Grandstream Networks DP715 User Manual page 48

message
Check SIP User ID for
incoming INVITE
Authenticate incoming
INVITE
Allow Incoming SIP
Messages
from SIP Proxy Only
Caller ID Display
Use Privacy Header
Use P-Preferred-Identity
Header
SIP T1 Timeout
SIP T2 Interval
DTMF Payload Type
Preferred DTMF method
Disable DTMF
Negotiation
Send Hook Flash Event
Enable Call Features
Proxy-Require
Use NAT IP
Use SIP User-Agent
Header
Do Not Escape '#' as
%23 in SIP URI
Disable Multiple m line
in SDP
Ring Timeout
Hunting Group Ring
Timeout
Hunting Group Type
Delayed Call Forward
Wait Time
No Key Entry Timeout
Firmware version 1.0.0.31
RFC rules. If message will not pass validation process, call will be rejected.
Default is No. Check the incoming SIP User ID in Request URI. If they don't match, the
call will be rejected. If this option is enabled, the device will not be able to make direct IP
calls.
After enable, unit will chanllenge any incoming INVITE with the SIP account password.
Default is No. Check the incoming SIP messages. If they don't come from the SIP proxy,
they will be rejected. If this option is enabled, the device will not be able to make direct IP
calls.
Default is Auto.
Caller ID will be found in order:
1) Use the caller id in P-Asserted Identity Header, if not found:
2) Use the caller id in Remote-Party-ID Header, if not found:
3) Use the caller id in "From" header.
If set to Default, it will only add Privacy header when special feature is not Telkom SA or
CBCOM.
If set to Default, it will only add PPI header when special feature is not Telkom SA or
CBCOM.
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. Default is 0.5
Sec.
Maximum retransmission interval for non-INVITE requests and INVITE responses. Default
is 4 Sec.
Sets the payload type for DTMF using RFC2833. Default is 101.
The DP715 supports up to 3 different DTMF methods including in-audio, via RTP
(RFC2833) and via Sip Info using SIP INFO messages. The user can configure DTMF
method in a priority list.
Default is No. If set to yes, use above DTMF order without negotiation
Default is No. If set to yes, flash will be sent as DTMF event.
Default is Yes. (If Yes, call features using star codes will be supported locally)
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
NAT IP address used in SIP/SDP message. Default is blank.
Used to replace the SIP User-Agent Header. Default is NO.
If set to "Yes", device will use '#' instead of %23 in the send URI.
Default is No. If set to Yes, device will send only one m line in SDP, regardless how many
m field in the incoming SDP.
Default value is 60 Sec. Incoming call will stop ringing when not picked up given a specific
period of time.
Default value is 20 Sec. If call is not answered within this designated time period, the
callwill be forwarded to the next member of a Hunt Group.
Linear, parallel and Shared line.

Linear style will sort the call to the lowest-numbered available line; this is also
called "serial hunting".

Parallel Style will ring all the handsets on incoming calls.

Shared line Style will ring all the handsets on incoming calls, any handset
member of the same hunt group that try to dial out while there is an ongoing call
it will be automatically conference in the ongoing call.
Default value is 20 seconds. In case this feature activated using * codes (*92 code), the
call will be forwarded after this preconfigured amount of time.
Default is 4 seconds. Dialing process is completed and outgoing call is initiated if no key
entry occurs during this preconfigured interval.
DP715/DP710 User Manual Page 46 of 56