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Grandstream GXP1610 Administration Guide User Manual

Page 29

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GXP1610/GXP1620/GXP1625/GXP1628

Administration Guide

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[345]xx: 3-digit number with the leading digit of 3, 4 or 5;
[6-9]xx: 3-digit number with the leading digit from 6 to 9.

Alert Info text
Users could configure the matching rule as certain text (e.g., priority) and
select the custom ring tone mapped to it. The custom ring tone will be
used if the phone receives SIP INVITE with Alert-Info header in the
following format:
Alert-Info: ; info=priority

Selects the distinctive ring tone for thematching rule. When the incoming
caller ID or Alert Info matches the rule, the phone will ring with the selected
ring.

Ring Timeout

Defines the timeout (in seconds) for the rings on no answer. The default
setting is 60 seconds.

Send Anonymous

If set to "Yes", the "From" header in outgoing INVITE messages will be set to
anonymous, essentially blocking the Caller ID to be displayed.

Anonymous Call
Rejection

If set to "Yes", anonymous calls will be rejected. The default setting is "No".

Auto Answer

If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep.

Allow Auto Answer by
Call-Info

If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep, based on the SIP info
header sent from the server/proxy. The default setting is "No".

Custom Call-Info for
Auto Answer

Used in addition to match the contents of the info parameter in the Call-Info
header for auto answer.

Refer-To Use Target
Contact

If set to "Yes", the "Refer-To" header uses the transferred target's Contact
header information for attended transfer. The default setting is "No".

Transfer on Conference
Hangup

Defines whether or not the call is transferred to the other party if the initiator of
the conference hangs up. The default setting is"No".

No Key Entry
Timeout(s)

Defines the timeout (in seconds) for no key entry. If no key is pressed after the
timeout, the digits will be sent out. The default value is 4 seconds.

Use # as Dial Key

Allows users to configure the "#" key as the "Send" key. If set to "Yes", the "#"
key will immediately dial out the input digits. In this case, this key is
essentially equivalent to the "Send" key. If set to "No", the "#" key is included
as part of the dialing string.

DND Call Feature On

Defines the feature code to turn on DND.

DND Call Feature Off

Defines the feature code to turn off DND.

SETTINGS PAGE DEFINATIONS

Settings -> General Settings

Local RTP Port

This parameter defines the local RTP port used to listen and transmit. It is the
base RTP port for channel 0. When configured, channel 0 will use this port
_value for RTP; channel 1 will use port_value+2 for RTP. Local RTP port
ranges from 1024 to 65400 and must be even. The default value is 5004.

Use Random Port

When set to "Yes", this parameter will force random generation of both the
local SIP and RTP ports. This is usually necessary when multiple phones are
behind the same full cone NAT. The default setting is "Yes" (This parameter
must be set to "No" for Direct IP Calling to work).

Keep-alive Interval

Specifies how often the phone sends a blank UDP packet to the SIP server in