Grandstream Networks GXP2020 UAS Specify Refresher, Force Invite, Enable 100rel, Ring Timeout

Page 40

 

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 ring 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 Method

 

Whether to use “<sip:anonymous@anonymous.invalid>” in the From Header or P-

 

 

 

 

Asserted-Identity header.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Anonymous Call

 

Default is NO. If set to YES, anonymous call will be rejected

 

 

Rejection

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Auto Answer

 

Default is No. If set to “Yes”, GXP 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

 

 

Call-Info

 

automatically (so called paging mode).

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Turn off speaker on

 

When BYE is received, the phone will turn off its speaker automatically.

 

 

remote disconnect

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Check SIP User ID for

 

Check the SIP User ID in Request URI. If they don’t match, the call will be

 

 

incoming INVITE

 

rejected.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Refer-To Use Target

 

Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the

 

 

Contact

 

transferred target’s Contact header information.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Disable Multiple Media

 

Default is No.

 

 

 

 

Attribute in SDP

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Preferred Vocoder

 

GXP supports up to 7 different Vocoder types including G.711(a/µ) (also known

 

 

 

 

as PCMU/PCMA), GSM, G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).

 

 

 

 

Configure Vocoders in a preference list that is included with the same preference

 

 

 

 

order in SDP message. Enter the first Vocoder in this list by choosing the

 

 

 

 

appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by

 

 

 

 

choosing the appropriate option in “Choice 8”.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

SRTP Mode

 

Enable SRTP mode based on selection. Default is No.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

GXP User Manual

Page 40 of 44

 

 

 

 

Firmware 1.2.5.3

Last Updated: 03/2011

 

Image 40
Contents Grandstream Networks, Inc Table of Figures GUI Interface Examples Welcome Installation Connecting Your PhoneEquipment Packaging 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 Getting Familiar with the LCD Using the GXP SIP Enterprise PhoneLCD Buttons LogIn LCD IconsAM PM Line Buttons Handset, Speakerphone and Headset Mode Multiple SIP Accounts and LinesCompleting Calls Making Phone CallsSpeed Dial Making Calls using IP Addresses For exampleReceiving Calls Do Not DisturbCall Waiting/ Call Hold Mute/DeleteInitiate a Conference Call Way ConferencingEnd Conference Voice Messages Message Waiting Indicator Busy Lamp FieldGXP Call Features Call FeaturesCustomized LCD Screen & XML Configuration VIA Keypad Configuration GuideKey Pad Configuration Menu Factory Functions Reboot Exit UpgradeLayer 2 QoS Configure Vlan Tags Diagnostic Mode All LEDs will light upSIP Configuration VIA WEB Browser Access the Web Configuration MenuDefinitions Device Configuration Status Device Configuration Basic SettingsMulti Purpose Key Time ZoneLCD Backlight Always Time Display FormatDisable in-call Dtmf Disable Missed CallDaylight Savings Time LCD BacklightAdvanced Settings No Key Entry Timeout Voice Frames per TXLayer 3 QoS Layer 2 QoSAuthenticate Conf File Keep-alive intervalUse NAT IP Stun ServerAutomatic Upgrade Phonebook XMLIdle Screen XML DownloadPublic Mode Disable Call WaitingDisable Call Waiting Tone Disable Direct IP CallsUse Quick IP Call Mode Disable ConferenceLock keypad update Enable MPK SendingSIP Account Settings Authenticate Password Un-register on RebootRetry Wait Time SIP T1 Timeout Use DNS SRVDelayed Call Forward Enable Call FeaturesCaller Request Timer Callee Request TimerEnable 100rel Ring TimeoutDisable Multiple Media Srtp ModeSpecial Feature Saving the Configuration ChangesRebooting the Phone Remotely Eventlist BLF URISoftware Upgrade & Customization Web Configuration InterfaceFirmware Upgrade Through TFTP/HTTP Key Pad MenuInstructions for local Tftp Upgrade Managing Firmware and Configuration File DownloadConfiguration File Download 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.