Re: [Asterisk-Users] ATA-488 FXO
On Nov 8, 2005, at 10:39 AM, Bill Michaelson wrote: 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? I am a newbie, and am also trying to get this device to work. I have it working well enough that I can manually dial the extension of the FXO and get my PSTN dial tone via VOIP. This seems to work reliably and I have been testing it with softphones for several days. Also I have the FXS working ok so yes you can use them both independently. My current struggle is to figure out how make an appropriate dialplan, so that dialing a regular old 7 digit (or 10) phone number will route the call through the FXO. This appears to be a relevant thread, but I am still deciphering it. http://voxilla.com/PNphpBB2-viewtopic-t-4555.html Thanks for posting this topic ! I was too chicken. Marty ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-488 FXO
Bill, check the following thread to see if you can find some answers: http://lists.digium.com/pipermail/asterisk-users/2005-August/123548.html - Original Message - From: Bill Michaelson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, November 08, 2005 8:39 PM 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? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-488 FXO
I tried this unsuccessfully with an early (pre-release) version of the 488 firmware. I haven't tried it recently though. I'll have a play later in the week and let you know... == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == Bill Michaelson wrote: 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? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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)? I tried 2 of them at a client's site here in the UK. Is it possible to have such a beast operate reasonably? I was unsuccessful. The device would answer the line quite happily (and remarkably echo-free) for a few hours, after which it would refuse to answer more than about 50% of incoming calls on the line. A reset would fix the unit, but every 3-4 hours was rather impractical (and shouldn't be necessary). If so, is it also possible to use the FXS port concurrently and independently? I didn't have any problem with the FXS port at all - this worked perfectly and independently of the FXO port (i.e. even when the FXO port was being tempramental, I never had any problems with the DECT phone connected to the FXS port). Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA-488 FXO
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Michaelson Sent: den 8 november 2005 19:40 To: asterisk-users@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? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
