Grandstream Networks GXP-280 Local SIP Port, Retry Wait Time SIP T1 Timeout, SIP T2 Interval

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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.

 

 

 

 

 

 

 

 

 

 

 

 

PUBLISH for Presence

 

Enable Presence feature.

 

 

 

 

 

 

 

 

 

 

 

 

Proxy-Require

 

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

 

 

 

 

 

 

 

 

 

 

 

 

Voice Mail UserID

 

When configured, user can access messages by pressing “MSG” button. This ID

 

 

 

 

is usually the VM portal access number.

 

 

 

 

 

 

 

 

 

 

 

 

Send DTMF

 

This parameter specifies the mechanism to transmit DTMF digit. There are 3

 

 

 

 

supported modes: in audio which means DTMF is combined in audio signal (not

 

 

 

 

very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.

 

 

 

 

 

 

 

 

 

 

 

 

Early Dial

 

Default is No. Use only if proxy supports 484 responses.

 

 

 

 

 

 

 

 

 

 

 

 

Dial Plan Prefix

 

Sets the prefix added to each dialed number.

 

 

 

 

 

 

 

 

 

 

 

 

Delayed Call Forward

 

Time waited before the call is forward to a number or VM.

 

 

Wait Time

 

Default is 20 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Enable Call Features

 

Default is No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are

 

 

 

 

supported locally provided ITSP support those features.

 

 

 

 

 

 

 

 

 

 

 

 

Call Log

 

User can choose to disable Call Log and what kind of calls to log.

 

 

 

 

 

 

Grandstream Networks, Inc.

GXP User Manual

Page 33 of 38

 

Firmware 1.1.6.44

Last Updated: 12/2008

Image 33
Contents Page Table of Figures GUI Interface Examples Welcome Installation Connecting the GXP-2000 GXP-2000 w/GXP-ExtensionGXP-Extension Connection plate GXP-2000 Internal Headset Wiring Schema GXP Product Models Product OverviewGXP Hardware Specifications GXP Key Features in a GlanceGXP Technical Specifications Features Audio FeaturesUpgrades Server FeaturesUsing the GXP SIP Enterprise Phone LCD ButtonsLCD Icons AM PM Line Buttons Handset, Speakerphone and Headset Mode Multiple SIP Accounts and LinesCompleting Calls Speed Dial For example Making Calls using IP AddressesDo Not Disturb Receiving CallsCall Waiting/ Call Hold Mute/DeleteWay Conferencing Voice Messages Message Waiting IndicatorBusy Lamp Field Initiate a Conference CallKey Call Features GXP Call FeaturesKey Pad Configuration Menu Configuration GuideUpgrade Configure Factory Functions Reboot ExitFactory Reset Layer 2 QoS Configure Vlan TagsSIP Definitions Access the Web Configuration MenuDevice Configuration Status Device Configuration Basic SettingsEnd User Password IP Address Multi Purpose Key Time Zone Mute Speaker Ringer in Headset Mode Advanced Settings Voice Frames Timeout Use # asPer TX Layer 3 QoSStun Server AuthenticateVia Http ServerApplication Offhook Auto DownloadDial Dtmf Payload Type OnhookWaiting Disable Call Disable CallWaiting Tone Disable Direct Call ModeSIP Account Settings Delayed Call Forward Retry Wait Time SIP T1 TimeoutEnable Call Features Local SIP PortCaller Request Timer Ring TimeoutCallee Request Timer Force TimerSrtp Mode Disable Multiple MediaSpecial Feature Allow Auto Answer byWeb Configuration Interface Software Upgrade & CustomizationKey Pad Menu No Local Tftp ServerInstructions for local Tftp Upgrade Managing Firmware and Configuration File DownloadInstructions for Restoration Restore Factory Default Setting

GXP-280 specifications

The Grandstream Networks GXP-280 is a well-regarded model in the realm of VoIP (Voice over Internet Protocol) telephony, designed particularly for small to medium-sized businesses. This feature-rich IP phone is recognized for its reliability, ease of use, and versatile functionalities, making it a preferred choice among business users seeking a robust and effective communication solution.

One of the standout features of the GXP-280 is its support for up to four SIP accounts, enabling users to manage multiple lines simultaneously. This is particularly beneficial for businesses that require efficient call handling across different departments or clients. The phone sports an intuitive LCD display that provides clear visibility of caller information, call status, and feature access, ensuring user-friendly operation.

The GXP-280 is powered by advanced technologies such as HD Voice, which provides crystal-clear audio quality during calls. This clarity is further enhanced by the phone’s wideband audio support, allowing for richer sound reproduction that is crucial for effective communication. The phone is equipped with dual Ethernet ports, one of which supports Power over Ethernet (PoE), simplifying the installation process by allowing power and data to be transmitted through a single cable.

In terms of flexibility and personalization, the GXP-280 comes with programmable soft keys, allowing users to customize their interface to suit their specific workflow preferences. Additionally, the phone includes built-in Bluetooth support, which extends its capabilities to integrate with wireless headsets, enabling hands-free operations.

Security is also a pivotal aspect of the GXP-280, as it features encrypted signaling and media, ensuring that business communications remain secure from potential threats. The device supports standard security protocols such as SRTP and TLS, making it compliant with contemporary security standards.

Moreover, the GXP-280 offers a variety of codec options, including G.722, G.711, and G.729, allowing it to adapt to various network conditions and requirements. Its compatibility with a wide range of SIP platforms adds to its appeal, making deployment in existing infrastructures seamless.

To summarize, the Grandstream Networks GXP-280 is a versatile, reliable, and feature-rich IP phone that stands out for its clarity, flexibility, and security, making it an excellent choice for businesses looking to enhance their communication systems. With its impressive array of functionalities and support for modern technologies, the GXP-280 is a representation of Grandstream's commitment to providing high-quality communication solutions.