similar to: AGI Application Hangup when using AGI->getdata

hi list, i am making some diagnostic plots using plot.lme(), my lme model was fit with the option na.action='na.exclude'. On the following (or similar) commands: plot(flu.trees.42.w25.10.2, resid(., type="p") ~ fitted(.)|species, abline=0) R returns an error message: Error in getData.lme(object) : couldn't find function "naAct" It turns out that the function

Hi , I'm having a problem that really bothers me , I have looked for similar cases but couldn't really find an answer . I keep getting messages which says that -- B-channel xx successfully restarted on span x and this causes the calls to be disconnected if somone is already using the channel that is being restarted, plz notice below the under lined text in which I was testing and

Hi, following I would like to implement: 1. I receive a call. 2. I hang up the call. 3. I execute a macro I thought about using call files first... but they don't support macros, or? Then I figured I could use php agi after I receive the call, hang up the call with php agi and execute the macro with "exec ...". Unfortunately, the php agi seems to die with the hangup. Does

All; I wrote a PERL AGI script that prompts a caller to leave a message using print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n"; When the caller is done, they need to press the # key. The message is then delivered. However, the message is not delivered if the caller simply hangs up when finished. If the user hangs up, the script ends right then. How do I keep on processing the

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten =>

Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl, configuration in extension.conf is: exten =>_numbera,1,AGI,blacklist_2_in.agi exten =>_numbera,2,Answer exten =>_numbera,3,AGI,xisco_1.agi exten =>_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in

I know that hangup handlers ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers) have to finish quickly. So it's no surprise that my speech to text agi which takes 8 seconds gets killed. However, can anyone think of a way round this? So, once the caller has hung up, I need to take one of the channel variables, and pass it to a python agi script which then does speech to text.

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

I am trying to write a loop to forecast realized volatility over successive days for the purpose of VaR prediction using the HAR-RV-CJ model which is as follows: log(RV_t+1) = ?_0 + ?_CD log(CV_t) + ?_CW log(CV_t-5) + ?_CM log(CV_t-22) + ?_JD log(J_t + 1) + ?_JW log(J_t-5 + 1) + ?_JM log(J_t-22 + 1) + e_t where RV is realized volatility, CV is continuous volatility and J is the jump which is

I think this is an issue of serious interest to many of the subscribers of these lists; it would effectively ban a lot of security-related tools that many of use now find indispensable, e.g. ssh, pgp. ------- Start of forwarded message ------- Date: 21 Mar 1997 10:11:57 GMT From: rja14@cl.cam.ac.uk (Ross Anderson) Approved: R.E.Wolff@BitWizard.nl Subject: DTI proposals on key escrow The British

> In Florida some new subdivision developers have sold the > phone/cable/internet rights to a provider. They run fiber to each house > and then have the uplink to provider which isn't a traditional telco. > You can't get another provider as satellite dishes are limited in > covenants and restrictions (CCR). Those CC&Rs may very well be legally void and unenforceable.

I have set up an application which records incoming calls (customer problems), and then places outgoing calls (handing the problems to support people). I have written some AGI programs (in C) to implement parts of the functionality. So far I am very pleased with the overall application. But I have one very annoying, very nasty problem: The C AGI is handling a dialog with a user on an outgoing

Hi, I am trying to insert a system() call right after the call to Voicemail() in order to notify people that they have received new voicemail. (In case you are interested, I was following the tips here, but I have had to change a few things: http://www.voip-info.org/wiki-Asterisk+tips+callback ). My setup works when someone leaving a message hits # to exit the voicemail after speaking, or

Greetings all; I have an odd problem - Hangup isn't hanging up, instead Asterisk carries the flow going in the extensions.conf, and the next matching extension gets run. Not good. My extensions.conf (highly simplified) looks like this: [pri] include => dids include => SIPlookup [dids] exten => 13015555555,1,Wait,3 exten => 13015555555,2,Answer exten =>

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

hey i want to implement phone card application based on PIN......... for this i am planning to use the AGI......... which programming language ( c , python, java .etc) should i use? i mean which one is effective. please suggest me. _________________________________________________________________ The new MSN 8: smart spam protection and 2 months FREE*

Greetings All, I've created a sample trouble ticketing management script in perl that allows the management of trouble tickets, call routing, etc that interfaces with the perldesk helpdesk application. It is mainly for an example of what integration between asterisk and real world apps is possible. Basic instructions are included in the script itself including instructions for accessing

Hi all, Sorry for this elementary question (I'm a newbie). I'm trying to write an agi script (test.agi) and run it when I call in. However, I'm getting an error that says application agi isn't being found. I've put test.agi into agi-bin with permissions 755. Do I have to compile agi support into Asterisk, or is it built in? My test.agi script is php, but not using anything

I have placed the custom-cdr-V1.0.tar for download http://www.itsngroup.com/software/asterisk/downloads/ Thanks > Dear Ben, > I've also the problems as Chris Mason, Can you send us your own AGI CDR > logging application? > Best regards, > Jian Hong Guan > France > www.directcentrex.com > > >

I am running a prepaid application with Asterisk. When authentication has to be done by DTMF everything works fine. However when the user is authenticated directly from the sip phone, the channel variables seems to disappear. Trying to retrieve the channel status always returns -1 instead of the 6 that happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME variables. The