Operation Manual

FIRMWARE 1.0.0.49 GXW42XX USER MANUAL PAGE 41 OF 54
Validate incoming
SIP message
Default is No. If set to yes all incoming SIP messages will be strictly validated according to
RFC rules. If message does not pass validation process, call will be rejected.
Check SIP user ID
for incoming
INVITE
Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will
be rejected. (no direct IP calling if Yes)
Allow Incoming
SIP Messages
from SIP Proxy
Only
Default is No. If incoming SIP message does not match with SIP Server, it will be rejected.
(no direct IP calling if Yes)
SIP T1 Timeout
T1 is an estimate of the round-trip time between the client and server transactions.
If the network latency is high, select larger value for more reliable usage.
SIP T2 Interval
Maximum retransmission interval for non-INVITE requests and INVITE responses.
DTMF Payload
Type
Sets the payload type for DTMF using RFC2833.
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.
Disable DTMF
Negotiation
Default is No. If set to yes, use above DTMF order without negotiation
Send Hook Flash
Event
Default is No. If set to yes, flash will be sent as a DTMF event.
Enable Call
Features
Default is Yes. (If Yes, call features using star codes will be supported locally)
Proxy Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.