similar to: Sending e-mail from dialplan

My data (slength) look like this: Plant Block Treat Genotype Source MPH 1 1 1 w 05-AZM sp 160 NA 2 2 1 w 12-50 463 NA 3 3 1 w 12-51 sp 150 0.0508982 4 5 1 w 29-42 567 0.9017094 5 6 1 w KNG-KNG 811 NA 6 7 1 w 02-02 1 NA Treat column

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

What is the most efficient way to allow inbound callers to dial internal users yet restrict them from outbound PSTN calls? Today I have a basic greeting that after a welcome message allows inbound callers the ability to dial any of my users. However, it seems that since I transfer the inbound caller to a context that allows them the ability to call my internal users they have the same rights as

Is there any way to use Festival from script called by the ExternalIVR() dialplan function? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david@safedatausa.com

Hi, I want to prevent Asterisk from sending the h extension across to the SIP provider or to prevent it from hitting the script at all. The SIP Provider does not know what to do with the h extensions once it receives it. My SIP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

Hi All Curious will this work Std. PSTN line ---x------ X100p | ------ Fax Machine Using a standard "home phone" pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten => s,1,Wait(1) exten => s,2,IfFax continue to ring, so that the Fax Machine gets it exten

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the *

Does anything have to be done at compile time in order for Asterisk to take advantage of 2 CPU's? Thanks

Hello, I am trying to run festival, it is running but I am getting this when I run tts_ping.agi WARNING[278546]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Enter the eye-p address you wish to ping. WARNING[278546]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... WARNING[278546]: File app_festival.c, Line 400 (festival_exec):

I have done research and reading around text to speech, and was wanting to get an updated query of where everyone is with this. I have installed festival 1.4 on CentOS 3.5 ( system was installed using Asterisk at Home ISO ). I also changed the directory php application to use the festival.pl to read the names of those who have not installed a greeting. It works well enough, though the only voices

hey guys.. im a prototype rookie here.. the concept of "a framework for javascript" seemed mouth watering.. but my rails got stuck when i started using it. here is the guinea pig im using to experiment on prototype. <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> <html

We have the ability to create random UID's on own system through a custom CGI API. These UID's are written to individual sip configuration files based on the account name, so for instance sip_TEST.conf, sip_TEST2.conf, and sip_TEST3.conf, etc. Many of these UID's are created on the fly and at random times throughout the day. Right now, I have it setup to do a reload every night.

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to be hands on for each new phone number deployed... so I would like to set up some administrative extensions that can record greetings... lets say: [admin] exten => 8(NXXNXXXXXX),1,Record($1|-greeting.gsm) [incoming] exten => _(NXXNXXXXXX),1,Playback($1|-greeting) exten => _(NXXNXXXXXX),2,Goto($1,1000) exten

Hi there, I just purchased a CyberPower BR850ELCD and while setting NUT up with the install instructions from the homepage was pretty straightforward, and the UPS was detected by the usbhid-ups driver, I'm having problems getting the UPS to work properly. First issue: whatever I set as the offdelay, seems to be ignored and the UPS just cuts the power about 2sec after receiving the

Hi, I have a requirement where an IVR application on asterisk has to play a audio file in g729 and when a digit is pressed, the call should switch to another codec (say ulaw). So, What can I do in the extensions.conf to trigger a re-negotiation of codec? I used exten => 55xx,n,Set(SIP_CODEC=ulaw) but, I suppose this affects the next call and not the current one. Please help ASAP Thanks,

Hi, When I do a sip show peers in the cli, the status is lagged. This peer its behind a satellite link with 600/900ms of delay. May I change some parameter in the Asterisk? Some times I cant make a phone call from the remote site to my central site. Thanks Este mensaje ha sido analizado por C4I Mail Server en busca de virus y otros contenidos peligrosos, y se considera que esta limpio.