similar to: Prompt for extension with standard dial-tone.

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

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 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, 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 =>

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, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over