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| 1. |
How
many SIP servers may SG-183 register simultaneously?
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| 2. |
How
can I know the SG-183¡¯s IP address?
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| 3. |
How to use SG-183¡¯s Lifeline function?
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| 4. |
Why the settings vanish after reboot? |
| 5. |
How
to use the dial rule? |
| 6. |
How
to use speed-dial function? |
| 7. |
How
to configure digital map? |
| 8. |
How
to use Call Forward, Call Transfer and 3-way Conference calls?
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| 1. |
How
many SIP servers may SG-183 register simultaneously? |
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SG-183
is able to register two SIP servers simultaneously and one redundancy
server, you can configure the dial rule to route the call between
the sip servers. Please see ¡°How to use the dial rule?¡±
for detail. |
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| 2. |
How
can I know the SG-183¡¯s IP address?
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Pick
up the handset and then dial ¡°#*111#¡±,
and the SG-183 will promote you its IP address. |
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| 3. |
How
to use SG-183¡¯s Lifeline function? |
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SG-183
supports Lifeline function, you can use the same handset to place
PSTN and VoIP calls. First, you need to set up the Lifeline with
the accessory send with the SG-183, connect this accessory to SG-183¡¯s
FXS port, and then connect the handset to the accessory¡¯s
phone port, connect the landline to the accessory¡¯s
line port. You can receive PSNT and VoIP calls simply with configuration.
To place the PSTN call, you need to set up as follow:
a. Add a new dial rule in the Dial-Peer setting: set the phone number
to *T, and choose the Lifeline as the Call mode.
b. Add new Digital map item in the Advance ?Digital Map: set Prefix
Number to and *, and the length to 1.
Then when you want to place a PSTN calls, you can first press *
to switch to the PSTN line and then place your call as you normal
do. |
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| 4. |
Why
the settings vanish after reboot?
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Please
go to Config Manage > Save Config to save your setting always. |
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| 5. |
How
to use the dial rule? |
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SG-183
provide flexible dial rule, with different dial-rule configure,
user can easily implement the following function:
----Replace, delete or add prefix of the dial number.
----Make direct IP to IP call
----Place the call to different SIP server according the prefix.
----Make PSTN calls use Lifeline function (Please refer ¡°How
can use the Lifeline function of SG-183?¡±).
You can click ¡°Add¡± to add a new dial
rule. Below is the detail setting of the dial-rule:
Phone Number: The Number suit for this dial rule, you can the
number as full match or prefix match. Full match means that if
the number use dials is the completely same as this number, the
call will use this dial-rule. Prefix match means that if prefix
of the number that the user dials is the same as the prefix, the
call will use this dial-rule, to distinguish from the full match
case, you need to add ¡°T¡± after the prefix
number in the phone number setting.
Call Mode: support SIP and Lifeline£SIP means the call will
use sip protocol£Lifeline means the call will use the PSTN
line.
Destination (optional): call destination, can be IP or domain.
Default is 0.0.0.0, in this case the call will be routed to the
Public SIP server. If you set the destination to 255.255.255.255,
then the call will be routed to the private SIP server. Also you
can key other address here to make direct IP calls
Port (optional): Configure the port of the destination, default
is 5060
Alias (optional):Set up the Alias. We support four Alias as below.
Alias need to co-work with the Del Length:
add:xxx, add prefix to the phone number, can set to reduce the
dial length.
all: xxx, replace the phone number with the xxx, can use as speed
dial function.
del, delete the first N numbers. N is set in the Del Length
rep:xxx£replace the first N numbers. N is set in the Del
Length. For Example: Use wants to place a call 8610-62281493,
then you can set the phone number in the dial rule as 010T, and
set the Alias as rep:8610, and set the Del Length to 3. Then all
calls begin with 010 will be changed to 8610 xxxxxxxx.
Suffix (optional):Configure suffix, show no suffix if not set
Instance:
2T rule: If the call starts with 2, the first 2 will be deleted,
and the rest number will be sent to private server.
3T rule: If the call starts with 3, the first 3 will be deleted,
and the rest number with be sent to public server.
123 rule: Dial 123 and will send 8675583018049 to your server.
Used as speed dial function.
0T rule: If the calls is begin with 0, the first 0 will be replace
by 86. Means that if you dial 075583018049 and SG-183 will send
8675583018049 to your server.
*T rule: Dial the * and the line with switch to PSTN. Note that
you need to set another rule ¡°Prefix Number: *; Length:
1¡± in the Digital Map. (Refer ¡°How to
use SG-183¡¯s Lifeline Function?¡±)
179 rule: when you dial 179 , the call with send to 192.168.1.179,
suit for LAN application without set up a sip server. |
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| 6. |
How
to use speed-dial function? |
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Please
refer to ¡°How to use dial rule?¡±. |
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| 7. |
How
to configure digital map? |
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Digit
map is a set of rules to determine when the user has finished
dialing.
SG-183 support below digital map:
Digital Map is based on some rules to judge when user end their
dialing and send the number to the server. SG-183 support following
digital map:
----End With ¡°#¡±: Use # as the end of
dialing.
----Fixed
Length: When the length of the dialing match, the call will be
sent.
----Timeout: Specify the timeout of the last dial digit. The call
will be sent after timeout
----Prefix: User define digital map:
[ ] represents the range of digit, can be a range such as [1-4],
or use comma such as [1,3,5], or use a list such as [234]
x represents any one digit between 0~9
Tn represents the last digit timeout. n represents the time from
0~9 second, it is necessary. Tn must be the last two digit in
the entry. If Tn is not included in the entry, we use T0 as default,
it means system will sent the number immediately if the number
matches the entry.
Example:
[1-8]xxx All number from 1000 to 89999 will be sent immediately.
9xxxxxxx 8 digits numbers begin with 9 will be sent immediately.
911 Number 911 will be sent will be immediately
99xT4 3 digits numbers begin with 99 with be sent after four seconds. |
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| 8. |
How
to use Call Forward, Call Transfer and 3-way Conference calls? |
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User
may set up the configuration in the Call Service page to use these
value add service.
Call
Forward:
----Forward when busy: select Busy in the Call Forward Field,
and Key in the destination phone number in the Forward Number.
If some one calls you when you having a call, the caller will
be forwarded to the destination number.
----Forward no answer: Select No Answer in the Call Forward Field,
and Key in the destination phone number in the Forward Number,
fill the time in the No Answer Time. If some one calls you and
no one answer the caller during the No Answer Time, the call will
be forward to the destination number.
----Forward Always: Select Always in the Call Forward Field, and
Key in the destination phone number in the Forward Number, then
any one calls this gateway will be forward to the destination
number.
Call
Transfer:
Check
the Enable Call Transfer.
If A is the SG-183 user, and B calls and talking with A through
VoIP. A can press the Hook-Flash to hold the call with B, and
then enter C¡¯s number. B will be transferred to C
and can talk with C.
3-Way
Conference Calls
Check
Enable Three Way Call
Assume A is the SG-183 user, and B calls and talking with A through
VoIP. A can press Hook-Flash to hold the call with B, then enter
* and then enter C¡¯s number to talk with C, and then
press Hook-Flash again to make 3-way conference calls. |
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