[GrandStream HT-488] How to configure?

Hello

After a bit of fiddling, I got to the web interface and can more or less successfully receive a call through an Asterisk server using this SIP gateway.

I have the following questions, though:

  1. HT488 doesn't detect that the remote end has hung up: What should I use for France as regional settings in the FXO Port section? Here are the default settings that show up in the web interface: PSTN AC Termination: 320 Ohm + (1050 Ohm || 230 nF)) PSTN Disconnect Tone: Frequency: f1 480 f2 620 PSTN Disconnect Tone Cadence: All 0's PSTN Silence Timeout : 60
  2. No CID is returned, even though it works OK if I plug a caller ID-capable modem on the phone line. HT488 doesn't handle CID, or is it linked to regional settings above?
  3. How to secure access to PSTN line so that unauthorized users from the Net don't use my PSTN line?

Thank you.

Reply to
Vincent Delporte
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I don't live in France but as far as I know, the busy tone in France is 440 Hz with 0.5/0.5 sec for on/off rate. Try to set your equipment accordingly.

If it is not the busy tone what is played by your exchange when the remote end has hung up, and nobody knows what is played actually, (=even the telco guys can't help you) than the actual tone is to be analysed, and you should configure the HT488 accordingly.

Reply to
ldemsp

Thanks, I'll try to change the settings.

What about caller ID from the FXO port? Back in Oct 2005, someone said that CID is only available for the FXS port, but support for FXO might be added in future firmware... but the unit is already up to date.

If no CID, I can't use that equipment :-/

Reply to
Vincent Delporte

I don't even understand the question. CID can only work if it is received from the FXO port, and it is relayed to either the FXS port, or forwarded as a source ID to the VoIP termination target. Or the alternate direction, when the call can come from the VoiP leg, and the CID is shown on the FXS port again.

I really don't understand what are you waiting for. You will never have the equipment transfer CID from the FXS or from VoIP towards the FXO side. It is nonsense: the switch (PBX, or local exchange) will give you the CID according to your call number, not by the info it gets from the station.

It expects and understands hook on/hook off/DTMF/hook-flash from the station side, and nothing else.

Reply to
ldemsp

If I hook up the HT to the network and call in from the PSTN, it doesn't forward CID to the Asterisk server. There is CID-related items in the FXS section, but nothing in the FXO section. I'll see if the former is enough to have the unit send CID information to Asterisk, or if I have to change regional settings.

Thanks.

Reply to
Vincent Delporte

I see. You are asking if the equipment can understand the CID from the FXO interface in order to translate it to FXS interface (and to the network) or it just rings thru.

Theoretically it understands CID, and should forward it with the call. In France you should use ETSI-FSK type setting. Check if you have unset block caller ID. *31 is the unsetting from the phone, and you should set the "Send Anonymous" option to No.

Reply to
ldemsp

No. I don't use the FXS port at all (no handset connected to it). I just use the FXO port and the WAN port so that the HT488 is used as an SIP gateway for an Asterisk server.

CID works on this line, but I'll see if making regional changes in the FXS section has an effect on the FXO section.

Thanks.

Reply to
Vincent Delporte

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