Can you please show your dialplan? with and withou the www? Please only the dialplan that doesnt work. Also include /etc/asterisk/zapata.conf
On 8/3/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
CF, Adding www after Dial doesn't solve the trouble. I think we are talking the same but I don't express correctly. Did you saw my dialplan? I don't think I would have to add r. Yes, I have installed a 4 FXO Card, with fxsks signalling. What I mean is I understand FXO doesn't give the tone, but Panasonic. The cadence of ringing on Panasonic is a little different to the PSTN's cadence, FXO detects properly PSTN cadence when a call goes to or come from PSTN, and when a call goes from Panasonic to Asterisk, but doesn't make same job with a call going from Asterisk to Panasonic. The Sip phone behind Asterisk make the call, keeps ringing until Panasonic extension answer…. It's normal, but even Panasonic user pick up the phone the Sip phone keeps ringing… to user on Sip pone, nobody answer his call, however user on Panasonic pick up and doesn't hear anything. I'm going crazy… _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
