Revision History ............................................................................................................................................ 4
1.2 Features .......................................................................................................................................... 4
1.3 Monitor Interface and Description ................................................................................................. 5
2.3 Language ......................................................................................................................................... 9
2.8.3 Version ........................................................................................................................... 12
2.9 More .............................................................................................................................................. 13
3.7 Live View ....................................................................................................................................... 22
Chapter 4 Web Configuration ..................................................................................................................... 23
4.1 Web Login ..................................................................................................................................... 23
4.2 Login using Web Browser .............................................................................................................. 23
4.3 Status ............................................................................................................................................ 24
AMCSM7 is a multifunctional communicator, with an Android operating system, provides audio and
video communication with Door phones via SIP 2.0 protocol. It delivers the ultimate touch screen
experience in an unobtrusive, space-saving design featuring a brilliant 7 inch capacitive touch screen
display and 5 soft-touch buttons.
1.2 Features
Highlights
Ultra-clear, stylish appearance
Thin profile and small footprint
7” capacitive touch screen with energy saving mode
Reception of a HD picture from the door communicator
Two-way communication with other units in the network, based on SIP v2.0 protocol
Powered by PoE or external source v Supports US or European electrical wall box mounting
Touch Screen Display
Display Type: Active matrix TFT, transmissive type
Size: 7 inch (176 mm) diagonal
Aspect Ratio: 16:9
Resolution: 1024 x 600
Brightness: 30 0nits (cd/m²)
Contrast: 800:1
Color Depth: 24-bit, 16M colors
Illumination: Edge lit LED
Viewing Angle: ± 80° Left, ±80° Right, ± 60° Upper, ±70° Lower
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Here are some safety recommendations about the installation and the usage:
Do not use this product near water, such as: bath, washbasin, kitchen sink and other
damp places and so on.
Place the device in a place away from heat.
Place the device away from traffic areas to prevent collisions.
Please use the equipment with the matching power adapter or any POE.
Chapter 2 Interface Configuration
2.1 Main interface settings
AMCSM7 support two layout interfaces, Phone and Launcher, by default it will be set to Phone which is
the layout below.
2.2 Display
Go to Settings -> Display, you can modify Brightness, Sleep time, Screen Saver or clear the screen saver.
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More settings allow you to modify network, account, live view, arm code, etc, default password is
“123456”
2.9.1 Network
DHCP/Static IP
Indoor monitor unit is set to DHCP by default, plugin the monitor into the network to get an IP, you can
go to Settings -> More -> Network, default password is “123456” you can setup the Static IP by changing
the LAN type first and entering the Lan IP, Subnet, gateway, etc.
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SIP account is provided by the SIP server, go to Settings -> More -> Account, all information must be
provided from the sip server, if using only for IP to IP calling then sip account can be blank.
2.9.3 Live View
Live view is used to monitor to display the live view of the door phone or outdoor intercom using RTSP
streaming, go to Settings -> More -> Live View, sample RTSP URL address is:
Rtsp://192.168.35.26/live/ch00_0
The highlighted IP is the intercom’s IP address.
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Assistance is where you can setup a quick dial to dial an emergency number by tapping the assistance
button on the main screen.
2.9.5 Intruder
Intruder settings are where the configuration for the Input/output for the alarm system, each zone can
be configure by name, type, trigger mode and status.
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In System code you can modify or edit the arm and disarm code for alarm system.
2.9.7 Unlock Code
Indoor monitor unit can open the outdoor phone by pressing the unlock button during a call
conversation, soft and hardware key is set to “Auto” by default.
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In Folder menu you can open files or picture and view it.
Chapter 4 Web Configuration
4.1 Web Login
Obtaining IP Address – the SM7 is set to DHCP by default, make sure the SIP Door Station is plugin to
the network, to get the IP address go to Settings -> System Info -> Network
4.2 Login using Web browser
Open web browser and enter the IP address of the door station into the address bar and type in the
username and password.
User name: admin
Password: admin
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SIP Account – Sip Account is provided by the SIP server.
SIP Server 1 – Sip Server URL provided by the SIP server
SIP Server 2 – Secondary SIP Server, indoor unit can be connected to two different SIP Servers.
Outbound Proxy Server – outbound proxy server is used to receive all initiating request messages
and route them to the designated SIP server.
Transport Type – Select transport type used by the sip server
o UDP - is an unreliable but very efficient transport layer protocol.
o TCP - Reliable but less-efficient transport layer protocol.
o TLS - Secured and Reliable transport layer protocol.
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SIP Account – Select which account to modify
Audio Codecs – Is a coder-decoder which is used to transfer analog signal to digital signal or vice
versa. Familiar codecs are PCMU (G711U), PCMA (G711A), G722 (wide-band codecs), G729 and so on.
Video Codecs - this setting allows you to modify the video quality.
o Codec Name – The default video codec is H264.
o Codec Resolution - It can support QCIF, CIF, VGA, 4CIF, 720P.
o Codec Bitrate - The lowest bitrate is 128, the highest bitrate is 2048.
o Codec payload - From 90-119.
Subscribe – Allows you to configure MWI, BLF, ACD subscription settings.
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oMWI - Message Waiting Indicator which is used to indicate whether there is unread new
voice message.
oBLF - BLF is short for Busy Lamp Field which is used to monitor the designated extension
status.
oACD - 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 – Select the DTMF settings.
o Type: Supports Inband, Info, RFC2833 or their combination.
o How To Notify DTMF: Only available when DTMF Type is Info.
o DTMF Payload: To configure payload type for DTMF.
Call – Settings to configure call related features.
o Max Local SIP Port: To configure maximum local sip port for designated account.
o Min Local SIP Port: To configure minimum local sip port for designated account.
o Caller ID Header: To configure which Caller ID format to fetch for displaying on Phone
UI.
oAuto Answer: If enabled, IP phone will be auto-answered when there is an incoming call
for designated account.
o Ringtones: Choose the ringtone for each account.
o Provisioning Response ACK: 100% reliability for all provisional messages, this means it
will send ACK every time the IP phone receives a provisional SIP message from SIP
server.
o User=phone: If enabled, IP phone will send user=phone within SIP message.
o PTime: Interval time between two consecutive RTP packets.
o Anonymous Call: If enabled, all outgoing call for the designated account will be
anonymous number.
oAnonymous Call Rejection: If enabled, all incoming anonymous-out call for the
designated account will be rejected.
o Is escape non Ascii character: To transfer the symbol to Ascii character.
o Missed Call Log: To display the miss call log.
o Prevent SIP Hacking: Enable to prevent SIP from hacking.
Session Timer – Configuration settings for session timer.
oActive: To enable or disable this feature, If enable, the ongoing call will be disconnected
automatically once the session expired unless it’s been refreshed by UAC or UAS.
o Session Expire: Configure session expire time.
o 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.
Encryption – Settings to enable or disable the SRTP feature, default is set to disable, if enabled, all
audio signal (technically speaking its RTP streams) will be encrypted for more security.
NAT – Settings for NAT-Related configurations.
oUDP Keep Alive message: If enabled, IP phone will send UDP keep-alive message
periodically to router to keep NAT port alive.
o UDP Alive Msg Interval: Keep alive message interval.
o Rport: Remote Port, if enabled, it will add Remote Port into outgoing SIP message for
designated account.
Conference – Settings to select local or network conference
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o Type: To select desired conference type
o Conference URI: If network conference is selected, a network conference URI is needed
to be input.
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
4.5 Network
Settings to configure all network related configuration, such as IP Address, subnet mask, default
gateway and others.
4.5.1 Basic – Configuration settings to setup static IP or DHCP.
4.5.2 Advanced – To cconfigure Local RTP settings.
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o Feature Key Sync: To enable or disable feature key sync.
o Mode: Select the desired mode.
Forward Transfer – This feature has three options; always Forward, Busy Forward and No Answer
Forward.
o Always Forward: Any incoming call will be forwarded in any situation.
o Busy Forward: Any incoming call will be forwarded if IP phone is busy.
o No answer Forward: Any incoming call will be forwarded if it’s no answer after a specific
time.
DND – DND or Do Not Disturb allows IP phones to ignore any incoming calls.
oDND Emergency: the phone from the Authorized number can still be received after
enable this function.
o DND Authorized Number: Setup authorized numbers for DND Emergency.
o Account: Select an account for DND
o DND: Disable by default.
o Return Code when DND: Determine what responses code should be sent back to server
when there is an incoming call if DND on.
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oDND 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.
oDND 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 indoor unit.
o Active: To enable or disable Intercom feature.
o Intercom Mute: If enabled, once the call established, the indoor unit will be muted.
Remote Control – Remote Control allows specific host to interact with IP phone by sending HTTP or
HTTPS requests. The specific action could be answering an incoming call, hang-up an ongoing call and
so on
o Allowed Access IP List: To configure the allowed host address.
Note: For now, IP phone can only support IP address, IP address list and IP address pattern as allowed
hosts
UACSTA – Using CSTA for SIP phone user agents. It can control some features of calling. UACSTA is
used to send ECMA-323(CSTA XML) information during SIP calling. The default status is disabled.
Open Lobby – User can choose which types you need for each key. Indoor unit supports 2 types to
unlock-DTMF and Relay. DTMF is used to unlock the lobby door remotely; Relay is used to open the
local door.
o Softkey: During the talking, user can press Unlock key to open the door.
o Hardware Key: User can also press hardware key to unlock the door.
o DTMF: If you choose DTMF code for one unlock key. Please setup the DTMF code.
Door Phone – When there is an incoming call from door phone, setup the delay auto answer time,
Indoor unit will auto answer the call after the timeout.
Note: if you enable Auto Answer function, this feature will be not available.
Others
oReturn Code When Refuse: Allows user to assign specific code as return code to SIP
server when an incoming call is rejected.
oAuto Answer Delay: To configure delay time before an incoming call is automatically
answered.
o Auto Answer Mode: To set video or audio mode for auto answer by default.
o Early DTMF: Enable or disable early DTMF function
o Direct IP: Direct IP call without SIP proxy.
4.6.3 Audio – Audio settings allows you to configure all audio related settings.
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Echo Canceller – Is to remove acoustic echo from a voice communication in order to improve the
voice quality.
oVAD (Voice Activity Detection): Allow IP phone to detect the presence or absence of
human speech during a call. When detecting period of “silence”, VAD replaces that
silence efficiently with special packets that indicate silence is occurring. It can facilitate
speech processing, and deactivate some processes during non-speech section of an
audio session. It can avoid unnecessary coding or transmission of silence packets in VoIP
applications, saving on computation and network bandwidth.
oCNG (Comfort Noise Generation): Allow IP phone to generate comfortable background
noise for voice communications during periods of silence in a conversation. It is a part of
the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with
VAD algorithms, quickly responds when periods of silence occur and inserts artificial
noise until voice activity resumes. The insertion of artificial noise gives the illusion of a
constant transmission stream, so that background sound is consistent throughout the
call and the listener does not think the line has released.
Automatic Generation Control – Indoor unit will auto adjust amplification circuit gain via signal.
o Automatic Gain Control(Sending-side): Disabled by default
o Automatic Gain Control(Receiving-side): Disabled by default
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oHardware Endec Acceleration: This function is used to solve the image issue. User can
enable the Hardware Endec Acceleration when you need.
o Hardware Decodec Acceleration: Disabled by default.
o Color Enhancement: To increase the phone display color. Enabled by default.
o Image Quality: User can select Low, Middle or High mode.
o Camera Priority: Indoor unit can connect extra camera.
o Video Call Status: Disabled by default.
4.7 Phone Book – Phone book has two options, local and call log.
4.7.1 Local Book – This setting allows you to input contact numbers of different intercoms.
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o All Contacts: To display or edit all local contacts.
o Black List: To display black list contacts.
Search – To search designated contacts from local phonebook.
Dial – To dial out a number remotely from the browser.
Contact Setting – To input a new contact.
Import/Export - To import or export the contact or blacklist file.
4.7.2 Call Log – to display the call history of the indoor unit.
4.8 Upgrade – upgrade settings allows you to upgrade firmware, factory default indoor unit or setup
the advanced setting for indoor unit.
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Firmware Version – Displays the current firmware version of the indoor unit.
Hardware Version – Displays the actual hardware version of the indoor unit.
Upgrade – To upgrade the firmware to a latest firmware if available
Reset to Factory Setting – To reset the indoor unit to factory default
Reset Config To Factory Setting - To reset the indoor unit’s configuration to default
Reboot – Remotely reboots the indoor unit from the browser
4.8.2 Advanced
Note: Only change settings if needed, otherwise leave all by default.
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PNP Option - To display and configure PNP setting for Auto Provisioning.
oPNP: 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.
oDHCP option: If configured, IP Phone will use designated DHCP option to get Auto
Provisioning server’s address via DHCP. This setting require DHCP server to support
corresponding option.
Manual Autop - To display and configure manual update server’s settings.
o URL: Auto provisioning server address.
o User name: Configure if server needs a username to access, otherwise left blank.
o Password: Configure if server needs a password to access, otherwise left blank.
o Common AES Key: Used for IP phone to decipher common Auto Provisioning
configuration file.
oAES 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). Enter the URL address, and then click the AutoP Immediately
label, the phone will according the URL to ask for configuration file to update.
Note: AES is one of many encryptions, it should be configured only if ciphered is filled with AES,
otherwise left blank.
Automatic 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 syslog level and export syslog file.
oSyslog level: From level 0~7.The higher level means the more specific syslog is saved to
a temporary file. By default, it’s level 3.
o Export Log: Click to export temporary syslog file to local PC.
o Remote System Log: To enable or disable Remote System Log.
o Remote System Server: To input the syslog server address.
PCAP – To start, stop packets capturing or to export captured Packet file.
o Start: To start capturing all the packets file sent or received from IP phone.
o Stop: To stop capturing packets.
o Export: To export the capture packet file, use capture tool to open the file.
Note: IP phone will save captured packets file to a temporary file, this file maximum size is 1M
(megabytes), and will top capturing once reaching this maximum size.
Others – To display or configure others features from this page.
o Config file: To export or import configure file for indoor unit.
4.9 Security – To modify web password and other security related settings.
4.9.1 Basic – To configure the web login password.
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Web Password Modify – to modify user’s login password.
Session Time Out – if the user is been idle for the set time user would need to login again after the
time set has expired.
4.9.2 Advanced
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Web Server Certificate – To display or delete Certificate which is used when IP phone is connected
from any incoming HTTPs request.
Note: The default certificate could not be deleted.
Web Server Certificate Upload – To upload a certificate file which will be used as server certificate.
Client Certificate – To display or delete Certificates which is used when IP phone is connecting to
any HTTPs server.
Client Certificate Upload - To upload certificate files, this is used as client certificate.
oOnly Accept trusted Certificates: If this option is enabled, only trusted certificates will
be accepted.
4.10 Cloud Server – Not yet available.
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