On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes <[email protected]> wrote: >I think this is a very common situation, so I'm not really sure what >your problem is. Perhaps it's because I don't use an internal card, >but in my situation it works just fine. I dial a number on my SIP >phone, Asterisk goes through the dialplan, and puts the call out via >the SPA3102. In my ear I hear ringing sounds, busy, wrong number or >someone talking to me just like if I had connected a "normal" phone to >the PSTN line.
I haven't done this yet, and was looking for information. I was under the (apparently false) impression that Asterisk/Dahdi didn't connect the two legs until the callee had gone off-hook. So it looks like there's really no issue in connecting a remote SIP client with a PSTN number through an FXO port + ADSL VoIP modem to take advantage of free phone calls. Thanks everyone. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
