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...xtension.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp [default] include 4918 [4918] exten => 4918,1,Dial(SIP/4918,15,t) ; see "show application dial" for options and formats exten => 4918,2,Voicemail2(u4918) ; go to Voicemail2 if phone is "U"nanswered exten => 4918,102,Voicemail2(b4918) ; go to Voicemail2 if phone is "B"usy exten => 4918,103,Hangup ; and then hangup Thanks Deepak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/200312...

Hello! I've looked at the reference examples they are all for SIP. I have two X100p and a TDM400P. Can someone send me a working example so I can receive calls and make them. I'm stuck at first base. [I'm using standard phones - not SIP] Help please! Thanks, Bill Flood

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show