intercom devices using the SIP protocol. They can be third-party servers or built-in PBX in Akuvox indoor monitor.

To do this configuration on web Account > Basic > SIP Server interface.
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Parameter Set-up:

  • Server IP: enter the primary server IP address number or its URL.
  • Server IP: enter the backup SIP server IP address or its URL.
  • Port: set up SIP server port for data transmission.
  • Registration Period: set up SIP account registration time pan. SIP re-registration will start automatically if the account registration fails during the registration time span. The default registration period is 1800, ranging from 30-65535s.

Configure Outbound Proxy Server

An outbound proxy server is used to receive all initiating request messages and route them to the designated SIP server in order to establish a call session via port-based data transmission.

To set it up on the device web Account > Basic > Outbound Proxy Server Interface.
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Parameter Set-up:

  • Server IP: enter the SIP address of the primary outbound proxy server.
  • Port: enter the Port number to establish a call session via the primary outbound proxy server.
  • Backup Server IP: set up backup server IP for the backup outbound proxy server.
  • Port: enter the port number for establishing call sessions via the backup outbound proxy server.

Configure Data Transmission Type

Akuvox intercom devices support four data transmission protocols: User Datagram Protocol(UDP)Transmission Control Protocol(TCP), Transport Layer Security(TLS), and DNS-SRV.

To do this configuration on web Account > Basic > Transport Type interface.
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Parameter Set-up:

  • UDP: select UDP for unreliable but very efficient transport layer protocol. UDP is the default transport protocol.
  • TCP: select TCP for a reliable but less-efficient transport layer protocol.
  • TLS: select TLS for secured and Reliable transport layer protocol.
  • DNS-SRV: select DNS-SRV to obtain a DNS record for specifying the location of servers. And SRV not only records the server address but also the server port. Moreover, SRV can also be used to configure the priority and the weight of the server address.

Configure Calling Feature

DND

The Do Not Disturb(DND) feature prevents unwanted incoming SIP calls, ensuring uninterrupted focus. It also allows you to set a code to be sent to the SIP server when rejecting a call.

Go to Phone > Call Feature > DND interface.
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Parameter Set-up:

  • Return Code When DND: select what code should be sent to the calling device via the SIP server. 404 for not found; 480 for temporary unavailable; 486 for busy here; 606 for decline.

Enable Prevent SIP Hacking

Internet phone eavesdropping is a network attack that allows unauthorized parties to intercept and access the content of the communication sessions between intercom users. This can expose sensitive and confidential information to the attackers. SIP hacking protection is a technique that secures SIP calls from being compromised on the Internet.

You can enable Prevent SIP Hacking if you only want to receive the calls made by the callers in your contact list. To enable it, go to Account > Advanced > Call.
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Speed Dial Call

Speed Dial is used to quickly initiate the pre-configured numbers by pressing the Dial key. You can create up to 16-speed dial numbers. To do the configuration on the web Intercom > Basic > Speed Dial interface.
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Group Call

You can make calls to a group of numbers by pressing the on the device dial pad. To set the group call, go to Intercom > Basic > Manager Dial.
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Parameter Set-up:

  • Call type: select Group Call.
  • Group Call Number (Local): enter the group call number. If you fill in the local group call number, then the local group number will be called instead of the SmartPlus group call number.

Sequence Call

Sequence Call is a feature that allows you to dial a group of numbers in a predefined order until one of them answers. This feature is supported by Akuvox SmartPlus, which provides a set of sequence call numbers for the application.

To do the configuration on the web Intercom > Basic > Manager Dial interface.
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Parameters Set-up:

  • Call Type: select Sequence Call.
  • Call Timeout (Sec): set the call timeout before calling the next called party when the first called party does not receive the call within the timeout.
  • When Refused: if you select Do Not Call Next, then the sequence call will be terminated if the call is rejected by the called party. If you select Call Next, then the sequence call will be continued to the next called party if it is rejected by the first called party.

Web Call

The web call feature allows for making calls via the device’s web interface, commonly used for remote call testing purposes.

To do the configuration on the web Intercom > Basic > Web Call interface.
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Parameters Set-up:

  • Auto/Account1/Account2: to choose a suitable SIP account to make a web call. If you call using an IP address, account selection is not needed here.

Dial Plan

The dial number replacement feature simplifies long and complex dial numbers of the device, providing shorter and more user-friendly alternatives for making calls. It allows the substitution of multiple dial numbers, such as IP addresses or SIP numbers, with a single, simplified number.

To configure the number replacement on the device, navigate to Phone > Dial Plan, then click Add. To replace the number in batch, you can import the .xml file to the door phone. And the file from the door phone can be exported out before importing them to other door phones.
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Auto Answer

Auto-answer feature allows the device to automatically pick up incoming calls without any manual intervention. You can also customize this feature by setting the time duration for auto-answering and choosing the communication mode between audio and video.

To enable this feature on web Account > Advanced > Call interface, you can set up the related parameters on web Phone > Call Feature> Auto Answer.

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Parameters Set-up:

  • Auto Answer Delay: set up the delay time (from 0-5 sec.) before the call can be answered automatically. For example, if you set the delay time as 1 second, then the call will be answered in 1 second automatically.
  • Mode: set up the video or audio mode you preferred for answering the call automatically.

Multicast

The Multicast function allows one-to-many broadcasting for different purposes. For example, it enables the indoor monitor to announce messages from the kitchen to other rooms, or to broadcast notifications from the management office to multiple locations. In these scenarios, indoor monitors can either listen to or send audio broadcasts.

To do the configuration on the web Phone > Multicast interface.
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Parameters Set-up:

  • Multicast Priority Paging Barge: multicast or how many multicast calls are higher priority than SIP calls, if you disable paging priority, SIP calls will have high priority.
  • Paging Priority Enabled: multicast calls are called in order of priority or not.
  • Listening Address: enter the multicast IP address you want to listen to. The multicast IP address needs to be the same as the listened part and the multicast port cannot be the same for each IP address. Multicast IP address is from 224.0.0.0 to 239.255.255.255.

Configure Maximum Call Duration

The door phone allows you to set up the call time duration in receiving the call from the calling device as the caller side might forget to hang up the intercom device. When the call time duration is reached, the door phone will terminate the call automatically.

To do this configuration on the web Intercom > Basic > Max Call Time interface.
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Parameters Set-up:

  • Max Call Time: enter the call time duration according to your need (ranging from 0-120 min). The default call time duration is 5 min.
Note:
  • Max call time of the device is also related to the max call time of the SIP If using a SIP account to make a call, please pay attention to the max call time of the SIP server. If the max call time of the SIP server is shorter than the max call time of the device, the shorter one is available.

Maximum Dial Duration

Maximum Dial Duration is the time limit for incoming- and/or outgoing calls on the door phone. If configured, the door phone will automatically terminate the call if no one answers the call within the preset time, whether it is incoming or outgoing.

To do this configuration, go to Intercom > Basic > Max Dial Time interface.
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Parameters Set-up:

  • Dial in Time: enter the dial-in time duration for your door phone (ranging from 5-120 sec). For example, if you set the dial-in time duration as 60 seconds on your door phone, then the door phone will hang up the incoming call automatically if the call is not answered by the door phone in 60 seconds. 60 seconds is the dial-in time duration by default.
  • Dial out Time: enter the dial-in time duration for your door phone (ranging from 5-120 sec). For example, if you set the dial-out time duration as 60 seconds on your door phone, then the door phone will hang out the call it dialed out automatically if the call is not answered by the device being called.
Note:
  • Max dial time of device is also related with max dial time of SIP server. If using SIP account to make a call, please pay attention to the max dial time of SIP server. If the max dial time of SIP server is shorter than the max dial time of device, the shorter one is available.

Hang Up After Open Door

This feature automatically ends the call once the door is released, allowing for the seamless reception of subsequent calls.

To do this configuration on the web Intercom > Basic> Hang Up After Open Door.
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Parameter Set-up:

  • Type: select the open door type. The door can be unlocked via the DTMF, HTTP, DTMF or HTTP, and Input, DTMF, or HTTP.
  • Timeout: set up from 1 second to 15 seconds. 5 seconds is the default. If you set it 5 seconds, then the call will be hung up 5 seconds after the door is opened. If you want to disable the feature, set the timeout as 0.

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