Support SIP Instance ID

 

Selects whether or not SIP Instance ID is supported.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

NAT Traversal

 

This parameter activates the NAT traversal mechanism. It has options: No, STUN,

 

 

 

 

 

 

 

Keep-Alive, UPnP, Auto, VPN.

 

 

 

 

If selecting STUN 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 selecting Keep-Alive with no specified STUN server, the GXP1400/1405 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.

 

 

 

 

 

 

 

 

 

 

 

 

SUBSCRIBE for

 

Default is “No”. When set to “Yes” a SUBSCRIBE for Registration will be sent

 

 

 

 

 

Registration

 

periodically.

 

 

 

 

 

 

 

 

 

 

 

 

Feature Key

 

Default is “No”. This option is to synchronize DND/Call Forward features with

 

 

 

 

 

Synchronization

 

Broadsoft. When set to “Yes”, a SUBSCRIBE will be sent out periodically to the

 

 

 

 

server. Then when DND/Call Forward features (Call Forward No Answer,

 

 

 

 

Unconditional Call Forward and Call Forward on Busy) are configured or changed

 

 

 

 

on the phone and the Broadsoft server side, those features will be synchronized on

 

 

 

 

the phone side and the Broadsoft server side.

 

 

 

 

 

 

 

 

 

 

 

 

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.

 

 

 

 

 

 

 

 

 

 

 

 

DTMF Payload Type

 

Sends DTMF using RFC2833. The default is 101.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Early Dial

 

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

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Dial Plan Prefix

 

Sets the prefix added to each dialed number.

 

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

GXP1400/1405 User Manual

Page 29 of 36

 

Firmware version: 1.0.1.83

Last Updated: 08/2011

Page 29 Page 31
Page 30
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Grandstream Networks GXP1400, GXP1405 Support SIP Instance ID, NAT Traversal, Subscribe for MWI, Registration, Feature Key
Page 29 Page 31

GXP1405, GXP1400 specifications

Grandstream Networks is a renowned provider of communication solutions, offering a wide range of innovative products for the VoIP market. Among their popular offerings are the GXP1400 and GXP1405 IP phones, designed specifically for users seeking affordable and reliable communication tools without compromising on quality.

The GXP1400 is an entry-level IP phone that boasts a sleek design and robust functionality. It features a 128x40 pixel LCD display, providing users with clear visibility for dialing, call management, and access to important information. Equipped with two lines, the GXP1400 allows users to manage multiple calls efficiently, making it suitable for small businesses or home office environments.

On the other hand, the GXP1405 is an enhanced version of the GXP1400, offering additional features for the same targeted market. The GXP1405 provides up to four lines, which is ideal for users who require more versatility in their calling options. Additionally, this model comes with a larger LCD screen with the same resolution, offering an extended view for enhanced usability.

Both models support advanced telephony technologies, including SIP (Session Initiation Protocol), enabling seamless connectivity with various VoIP services. They are compatible with a wide range of SIP-based services, ensuring that users can easily integrate them into their existing phone systems. Furthermore, the GXP1400 and GXP1405 support HD audio for crystal-clear voice quality, improving the overall communication experience.

In terms of features, both models include a built-in speakerphone, which allows hands-free calling, and programmable DSS keys for quick access to frequently dialed numbers. They also come with features like call hold, call transfer, and three-way conferencing, making them suitable for dynamic office environments.

Security is a priority for Grandstream, and both the GXP1400 and GXP1405 include encryption protocols to protect users' communications, ensuring privacy and data integrity.

Overall, the Grandstream Networks GXP1400 and GXP1405 are exceptional choices for users looking for cost-effective IP phones that offer essential features and reliable performance. Their combination of user-friendly functionality, advanced SIP support, HD audio quality, and robust security makes them an excellent addition to any office setup.
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