June 25, 2013 at 20:29Review IP-Phone Escene US102
Hello, 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 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 an integrated bridge that can be switched to routing mode eliminate the need to lay another cable to the employee’s workstation.
The buyer receives all the benefits of IP-telephony, without spending any extra money. The cost of modification without PoE is about 2000 rubles. And about 2,300 rubles is a phone with PoE support, which is cheaper than analogues from other manufacturers.
VoIP
Data transfer
Physical parameters
Additional types of service (additional functions)
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:
A 5-volt power supply Escene AD200 is included only for US102-YN (without PoE), for US102-PYN it needs to be ordered separately.
Phone front panel
Navigation through the on-screen menu is carried out using five navigation buttons and two buttons located to the left and right of the navigation block. Using the navigation buttons, you can see missed, outgoing and incoming calls with one click, as well as decrease or increase the volume.
Buttons signed with three dots are used to navigate the menu, their names change while moving around the phone menu. The Line 1 and Line 2 buttons are used to select a line (SIP account) for an outgoing call.
The button block of additional functions contains:
The phone has a monochrome LCD screen with a resolution of 128x64 pixels. The backlight is not very bright. But this is enough to read messages on the screen without difficulty.
For each line, in addition to its number, you can write an arbitrary name. In this case, “Line 1” is an arbitrary name.
Conversation status
Incoming call
When an incoming or outgoing call, the indicator of the corresponding line lights up.
The phone has two connectors for connecting to a LAN and power over PoE (US102-PYN), a second connector on the right, for connecting to a computer and a socket 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 that leave a minimum of settings in the phone’s menu and a larger number of them only through the web interface, Escene developers decided to make available from the phone’s menu, in addition to the standard settings, settings related to SIP accounts.
This step is justified, in some cases, you can set up your phone faster. In addition, sometimes there may be problems with accessing the phone via the web interface or it may be necessary to remotely explain to the employee how to reconfigure his phone. It will be easier for an untrained person to use the phone menu than the web-based interface.
Initial setup using phone buttons
So, we turned on the phone, connected the LAN port to the local network, which has access to the IP PBX. The employee’s computer was connected to the PC port through a cable.
Now we need to turn on the Russian language in the menu:
Press the “Menu” button, it corresponds to the button on the left, marked with three dots on top, the “Function Settings” menu opens , use the up or down navigation buttons to find the “System Settings” menu , press "Enter" . Press the number 1 (or the Enter button), which corresponds to “Phone Settings” , then select the “Language” menu , press the number 1 (or the Enter button), use the up or down navigation buttons to select “Russian”and click OK . Then press the back button until you exit the menu.
Now you need to configure the network settings:
Press "Menu" , then select the "Settings" menu , number 2 - "Advanced Settings" , the default password is blank, just click "OK" . If you need to configure VLAN (menu item 2) go to the corresponding menu and set its ID and priority, select “Network” , then “LAN port”, by default, after the phone boots up, a DHCP client is turned on, which tries to get an IP address, so there must be a DHCP server in the network where the IP phone is located. If all the settings are made correctly, the phone will receive an IP address and will be ready for further configuration.
If you need to use a static IP address, press the number 1 - “Type” and select “Static” and press “OK” . By default, IP 192.168.0.200 is configured on the phone, to change the IP address, mask, gateway and DNS settings, use the menu buttons and navigation keys, after saving the settings, the phone will reboot. Please 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 access to the phone via telnet.
Of particular note is the PC Port setting ( Menu -> Network -> Advanced Settings -> PC Port ). Here you can configure the network mode between the PC and LAN ports. In bridge mode, it 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, then the IP port and mask are assigned to the PC port, NAT address translation is enabled between the LAN and PC, you can also enable the DHCP server. Thus, the phone also becomes a NAT-enabled router.
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 the number 1 - “Network” , in my case the IP address assigned by DHCP: 192.168.1.253
Setting additional phone functions
All these settings are made in the “Menu” -> “Functions”
Support for additional types of service (VAS) and programmable buttons The
phone supports two independent SIP accounts, that is, registration on two different IP PBXs. When simultaneously registering both lines, the first line will be used by default, to switch to the second line to make an outgoing call or return to the first, use the “Line 1” and “Line 2” buttons.
As for the Far East Region, 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 levels on the phone access: administrator level, which can change any settings and user, which can perform a limited number of settings.
We get to the main menu of the phone’s web configurator. For convenience, we immediately select the Russian language in the lower left menu: The
menu is divided into several groups:
Consider the most important phone menu items. The “Setup Wizard” menu serves for quick basic phone setup, allows you to sequentially configure two tabs: the “Network” menu -> “LAN port” and the basic setup of the SIP account located 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: via DHCP, static IP or PPPoE. An important setting is HTTP and Telnet ports, they should be made non-standard if the phone is in an untrusted network.
Menu "Network" -> "PC port"
Between the LAN and PC ports of the phone, L2 switching is switched on by default - "Bridge" mode . The phone can switch to L3 routing mode - the NAT address translation will turn on on the LAN port, the IP address will need to be configured on the PC port, and if necessary, enable the DHCP server in which to register the pool of IP addresses for clients.
Network Menu -> VLAN Settings
In a corporate network, it is recommended to isolate computer network traffic from voice network traffic, this is most often done using two VLANs. The phone supports VLAN on both ports.
Menu "Network" -> "VPN Settings"
If you need to connect the phone through a secure VPN channel, you can do this directly from the phone, without buying additional equipment (such as a VPN router), the phone supports L2TP and SSL VPN.
The phone allows you to manage a large number of SIP alarm settings and settings for RTP media traffic.
Menu “SIP Accounts” -> “Account 1”
In addition to standard SIP account settings - Username (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’s screen .
In addition to the main IP address of the SIP server, you can add an additional IP SIP server; if registration fails during the timeout, which is 32 seconds by default, its address of the additional SIP server will be used for registration. The phone supports encryption of RTP and SIP signaling traffic via the TLS protocol.
Sound Menu
By default, when calling, the phone declares all possible codecs. If necessary, unused codecs can be disabled.
Various volume parameters are configured in the menu: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your ringtone.
Menu “Advanced Settings” -> “Global SIP Settings”
If you set the SIP settings here, they will be applied for both lines automatically, except for the settings “Local SIP Port” and “RTP Port Range” , which can be useful for proper network settings screen.
Menu "Advanced Settings" -> "Phone Settings"
In this menu, additional functions are configured. Such as “Hot line” - when you pick up the handset, the preset number is automatically dialed, you can enable auto search by address book during dialing and auto answer a call.
If you want to transfer a call using a special combination of buttons (starcode), instead of the standard SIP message 302, this can be specified in the “Special code for transferring calls” setting . A useful setting that allows you to keep the connection in the conference if the initiator leaves it. You can set call forwarding by condition (busy and unanswered) and unconditional.
Menu "Phonebook"
The phone has a built-in phone book, and quite advanced. It allows you to store up to 300 records of contacts, in each of which you can save up to 3 phone numbers. Entries can be made through the on-screen menu of the phone, using the web interface. To download or save a ready-made phonebook in XML format, use the menu “Phone Service” -> “Update via HTTP” -> “XML Phonebook” , here you can save or download the phonebook in xml format.
If the company uses an LDAP server, you can connect a telephone to it and synchronize corporate contacts. Version 2 and 3 of the protocol are supported, also using the settings “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial”You can apply for a contact name for incoming and outgoing calls, if the contact is in the LDAP directory, the name will be automatically added to the number.
The phone also supports blacklists or ban lists: an unwanted phone number is added to this list and can no longer get through.
Programmable Buttons Menu
Your phone has a panel with 9 buttons that you can program to quickly dial numbers, 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 linked to a specific SIP account.
The Debug menu.
To debug the phone, you can enable logging by specifying the necessary logs (Menu “Phone service” -> Log). You can view them in two ways:
Menu “Phone Service” -> “Auto Provision” (Auto update).
Using this menu, you can configure the phone to automatically download configuration, firmware and notebook. You can download using one of several protocols: http / https / ftp / tftp.
Backing up and updating software You can
copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update.
Phone status and system software can be viewed in the menu items “Status” and “System Information”.
Setting up connection to Asterisk IP PBX using the web interface
Suppose we need to configure two extension numbers (two SIP accounts). For example, the first record on the Asterisk IP PBX, the second on the virtual IP PBX:
Server IP address 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:
[ 10]
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 setting up free-pbx in the web interface, using the first line as an example:
Go to the line settings for the phone:
just use Asterisk to configure 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 phone screen, in this case, "Line 1".
You can reduce the time of re-registration from the 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 located on the local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click the “Submit” button.
Exactly the same thing must be done with the second line, for example, city number 78126470011 on the West Call SIP server. We will 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 indication will appear on the phone screen, so information about the line registration status is available on the Status menu page:
Account 1: Registered
Account 2: Registered
In order to use the DVO buttons (transfer, hold, conference) of additional settings not required.
The key features of the phone include:
If you are interested in checking Escene US102 on your network, we are ready to provide you with a device for testing. Please use the contact information provided on the official Escene website .
Digital Angel is the exclusive distributor of Escene equipment in Russia, and we decided to start our blog with a detailed review of the budget phone of this brand.
The base models of Escene IP phones are equipped with the full range of features that are required for voice communications. US Series Universal Phones 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 an integrated bridge that can be switched to routing mode eliminate the need to lay another cable to the employee’s workstation.
To date, the model is represented by two modifications that differ in the possibility of power supply over PoE: US102-YN (without PoE) and US102-PYN (with PoE).
Features
- Easy setup through a clear interface.
- Russified web interface and on-screen menu.
- The ability to fully customize the phone using the screen and buttons, including SIP accounts.
- The ability to adapt the phone to work with SIP-compatible equipment.
- Functionality is greater than most IP-PBXs and telecom operators currently support.
The buyer receives all the benefits of IP-telephony, without spending any extra money. The cost of modification without PoE is about 2000 rubles. And about 2,300 rubles is a phone with PoE support, which is cheaper than analogues from other manufacturers.
Functionality
- Support for high definition audio Voice HD (G.722 codec).
- Direct SIP connection to Virtual IP PBXs (e.g. Broadworks, MFI RTU) and to office IP PBXs (e.g. Asterisk, 3CX IP PBX).
- Two Ethernet (PC / LAN) ports with VLAN support and the ability to work in switching or routing mode.
- Easy installation and operation, the possibility of advanced settings (including SIP and DVO functions) through 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.
- Support for encryption using a VPN tunnel or encryption of signal SIPS and SRTP media traffic.
- Support for corporate notebook using LDAP or XML or personal notebook.
- Russified OSD and web-based phone.
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).
- Support for DNS SRV.
- Two SIP accounts with the ability to register on two independent SIP servers and the ability to automatically switch in case of loss of registration.
Data transfer
- 2 * RJ45 10 / 100M Ethernet interfaces (LAN / PC) with PoE support (for US102-PYN model).
- Support VLAN / QoS.
- IP Addressing: DHCP client or static IP assignment.
- 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 backlight 128 * 64 characters.
- 5 navigation buttons.
- Sound adjustment buttons (9 levels).
- Buttons of additional services: mute the microphone, headset, phone, messages, menus, hold, repeat, conference and translation.
- 9 programmable buttons + Hold call button Hold.
- Speakerphone.
Additional types of service (additional functions)
- Waiting for a second call, queue (if it supports IP PBX), transferring a call, transferring a call, holding a call, intercepting a call, returning a call, retrying a call, answering automatically.
- Speed dialing, a button to start recording a conversation using the old code (if it supports IP PBX).
- Multilateral conference (if supports IP PBX), 3-way conference on the phone.
- Do Not Disturb (DND).
- Voicemail (if the function is supported by IP PBX).
- Personal Notebook, Corporate Notebook (LDAP).
Control
- Protocol update: HTTP / TFTP / (PnP auto-provisioning) 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 Volts, output DC 5 Volts / 1A).
- Power Over Ethernet (IEEE 802.af) LAN port - for modification of US102-PYN.
Scope of delivery, 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.
A 5-volt power supply Escene AD200 is included only for US102-YN (without PoE), for US102-PYN it needs to be ordered separately.
Phone front panel
Navigation through the on-screen menu is carried out using five navigation buttons and two buttons located to the left and right of the navigation block. Using the navigation buttons, you can see missed, outgoing and incoming calls with one click, as well as decrease or increase the volume.
Buttons signed with three dots are used to navigate the menu, their names change while moving around the phone menu. The Line 1 and Line 2 buttons are used to select a line (SIP account) for an outgoing call.
The button block of additional functions contains:
- Сonf - to create a 3-way conference (initiator, and two participants), to create a conference with a large number of participants, support for such a function on the IP PBX is required.
- Trans - transfer a call.
- Redial - redial the last number.
- HF - enable or disable the speakerphone (speakerphone).
- A block of 10 buttons , of which one Hold button - hold a call.
- 9 programmable buttons , each of which can be assigned one of the following functions:
- One touch dialing.
- One-touch prefix dialing.
- Sending a combination of DTMF tones.
The presence of these buttons is extremely convenient, because each employee has a list of “favorite” numbers. To quickly dial them, you do not need to buy an additional extension console separately. There is also a paper insert to indicate buttons.
The phone has a monochrome LCD screen with a resolution of 128x64 pixels. The backlight is not very bright. But this is enough to read messages on the screen without difficulty.
For each line, in addition to its number, you can write an arbitrary name. In this case, “Line 1” is an arbitrary name.
Conversation status
Incoming call
When an incoming or outgoing call, the indicator of the corresponding line lights up.
The phone has a removable stand. If the phone needs to be hung on a wall, it is easy to remove. There are mounting holes on the case for attaching the phone to the wall.
The phone has two connectors for connecting to a LAN and power over PoE (US102-PYN), a second connector on the right, for connecting to a computer and a socket 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 that leave a minimum of settings in the phone’s menu and a larger number of them only through the web interface, Escene developers decided to make available from the phone’s menu, in addition to the standard settings, settings related to SIP accounts.
This step is justified, in some cases, you can set up your phone faster. In addition, sometimes there may be problems with accessing the phone via the web interface or it may be necessary to remotely explain to the employee how to reconfigure his phone. It will be easier for an untrained person to use the phone menu than the web-based interface.
Initial setup using phone buttons
So, we turned on the phone, connected the LAN port to the local network, which has access to the IP PBX. The employee’s computer was connected to the PC port through a cable.
Now we need to turn on the Russian language in the menu:
Press the “Menu” button, it corresponds to the button on the left, marked with three dots on top, the “Function Settings” menu opens , use the up or down navigation buttons to find the “System Settings” menu , press "Enter" . Press the number 1 (or the Enter button), which corresponds to “Phone Settings” , then select the “Language” menu , press the number 1 (or the Enter button), use the up or down navigation buttons to select “Russian”and click OK . Then press the back button until you exit the menu.
Now you need to configure the network settings:
Press "Menu" , then select the "Settings" menu , number 2 - "Advanced Settings" , the default password is blank, just click "OK" . If you need to configure VLAN (menu item 2) go to the corresponding menu and set its ID and priority, select “Network” , then “LAN port”, by default, after the phone boots up, a DHCP client is turned on, which tries to get an IP address, so there must be a DHCP server in the network where the IP phone is located. If all the settings are made correctly, the phone will receive an IP address and will be ready for further configuration.
If you need to use a static IP address, press the number 1 - “Type” and select “Static” and press “OK” . By default, IP 192.168.0.200 is configured on the phone, to change the IP address, mask, gateway and DNS settings, use the menu buttons and navigation keys, after saving the settings, the phone will reboot. Please 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 access to the phone via telnet.
Of particular note is the PC Port setting ( Menu -> Network -> Advanced Settings -> PC Port ). Here you can configure the network mode between the PC and LAN ports. In bridge mode, it 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, then the IP port and mask are assigned to the PC port, NAT address translation is enabled between the LAN and PC, you can also enable the DHCP server. Thus, the phone also becomes a NAT-enabled router.
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 the number 1 - “Network” , in my case the IP address assigned by DHCP: 192.168.1.253
Setting additional phone functions
All these settings are made in the “Menu” -> “Functions”
- Auto answer allows you to set up an automatic answer to a call without picking up the handset.
- DND allows you to reject all calls in case of busy subscriber.
- VM Number — Set the number for accessing voicemail.
- The hot line 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 - to program speed dial keys.
Support for additional types of service (VAS) and programmable buttons The
phone supports two independent SIP accounts, that is, registration on two different IP PBXs. When simultaneously registering both lines, the first line will be used by default, to switch to the second line to make an outgoing call or return to the first, use the “Line 1” and “Line 2” buttons.
As for the Far East Region, 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 is today almost all IP PBXs on the market.
- The Trans button - call transfer with consultation and blind, also uses the SIP 302 Moved Temporarily.
- When you click on the Redial button , the last number dialed is redialled.
- HF button - allows you to turn the speakerphone on or off, answer a call with the speakerphone turned on, or end a call if the conversation is through a 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 levels on the phone access: administrator level, which can change any settings and user, which can perform a limited number of settings.
We get to the main menu of the phone’s web configurator. For convenience, we 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 entries and advanced features for signaling and media traffic)
- Settings for additional phone functions (settings for the phone book, programmable buttons, dial plan, sound, etc.)
- Service settings (logging, reset and reboot, configuration management and software updates, etc.)
Consider the most important phone menu items. The “Setup Wizard” menu serves for quick basic phone setup, allows you to sequentially configure two tabs: the “Network” menu -> “LAN port” and the basic setup of the SIP account located 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: via DHCP, static IP or PPPoE. An important setting is HTTP and Telnet ports, they should be made non-standard if the phone is in an untrusted network.
Menu "Network" -> "PC port"
Between the LAN and PC ports of the phone, L2 switching is switched on by default - "Bridge" mode . The phone can switch to L3 routing mode - the NAT address translation will turn on on the LAN port, the IP address will need to be configured on the PC port, and if necessary, enable the DHCP server in which to register the pool of IP addresses for clients.
Network Menu -> VLAN Settings
In a corporate network, it is recommended to isolate computer network traffic from voice network traffic, this is most often done using two VLANs. The phone supports VLAN on both ports.
Menu "Network" -> "VPN Settings"
If you need to connect the phone through a secure VPN channel, you can do this directly from the phone, without buying additional equipment (such as a VPN router), the phone supports L2TP and SSL VPN.
VoIP Settings
The phone allows you to manage a large number of SIP alarm settings and settings for RTP media traffic.
Menu “SIP Accounts” -> “Account 1”
In addition to standard SIP account settings - Username (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’s screen .
In addition to the main IP address of the SIP server, you can add an additional IP SIP server; if registration fails during the timeout, which is 32 seconds by default, its address of the additional SIP server will be used for registration. The phone supports encryption of RTP and SIP signaling traffic via the TLS protocol.
Sound Menu
By default, when calling, the phone declares all possible codecs. If necessary, unused codecs can be disabled.
Various volume parameters are configured in the menu: handset, ringer, microphone, speakerphone. You can enable echo cancellation and VAD. Moreover, you can download your ringtone.
Menu “Advanced Settings” -> “Global SIP Settings”
If you set the SIP settings here, they will be applied for both lines automatically, except for the settings “Local SIP Port” and “RTP Port Range” , which can be useful for proper network settings screen.
Settings for advanced phone features
Menu "Advanced Settings" -> "Phone Settings"
In this menu, additional functions are configured. Such as “Hot line” - when you pick up the handset, the preset number is automatically dialed, you can enable auto search by address book during dialing and auto answer a call.
If you want to transfer a call using a special combination of buttons (starcode), instead of the standard SIP message 302, this can be specified in the “Special code for transferring calls” setting . A useful setting that allows you to keep the connection in the conference if the initiator leaves it. You can set call forwarding by condition (busy and unanswered) and unconditional.
Menu "Phonebook"
The phone has a built-in phone book, and quite advanced. It allows you to store up to 300 records of contacts, in each of which you can save up to 3 phone numbers. Entries can be made through the on-screen menu of the phone, using the web interface. To download or save a ready-made phonebook in XML format, use the menu “Phone Service” -> “Update via HTTP” -> “XML Phonebook” , here you can save or download the phonebook in xml format.
If the company uses an LDAP server, you can connect a telephone to it and synchronize corporate contacts. Version 2 and 3 of the protocol are supported, also using the settings “LDAP Lookup For Incoming Call” and “LDAP Lookup For PreDial / Dial”You can apply for a contact name for incoming and outgoing calls, if the contact is in the LDAP directory, the name will be automatically added to the number.
The phone also supports blacklists or ban lists: an unwanted phone number is added to this list and can no longer get through.
Programmable Buttons Menu
Your phone has a panel with 9 buttons that you can program to quickly dial numbers, 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 linked to a specific SIP account.
Service Settings
The Debug menu.
To debug the phone, you can enable logging by specifying the necessary logs (Menu “Phone service” -> Log). You can view them in two ways:
- In the same menu, enable sending logs to the syslog server.
- In the menu “Phone Service” -> “Update via HTTP” download the log file.
Menu “Phone Service” -> “Auto Provision” (Auto update).
Using this menu, you can configure the phone to automatically download configuration, firmware and notebook. You can download using one of several protocols: http / https / ftp / tftp.
Backing up and updating software You can
copy configuration files using three different protocols. FTP, TFTP and HTTP - the choice of a protocol is a matter of taste and convenience. Software update is extremely simple, you need to select the firmware file, then click update.
Phone status and system software can be viewed in the menu items “Status” and “System Information”.
Setting up connection to Asterisk IP PBX using the web interface
Suppose we need to configure two extension numbers (two SIP accounts). For example, the first record on the Asterisk IP PBX, the second on the virtual IP PBX:
Server IP address 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:
[ 10]
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 setting up free-pbx in the web interface, using the first line as an example:
Go to the line settings for the phone:
just use Asterisk to configure 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 phone screen, in this case, "Line 1".
You can reduce the time of re-registration from the 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 located on the local network and the IP PBX is on the Internet, no special settings are usually required to overcome NAT. Next, click the “Submit” button.
Exactly the same thing must be done with the second line, for example, city number 78126470011 on the West Call SIP server. We will 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 indication will appear on the phone screen, so information about the line registration status is available on the Status menu page:
Account 1: Registered
Account 2: Registered
In order to use the DVO buttons (transfer, hold, conference) of additional settings not required.
conclusions
The phone is very attractive considering its cost and functionality. The device is easy to configure, works stably, does not lose registration, has good sound quality, additional functions (transfer, hold, call forwarding, etc.) also work stably.
The key features of the phone include:
- Support for two independent SIP accounts on the 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 parameters, SIP accounts, speed dial buttons and call forwarding directly from the phone screen.
- Built-in notebook for 300 entries.
- Nine programmable buttons.
If you are interested in checking Escene US102 on your network, we are ready to provide you with a device for testing. Please use the contact information provided on the official Escene website .
