| AG-188 FAQ |
 Go Back To Product Details |
| 1. How many SIP servers may AG-188 register simultaneously? |
| 2. How can I know the AG-188’s IP address? |
| 3. How to use AG-188’s Lifeline function? |
| 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? |
|
|
| 1. How many SIP servers may AG-188 register simultaneously? |
| AG-188 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. |
|
|
|
| 2. How can I know the AG-188’s IP address? |
| Pick up the handset and then dial “#*111#”, and the AG-188 will promote you its IP address. |
|
|
|
| 3. How to use AG-188’s Lifeline function? |
AG-188 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 AG-188, connect this accessory to AG-188’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:
----Add a new dial rule in the Dial-Peer setting: set the phone number to *T, and choose the Lifeline as the Call mode.
----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. |
|
|
|
| 4. Why the settings vanish after reboot? |
| Please go to Config Manage > Save Config to save your setting always. |
|
|
|
| 5. How to use the dial rule? |
AG-188 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 AG-188?”).
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 AG-188 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 AG-188’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. |
|
|
|
| 6. How to use speed-dial function? |
There are 9 speed dial keys in the AT-320 panel, Usage:
Set speed dial number: press the speed key and enter the speed dial number and then press Menu/OK key to save the setting.
Pick up the handset and press the speed dial key to dial the pre-define number. |
|
|
|
| 7. How to configure digital map? |
Digital Map is based on some rules to judge when user end their dialing and send the number to the server. AT-530 supports 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 + Length: f the Prefix and Length match, the call will be sent. |
|
|
|
| 8. How to use Call Forward, Call Transfer and 3-way Conference calls? |
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 call this gateway will be forward to the destination number.
Call Transfer:
Check the Enable Call Transfer.
If A is the AT-530 user, and B calls and talking with A through VoIP. A can press FWD button 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 AG-188 user, and B calls and talking with A through VoIP. A can press FWD button to hold the call with B, then enter * and then enter C’s number to talk with C, and then press FWD button again to make 3-way conference calls. |
|
