similar to: How do I configure asterisk?

Hi: I've configured an asterisk server with A102d sangoma's card and the E1 link.I want to dial outside number using the E1 Link.How can I dial a phone number?Is this true? exten => 123,1,Dial(ZAP/1/phone number) I'd appreciate any help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I configured an asterisk server with 2.4G cpu and 1G ram for conference call service but when 5 peoples about 20 minutes were talking together suddenly asterisk was disconnected.May it has happened because low cpu or ram?I saw var/log/asterisk/messages file but everything was going well apparently,asterisk was disconnected suddenly.What 's  your idea?Please  guide me.

Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: Please every that work with A102d say how about is it?Is it really difficult to install card for me new in asterisk? Best regards. --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

Hi: I want to have conference call(meetme) service with asterisk and 30 users.Now do I use 1E1 or 30 analog lines with due attention to high price of E1 line?And which interface card do I use? Best regards. --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was

Hi: I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference? Best regards. --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I want to have conference call service and I have A102d sangoma's card so I install asterisk 1.2.x or 1.4.x? Best regards. --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: Can I set 1 extension(i.e.6000) in extensions.conf file for several room for conference call service ? Or for every room I should set 1 special extension. Regards. --------------------------------- Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi: I want to receive a fax with an E1 link connected to A102d card from a fax machine,but after dialling the phone number, it connects then will be busy.In fact asterisk can't detect the fax.These are zapata.conf, extensions.conf filels and debug in console: extensions.conf: [from-pstn] exten => 9711315,1,Answer() exten => 9711315,2,Wait(10) exten =>

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi: I want to use asterisk1.2 but I don't know which version of asterisk1.2 and zaptel1.2 is best.Please offer me one version of asterisk and zaptel and libpri.How about asterisk1.2.24 and zaptel1.2.20.1 and libpri1.2.5?And do they work togather well? Best regards. --------------------------------- Pinpoint customers who are looking for what you sell. -------------- next part

Hi! I need to use text to speech, what is the best application? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/bc69eb9d/attachment.htm

Hello, I want to create php rate script and I'm using Deadagi. But I allways get billsec 0 , or nothing. Can you help me to solve this problem... My extension.conf: exten => _123,1,DeadAgi(rate.php) exten => _123,2,hangup And my simple test php script rate.php #!/usr/local/bin/php -q <?php include_once (dirname(__FILE__)."/phpagi.php"); $AGI = new AGI();

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get

Hello, Let's say I have a Database of my clients about 50 clients, I want to announce a new product or service to them, can asterisk do it for me? It is something like a appointment reminder for doctors. I want to know is there any software for this or I should Write a program for it using AGI or ruby on Rails. Thank you all, AA -------------- next part -------------- An HTML attachment

Hello, I feel like I understand how the dial plan works pretty well with one exception. It seems like queues are using the stdexen macro to ring the agents/extensions. Is this normal? Is there anyway to configure this differently? I realize this is a newbie question, but I have searched google/archives and haven't been able to find the answer. Thanks, Elliot --------------

Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using

Hi, Can anyone tell me why the below dialplan doesn't filter off dialed numbers for 01793520158, and jump to "local",priority1 If I change it to : exten => 01793520158,1,Goto(local,${EXTEN:-3},1) .... then it works fine (but that's too specific)... exten => _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1) exten =>