SUBSCRIBE for

 

Default is No. This is mainly used for IMS purposes. When enabled, the terminals

 

 

Registration Event

 

should store the Service-Route header values after successfully registered, and

 

 

 

 

thereafter add a route header with the values stored in the Service-Route when

 

 

 

 

initiating a request excluding REGISTER.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

BLF Call-pickup Prefix

 

Default is ‘**”. This prefix is prepended when answering call with BLF key.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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

 

 

 

 

supported locally provided ITSP support those features. In addition, “ForwardAll”

 

 

 

 

softkey will be hidden if call feature code is disabled for Account 1.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Call Log

 

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

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Session Expiration

 

The SIP Session Timer extension enables SIP sessions to be periodically

 

 

 

 

“refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval

 

 

 

 

expires, if there is no refresh via a UPDATE or re-INVITE message, the session is

 

 

 

 

terminated.

 

 

 

 

 

 

Session Expiration is the time (in seconds) at which the session is considered

 

 

 

 

timed out, provided no successful session refresh transaction occurs beforehand.

 

 

 

 

The default value is 180 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Min-SE

 

Defines the minimum session expiration (in seconds). Default is 90 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Caller Request Timer

 

If set to “Yes”, the phone will use session timer when it makes outbound calls if

 

 

 

 

remote party supports session timer.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Callee Request Timer

 

If selecting “Yes”, the phone will use session timer when it receives inbound calls

 

 

 

 

with session timer request.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Force Timer

 

If set to “Yes”, the phone will use session timer even if the remote party does not

 

 

 

 

support this feature. If set to “No”, the session timer is enabled only when the

 

 

 

 

remote party supports this feature. To turn off Session Timer, select “No” for

 

 

 

 

Caller Request Timer, Callee Request Timer, and Force Timer.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

UAC Specify Refresher

 

As a Caller, select UAC to use the phone as the refresher, or UAS to use the

 

 

 

 

Callee or proxy server as the refresher.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

GXP User Manual

Page 39 of 44

 

 

 

 

Firmware 1.2.5.3

Last Updated: 03/2011

 

Page 39
Image 39
Grandstream Networks GXP1200 Subscribe for, Registration Event, Publish for Presence, Proxy-Require, Voice Mail UserID

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.