Grandstream GXW410x User Manual - Manual (Page 22)

Grandstream Networks, Inc. GXW410x User Manual Page 22 of 32
Firmware Version 1.4.1.5 Last Updated: 5/2014
customer need an accurate indication of calls through a network.
Silence Timeout
Terminate call after long silence detected. Default is 60 seconds, max 65536.
Incoming Call Timeout Default value is 6 seconds. Incoming call will stop ringing when not picked up given a
specific period of time.
AC Termination Impedance Selects the impedance of the analog line connected to the FXO port on the
GXW410x. Here is some basic information which may be helpful for initial
configuration:
600 Ohm
– North America;
270 Ohm + (750 Ohm || 150 nF) -- Most of Europe
220 Ohm + (820 Ohm || 120 nF)
– Australia, New Zealand
220 Ohm + (820 Ohm || 115 nF) – Austria, Bulgaria, Germany, Slovakia, South Africa
370 Ohm + (620 Ohm || 310 nF) – UK., India
If this parameter is not configured properly you may experience echo or static in the
line. Please refer to Table 9 (FXO Lines Test Tab Definition). This tool will run an
automated test to determine the correct impedance value to match your lines
Number of Rings Before
Pickup Default is 4. This is the number of rings the gateway will wait to send the call
to the VOIP side in case the Caller ID has yet to be detected. If there's CID
information the call will be sent right away. If your lines don't have the CID
service set this to 1.
Caller ID Scheme The GXW410x supports 5 different types of schemes:
1. Bellcore (US standard)
2. ETSI-FSK during ringing
3. ETSI-FSK prior to ringing with DTAS
4. ETSI-FSK prior to ringing with LR
5. ETSI-FSK prior to ringing with PR
6. ETSI-DTMF during ringing
7. ETSI-DTMF prior to ringing with DTAS
8. ETSI-DTMF prior to ringing with PR
9. ETSI-DTMF prior to ringing with PR
10. SIN 227 - BT
11. NTT (Japanese standard)
Please check with your PSTN service provider (or traditional PBX specs) for which
caller ID scheme they/it support. If you are not sure about which to use please refer
to Table 9 (FXO Lines Test Tab Definition). This tool will run an automated test to
determine the proper Caller ID Scheme so the gateway can properly detect the Caller
ID.
Similarly to the cases explained above we can specify a caller ID scheme for each
channel independently.
Caller ID Transport type Default is “relay via From header”. You may also select :
“relay via P_Asserted_Identity header”
“Disable” : Caller ID feature will be disabled.
“Send anonymous” : All calls forwarded to VOIP end will be sent as anonymous.
DIALING
Wait for Dial-tone Default is Yes. When set to Yes, the gateway will recognize dial-tone from the Central
Office (CO) before it completes call.
If you can’t make an outbound call, set this is
No.
Stage Method Syntax - ch1-8:1; {all channels 1 to 8 are set to value 1 or 2}
Stage method can be set to either 1 or 2.
Set this parameter to 1 if you need to make a direct PSTN call from a VOIP endpoint.
When you set it to 2, you will first dial one of the VOIP channel accounts from the
VOIP endpoint, this will result in getting a PSTN line dial-tone to then dial out the
destination PSTN number.