Grandstream Networks GXP21XX Session Expiration, Min-SE, Caller Request Timer, Force Timer

Page 37

 

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.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

UAS Specify Refresher

 

As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to

 

 

 

 

 

 

 

use the phone as the refresher.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Force INVITE

 

Session Timer can be refreshed using INVITE method or UPDATE method. Select

 

 

 

 

 

 

 

“Yes” to use INVITE method to refresh the session timer.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Enable 100rel

 

PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional

 

 

 

 

 

 

 

responses (1xx series). This is required to support PSTN inter-networking.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Account Ring Tone

 

There are 4 uniquely defined ring tones:

 

 

 

 

 

 

 

 

 

 

 

• One (1) System Ring Tone: when selected, all calls will ring with system

 

 

 

 

ring tone.

 

 

 

 

 

 

• Three (3) Customer Ring Tones: when selected, incoming calls from

 

 

 

 

designated account will play selected ring tone.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Ring Timeout

 

Defines how long the phone will ring when receiving a call. Default is 60 seconds.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Send Anonymous

 

If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will

 

 

 

 

 

 

 

be set to anonymous, essentially blocking the Caller ID from displaying.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Anonymous Call

 

Default is “No”. If set to “Yes”, anonymous call will be rejected.

 

 

 

 

 

Rejection

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Auto Answer

 

Default is “No”. If set to “Yes”, GXP21xx will automatically switch on speaker to

 

 

 

 

 

 

 

answer the incoming call. Set to Intercom/Paging mode, it will answer the call based

 

 

 

 

on the SIP info header from the server.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Allow Auto Answer by

 

If the Call-Info header contains answer-after=0, the call be answered automatically

 

 

 

 

 

Call-Info

 

(so called paging mode).

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

 

GXP21xx User Manual

Page 36 of 41

 

 

 

 

Firmware version: 1.0.1.66

Last Updated: 05/2011

 

Page 36 Page 38
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Page 36 Page 38
Contents Grandstream Networks, Inc Table of Figures GUI Interface Examples Welcome Connecting Your Phone InstallationEquipment Packaging EXTConnecting the GXP2120/2110 and the GXP Extension Warranty Safety CompliancesGXP21xx Product Models Product OverviewGXP21xx Comparison Guide GXP21xx Hardware Specifications GXP21xx Key Features in a GlanceLED Lines Protocol Support Display Feature Keys GXP21xx Technical SpecificationsAudio Features HumidityFirmware Upgrades Telephony FeaturesFeatures NetworkGetting Familiar with the LCD Using the GXP21xx SIP Enterprise PhoneLCD Buttons GXP21xx Keypad Buttons LCD IconsMultiple SIP Accounts and Lines Handset, Speakerphone and Headset ModeMaking Phone Calls Completing Calls Speed Dial Making Calls using IP AddressesReceiving Calls Setting up the phone to make Quick IP callsDo Not Disturb For exampleMute Call Waiting/ Call HoldCall Transfer Way ConferencingBusy Lamp Field Voice Messages Message Waiting IndicatorShared Call Appearance SCA End ConferenceGXP21xx Call Features Call FeaturesCustomized LCD Screen & XML Configuration VIA Keypad Configuration GuideKey Pad Configuration Menu SIP SIP Configuration VIA WEB Browser Access the Web Configuration MenuDefinitions Device Configuration Settings/Basic Settings Device Configuration StatusMTZ+6MDT+5 Stock Update Weather UpdateCurrency Update Disable in-call DtmfDevice Configuration Settings /Advanced Settings ACS URL Port Phonebook XML At Boot-upDownload Phonebook XML ServerSyslog Level Password Distinctive Ring ToneSend SIP Log NTP serverDtmf SIP Account Settings Authenticate Password Authenticate IDDNS Mode Backup IPSubscribe for MWI NAT TraversalPublish for Presence Proxy-RequireEnable Call Features Delayed Call ForwardDial Plan Wait TimeCallee Request Timer Caller Request TimerForce Timer Enable 100relNo Key Entry Timeout Srtp ModeRefer-To Use Target ContactRebooting the Phone Remotely Saving the Configuration ChangesWeb Configuration Interface Software Upgrade & CustomizationFirmware Upgrade Through TFTP/HTTP Key Pad MenuInstructions for Local Tftp Upgrade Managing Firmware and Configuration File DownloadConfiguration File Download Instructions for Restoration Restore Factory Default Setting
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GXP21XX specifications

Grandstream Networks GXP21XX series is a line of high-performance VoIP (Voice over Internet Protocol) desktop IP phones designed to deliver a versatile communication experience for both small and medium-sized businesses as well as large enterprises. The GXP21XX series encompasses a range of models, including the GXP2130, GXP2140, and GXP2160, each offering distinct features and capabilities to cater to various user needs.

One of the main features of the GXP21XX series is its support for high-definition (HD) voice technology. This ensures crystal-clear audio quality during calls, enhancing the overall communication experience. HD voice is complemented by advanced echo cancellation and noise reduction technologies, making it easy to communicate even in bustling environments.

The GXP21XX series boasts a user-friendly interface supported by a large LCD display, which is designed for easy navigation and menu access. These phones offer programmable keys, allowing users to customize their devices for quick access to frequently used features, including speed dials and call transfers.

In terms of connectivity, the GXP21XX series is equipped with dual Gigabit Ethernet ports, facilitating high-speed connections to both the local network and the internet. This allows for seamless integration into existing infrastructures without compromising call quality or network performance. Additionally, the phones support Power over Ethernet (PoE), enabling straightforward installation without the need for additional power sources.

The phones come pre-loaded with a multitude of voice codecs, including G.722, G.711, and G.729, offering flexibility to adapt to various network conditions. Users can enjoy secure communication through built-in security encryption features, such as TLS and SRTP, ensuring sensitive data remains safeguarded.

Moreover, the GXP21XX series incorporates advanced functionalities, such as support for up to six SIP accounts, call presence, and a full-duplex speakerphone, making them ideal for multi-tasking and collaborative environments. Interoperability with major VoIP platforms enhances compatibility, ensuring that users can easily integrate their devices into diverse communication systems.

In conclusion, the Grandstream GXP21XX series stands out as a robust and feature-rich solution for VoIP communication. With its focus on HD voice quality, user-friendly interface, and advanced connectivity options, it provides a comprehensive tool for enhancing both individual and team productivity. Whether used in an office environment or for remote work, the GXP21XX series delivers reliability and performance tailored to the needs of today’s business communication landscape.
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