similar to: Autodial feature doesn't return $DIALSTATUS values

ciao a tutti.     come al solito, sto andando avanti a piccoli passi nel creare l''applicazione depot del libro "Sviluppare Applicazioni Web con Rails". Ora mi trovo nella fase di creazione dei test automatizzati. Ma c''è una cosa che proprio non capisco: Ho la seguente classe in ~/depot/test/unit/product_test.rb require File.dirname(__FILE__) +

The ade4 package (v. 1.2-1) has been updated on CRAN. New features include: - functions based on Rao's axiomatization of diversity measures : Rao's diversity coefficient and dissimilarity coefficient (divc and disc) - functions based on Excoffier et al. analysis of molecular variance with tests of the difference among the factors (amova). - functions introducing double principal

The ade4 package (v. 1.2-1) has been updated on CRAN. New features include: - functions based on Rao's axiomatization of diversity measures : Rao's diversity coefficient and dissimilarity coefficient (divc and disc) - functions based on Excoffier et al. analysis of molecular variance with tests of the difference among the factors (amova). - functions introducing double principal

I'm looking to build an application which requires a phone to autodial when the handset is lifted off of the hook. Since the system will be available to the general public, I want a solid phone that will withstand abuse, but not be too expensive. I'm leaning towards using a Sipura adapter with an analog outdoor handset. However, I don't nor ever have had a sipura adapter, so I'm

I'd like to setup an autodial that will do the following: 1. First try call my multiple SIP phones at my house - if no answer within n seconds, try my mobile phone. Alternatively call both SIP phones at home + mobile at the same time. 2. Once someone answer in step 1 (on any phone) initiate calling to a specific phone number/extension Currently I have it working except that in step 1 I can

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

Hello, I am using asterisk to autodial a phone number once an hour to verify an answering IVR script is working by listening for a beep played by the IVR, and that is working well. The one thing I have not been able to find is a way to trap if the call didn't connect at all, like if the entire IVR is down or the phone line is dead, since it doesnt enter the menu until the call is

I have tried to get my TDM400P card to automatically dial a number or run an application when I pick up the phone without much luck. After reviewing the email archives, config files and source to chan_zap.c it appeared that all I had to do was set "immediate=yes" in the zapata.conf file and have a default number in the TDM400P's context (ex. s,1,Directory(default)). So far I have

Hello, Current moment, I've successfully put the incoming calls into Queues and dial to an idle agents. When the agents answer the calls, the agents can hear the pre-recorded message to incidate what's the service that the call is calling. But there one problem that I'm not able to make it having the Caller ID display on the X-Lite. Even I try to make a call direct transfer from

Hi :) I'm doing same test with autodialout, placing a file in /var/spool/asterisk/outgoing, it's work good when I want to dial a number from a phone.... --MY-FILE------------------------------------ Channel: SIP/BGT100 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: SIP_inbound Extension: 600 Priority: 2 --------------------------------------------------- Can I create a call from a

Hi All - I'm setting up a corporate emergency broadcast system that uses an autodialer to contact all company employees. Everything works fine except if the auto-dialed calls go to the end users' voicemail. If that happens, asterisk starts playback of the emergency message while the voicemail system on the other end is playing its outgoing message. The result is that the beginning (or

Hi to all, my question is: Is possible implementing SIS (with hardlink) with maildir instead of *dbox format? If yes in dovecot.conf it's only necessary the below parameters or what else? mail_attachment_dir = /var/qmail/attachments I have also acting the zlib plugin it is not a problem isn't it? Thanks in advance for any response -- */Davide Marchi /Teorema Ferrara Srl /(Tel:

Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time.

Hi to all, my question is what is benefit implementing LMTP service replacing LDA i have dovecot 2.1.8 with vpoipmail+qmail and about 500 users now i'm using LDA and i'm interested on LMTP service. Thanks in advance -- */Davide Marchi /Teorema Ferrara Srl /(Tel: /**/+39 0532 783161)/**/ (Fax: +/**/39 0532 783368/**/)/**//**/ /**//**/Davide.Marchi at mail.cgilfe.it

--- -------------------------- CISTI'2016 ----------------------------- 11? Conferencia Ib?rica de Sistemas y Tecnolog?as de Informaci?n 15 a 18 junio de 2016, Isla Gran Canaria, Espa?a http://www.aisti.eu/cisti2016/ ------------------------------------------------------------------- Nos satisface invitar a la comunidad acad?mica y empresarial a presentar trabajos a

--- -------------------------- CISTI'2016 ----------------------------- 11? Conferencia Ib?rica de Sistemas y Tecnolog?as de Informaci?n 15 a 18 junio de 2016, Isla Gran Canaria, Espa?a http://www.aisti.eu/cisti2016/ ------------------------------------------------------------------- Nos satisface invitar a la comunidad acad?mica y empresarial a presentar trabajos a

Hi to all, sorry in advance for my poor english, this is the first time that i wrote to a list if i make mistake .... excuseme. My problem is this: i have dovecot 2.1.8 installed and functioning from 2 years one week ago i have installed crm114 for my last spam detection filter "version 20100106-BlameMichelson (TRE 0.8.0 (BSD))" My mail system is qmail that through .qmail default

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

Hi, according to the wiki the value NOANSWER for the channel variable DIALSTATUS means: No answer. The dial command reached its number, the number rang for too long, then the dial timed out. In out dialplan we grap all these events with exten => s-NOANSWER,1,Playback(sometext) exten => s-NOANSWER,2,WAIT(1) exten => s-NOANSWER,3,Hangup() The dial commands for internal and external

Hi there, could anybody tell me if the info below is still correct: Note: In order to obtain useful DIALSTATUS information when dialing a peer you will need to have qualify=yes in that peer's definition (e.g. in sip.conf or iax.conf). http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS THANKS!! -- Regards, Rustam Kovhaev