Infinity 2500 Hospitality IP Phone Administration Guide
Accessing the Web Interface
Connect the phone to a network that is setup for DHCP. After initialization is complete, perform the following steps to obtain the IP address assigned to the phone:
-
While the phone is in idle mode, long press the ' # ' sign and the screen will display the IP address for a few seconds.
OR - Dial ' # * 111 ' and the phone will voice announce the IP address.
From a browser, enter the IP address of the phone.
Login to the web interface using an administrator account. (Default username admin and default password admin)
System
System - Information
This page displays the system information of the device including:
- Model
- Hardware Version
- Software Version
- Uptime
- Last Uptime
- MEMInfo
It also shows a summary of the network status:
- Network Mode
- MAC
- IP
- Subnet Mask
- Default Gateway
Plus a summary of the SIP account status:
- SIP User
- SIP account status ( Registered / Unapplied / Trying / Timeout )
System - Account
An account is required to access the phone’s web interface. There are two types of accounts: Administrators and Users.
- A User account has limited access to the device and cannot change certain device settings.
- An Administrator account has full access and can add/delete and manage other accounts and set passwords.
NOTE: The device is shipped with a default Administrator account. The default username and password for this account is ‘ admin ’.
System - Configurations
Administrators can export or import the device configuration on this page and reset the device to factory setting.
System - Upgrade
The device supports automatic upgrades by periodically checking the software release version on the cloud server. Software can also be downloaded and updated manually if the device is unable to connect to the cloud server.
System - Auto Provision
The Auto Provision page allows administrators to easily deploy and manage devices in bulk.
System - Tools
System tools provided on this page help administrators identify and troubleshoot issues.
Network
Network Basic
Network connection type and parameters are configured on this page.
Network Advanced
Network VPN
Line
Line - SIP
The SIP settings of the line are configured on this page.
Basic Settings
BASIC SETTINGS | DESCRIPTION |
---|---|
Line Status | Display the current line status at page loading. To get the up to date line status, user has to refresh the page manually. |
Username | Enter the username of the service account. |
Display Name | Enter the display name to be sent in a call request. |
Authentication Name | Enter the authentication name of the service account |
Authentication Password | Enter the authentication password of the service account |
SIP Proxy Server Address | Enter the IP or FQDN address of the SIP proxy server |
SIP Proxy Server Port | Enter the SIP proxy server port, default is 5060 |
Outbound Proxy Address | Enter the IP or FQDN address of outbound proxy server provided by the service provider |
Outbound Proxy Port | Enter the outbound proxy port, default is 5060 |
Realm | Enter the SIP domain if requested by the service provider |
Activate | Whether the service of the line should be activated |
Codec Settings | Set the priority and availability of the codecs by adding or remove them from the list. |
Advanced Settings
ADVANCED SETTINGS | DESCRIPTION | |
---|---|---|
Call Forward Unconditional | Enable unconditional call forward, all incoming calls will be forwarded to the number specified in the next field | |
Call Forward Number for Unconditional | Set the number of unconditional call forward | |
Call Forward on Busy | Enable call forward on busy, when the phone is busy, any incoming call will be forwarded to the number specified in the next field | |
Call Forward Number for Busy | Set the number of call forward on busy | |
Call Forward on No Answer | Enable call forward on no answer, when an incoming call is not answered within the configured delay time, the call will be forwarded to the number specified in the next field | |
Call Forward Number for No Answer | Set the number of call forward on no answer | |
Call Forward Delay for No Answer | Set the delay time of not answered call before being forwarded | |
Enable Hotline | Enable hotline configuration, the device will dial to the specific number immediately at audio channel opened by off-hook handset or turn on hands-free speaker or headphone | |
Hotline Number | Set the hotline dialing number | |
Hotline Delay | Set the delay for hotline before the system automatically dialed it | |
Enable Auto Answering | Enable auto-answering, the incoming calls will be answered automatically after the delay time | |
Auto Answering Delay | Set the delay for incoming call before the system automatically answered it | |
Subscribe For Voice Message | Enable the device to subscribe a voice message waiting notification, if enabled, the device will receive notification from the server if there is voice message waiting on the server | |
Voice Message Number | Set the number for retrieving voice message | |
Voice Message Subscribe Period | Set the interval of voice message notification subscription | |
Enable DND | Enable Do-not-disturb, any incoming call to this line will be rejected automatically | |
Blocking Anonymous Call | Reject any incoming call without presenting caller ID | |
Use 182 Response for Call waiting | Set the device to use 182 response code at call waiting response | |
Anonymous Call Standard | Set the standard to be used for anonymous | |
Dial Without Registered | Set call out by proxy without registration | |
User Agent | Set the user agent, the default is Model with Software Version. | |
Use Quote in Display Name | Whether to add quote in display name | |
Ring Type | Set the ring tone type for the line | |
Conference Type | Set the type of call conference, Local=set up call conference by the device itself, maximum supports two remote parties, Server=set up call conference by dialing to a conference room on the server | |
Server Conference Number | Set the conference room number when conference type is set to be Server | |
Transfer Timeout | Set the timeout of call transfer process | |
Enable Long Contact | Allow more parameters in contact field per RFC 3840 | |
Enable Missed Call Log | If enabled, the phone will save missed calls into the call history record. | |
Response Single Codec | If setting enabled, the device will use single codec in response to an incoming call request | |
Use Feature Code | When this setting is enabled, the features in this section will not be handled by the device itself but by the server instead. In order to control the enabling of the features, the device will send feature code to the server by dialing the number specified in each feature code field. | |
Enable DND | Set the feature code to dial to the server | |
Disable DND | Set the feature code to dial to the server | |
Enable Call Forward Unconditional | Set the feature code to dial to the server | |
Disable Call Forward Unconditional | Set the feature code to dial to the server | |
Enable Call Forward on Busy | Set the feature code to dial to the server | |
Disable Call Forward on Busy | Set the feature code to dial to the server | |
Enable Call Forward on No Answer | Set the feature code to dial to the server | |
Disable Call Forward on No Answer | Set the feature code to dial to the server | |
Enable Blocking Anonymous Call | Set the feature code to dial to the server | |
Disable Blocking Anonymous Call | Set the feature code to dial to the server | |
Specific Server Type | Set the line to collaborate with specific server type. | |
Registration Expiration | Set the SIP expiration interval | |
Use VPN | Set the line to use VPN restrict route | |
Use STUN | Set the line to use STUN for NAT traversal | |
Convert URI | Convert not digit and alphabet characters to %hh hex code | |
DTMF Type | Set the DTMF type to be used for the line | |
DTMF SIP INFO Mode | Set the SIP INFO mode to send ‘*’ and ‘#’ or ‘10’ and ‘11’ | |
Transport Protocol | Set the line to use TCP or UDP for SIP transmission | |
SIP Version | Set the SIP version | |
Caller ID Header | Set the Caller ID Header | |
Enable Strict Proxy | Enables the use of strict routing. When the phone receives packets from the server, it will use the source IP address, not the address in via field. | |
Enable user=phone | Sets user=phone in SIP messages. | |
Enable SCA | Enable/Disable SCA (Shared Call Appearance ) | |
Enable BLF List | Enable/Disable BLF List | |
Enable DNS SRV | Set the line to use DNS SRV which will resolve the FQDN in proxy server into a service list | |
Keep Alive Type | Set the line to use dummy UDP or SIP OPTION packet to keep NAT pinhole opened | |
Keep Alive Interval | Set the keep alive packet transmitting interval | |
Sync Clock Time | Time Sycn with server | |
Enable Session Timer | Set the line to enable call ending by session timer refreshment. The call session will be ended if there is not new session timer event update received after the timeout period | |
Session Timeout | Set the session timer timeout period | |
Enable Rport | Set the line to add rport in SIP headers | |
Enable PRACK | Set the line to support PRACK SIP message | |
Keep Authentication | Keep the authentication parameters from previous authentication | |
Auto TCP | Using TCP protocol to guarantee usability of transport for SIP messages above 1500 bytes | |
Enable Feature Sync | Feature Sycn with server | |
Enable GRUU | Support Globally Routable User-Agent URI (GRUU) | |
BLF Server | The registered server will receive the subscription package from ordinary application of BLF phone. Please enter the BLF server, if the sever does not support subscription package, the registered server and subscription server will be separated. | |
BLF List Number | BLF List allows one BLF key to monitor the status of a group. Multiple BLF lists are supported. | |
SIP Encryption | Enable SIP encryption such that SIP transmission will be encrypted | |
SIP Encryption Key | Set the pass phrase for SIP encryption | |
RTP Encryption | Enable RTP encryption such that RTP transmission will be encrypted | |
RTP Encryption Key | Set the pass phrase for RTP encryption |
Line - Dial Peer
Line - Dial Plan