Grandstream Networks GXP2000 Authenticate Password, Use DNS SRV, User ID is Phone, Number

Page 38

 

Authenticate Password

 

SIP service subscriber’s account password for GXP to register to (SIP) servers of

 

 

 

 

ITSP.

 

 

 

 

 

 

 

 

 

 

 

 

Name

 

SIP service subscriber’s name that is used for Caller ID display.

 

 

 

 

 

 

 

 

 

 

 

 

Use DNS SRV:

 

Default is No. If set to “Yes”, the client will use DNS SRV to look up server.

 

 

 

 

 

 

 

 

 

 

 

 

User ID is Phone

 

If the phone has an assigned PSTN telephone number, this field should be set to

 

 

Number

 

“Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be

 

 

 

 

attached to the “From” header in SIP request

 

 

 

 

 

 

 

 

 

 

 

SIP Registration

 

This parameter controls sending REGISTER messages to the proxy server. The

 

 

 

 

default setting is “Yes”.

 

 

 

 

 

 

 

 

 

 

 

Un-register on Reboot

 

Default is “No”. If set to “Yes”, the SIP user’s registration information will be

 

 

 

 

cleared from the server when the phone reboots.

 

 

 

 

 

 

 

 

 

 

 

SIP Instance ID

 

Default is set “No.” If set to Yes it will be enabled and will add reg-ID and Instance

 

 

 

 

ID on contact header in the REGISTER messages. This feature is mainly

 

 

 

 

provided for servers that don't support SIP Instance ID feature, but will still allow

 

 

 

 

phones to register.

 

 

 

 

 

 

Register Expiration

This parameter allows user to specify the time frequency (in minutes) that GXP refreshes its registration with the specified registrar. The default interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days).

 

Local SIP Port

 

This parameter defines the local SIP port used to listen and transmit. The default

 

 

 

 

value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and

 

 

 

 

Account 4 respectively.

 

 

 

 

 

 

 

 

 

 

 

 

SIP Registration Failure

 

Retry registration if the process failed. Default is 20 seconds.

 

 

Retry Wait Time

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SIP T1 Timeout

 

RFC 3261 SIP T1 timer. Default is 1 second.

 

 

 

 

 

 

 

 

 

 

 

 

SIP T2 Interval

 

RFC 3261 SIP T2 timer. Default is 0.5 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

SIP Transport

 

Choose SIP Transport between UDP and TCP. Default is UDP.

 

 

 

 

 

 

 

 

 

 

 

 

Use RFC3581

 

Default No. When selected the phone will follow the routing procedures specified

 

 

Symmetric Routing

 

in RFC3581.

 

 

 

 

 

 

 

 

 

 

 

 

NAT Traversal (STUN)

 

This parameter activates the NAT traversal mechanism. If activated (by choosing

 

 

 

 

“Yes”) and a STUN server is also specified, the phone performs according to the

 

 

 

 

STUN client specification. Using this mode, the embedded STUN client detects if

 

 

 

 

and what type of NAT/Firewall configuration is used. If the detected NAT is a Full

 

 

 

 

Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped

 

 

 

 

public IP address and port in all of its SIP and SDP messages. If the NAT

 

 

 

 

Traversal field is set to “Yes” with no specified STUN server, the GXP will

 

 

 

 

periodically (every 20 seconds or so) send a blank UDP packet (with no payload

 

 

 

 

data) to the SIP server to keep the “hole” on the NAT open.

 

 

 

 

 

 

 

 

 

 

 

 

SUBSCRIBE for MWI:

 

Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication

 

 

 

 

will be sent periodically.

 

 

 

 

 

 

Grandstream Networks, Inc.

GXP User Manual

Page 38 of 44

 

Firmware 1.2.5.3

Last Updated: 03/2011

Image 38
Contents Grandstream Networks, Inc Table of Figures GUI Interface Examples Welcome Equipment Packaging InstallationConnecting Your Phone GXP-2000 Extension UnitConnecting the GXP-2000AND the GXP-EXTENSION Safety Compliances WarrantyProduct Overview GXP Product ModelsGXP Key Features in a Glance GXP Comparison GuideGXP Hardware Specifications 5mm RJ22GXP Technical Specifications Features LCD Buttons Using the GXP SIP Enterprise PhoneGetting Familiar with the LCD LogIn LCD IconsAM PM Line Buttons Completing Calls Handset, Speakerphone and Headset ModeMultiple SIP Accounts and Lines Making Phone CallsSpeed Dial Making Calls using IP Addresses For exampleCall Waiting/ Call Hold Receiving CallsDo Not Disturb Mute/DeleteEnd Conference Way ConferencingInitiate a Conference Call Voice Messages Message Waiting Indicator Busy Lamp FieldCustomized LCD Screen & XML Call FeaturesGXP Call Features Key Pad Configuration Menu Configuration GuideConfiguration VIA Keypad Layer 2 QoS Configure Vlan Tags Factory Functions Reboot ExitUpgrade Diagnostic Mode All LEDs will light upSIP Definitions Access the Web Configuration MenuConfiguration VIA WEB Browser Device Configuration Status Device Configuration Basic SettingsLCD Backlight Always Multi Purpose KeyTime Zone Time Display FormatDaylight Savings Time Disable in-call DtmfDisable Missed Call LCD BacklightAdvanced Settings Layer 3 QoS No Key Entry TimeoutVoice Frames per TX Layer 2 QoSUse NAT IP Authenticate Conf FileKeep-alive interval Stun ServerIdle Screen XML Automatic UpgradePhonebook XML DownloadDisable Call Public ModeDisable Call Waiting Waiting Tone Disable Direct IP CallsLock keypad update Use Quick IP Call ModeDisable Conference Enable MPK SendingSIP Account Settings Retry Wait Time SIP T1 Timeout Authenticate PasswordUn-register on Reboot Use DNS SRVCaller Request Timer Delayed Call ForwardEnable Call Features Callee Request TimerDisable Multiple Media Enable 100relRing Timeout Srtp ModeRebooting the Phone Remotely Special FeatureSaving the Configuration Changes Eventlist BLF URIFirmware Upgrade Through TFTP/HTTP Software Upgrade & CustomizationWeb Configuration Interface Key Pad MenuConfiguration File Download Managing Firmware and Configuration File DownloadInstructions for local Tftp Upgrade Restore Factory Default Setting Instructions for Restoration
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GXP1200, GXP2000, GXP280/GXP285, GXP2020, GXP2010 specifications

The Grandstream Networks GXP2010 is a versatile and reliable SIP telephone designed for small to medium-sized businesses. With its robust feature set and user-friendly interface, the GXP2010 stands out in the world of IP telephony. This model accommodates the growing demand for efficient communication tools while blending seamlessly into various business environments.

One of the defining characteristics of the GXP2010 is its ability to support multiple SIP accounts. It can manage up to four SIP accounts, allowing users to handle multiple lines of communication efficiently. This feature is especially beneficial for organizations that often deal with a high volume of calls or need to maintain different lines for various departments.

The GXP2010 boasts high-definition audio quality, utilizing Wideband CODEC technology to ensure crystal clear voice communication. This technology helps eliminate background noise, producing a more pleasant and professional experience during calls. The device also includes various audio codecs, providing flexibility to suit different network environments.

In terms of design, the GXP2010 features a user-friendly interface with a backlit LCD display that offers intuitive navigation. The phone is equipped with programmable keys, enabling users to customize their experience and access frequently used functions quickly. The inclusion of HD handset and speakerphone capabilities ensures high audio quality regardless of the speaking method preferred by the user.

Furthermore, the GXP2010 supports Power over Ethernet (PoE), simplifying the installation process by reducing the number of cables needed. This feature allows the phone to receive power and network connectivity through a single Ethernet cable, resulting in a more organized workspace.

Security is another priority for Grandstream, and the GXP2010 reflects this focus with advanced encryption protocols for secure communications. The device supports TLS and SRTP for secure signaling and media encryption, safeguarding sensitive business conversations.

The GXP2010 is not only compatible with a variety of SIP-based platforms, but it also includes support for advanced telephony features such as call forwarding, call transfer, and call waiting. This versatility makes it an excellent choice for businesses seeking to streamline their communication processes.

In conclusion, the Grandstream Networks GXP2010 is an ideal solution for businesses looking for a reliable, high-quality SIP phone that combines versatility and advanced technology. Its multiple SIP account support, high-definition audio capabilities, user-friendly design, and strong security measures make it a valuable addition to any modern office environment.