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...dialplan to see where yo= u have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o 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 Pro...

Hi everyone. Newbie to statistics. I have 40 matrices of ~400 values. how may I determine whether the distribution follows zips law? response <-sample (1:20,400*4, replace= TRUE) Thank you vry much. -- View this message in context: http://r.789695.n4.nabble.com/A-introductory-question-about-Zips-law-Newbie-to-statistics-tp4521190p4521190.html Sent from the R help mailing list archive at Nabble.com.

Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

...dialplan to see where yo= u have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o 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 Pro...

...in effect, be taking a tutorial from me on these subjects. (If you've seen some of my questions on this list, please don't laugh.) As an undergrad he majored in philosophy, so this will be his first foray into computer programming and statistics. I'm thinking of having him use "Introductory Statistics with R" by Peter Dalgaard, but I'm unable to tell if the book requires calculus. I don't think this student knows calculus, so this would be a deal breaker. Can someone tell me if my student can get through this book starting out with just knowledge of algebra? Also, do...

...dialplan to see where yo= u have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o 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 Pro...

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I

Hi-- I'm trying to teach myself R using Peter Dalgaard's "Introductory Statistics with R." Does anyone know if there are solutions available for the exercises? I'm really just hoping to learn by going through exercises and checking myself, so alternatively, does anyone know of an introductory textbook to R that does have answers available either in the back...

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register => 908366554:396444 at carrier.jazzey.com register => 908366554:396444 at sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

...dialplan to see where yo= u have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New t= o 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 Pro...

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello guys, i would like to implement authentication for my sip extension with an openldap server. Following this guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html i see a template named [sip] to map the information of sip peers into ldap. But i'm not interested to create a template, i would only authenticate sip extensions using username

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

...gt; Thanks, >>>> Sonny. >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >...

...I might use if push comes to shove, but I wanted to ask whether anyone of you knows of a "official version" by one of the Core Project members ? Christian ----- Christian Langkamp christian.langkamp-at-gmxpro.de -- View this message in context: http://r.789695.n4.nabble.com/Standard-introductory-presentation-tp4640199.html Sent from the R help mailing list archive at Nabble.com.

Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |