User's Manual
Table Of Contents
P a g e | 53
GXW42xx User Manual
Version 1.0.5.43
Check SIP User ID for
Incoming INVITE
Default is No. 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. Direct IP calling will also be disabled.
Accept Incoming SIP from
Proxy Only
Default is No. 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. Direct IP calling will also be disabled.
Authenticate Incoming INVITE
Default is No. If set to Yes, the phone will challenge the incoming
INVITE for authentication with SIP 401 Unauthorized response.
Authenticate server certificate
domain
Default is No. If this is set to Yes, device will check the server TLS
certificate to ensure that the Common Name matches the configured
SIP server
Authenticate server certificate
chain
Default is No. If this is set to Yes, device will check the server TLS
certificate to ensure that it is authorized by a known Certificate Authority
Fax Settings
Fax Mode
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec
PCMU/PCMA)
Fax Tone Detection Mode
Default is Callee. This decides whether Caller or Callee sends out the
re-INVITE for T.38 or Fax Pass Through.
Send Re-INVITE After Fax
Completion
Default is No, If set to “Yes”, device will send an INVITE with audio
vocoders upon competition of Fax to continue session in audio only.
Send Re-INVITE After Fax
Tone
If set to “Yes”, device will send a Re-INVITE after Fax tone is detected,
disabling will only work under Broadsoft feature.
Enable Silence Detection for
Fax Disconnect
For fax machines that do not send a Disconnect when fax is done. This
option Enables/Disables the detection of silence in order to know the
fax has finished. The silence period is non-configurable and fixed to 7
seconds.
Audio Settings
Preferred DTMF method (in
listed order)
The GXW42XX supports up to 3 different DTMF methods including in-
audio, via RTP (RFC2833) and via Sip Info. The user can configure
DTMF method in a priority list.