Escene US102 IP Phone Review
Hi, Habr!
Digital Angel is the exclusive distributor of Escene equipment in Russia, and we decided to start our blog with a detailed review of the most budget phone of this brand.
According to its functional characteristics, the series belongs to the class of corporate IP-phones with the presence of advanced DVO. PoE support and a built-in bridge, which can be switched to routing mode, eliminates the need to lay another cable to the workplace of the employee.
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The buyer receives all the benefits of IP-telephony, without spending extra money. The cost of modification without PoE is about 2000 rubles. And about 2300 rubles is a PoE-enabled phone, which is cheaper than counterparts from other manufacturers.
VoIP
Data transfer
Physical parameters
Additional services (additional features)
Control
Nutrition
The phone comes in a simple white cardboard box, on the side there is a sticker with the model number and barcode of the device.
The box contains standard equipment, which includes:
The 5 volt Escene AD200 power supply is bundled only for US102-YN (without PoE), for US102-PYN it must be ordered separately.
Front panel of the phone
Navigation through the on-screen menu is performed using five navigation buttons and two buttons to the left and right of the navigation block. Using the navigation buttons, you can see missed, outgoing and incoming calls with one touch, as well as reduce or increase the volume.
Buttons signed by three dots are used to navigate the menu, their names change as you navigate the phone menu. The Line 1 and Line 2 buttons are used to select a line (SIP account) for an outgoing call.
The block of buttons for additional functions contains:
The phone has a monochrome 128x64 pixel LCD screen. The screen backlight is not very bright. But this is enough to easily read the messages on the screen.
For each line, in addition to its number, you can write an arbitrary name. In this case, “Line 1” is an arbitrary name.
Conversation state
Incoming call
For an incoming or outgoing call, the corresponding line indicator lights up.
The phone has two connectors for connecting to the LAN and power over PoE (US102-PYN), a second connector on the right for connecting to a computer and a jack for connecting a 5 volt power adapter (AD200).
The phone can be configured either using the phone menu, or using the web interface. Unlike most phones from other manufacturers, which leave a minimum of settings in the phone menu and a larger number only through the web interface, the Escene developers decided to make accessible from the phone menu in addition to the standard settings, also settings related to SIP accounts.
Such a move is fully justified, in some cases, you can configure the phone faster. In addition, sometimes there may be problems with access to the phone through a web-interface or it may be necessary to explain to an employee remotely how to reconfigure his phone. It will be easier for an unprepared person to use the phone menu, rather than a web interface.
Initial setup using phone buttons
So, we turned on the phone, connected the LAN port to the local network that has access to the IP PBX. Employee's computer, through the cable connected to the PC port.
Now we need to include the Russian language in the menu:
Press the “Menu” button, the button on the left corresponding to it, marked with three dots on top, the Function Settings menu will open. Using the up or down navigation buttons, find the System Settings menu, press Enter . Press the number 1 (or the Enter button), which corresponds to the “Phone Settings” , then select the Language menu, press the number 1 (or the Enter button), use the up or down navigation buttons, select the Russian and press OK " . Then click the back button until you exit the menu.
Now you need to configure the network settings:
Click "Menu" , then select the "Settings" menu, the number 2 - "Advanced Settings" , the default password is empty, just click "OK" . If you need to configure a VLAN (menu item 2), go to the appropriate menu and configure its ID and priority, select “Network” , then “LAN port” , by default, after the phone is loaded, a DHCP client is activated that tries to obtain an IP address, therefore in the network where the IP phone is located must be a DHCP server. If all settings are correct, the phone will receive an IP address and will be ready for further configuration.
If you need to use a static IP address, click the number 1 - “Type” and select “Static” and click “OK” . By default, the phone is configured with IP 192.168.0.200, to change the settings of the IP address, mask, gateway and DNS, use the menu buttons and navigation keys, after saving the settings, the phone will reboot. I note that in this menu “LAN port” you can configure the port for access to the web interface, by default it is 80, as well as the port for accessing the phone via telnet.
Special attention is given to the setting of the PC port ( Menu -> Network -> Advanced Settings -> PC Port ). Here you can configure the network operation mode between the PC and LAN ports. In bridge mode, this is a two-port switch with support for a separate tagged or untagged VLAN for a LAN or PC port. If you set the Router mode, the IP port and mask are assigned to the PC port, the NAT address translation is enabled between the LAN and the PC, you can also enable the DHCP server. Thus, the phone becomes also a router with NAT support.
Now you need to check the correctness of the network settings and see the IP address that was assigned by DHCP, to do this, click "Menu" , select "Status" , then figure 1 - "Network" , in my case the IP address assigned by DHCP: 192.168.1.253
Setting up additional phone features
All these settings are made in the "Menu" -> "Functions"
Support for additional services (DVO) and programmable buttons
The phone supports two independent SIP accounts, that is, registration on two different IP PBXs. When registering both lines at the same time, the first line will be used by default, to switch to the second line to make an outgoing call or return to the first line, use the “Line 1” and “Line 2” buttons.
As for the DVO, they all work correctly:
The call log contains records of recent outgoing, incoming, and missed calls.
To access the web interface from a computer with access to the network where the phone is located, enter the IP address of the phone in the address bar of the web browser, in my case it is 192.168.1.253
192.168.1.253
Default login and password:
root
root
There are two access levels on the phone: the administrator level, which can change any settings and the user, who can perform a limited number of settings.
We get to the main menu of the web configurator of the phone. For convenience of setup, immediately select the Russian language in the lower left menu:
The menu is divided into several groups:
Consider the most important menu items phone. The “Setup Wizard” menu is used to quickly set up the phone, allows you to configure two tabs in sequence: the “Network” menu -> “LAN port” and the basic SIP account settings in the “SIP accounts” -> “Account1” menu . These tabs will be discussed in more detail below.
Menu "Network" -> "LAN port"
You can set one of three connection methods: DHCP, static IP, or PPPoE. An important setting is the HTTP and Telnet ports, they should be made non-standard if the phone is on an untrusted network.
Menu "Network" -> "PC port"
L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode. The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.
Menu “Network” -> “VLAN Settings”
In a corporate network, it is recommended to isolate the computer network traffic from the voice network traffic, this is most often implemented using two VLANs. The phone supports VLAN on both ports.
Menu "Network" -> "VPN Settings"
If you need to connect your phone via a secure VPN channel, this can be done directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and SSL VPN.
The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.
Menu "SIP accounts" -> "Account 1"
In addition to the standard SIP account settings - User Name (UserID), Name (AuthID) and password, there is a “Label” field, it allows you to insert an arbitrary line description that will be displayed on the phone screen.
In addition to the main IP address of the SIP server, you can add an additional IP SIP server, in case of unsuccessful registration during the timeout, which is 32 seconds by default, its registration address will be used for the additional SIP server. The phone supports encryption of RTP and SIP signaling traffic over the TLS protocol.
Sound menu
By default, when calling, the phone claims all possible codecs. If necessary, unused codecs can be disabled.
In the menu, you can adjust various volume parameters: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
Menu "Advanced Settings" -> "Global SIP Settings"
If you set up the SIP settings here, they will be applied to both lines automatically, except for the settings “Local SIP port” and “RTP port range” , which can be useful for properly configuring the firewall.
Menu "Advanced Settings" -> "Phone Settings"
In this menu, additional functions are configured. Such as the “Hotline” - when you pick up the handset, the preset number is automatically dialed, you can turn on auto search in the address book during dialing and auto answer the call.
In case the call transfer is required to be performed with a special combination of buttons (starcode), instead of the standard SIP message 302, this can be specified in the setting “Special code for call transfer” . A useful setting that allows you not to break the connection in the conference, if it left the initiator. You can set call forwarding by condition (busy and non-response) and unconditional.
Menu "Phonebook"
The phone has a built-in phone book, which is quite advanced. It allows you to store up to 300 contact entries, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web interface. To download or save the phone book in XML format, use the “Phone maintenance” menu -> “Update by HTTP” -> “XML Phone book” ; here you can save or load the phone book in xml format.
If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts. 2 and 3 protocol versions are supported, using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can also search for the contact name for an incoming and outgoing call, if the contact is in the LDAP directory, the number will be his name is automatically added.
The phone also supports blacklists or ban lists: an unwanted phone number is added to such a list and can no longer be reached.
Menu "Programmable buttons"
The phone has a panel with 9 buttons that can be programmed for speed dialing, prefix before dialing or sending DTMF tones. For example, if you want to log in to the contact center of the bank. This functionality is extremely convenient. Moreover, the speed dial buttons can be tied to a specific SIP-account.
Debug menu
To debug the phone, you can turn on logging by specifying the necessary logs (Menu “Phone maintenance” -> Log). You can view them in two ways:
Menu "Phone maintenance" -> "Auto Provision" (Automatic update)
Using this menu, you can configure automatic downloading of configuration, firmware and address book to your phone. You can download one of several protocols: http / https / ftp / tftp.
Backup and Software Update
You can copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a particular protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update.
The status of the phone and system software can be viewed in the menu items "Status" and "System Information"
Setting up connection to IP PBX Asterisk using web-interface
Suppose we need to configure two extensions (two SIP accounts). For example, the first entry on the IP PBX Asterisk, the second on the virtual IP PBX:
IP address of the server with Asterisk = 10.10.10.1
UserID = 10
password = kRcB7zT3
SIP server (Asterisk) = 10.10.10.1
In the Asterisk sip.conf configuration, this will be equivalent to:
[ten]
deny = 0.0.0.0 / 0.0.0.0
secret = 7zT3kRcB
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = g722
dial = SIP / 10
mailbox = 10 @ device
permit = 0.0.0.0 / 0.0.0.0
callerid = device <10>
callcounter = yes
faxdetect = no
Equivalently, when configured in the free-pbx web interface, using the first line as an example:
Go to the line settings for the phone:
To work with Asterisk, it is enough to configure the Username (Username) = 10, password (Password) = 7zT3kRcB and SIP Server (SIP Server) = 10.10.10.10. You can add a label (Label) that will be displayed on the screen of the phone, in this case “Line 1”.
You can reduce the re-registration time from standard 3600 seconds to 600 seconds, this is especially true if the IP PBX is located outside the office, for example, Virtual PBX. If the phone is on a local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click "Submit"
Exactly the same must be done with the second line, for example, city number 78126470011 on the SIP server West Call. Let's register it on a virtual PBX with a non-standard SIP port 9966:
userid = 78126470011
authid = 78126470011
password = Bxu5qnXsXMAgG3l
sip proxy = uc.westcall.net
port = 9955
To specify a non-standard SIP port (other than 5060), you must explicitly specify it in the SIP server line: uc.westcall.net:9955 Next, click the "Apply" button.
In case of successful registration, the corresponding display will appear on the phone screen, so information on the status of line registration is available on the Status menu page:
Account 1: Registered
Account 2: Registered
In order to use the DVO buttons (transfer, hold, conference) no additional configuration is required.
Key features of the phone include:
If you are interested in checking Escene US102 in your network, we are ready to provide you with a testing machine. Contact the contact details listed on the official website of Escene .
Digital Angel is the exclusive distributor of Escene equipment in Russia, and we decided to start our blog with a detailed review of the most budget phone of this brand.
The basic models of Escene IP phones are equipped with a full range of functions that are required for voice communications. Universal phones of the US series are ideal for small businesses and home offices. They are affordable, reliable and easy to use.
According to its functional characteristics, the series belongs to the class of corporate IP-phones with the presence of advanced DVO. PoE support and a built-in bridge, which can be switched to routing mode, eliminates the need to lay another cable to the workplace of the employee.
')
Today, the model is represented by two modifications that differ in the power supply capability of PoE: US102-YN (without PoE) and US102-PYN (with PoE).
Special features
- Simplicity of setup due to the clear interface.
- Russified web interface and OSD menu.
- The ability to fully customize the phone using the screen and buttons, including SIP accounts.
- Ability to adapt the phone to work with SIP-compatible equipment.
- The functionality is more than most of the IP PBX and telecom operators currently support.
The buyer receives all the benefits of IP-telephony, without spending extra money. The cost of modification without PoE is about 2000 rubles. And about 2300 rubles is a PoE-enabled phone, which is cheaper than counterparts from other manufacturers.
Functionality
- High definition audio support Voice HD (G.722 codec).
- Direct SIP connection to Virtual IP PBXs (for example, Broadworks, MFI RTU) and to office IP PBXs (for example, Asterisk, 3CX IP PBX).
- Two Ethernet ports (PC / LAN) with VLAN support and the ability to work in the switching or routing mode.
- Easy installation and operation, advanced configuration options (including SIP and DVO functions) via the on-screen menu or via the web interface.
- Full duplex speakerphone, caller ID, call hold, call transfer and call forwarding, as well as other additional functions.
- Encryption support using VPN tunnel or signal encryption SIPS and SRTP media traffic.
- Support for a corporate notebook using the LDAP or XML protocol or a personal notebook.
- Russified on-screen menu and web phone interface.
Specifications
VoIP
- RFC 3261 standard SIP server, Asterisk, Broadsoft, RTU MFI, 3CX IP PBX, Panasonic SIP PBX and others.
- SIPS and SRTP encryption.
- Audio codecs: G.711 u / a, G.722 (HD Voice), G.729a, G.723.
- QoS: TOS, Jiffer Buffer, VAD, CNG, G.168 (32ms).
- DNS SRV support.
- Two SIP accounts with the possibility of registration on two independent SIP servers and the possibility of automatic switching in case of loss of registration.
Data transfer
- 2 * RJ45 10 / 100M Ethernet interfaces (LAN / PC) with PoE support (for model US102-PYN).
- VLAN / QoS support.
- IP addressing: DHCP client or static IP destination.
- Built-in VPN client L2TP or SSL VPN.
- Network protocols HTTP, BOOTP, FTP, TFTP, IEEE 802.1Q, IEEE 802.1X.
Physical parameters
- Monochrome LCD screen with backlit 128 * 64 characters.
- 5 navigation buttons.
- Sound adjustment buttons (9 levels).
- Buttons of additional services: turn off the microphone, headset, phone, messages, menu, hold, repeat, conference and transfer.
- 9 programmable buttons + Hold call button.
- Speakerphone.
Additional services (additional features)
- Waiting for a second call, a queue (if it supports an IP PBX), call transfer, call forwarding, call hold, call pickup, callback, call recall, auto answer.
- Speed ​​dialing, button to start recording a conversation using the old code (if it supports IP PBX).
- Multilateral conference (if it supports IP PBX), 3-way conference on the phone.
- Do not disturb (DND).
- Voice mail (if the function is supported by IP PBX).
- Personal note book, corporate note book (LDAP).
Control
- Protocols update: HTTP / TFTP / (PnP auto-tuning) PnP auto-provision.
- Configuration: via the phone's on-screen menu / web-interface / auto-provision.
- Debugging: telnet / phone screen / web-interface.
Nutrition
- Adapter model AD200 (AV 220/110 Volt, output DC 5 Volt / 1A).
- Power Over Ethernet (IEEE 802.af) LAN port - for modification of US102-PYN.
Package, appearance and packaging
The phone comes in a simple white cardboard box, on the side there is a sticker with the model number and barcode of the device.
The box contains standard equipment, which includes:
- Telephone set.
- Handset.
- Handset cord.
- Stand (removable, if you need to hang the phone on the wall).
- Instruction and warranty card.
The 5 volt Escene AD200 power supply is bundled only for US102-YN (without PoE), for US102-PYN it must be ordered separately.
Front panel of the phone
Navigation through the on-screen menu is performed using five navigation buttons and two buttons to the left and right of the navigation block. Using the navigation buttons, you can see missed, outgoing and incoming calls with one touch, as well as reduce or increase the volume.
Buttons signed by three dots are used to navigate the menu, their names change as you navigate the phone menu. The Line 1 and Line 2 buttons are used to select a line (SIP account) for an outgoing call.
The block of buttons for additional functions contains:
- onf - to create a 3-party conference (initiator, and two participants), to create a conference with a large number of participants, support of such a function on an IP PBX is required.
- Trans - call transfer.
- Redial - redial the last number.
- HF - enable or disable the speakerphone (speakerphone).
- A block of 10 buttons , one of them Hold button - hold call.
- 9 programmable buttons , each of which can be assigned one of the following functions:
- One touch dialing.
- One-touch prefix set.
- Send a combination of DTMF tones.
The presence of these buttons is extremely convenient, because each employee has a list of "favorite" numbers. To quickly recruit them you do not need to separately buy an additional extension console. There is also a paper insert for the buttons.
The phone has a monochrome 128x64 pixel LCD screen. The screen backlight is not very bright. But this is enough to easily read the messages on the screen.
For each line, in addition to its number, you can write an arbitrary name. In this case, “Line 1” is an arbitrary name.
Conversation state
Incoming call
For an incoming or outgoing call, the corresponding line indicator lights up.
The phone has a removable stand. If the phone needs to be hung on the wall, it is easy to remove. On the case there are mounting holes for mounting the phone to the wall.
The phone has two connectors for connecting to the LAN and power over PoE (US102-PYN), a second connector on the right for connecting to a computer and a jack for connecting a 5 volt power adapter (AD200).
Phone setup
The phone can be configured either using the phone menu, or using the web interface. Unlike most phones from other manufacturers, which leave a minimum of settings in the phone menu and a larger number only through the web interface, the Escene developers decided to make accessible from the phone menu in addition to the standard settings, also settings related to SIP accounts.
Such a move is fully justified, in some cases, you can configure the phone faster. In addition, sometimes there may be problems with access to the phone through a web-interface or it may be necessary to explain to an employee remotely how to reconfigure his phone. It will be easier for an unprepared person to use the phone menu, rather than a web interface.
Initial setup using phone buttons
So, we turned on the phone, connected the LAN port to the local network that has access to the IP PBX. Employee's computer, through the cable connected to the PC port.
Now we need to include the Russian language in the menu:
Press the “Menu” button, the button on the left corresponding to it, marked with three dots on top, the Function Settings menu will open. Using the up or down navigation buttons, find the System Settings menu, press Enter . Press the number 1 (or the Enter button), which corresponds to the “Phone Settings” , then select the Language menu, press the number 1 (or the Enter button), use the up or down navigation buttons, select the Russian and press OK " . Then click the back button until you exit the menu.
Now you need to configure the network settings:
Click "Menu" , then select the "Settings" menu, the number 2 - "Advanced Settings" , the default password is empty, just click "OK" . If you need to configure a VLAN (menu item 2), go to the appropriate menu and configure its ID and priority, select “Network” , then “LAN port” , by default, after the phone is loaded, a DHCP client is activated that tries to obtain an IP address, therefore in the network where the IP phone is located must be a DHCP server. If all settings are correct, the phone will receive an IP address and will be ready for further configuration.
If you need to use a static IP address, click the number 1 - “Type” and select “Static” and click “OK” . By default, the phone is configured with IP 192.168.0.200, to change the settings of the IP address, mask, gateway and DNS, use the menu buttons and navigation keys, after saving the settings, the phone will reboot. I note that in this menu “LAN port” you can configure the port for access to the web interface, by default it is 80, as well as the port for accessing the phone via telnet.
Special attention is given to the setting of the PC port ( Menu -> Network -> Advanced Settings -> PC Port ). Here you can configure the network operation mode between the PC and LAN ports. In bridge mode, this is a two-port switch with support for a separate tagged or untagged VLAN for a LAN or PC port. If you set the Router mode, the IP port and mask are assigned to the PC port, the NAT address translation is enabled between the LAN and the PC, you can also enable the DHCP server. Thus, the phone becomes also a router with NAT support.
Now you need to check the correctness of the network settings and see the IP address that was assigned by DHCP, to do this, click "Menu" , select "Status" , then figure 1 - "Network" , in my case the IP address assigned by DHCP: 192.168.1.253
Setting up additional phone features
All these settings are made in the "Menu" -> "Functions"
- Auto Answer allows you to set up an automatic answer to a call without lifting the handset.
- DND allows you to reject all calls if the subscriber is busy.
- VM number — set the number to access your voice mail.
- The hotline allows you to set automatic dialing of a given number immediately or with a set timeout.
- Forwarding allows you to set conditional and unconditional forwarding to the specified numbers.
- Button - program the speed dial keys.
Support for additional services (DVO) and programmable buttons
The phone supports two independent SIP accounts, that is, registration on two different IP PBXs. When registering both lines at the same time, the first line will be used by default, to switch to the second line to make an outgoing call or return to the first line, use the “Line 1” and “Line 2” buttons.
As for the DVO, they all work correctly:
- The Conf button allows you to transfer a call; call transfer is implemented using the SIP 302 Moved Temporarily message. This message today almost all IP PBX on the market.
- The Trans button - call transfer with consultation and blindly, also uses SIP 302 Moved Temporarily.
- Pressing the Redial button redials the last number dialed.
- HF button - allows you to turn on or off the speakerphone, answer the call with the speakerphone on, or end the call if the conversation took place over the speakerphone.
The call log contains records of recent outgoing, incoming, and missed calls.
Web Interface Overview
To access the web interface from a computer with access to the network where the phone is located, enter the IP address of the phone in the address bar of the web browser, in my case it is 192.168.1.253
192.168.1.253
Default login and password:
root
root
There are two access levels on the phone: the administrator level, which can change any settings and the user, who can perform a limited number of settings.
We get to the main menu of the web configurator of the phone. For convenience of setup, immediately select the Russian language in the lower left menu:
The menu is divided into several groups:
- Network settings (interfaces, VLAN, VPN, etc.)
- VoIP settings (SIP records and additional functions for signaling and media traffic)
- Settings for additional phone functions (phonebook settings, programmable buttons, dial plan, sound, etc.)
- Service settings (logging, reset and reboot, configuration management and software updates, etc.)
Consider the most important menu items phone. The “Setup Wizard” menu is used to quickly set up the phone, allows you to configure two tabs in sequence: the “Network” menu -> “LAN port” and the basic SIP account settings in the “SIP accounts” -> “Account1” menu . These tabs will be discussed in more detail below.
Network settings
Menu "Network" -> "LAN port"
You can set one of three connection methods: DHCP, static IP, or PPPoE. An important setting is the HTTP and Telnet ports, they should be made non-standard if the phone is on an untrusted network.
Menu "Network" -> "PC port"
L2 switching is switched on by default between the LAN and PC ports of the telephone - the “Bridge” mode. The phone can switch to L3 routing mode — a NAT address translation will start on the LAN port, an IP address will need to be configured on the PC port, and if necessary, a DHCP server must be enabled in which to set the pool of IP addresses for clients.
Menu “Network” -> “VLAN Settings”
In a corporate network, it is recommended to isolate the computer network traffic from the voice network traffic, this is most often implemented using two VLANs. The phone supports VLAN on both ports.
Menu "Network" -> "VPN Settings"
If you need to connect your phone via a secure VPN channel, this can be done directly from the phone, without buying additional equipment (for example, a VPN router), the phone supports L2TP and SSL VPN.
VoIP Settings
The phone allows you to manage a large number of SIP signaling settings and settings for RTP media traffic.
Menu "SIP accounts" -> "Account 1"
In addition to the standard SIP account settings - User Name (UserID), Name (AuthID) and password, there is a “Label” field, it allows you to insert an arbitrary line description that will be displayed on the phone screen.
In addition to the main IP address of the SIP server, you can add an additional IP SIP server, in case of unsuccessful registration during the timeout, which is 32 seconds by default, its registration address will be used for the additional SIP server. The phone supports encryption of RTP and SIP signaling traffic over the TLS protocol.
Sound menu
By default, when calling, the phone claims all possible codecs. If necessary, unused codecs can be disabled.
In the menu, you can adjust various volume parameters: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your own ringtone.
Menu "Advanced Settings" -> "Global SIP Settings"
If you set up the SIP settings here, they will be applied to both lines automatically, except for the settings “Local SIP port” and “RTP port range” , which can be useful for properly configuring the firewall.
Settings for additional phone features
Menu "Advanced Settings" -> "Phone Settings"
In this menu, additional functions are configured. Such as the “Hotline” - when you pick up the handset, the preset number is automatically dialed, you can turn on auto search in the address book during dialing and auto answer the call.
In case the call transfer is required to be performed with a special combination of buttons (starcode), instead of the standard SIP message 302, this can be specified in the setting “Special code for call transfer” . A useful setting that allows you not to break the connection in the conference, if it left the initiator. You can set call forwarding by condition (busy and non-response) and unconditional.
Menu "Phonebook"
The phone has a built-in phone book, which is quite advanced. It allows you to store up to 300 contact entries, each of which can store up to 3 phone numbers. Entries can be made via the phone's on-screen menu using the web interface. To download or save the phone book in XML format, use the “Phone maintenance” menu -> “Update by HTTP” -> “XML Phone book” ; here you can save or load the phone book in xml format.
If your company uses an LDAP server, you can connect a phone to it and synchronize corporate contacts. 2 and 3 protocol versions are supported, using the “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial” settings, you can also search for the contact name for an incoming and outgoing call, if the contact is in the LDAP directory, the number will be his name is automatically added.
The phone also supports blacklists or ban lists: an unwanted phone number is added to such a list and can no longer be reached.
Menu "Programmable buttons"
The phone has a panel with 9 buttons that can be programmed for speed dialing, prefix before dialing or sending DTMF tones. For example, if you want to log in to the contact center of the bank. This functionality is extremely convenient. Moreover, the speed dial buttons can be tied to a specific SIP-account.
Service Settings
Debug menu
To debug the phone, you can turn on logging by specifying the necessary logs (Menu “Phone maintenance” -> Log). You can view them in two ways:
- In the same menu, enable sending logs to the syslog server.
- In the “Phone maintenance” menu -> “HTTP update” download the file with logs.
Menu "Phone maintenance" -> "Auto Provision" (Automatic update)
Using this menu, you can configure automatic downloading of configuration, firmware and address book to your phone. You can download one of several protocols: http / https / ftp / tftp.
Backup and Software Update
You can copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a particular protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update.
The status of the phone and system software can be viewed in the menu items "Status" and "System Information"
Setting up connection to IP PBX Asterisk using web-interface
Suppose we need to configure two extensions (two SIP accounts). For example, the first entry on the IP PBX Asterisk, the second on the virtual IP PBX:
IP address of the server with Asterisk = 10.10.10.1
UserID = 10
password = kRcB7zT3
SIP server (Asterisk) = 10.10.10.1
In the Asterisk sip.conf configuration, this will be equivalent to:
[ten]
deny = 0.0.0.0 / 0.0.0.0
secret = 7zT3kRcB
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes
callgroup = 01
pickupgroup = 01
allow = g722
dial = SIP / 10
mailbox = 10 @ device
permit = 0.0.0.0 / 0.0.0.0
callerid = device <10>
callcounter = yes
faxdetect = no
Equivalently, when configured in the free-pbx web interface, using the first line as an example:
Go to the line settings for the phone:
To work with Asterisk, it is enough to configure the Username (Username) = 10, password (Password) = 7zT3kRcB and SIP Server (SIP Server) = 10.10.10.10. You can add a label (Label) that will be displayed on the screen of the phone, in this case “Line 1”.
You can reduce the re-registration time from standard 3600 seconds to 600 seconds, this is especially true if the IP PBX is located outside the office, for example, Virtual PBX. If the phone is on a local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click "Submit"
Exactly the same must be done with the second line, for example, city number 78126470011 on the SIP server West Call. Let's register it on a virtual PBX with a non-standard SIP port 9966:
userid = 78126470011
authid = 78126470011
password = Bxu5qnXsXMAgG3l
sip proxy = uc.westcall.net
port = 9955
To specify a non-standard SIP port (other than 5060), you must explicitly specify it in the SIP server line: uc.westcall.net:9955 Next, click the "Apply" button.
In case of successful registration, the corresponding display will appear on the phone screen, so information on the status of line registration is available on the Status menu page:
Account 1: Registered
Account 2: Registered
In order to use the DVO buttons (transfer, hold, conference) no additional configuration is required.
findings
The phone is very attractive considering its cost and functionality. The device is easily configured, it works stably, does not lose registration, has good sound quality, additional functions (transfer, retention, redirection, etc.) also work stably.
Key features of the phone include:
- Support two independent SIP accounts per phone.
- The presence of an additional Ethernet port for connecting to a computer and the ability to work in routing mode
- PoE support (model US102-PYN).
- LCD screen with backlight.
- The ability to configure in addition to network settings, SIP accounts, speed dial buttons and redirects directly from the phone screen.
- Built-in notebook with 300 entries.
- Nine programmable buttons.
If you are interested in checking Escene US102 in your network, we are ready to provide you with a testing machine. Contact the contact details listed on the official website of Escene .
Source: https://habr.com/ru/post/184494/
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