User's Manual

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GXP17xx Administration Guide
select UAS to use the phone as the refresher. The default setting is "UAC".
Force INVITE
The Session Timer can be refreshed using the INVITE method or the
UPDATE method. Select "Yes" to use the INVITE method to refresh the
session timer. The default setting is "No".
Account x  SIP Settings  Security Settings
Check Domain
Certificates
Choose whether the domain certificates will be checked or not when
TLS/TCP is used for SIP Transport. The default setting is "No".
Validate Certificate
Chain
Validate certification chain when TCP/TLS is configured. The default setting
is “No”.
Validate Incoming
Messages
Choose whether the incoming messages will be validated or not. The default
setting is "No".
Check SIP User ID for
incoming INVITE
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
Accept Incoming SIP
from Proxy Only
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
Authenticate Incoming
INVITE
If set to "Yes", the phone will challenge the incoming INVITE for
authentication with SIP 401 Unauthorized response. The default setting is
"No".
Account x  Audio Settings
Send DTMF
This parameter specifies the mechanism to transmit DTMF digits. There are 3
supported modes: in audio which means DTMF is combined in the audio
signal (not very reliable with low-bit-rate codecs), via RTP (RFC2833), or via
SIP INFO.
• In audio, which means DTMF is combined in the audio signal (not very
reliable with low-bit-rate codecs);
• RFC2833, which means to specify DTMF with RTP packet. Users could
know the packet is DTMF in the RTP header as well as the type of DTMF;
• SIP INFO, which use SIP info to carry DTMF. The defect of this mode is that
it's easily to cause desynchronized of DTMF and media packet for the reason
the SIP and RTP are transmitted respectively. The default setting is
"RFC2833".
DTMF Payload Type
This parameter sets the payload type for DTMF using RFC2833. Default is
101. The valid range is from 96 to 127.
Preferred Vocoder
Multiple vocoder types are supported on the phone, the vocoders in the list is
a higher preference. Users can configure vocoders in a preference list that is
included with the same preference order in SDP message.