Akuvox R23X User Manual

R23X Door Phone User Manual

WWW.ACTIVEONLINE.COM.AU

1300 816 742

About This Manual

Thank you for choosing Akuvox’s products. In user manual, we provides all functions and

configurations you want to know about R23X. Please verify the packaging content and

network status before setting. This manual applies to firmware 26.0.0.96 or lower version.

Note: The old firmware may be a little different from 26.0.0.96 about some configuration.

Please consult your administrator for more information.

Content

1. Overview

1.1. Product Description

1.2. Feature

1.3. Panel Description

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2. Configuration

2.1. Web login

2.1.1 Obtain the IP address

2.1.2 Login the web

2.2. Status

2.2.1 Basic

2.3. Intercom

2.3.1 Basic

2.3.2 LED Setting

2.3.3 Relay&Input

2.3.4 AEC Setting

2.3.5 Multicast

2.3.6 Card Setting(R23C only)

2.4. Account

2.4.1 Basic

2.4.2 Account-Advanced

2.5. Network

2.5.1 Basic

2.5.2 Advance

2.6. Phone

2.6.1 Time/Lang

2.6.2 Call Feature

2.6.3 Voice

2.6.4 Phone-Multicast

2.6.5 Phone-Call Log

2.7. Upgrade

2.7.1 Basic

2.7.2 Advance

2.8. Security

2.8.1 Basic

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1

1.Overview

1.1. Product Description

R23C R23P

Akuvox's Audio Doorphone R23X is an open, non-proprietary and IP-based door

station for two-way communication and remote entry control. It is a perfect

complement to any SIP system and offers new possibilities of effectively control entry

to your premises.It’s applicable in villas, office and so on.

2

FCC Caution:

Any Changes or modifications not expressly approved by the party responsible for

compliance could void the user's authority to operate the equipment.

This device complies with part 15 of the FCC Rules. Operation is subject to the
following two conditions: (1) This device may not cause harmful interference, and (2)
this device must accept any interference received, including interference that may
cause undesired operation.

Note: This equipment has been tested and found to comply with the limits for a Class
B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to
provide reasonable protection against harmful interference in a residential
installation. This equipment generates, uses and can radiate radio frequency energy
and, if not installed and used in accordance with the instructions, may cause
harmful interference to radio communications. However, there is no guarantee that
interference will not occur in a particular installation. If this equipment does cause
harmful interference to radio or television reception, which can be determined by
turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:

—Reorient or relocate the receiving antenna.

—Increase the separation between the equipment and receiver.

—Connect the equipment into an outlet on a circuit different from that to which the
receiver is connected.

—Consult the dealer or an experienced radio/TV technician for help.

Specific Absorption Rate (SAR) information

SAR tests are conducted using standard operating positions accepted by the FCC with
the device transmitting at its highest certified power level in all tested frequency
bands, although the SAR is determined at the highest certified power level, the
actual SAR level of the device while operating can be well below the maximum value.
Before a new product is a available for sale to the public, it must be tested and
certified to the FCC that it does not exceed the exposure limit established by the FCC,
tests for each phone are performed in positions and locations as required by the FCC.
For headset, this part has been tested and meets the FCC RF exposure guidelines
when used with an accessory designated for this product or when used with an
accessory that contains no metal.
For baseband, this equipment complies with FCC radiation exposure limits set forth
for an uncontrolled environment .This equipment should be installed and operated
with minimum distance 20cm between the radiator& your body.

3

1.2. Feature



Highlight

 Vandal resistant body, with a flush button

 POE(IEEE802.3af, Power-over-Ethernet)

 Two-way audio communication over IP network with Echo cancel feature

 Complied with SIP Standard for easy integration in each SIP PBXes

 Physical&Power

 Body material: all-aluminum

 Button: 1 call button

 Output Relay: 2 output relays for door opener

 802.3af Power-Over-Ethernet

 12 DC connector(if not using POE)

 RF Card Reader:13.56MHz Supported (R23C only)

 Power consumption: less than 12w

 Water proof&Dust proof: IP65

 Installation: Wall-mounted

 Dimension: 190x110x35mm

 SIP Endpoint

 SIP v1(RFC2543), SIP v2(RFC3261)

 Audio codecs: G.711a, G.711μ, G.722, G.729

 Speech Quality: 7kHz Audio

 Echo Cancellation

 Voice Activation Detection

 Comfort Noise Generator

 Door opened via DTMF post-dial

4



Network Features



1x10/100Mbps Ethernet Port



Protocols support: IPv4, HTTP, HTTPS, FTP, SNMP, DNS, NTP, RTSP, RTP, TCP, UDP,

ICMP, DHCP, ARP

 Application Scenarios



Office door phone with on-site or hosted IP-PBX



Remote site entry over Internet



Villa intercom with door access control

1.3. Panel Description

R23C R23P

5

2.Configuration

2.1. Web login

2.1.1 Obtain the IP address

The Akuvox R23X uses Static IP by default, the default IP address is 192.168.1.100. If

the IP address is unknown, press and hold the call button for a short period of

time(about 5s) after LED light turns blue, the phone will announce its IP.

2.1.2 Login the web

Open a Web Browser, enter the corresponding IP address. Then, type the default

user name and password as below to log in:

User name: admin

Password: admin

6

2.2. Status

2.2.1 Basic

Sections

Description

Product Information

To display the device’s information such as Model name,
MAC address (IP device’s physical address), Firmware version
and Hardware version.

Network Information

To display the device’s Networking status(LAN Port),such as
Port Type(which could be DHCP/Static/PPPoE), Link Status, IP
Address, Subnet Mask, Gateway, Primary DNS server,
Secondary DNS server, Primary NTP server and Secondary
NTP server(NTP server is used to synchronize time from
INTERNET automatically).

Account Information

To display device’s Account information and Registration
status (account’s username, registered server’s address,
Register result).

Status can be viewed from “Status -> Basic”,including the information of product,

network and account.

7

2.3. Intercom

2.3.1 Basic

Sections

Description

Basic

 Select Account: R23X supports 2 accounts. You can

choose one account or Auto mode for the following
Intercom basic settings.

 No Answer Call ： R23X will call to the No answer call

number in order when the ringtone is time out without
answer of the push button number. Disable by default.

Push Button

 Push Button: To configure the destination number or IP

you want to contact with.

 No Answer Call 1&2: To setup two no answer call

numbers or one no answer call number.

Web Call

To dial out or answer the phone from website.

Go to the path: Intercom-Basic

8

Max Call Time

To configure the max call time

Max Dial Time

 Dial in Time: When other phone calls to R23X, if ring

tone is over the Dial in Time without answer. The call will
be hang up.

 Dial out Time: When R23X calls to the other party, if the

ringtone is over the Dial out Time without answer. R23X
will continue calls to no answer call number in order.

Push to Hang up

To enable or disable the Push to Hang up function

2.3.2 LED Setting

Sections

Description

State

Including five states:Normal,Offline,Calling,Talking and Receiving.

Color Off

The default status is OFF.

Color On

It can support three color: Red, Green, Blue.

Blink Mode

To setup the different blink frequency.

To setup the LED lighting mode.

9

2.3.3 Relay&Input

Sections

Description

Relay

To configure some settings about unlock

 Relay Select: R23X supports 2 relays.

 Relay Type: Different locks use different relay types,

positive or negative. If you connect the Lock in NO
connector, select positive type. Otherwise using negative
type.

 Relay Delay(sec): Allows the door to remain “open” for

certain period . The range is from 1 to 5 seconds.

 DTMF Option: R23X support 1digit or 4 digits DTMF

unlock code. Please select one type and enter the
corresponding code.

 DTMF: Setup 1 digit DTMF code for remote unlock

 4 Digits DTMF : Setup 4 digits DTMF code for remote

unlock.

 Status: Different relay types will show different status.

10

Web relay

R23X can support extra web relay. This function is more
safety to use DTMF code to remote unlock.

 Type: Connect web relay and choose the type.

 IP Address: Enter web relay IP address.

 User name: it is an authentication for connecting web

relay

 password: it is an authentication for connecting web

relay

Note: Users can modify username and password in web relay
website.

Input

There is a sensor that used for anti-vandal in R23X. When
R23X is broken by violent means, the sensor will be triggered,
then the management center will receive the alarm.

 Service: Enable by default

 Call Number: To setup management center number for

alarm.

 Display Name: Which is sent to the other call party for

displaying.

 Call Timer: The interval of calling. For instant , the Call

timer is 5sec, if you hang up the calling in the third
second, the calling will auto call out after 2sec.

 Light Status: The status will change according to the

sensor. Once the sensor is triggered , the status will
show Warning. Normal by default.

11

2.3.4 AEC Setting

Sections

Description

AEC Level

AEC(Configurable Acoustic and Line Echo Cancelers) is used
to adjust the echo effect during the communication. The
default value is 700. Increase the level, the echo control is
better.

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2.3.5 Multicast

Sections

Description

Multicast Audio Receiving

To display and configure the Multicast setting.

 Multicast Receiver Enable: Enable receiver multicast

function.

 Receiver address : Setup the multicast address.

 Receiver port : setup the multicast address port.

Multicast Audio Sending

To setup the multicast parameters.

 Multicast Sending Enable: Enable sender multicast

function

 Send to Address: setup the multicast address.

 Send to port: setup the multicast address port.

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2.3.6 Card Setting(R23C only)

Sections

Description

Import/Export Card Data

To import or export the card data file. Only support .xml
format.

Card Status

 Normal: choose Normal mode when reading card.

 Card Issuing: Choose Card Issuing mode when writing

card

Card Setting

 IC Key DoorNum: R23X can support to connect 2 doors.

Choose one and add the valid card for unlock.

 IC Key Name: To setup corresponding name for the card.

 IC Key Code: Place the card in the Card-reading area,

click “obtain” to read the card code, click “Add” and the
card information will show in the Door Card
Management list.

Door Card Management

Valid card information will show in the list. Users can tick the
current card information then delete one or all in the list.

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2.4. Account

2.4.1 Basic

Sections

Description

SIP Account

To display and configure the specific Account settings.

 Status: To display register result.

 Display Name: Which is sent to the other call party for

display.

 Register Name: Allocated by SIP server provider, used for

authentication.

 User Name: Allocated by your SIP server provide, used

for authentication.

 Password: Used for authorization.

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SIP Server 1

To display and configure Primary SIP server settings.

 Server IP: SIP server address, it could be an URL or IP

address.

 Registration Period: The registration will expire after

Registration period, the IP phone will re-register
automatically within registration period.

SIP Server 2

To display and configure Secondary SIP server settings.
Used for redundancy, if registering to Primary SIP server fails,
the IP phone will go to Secondary SIP server for registering.
Note: Secondary SIP server is used for redundancy, it can be
left blank if there is not redundancy SIP server in user’s
environment.

Outbound Proxy Server

To display and configure Outbound Proxy server settings.
An outbound proxy server is used to receive all initiating
request messages and route them to the designated SIP
server.
Note: All SIP request messages from the IP phone will be sent
to the outbound proxy server forcefully when configured.

Transport Type

To display and configure Transport type for SIP message

 UDP: UDP is an unreliable but very efficient transport

layer protocol.

 TCP: Reliable but less-efficient transport layer protocol.

 TLS: Secured and Reliable transport layer protocol.

 DNS-SRV: A DNS RR for specifying the location of

services.

NAT

To display and configure NAT(Net Address Translator)
settings.

 STUN: Simple Traversal of UDP over NATS is a solution to

solve NAT issues.

Note: By default, NAT is disabled.

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2.4.2 Account-Advanced

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Sections

Description

SIP Account

Select an account to display the settings.

Codecs

To display and configure available/unavailable codecs list.
Codec means coder-decoder which is used to transfer analog
signal to digital signal or vice versa.
Familiar codecs are PCMU(G711U), PCMA(G711A), G722
(wide-bandth codecs), G729 and so on.

Subscribe

To display and configure MWI, BLF, ACD subscription settings.

 MWI: Message Waiting Indicator which is used to

indicate whether there is unread new voice message.

 BLF: BLF is short for Busy Lamp Field which is used to

monitor the designated extension status.

 ACD: Automatic Call Distribution is often used in offices

for customer service, such as call center. The setting
here is to negotiate with the server about expire time of
ACD subscription.

DTMF

To display and configure DTMF settings.

 Type:Support Inband,Info,RFC2833 or their combination.

 How To Notify DTMF: Only available when Type is Info.

 DTMF Payload: To configure payload type for DTMF.

Note: Type RFC2833 is set by default as a standard. Type
Inband uses inband frequency to indicate DTMF tone which is
most used to be compatible to traditional telephone server.
Type Info use SIP Info message to indicate DTMF message.

Call

To display and configure call-related features.

 Max Local SIP Port: To configure maximum local sip port

for designated account.

 Min Local SIP Port: To configure minimum local sip port

for designated account.

 Caller ID Header: To configure which Caller ID format to

fetch for displaying on Phone UI.

 Auto Answer: IP phone will answered the incoming call

for designated account automatically when enabled.

 Ringtones: Choose the ringtone for each account.

 Provisioning Response ACK: 100% reliability for all

provisional messages, this means it will send ACK every
time when the IP phone receives a provisional SIP
message from SIP server.

 User=phone: If enabled, IP phone will send user=phone

within SIP message.

 PTime: Interval time between two consecutive RTP

packets.

 Anonymous Call: If enabled, all outgoing call to the

designated account will be anonymous number.

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 Anonymous Call Rejection: If enabled, all incoming

anonymous-out call of the designated account will be
rejected.

 Missed Call Log: To display the miss call log.

 Prevent SIP Hacking: Enable to prevent SIP from hacking.

Session Timer

To display or configure session timer settings.

 Active: To enable or disable this feature. If enable, the

on going call will be disconnected automatically once
the session expired unless it’s been refreshed by UAC or
UAS.

 Session Expire: Configure session expire time.

 Session Refresher: To configure who should be response

for refreshing a session.

Note: UAC means User Agent Client, here stands for IP
phone. UAS means User Agent Server, here stands for SIP
server.

BLF List

To display or configure BLF List URI address.

 BLF List URI: BLF List is short for Busy Lamp Field List.

 BLF List Pick Up Code: To set the BLF pick up code.

 BLF List Barge In Code : To set the BLF barge in code.

Encryption

To enable or disabled SRTP feature.

 Voice Encryption(SRTP): If enabled, all audio signal

(technically speaking it’s RTP streams) will be encrypted
for more security.

NAT

To display NAT-related settings.

 UDP Keep Alive message: If enabled, IP phone will send

UDP keep-alive message periodically to router to keep
NAT port alive.

 UDP Alive Msg Interval: Keep alive message interval.

 Rport: Remote Port, if enabled, it will add Remote Port

into outgoing SIP message to designated account.

User Agent

One can customize User Agent field in the SIP message; If
user agent is set to specific value, user could see the
information from PCAP. If user agent is not set by default,
user could see the company name, model number and
firmware version from PCAP

19

2.5. Network

2.5.1 Basic

Sections

Description

LAN Port

To display and configure LAN Port settings.

 DHCP: If selected, IP phone will get IP address, Subnet

Mask, Default Gateway and DNS server address from
DHCP server automatically.

 Static IP: If selected, you have to set IP address, Subnet

Mask, Default Gateway and DNS server manually.

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2.5.2 Advance

Sections

Description

Local RTP

To display and configure Local RTP settings.

 Max RTP Port: Determine the maximum port that RTP

stream can use.

 Starting RTP Port: Determine the minimum port that RTP

stream can use.

SNMP

To display and configure SNMP settings.

 Active: To enable or disable SNMP feature.

 Port: To configure SNMP server’s port.

 Trusted IP: To configure allowed SNMP server address, it

could be an IP address or any valid URL domain name.

Note: SNMP (Simple Network Management Protocols) is an
Internet-standard protocol for managing devices on IP
networks.

21

VLAN

To display and configure VLAN settings.

 Active: To enable or disable VLAN feature for designated

port.

 VID: To configure VLAN ID for designated port.

 Priority: To select VLAN priority for designated port.

Note: Please consult your administrator for specific VLAN
settings in your networking environment.

TR069

To display and configure TR069 settings.

 Active: To enable or disable TR069 feature.

 Version: To select supported TR069 version (version 1.0

or 1.1).

 ACS/CPE: ACS is short for Auto configuration servers as

server side, CPE is short for Customer-premise
equipment as client side devices.

 URL: To configure URL address for ACS or CPE.

 User name: To configure username for ACS or CPE.

 Password: To configure Password for ACS or CPE.

 Periodic Inform: To enable periodically inform.

 Periodic Interval: To configure interval for periodic

inform.

Note: TR-069(Technical Report 069) is a technical
specification entitled CPE WAN Management Protocol
(CWMP).It defines an application layer protocol for remote
management of end-user devices.

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2.6. Phone

2.6.1 Time/Lang

Sections

Description

NTP

To configure NTP server related settings.

 Time Zone: To select local Time Zone for NTP server.

 Primary Server: To configure primary NTP server

address.

 Secondary Server: To configure secondary NTP server

address, it takes effect if primary NTP server is
unreachable.

 Update interval: To configure interval between two

consecutive NTP requests.

Note: NTP, Network Time Protocol is used to automatically
synchronized local time with INTERNET time, since NTP
server only response GMT time, you need to specify the
Time Zone for IP phone to decide the local time.

23

2.6.2 Call Feature

Sections

Description

Mode

 Mode: Select the desired mode.

DND

DND (Do Not Disturb) allows IP phones to ignore any
incoming calls.

 Return Code when DND: Determine what response code

should be sent back to server when there is an incoming
call if DND on.

 DND On Code: The Code used to turn on DND on

server’s side, if configured, IP phone will send a SIP
message to server to turn on DND on server side if you
press DND when DND is off.

 DND Off Code: The Code used to turn off DND on

server’s side, if configured, IP phone will send a SIP
message to server to turn off DND on server side if you
press DND when DND is on.

Intercom

Intercom allows user to establish a call directly with the
callee.

 Active: To enable or disable Intercom feature.

 Intercom Mute: If enabled, once the call established, the

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2.6.3 Voice

callee will be muted.

Others

 Return Code When Refuse: Allows user to assign specific

code as return code to SIP server when an incoming call
is rejected.

 Auto Answer Delay: To configure delay time before an

incoming call is automatically answered.

 Auto Answer Mode: To set video or audio mode for auto

answer by default.

 Direct IP: Direct IP call without SIP proxy.

Sections

Description

Mic Volume

To configure Microphone volume , from 1-15,8 by default.

Speaker Volume

To configure Speaker Volume,from 1-15,8 by default.

25

2.6.4 Phone-Multicast

Sections

Description

Multicast Setting

To display and configure the Multicast setting.

 Paging Barge: Choose the multicast number ,range

from 1 to 10.

 Paging priority Active: Enable o disable the multicast.

Priority List

To setup the multicast parameters.

 Listening Address: Enter the IP address you need to

listen

 Label： Input the label for each listening address

26

2.6.5 Phone-Call Log

Sections

Description

Call History

To display call history records.
Available call history types are All calls, Dialed calls, Received
calls, Missed calls, Forwarded calls.
Users can check the call history in detail. Tick the number to
delete or delete all logs. R23X supports 100 call logs.

27

2.7. Upgrade

2.7.1 Basic

Sections

Description

Upgrade

To select upgrading zip file from local or a remote server
automatically.

Note: Please make sure it’s right file format for right model.

Firmware version

To display firmware version, firmware version starts with
MODEL name.

Hardware Version

To display Hardware version.

Reset to Factory Setting

To reset IP phone’s setting to factory settings.

Reboot

To reboot IP phone remotely from Web UI.

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2.7.2 Advance

Sections

Description

PNP Option

To display and configure PNP setting for Auto Provisioning.

 PNP: Plug and Play, once PNP is enabled, the phone will

send SIP subscription message to PNP server automatically
to get Auto Provisioning server’s address.

By default, this SIP message is sent to multicast address

224.0.1.75(PNP server address by standard).

DHCP Option

To display and configure custom DHCP option.

 DHCP option: If configured, IP Phone will use designated

DHCP option to get Auto Provisioning server’s address via
DHCP.

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This setting require DHCP server to support corresponding
option.

Manual Update Server

To display and configure manual update server’s settings.

 URL: Auto provisioning server address.

 User name: Configure if server needs an username to

access, otherwise left blank.

 Password: Configure if server needs a password to access,

otherwise left blank.

 Common AES Key: Used for IP phone to decipher common

Auto Provisioning configuration file.

 AES Key (MAC): Used for IP phone to decipher

MAC-oriented auto provisioning configuration file(for
example, file name could be 0c1105888888.cfg if IP
phone’s MAC address is 0c1105888888).

Note: AES is one of many encryption, it should be configure
only configure file is ciphered with AES, otherwise left blank.

AutoP

To display and configure Auto Provisioning mode settings.
This Auto Provisioning mode is actually self-explanatory.
For example, mode “Power on” means IP phone will go to do
Provisioning every time it powers on.

System Log

To display system log level and export system log file.

 System log level: From level 0~7.The higher level means

the more specific system log is saved to a temporary file.
By default, it’s level 3.

 Export Log: Click to export temporary system log file to

local PC.

30

2.8. Security

2.8.1 Basic

Sections

Description

Web Password Modify

To modify user’s password.

 Current Password: The current password you used.

 New Password: Input new password you intend to use.

 Confirm Password: Repeat the new password.

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