similar to: Calling Clients or Tele Marketing

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, 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 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

Hi: Which one is better and easier for configure asterisk,directly or by GUI ? I'd appreciate any idea. Regards. --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 =>

I have several installations of Asterisk (several versions) where we have our own web interface that uses Mysql and Realtime. When we do modifications to Mysql we use a Manager connection in order to reload the configuration (we use Realtime static for extensions) sometimes Asterisk will crash. Not every time and not every X times we reload. Sometimes it takes ten reloads and other just one

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Thanks. -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu

Hi, I get the following after a call has finished: ERROR[6862]:mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away Does this error message only appear when asterisk makes a new connection to mysql, because the old connection was stale (and dropped) ? If so, is there a way to get asterisk to stop reporting this as an error seeing it seems to write the CDR to

Hi, BRIEF RESUME: Is there any other way to obtain the same result but being easier to configure?? Thanks! EXTENDED RESUME: i've configured a, rather difficult, macro that even for me without being documented is difficult. I ask for the help of the experts to know if the functionality it apports can be achieved better in another way. What i'm trying is to enable call a channel (e.g.

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

Hi all, I want to limit the outgoing trunk to certain extensions, so for example 6 extensions can call long distance, but 4 other extensions are not allowed to do so. How can I do it in FreePBX specially? Looking forward to hear from you soon, Seysan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? Best Regards, Seysan -------------- next part --------------

Just hit Slashdot: http://slashdot.org/article.pl?sid=06/06/08/1725215&threshold=1 http://money.cnn.com/2006/06/07/news/companies/pluggedin_fortune/index.htm >From TFA: "But what's really cool about what will happen in Cardiff - and eventually the rest of the U.K. - is that BT is creating an open, standards-based platform for which anyone can develop new applications. In other

Dear everyone, I've got difficulties in realizing the following task: I have two vectors: A <- c(1:10) B<- seq(1,10,2) Now I want to make a table form vectors A and B as rows, and if a value of A isn't present B, then I want to put a N/A symbol in it: Output should look like this: 1 2 3 4 5 6 7 8 9 10 1 0 3 0 5 0 7 0 9 0 How can I do this in R? Thank you. -- Andris