search for: rigas

Hi! We are in process of setting up an audio guide that will cover notable places of our capital Riga, Latvia. The target audience are tourists that dials a free phone number from a mobile handset to listen to a 3 minute introduction to historic place. All audio, 10+ languages are recorder in studio at 44KHz. The audio is stored on server in A-law 8KHz because we'll be pushing it through E1

Dear R-Experts, I have an empirical dataset with 150 subjects for 24 observations. In each observation, each subject can have a score in the range 0:3. I made then a simple index making the sum of the values in each row, so each subject have a score between 0 and 72. I was thinking about what kind of theoretical distribution such an index should have, so I try to make things random like:

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

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, 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, (Reposting as my previous post got zero replies.) We're running Dovecot 2.2.36 and we need to set up the mail-crypt plugin to encrypt all incoming and outgoing emails. Outgoing emails seem to get encrypted fine but the incoming ones don't. We tried everything including this config: mail_attribute_dict = file:%h/Maildir/dovecot-attributes mail_plugins = $mail_plugins mail_crypt

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 recently upgraded my pdc server(samba3.0.x+ldap) to debian lenny( samba 3.2.3). After the upgrade, the win2k join is no longer working and returns "Logon failure: the User Name unknown or bad password". The Xp join works properly. The same thing seems to be happen to other users; same problem and same logs:

...cago|'vm-received' q 'digits/at' H 'digits/hundred' M 'hours' [default] 1234 => 4242,Example Mailbox,root@localhost ;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central ;4300 => 3456,Ben Rigas,ben@american-computer.net ;4310 => -5432,Sales,sales@marko.net ;4069 => 6522,Matt Brooks,matt@marko.net,,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes ;4073 => 1099,Bianca Paige,bianca@biancapaige.com,,delete=1 ;4110 => 3443,Rob Fl...

Hello all, is it possible to 'restream' another server stream with liveice/mpg123 to another icecast server? we have to setup this really fast, so maybe someone can share with actual cmdline string for it? ;) -- Best regards, deep-z Riga/LV --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list,

...more. The only machine I can ping is the virtual ttylinux server. Therefore I cannot reach the XEN machine from outside using ssh or telnet. The ttylinux vm works fine. From there I can reach every other machine in my network using ping!!! So the nic itself should not be the problem. Regards Rigas --------------------------------- Gesendet von Yahoo! Mail - Jetzt mit 1GB kostenlosem Speicher _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

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 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();

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 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 --------------

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 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 =>

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