Grandstream Networks, Inc.
DP715-US/DP710-US User Manual
Page
35 of 43
Firmware 0.0.0.8
Last Updated: 03/2012
SIP Registration
Controls whether the DP715 needs to send REGISTER messages to the proxy server.
The default setting is
Yes
.
Unregister on Reboot
Default is
No
. If set to Yes, the SIP user’s registration information will be cleared on
reboot.
Outgoing Call without
Registration
Default is
No
. If set to “Yes,” user can place outgoing calls even when not registered (if
allowed by Internet Telephone Service Provider) but is unable to receive incoming calls.
Register Expiration
This parameter allows the user to specify the time frequency (in minutes) the DP715
refreshes its registration with the specified registrar. The default interval is
60
minutes (or
1 hour). The maximum interval is 65535 minutes (about 45 days).
Registration Retry Wait
Time
Retry registration if the process failed. Default is
20
seconds.
Local SIP port
Defines the local SIP port the DP715 will listen and transmit.
Local RTP port
Defines the local RTP-RTCP port pair the DP715 will listen and transmit. It is the base
RTP port for channel 0. When configured,
channel 0
uses this port _value for RTP and the por1 for its RTCP
Use Random Port
Default is
No
. This parameter forces the random generation of both the local SIP and RTP
ports when set to Yes. This is usually necessary when multiple DP715 are behind the
same NAT.
Refer to Use Target
Contact
Default is
No
. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
transferred target’s Contact header information.
Transfer on Conference
Hang up
Default is
No
. In which case if the conference originator hangs up the conference will be
terminated. When option YES is chosen,
originator will transfer other parties to each
other so that B and C can choose to either continue the conversation or hang up.
Enable Ring-Transfer
Default is
No
, this will create a Semi-Attendant Transfer. When set to Yes, device can
transfer the call upon receiving ring back tone or SIP message 180.
Disable Bellcore Style 3-
Way Conference
Default is
No
. you can make a Conference by pressing ‘Flash’ key. If set to
Yes
, you need
to dial *23 + second callee number.
Remove OBP from
Route Header
Default is
No
. When option YES is chosen, the Out Bound Proxy will be removed from
Route header.
Support SIP Instance ID
Default is
Yes
. If set to Yes, the contact header in REGISTER request will contain SIP
Instance ID as defined in IETF SIP Outbound draft.
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 will not pass validation process, call will be rejected.
Check SIP User ID for
incoming INVITE
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.
Allow Incoming SIP
Messages from SIP
Proxy Only
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.
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. Default is
0.5
Sec.
SIP T2 Interval
Maximum retransmission interval for non-INVITE requests and INVITE responses. Default
is
4 Sec.
DTMF Payload Type
Sets the payload type for DTMF using RFC2833. Default is
101.
Preferred DTMF method
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.
Disable DTMF
Negotiation
Default is
No
. If set to yes, use above DTMF order without negotiation
Send Flash Event
Default is
No.
If set to yes, flash will be sent as DTMF event.
Enable Call Features
Default is
Yes
. (If Yes, call features using star codes will be supported locally)
