[Asterisk-Users] ATA-488 FXO

[Anders Svensson]

Yes you can connect the fxo to a asterisk using sip

I have cut out a piece of the manual. It works for m

5.2.7 VoIP-to-PSTN Calls
To make a VoIP-to-PSTN call, users need to dial the FXO SIP account phone
number first. A ring tone is played once followed by a dial tone. At this
time, users can dial a PSTN telephone number or a mobile telephone number
then # (or wait for 4 seconds). The call will be established afterwards. If
no PSTN number is entered after the dial tone, HandyTone-488 will hang up
automatically in 10 seconds.
In the web configuration page, if the Route to PSTN field is configured, the
second stage dialing is eliminated. That is, after users dial the FXO SIP
account number, the PSTN number will be called automatically.
5.2.8 PSTN-to-VoIP Calls
To make a PSTN-to-VoIP call, PSTN callers need to originate a call to the
FXO port telephone number first. If no one answers the FXS phone after 4
(default value, can be configured) ring tones, a dial tone
is played. At this time, users can dial a VoIP telephone number then # (or
wait for 4 seconds). The call will be established afterwards. If no VoIP
number is entered after the dial tone, HandyTone-488 will hang up
automatically in 10 seconds.
In the web configuration page, if the Route to VoIp field is configured, the
second stage dialing is eliminated. That is, after users dial the FXO port
telephone number, the VoIP number will be called automatically.

Anders

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bill
Michaelson
Sent: den 8 november 2005 19:40
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] ATA-488 FXO

Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk 
to their Asterisk box (via SIP, of course)?

Is it possible to have such a beast operate reasonably?

If so, is it also possible to use the FXS port concurrently and 
independently?


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