similar to: Operating environment for *

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

Hi I work at a small company that has some IVR solutions that use Dialogic hardware for everything. Everything is written in C++ using MS VC++ using the Dialogic API and runs only on Windows. Being the rebel that I am, I would like free myself from Dialogic. To do this without porting all our existing code to run on Linux I was thinking of controlling the Asterisk from a Windows machine running

Can someone point me to some reasonable example / starting point to implement a basic IVR menu? Looking for something rather simple like the press 1 for sales, 2 for tech support, and probably an option to list the voicemail directory kind of thing. Nothing elaborate needed, just basic menu. (Yes, I did look at the wiki and google searched for "ivr menu".)

Do they fall under FCC certification if they're built to the same specifications as the ones from Digium? If I build my own T100Ps from the schematics and board layouts that are available, are they legal to plug into the PSTN?

Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it

Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use "modprobe zaptel", get a "FATAL modul zaptel not found". Thanks. Yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040722/4608e1e5/attachment.htm

Hi- I'm almost embarrassed to ask the following simple question, following John's excellent and rigorous bandwidth analysis (see earlier thread): I have a straightforward Asterisk application, IVR-only (no connections between channels). It will simply decode DTMF's and speak prompts (probably A-Law encoded) on a number of E1 circuits simultaneously. Realistically, how many

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing

Thank you for reading this, sorry to waste bandwidth otherwise. I am part of a US company looking for someone to setup a demo IVR system for us. I seem unable with my current knowledge to pull this off myself. The demo is the regular enter your id and validate/repeat/continue methodoligy you put up with in everyday life. I would like to have the validation and other parts done via database

Are IVR and "Auto Attendant" interchangeable terms? They both do the "Press 1 for" thing. Sales is asking me how to word it and I've always used both terms interchangeably.

I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing

Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can

Hi all- It's often mentioned here that recompiling the Linux kernel may improve the performance of asterisk. My question is: Does re-compiling the kernel have any effect if there are no changes to the configuration? What parameters will typically improve performance? Thanks for any insight! Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London,

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm

Hi all- HELP! This is actually a revisit of a problem that I had earlier with E400P's at a customer site. Customer still gets locked up channel problem, but has learned to live with it (channels clear themselves after several minutes). The symptoms, which I believe are directly related: I'm having problems with tons of framing and "read" errors on my E1 connections (and

I am trying to perform goodness of fit test using R. I am using this website http://wiener.math.csi.cuny.edu/Statistics/R/simpleR/stat013.html for help. However, I am unable to carry out the test successfully. My code follows. It is taken from the website just mentioned. freq=c(22,21,22,27,22,36) # frequencies obtained after rolling the dice 150 times. prob=c(1,1,1,1,1,1)/6 # specify expected

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

I am wondering what is the best way to send the smallest files with the vm to emai l integration? I am not sure what order the three lines of the format command take, so I have just tried trial and error swapping. I think when set to "gsm" I get the smallest sizes. I can get my Windows Media player to play at least part of the file (get missing codec message from Realplayer), but get a

Hi all, I want to make a contingency table in R. I want to tabulate two variables, one as the independent and second as the dependent variable. The IV has two categories, namely, birth complications, and no birth complications. The frequency of birth complication category is fifty, and the frequency of no birth complication category is 34. The categories and frequencies of DV follows.