Re: [asterisk-users] permit -- deny not working
Hi, It was already answered... You must always deny before allow. Regards, Carlos -- _ -- 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
Re: [asterisk-users] Answering machine answers after pickup a phone.
John is absolutly right. You should connect your phones to an FXS port. Otherwise you can't do what you wan't. Regards, Carlos M Cruz -- _ -- 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
Re: [asterisk-users] Connect asterisk to normal telephone PBX
Hi, Did you created your normal Inbound and Outbound routes in freepbx? For use with your zap channels? You'll problably have to change your routes on your pbx too... Regards, Carlos M Cruz 2011/7/28 michael k mich...@inapp.com Hello All, I don't even know the relevancy of my question. Please answer me if my question have some sense. I have recently implemented an asterisk server with freepbx. I have created 100 extentions and i can make successful calls between extensions from anywhere. But my office have three different land-line numbers and three of them are terminating into an internal PBX ( normal matrix telephone PBX) with more than 60 extensions. This internal PBX is the live PBX where we can call local, STD and ISD from extensions. At present i have some practical difficulties to configure telephone lines at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the normal telephone PBX. I have installed 1 port x100p FXO card in my asterisk PBX and detected by my freepbx. Then i removed my normal telephone extension cable from phone and connected to the FXO port of my asterisk PBX. Ultimately my intention is that 1) if somebody call to my normal telephone extension, that should reach to my asterisk server, and asterisk server should send this call to my asterisk extension. 2) if i am calling from my asterisk extension, call should go to the normal telephone PBX via FXO card in my asterisk server and ultimately the call should send outside via the telephone PBX. Is my approach is correct ? If it is wrong please somebody assist me to connect my asterisk PBX to normal telephone PBX. Michael.K -- _ -- 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 -- _ -- 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
Re: [asterisk-users] fail-over server
Hi, Thats very simple. Use sip realtime registration with mysql and heartbit to control switiching. Regards, Carlos M Cruz Em 2011/02/08 16:07, Vieri rentor...@yahoo.com escreveu: Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually on-line so calls can be routed to them. How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri -- _ -- 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 -- _ -- 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
