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R23X Door Phone User Manual
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
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.
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FCC Caution:
Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equ ipment.
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.
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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
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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
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R23P
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
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2.2. Status
2.2.1 Basic
Status can be viewed from “Status -> Basic”,including the information of product, network and account.
Sections
Product Information
Network Information
Account Information
Description
To display the device’s information such as Model name,
MAC address (IP device’s physical address), Firmware version and Hardware version.
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).
To display device’s Account information and Registration status (account’s username, registered server’s address,
Register result).
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2.3. Intercom
2.3.1 Basic
Go to the path: Intercom-Basic
Basic
Sections
Push Button
Web Call
Description
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: 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.
To dial out or answer the phone from website.
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Max Call Time
Max Dial Time
Push to Hang up
2.3.2 LED Setting
To setup the LED lighting mode.
To configure the max call 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.
To enable or disable the Push to Hang up function
Sections
State
Color Off
Color On
Blink Mode
Description
Including five states:Normal,Offline,Calling,Talking and Receiving.
The default status is OFF.
It can support three color: Red, Green, Blue.
To setup the different blink frequency.
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2.3.3 Relay&Input
Relay
Sections Description
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.
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Web relay
Input
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.
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.
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2.3.4 AEC Setting
AEC Level
Sections Description
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
Multicast Audio Receiving
Multicast Audio Sending
Description
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.
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
Import/Export Card Data
Card Status
Card Setting
Door Card Management
Description
To import or export the card data file. Only support .xml
format.
Normal: choose Normal mode when reading card.
Card Issuing: Choose Card Issuing mode when writing card
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.
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
SIP Account
Description
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
SIP Server 2
Outbound Proxy Server
Transport Type
NAT
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.
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.
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.
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.
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
SIP Account
Codecs
Subscribe
DTMF
Call
Description
Select an account to display the settings.
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.
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.
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.
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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Session Timer
BLF List
Encryption
NAT
User Agent
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.
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.
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.
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.
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.
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
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2.5. Network
2.5.1 Basic
LAN Port
Sections Description
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
Local RTP
Sections
SNMP
Description
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.
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.
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VLAN
TR069
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.
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
NTP
Sections Description
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.
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2.6.2 Call Feature
Mode
DND
Sections
Intercom
Description
Mode: Select the desired mode.
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 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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Others
2.6.3 Voice
callee will be muted.
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
Mic Volume
Speaker Volume
Description
To configure Microphone volume , from 1-15,8 by default.
To configure Speaker Volume,from 1-15,8 by default.
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2.6.4 Phone-Multicast
Sections
Multicast Setting
Priority List
Description
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.
To setup the multicast parameters.
Listening Address: Enter the IP address you need to listen
Label： Input the label for each listening address
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2.6.5 Phone-Call Log
Sections
Call History
Description
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.
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2.7. Upgrade
2.7.1 Basic
Upgrade
Sections
Firmware version
Hardware Version
Reset to Factory Setting
Reboot
Description
To select upgrading zip file from local or a remote server automatically.
Note: Please make sure it’s right file format for right model.
To display firmware version, firmware version starts with
MODEL name.
To display Hardware version.
To reset IP phone’s setting to factory settings.
To reboot IP phone remotely from Web UI.
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2.7.2 Advance
Sections
PNP Option
DHCP Option
Description
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).
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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Manual Update Server
AutoP
System Log
This setting require DHCP server to support corresponding option.
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.
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.
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.
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2.8. Security
2.8.1 Basic
Sections
Web Password Modify
Description
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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