User manual

16
GSM VOIP Gateway Series
GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [e-mail]
COPYRIGHT ©2011 WITURA CORPORATION SDN BHD
This item displays the adjusted time according to the selected time zone. The
gateway receive time and date information from the server through the Network
Time Protocol and the time difference will l be automatically adjusted. For
example, the pacific standard time (PST) is GMT-8 and the pacific daylight time
(PDT) is GMT-7.
The time zone indicates the zone where the gateway is used. You need to enter
the correct time zone, so that the time of the caller ID and charging information
can be displayed correctly. The time server is the address of the server that
obtains the network time through the Internet. The default time server is
timekeeper.isi.edu.
This parameter is used to set the minimum interval of two DTMF signals. Packets
may be lost during the data transmission over the GSM. As a result, a DTMF may
be incorrectly identified as two or multiple identical DTMFs when detected by the
GSM VOIP Gateway. The problem of repeated code can be solved effectively
through the modification of the parameter.
This parameter value ranges from 60ms to 120ms and the default value is 80ms.
When the value of this parameter is increased properly, the repeated DTMF can
be avoided efficiently. However, the packet loss may also be caused.
If the service provider provides the automatic setting, you can select “Enable” to
start the automatic setting feature and enter the address of the server. If the
service provider does provide the automatic setting, you need to select “Disable”
to speed up the startup time of the GSM VOIP Gateway
3.3.2 Time Zone and Time Server
3.3.3 DTMF Minimum Detection Interval
3.3.4 Automatic Setting
61
GSM VOIP Gateway Series
GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [e-mail]
COPYRIGHT ©2011 WITURA CORPORATION SDN BHD
GSM VOIP Gateway sends a call request through the SIP number of the GSM VOIP
Gateway, the GSM VOIP Gateway will automatically add the number of the short
message sender to the PSTN Forwarding Number in Cal Forwarding (VoIP
Incoming Call , Forwarding to the PSTN Immediately);
In this mode, when the GSM VOIP Gateway receives the call from the SIP server,
the GSM VOIP Gateway will forward the call to the short message sending
equipment through the GSM network.
The SMS dial prefix is still valid in this mode;
The call request signaling in this mode is as follows:
SendingMessage to 192.168.2.1:5060:
INVITE
sip:8675588228822*8613800000000@192.168.2.1:5060;transport=udp
SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813
From: <sip:20001@192.168.2.1:5060>;user=phone;tag=65248630
To: <sip:8675588228822*8613902994477@192.168.2.1>
Call-ID: 117025903@192.168.2.237
CSeq: 2 INVITE
Contact: <sip:20001@192.168.2.237:5060>
Max-Forwards: 30
User-Agent:H
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER,
MESSAGE, INFO, SUBSCRIBE
Content-Type: application/sdp