similar to: How to play DTMF digits without blocking

Tenho o wine instalado no Fedora 6 e quero rodar o VFP, mas quando executo o VFP aparece "erro ao conectar com SQL. No nodo texto aparece "err:CoGetClass {854d7-bc3d-11d0-b421-00a0c90f9dc4} not registred. O que eu posso fazer? cid:image001.jpg at 01C8625E.FF0BF240 Reginaldo Mendes da Cruz Analista de Suporte e Sistemas Fone:(011) 5503-4206 <mailto:reginaldo at

Hi all, I just installed Cepstral and app_swift version 1.4.2 on my Asterisk 1.4.22.1 box. It seems to work great with one exception. If I play a test message with instructions to collect a maximum of 5 digits, it collects those 5 digits correctly if the user waits for the message to complete before entering them. But if the user barges in with digits before the message completes, the detected

-- Executing DateTime("SIP/phone1-07ff", "") in new stack -- Playing '/var/lib/asterisk/sounds/digits/day-1' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/h-19' (language 'en') This works - the pathname is complete - Joy.

Hi list. I was googling about this subject and found a message to this list asking the same thing I want, but see no response. So, I'm reposting it in case someone has a solution for this. I'd like to know if there's some way to "detect" (not block) a collect call in a ISDN E1? I need this cause I have my own billing system and I need to charge my customers for collect calls

Hi list! I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and some PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and Asterisk to INFO too. At first, is INFO method different from RFC2833?? Well, I have two problems. The first is that when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems

Full_Name: Reginaldo Constantino Version: 2.8.0 OS: Ubuntu Hardy (32 bit, kernel 2.6.24) Submission from: (NULL) (189.61.88.2) For many tables, chisq.test with simulate.p.value=TRUE gives a p value that is obviously incorrect and inversely proportional to the number of replicates: > data(HairEyeColor) > x <- margin.table(HairEyeColor, c(1, 2)) >

Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

Dear R users, I have been plotting several graphics in only one page. I intend to plot only one legend in each page. Since the graphics are inside a loop I have to do it automatically. The main problem is that inside the loop I work with differente types of variables, I mean , the range is different for each one. I have already tried : par(xpd=TRUE) pos<-par("usr")

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

I have the following in my extensions.conf: exten => 21,1,NoOp(${CALLERIDNUM}) exten => 21,2,GotoIf($[${CALLERIDNUM} = ""]?21|4:21|9) exten => 21,4,Playback(/etc/asterisk/interactive-services/no-callerid) exten => 21,5,Wait(1) exten => 21,6,Playback(/etc/asterisk/interactive-services/no-callerid) exten => 21,7,Wait(1) exten => 21,8,Goto(10,4) exten =>

hello, I use trixbox.I had define a feature code testfeature: [applicationmap] #include features_applicationmap_additional.conf testfeature => *3,callee,Macro,vote [featuremap] blindxfer => ## ; Blind Transfer disconnect => ** ; Disconnect Call automon => *1 ; One Touch Record atxfer => *2 ; Attended Xfer testfeature => *3 here is my macro-vote: [macro-vote] exten

Dear all i'm creating an outgoing call to number xxx with this command: http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external &Exten=testDTMF&Context=cRETEUNICA&Priority=1 wich points correctly to this portion of dialplan: [cRETEUNICA] exten => testDTMF,1,Answer exten => testDTMF,n,Read(digito,,1) exten => testDTMF,n,SayDigits(${digito}) The

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

I have a user who wants to receive an ANI spitback in DTMF. Right now, the "SayDigits(${CALLERIDNUM})" command works fine with voice. But I'd like to end up doing both. Something along the lines of: exten => 34,1,Answer exten => 34,2,Wait(1) exten => 34,3,Playback(vm-extension) exten => 34,4,SayDigits(${CALLERIDNUM}) exten => 34,5,Wait(2) exten =>

Hello, I'm an MD working in an eye clinic. I'm learning by myself to use R for use in my research works and for implementation in a software project. There are some authors who recomends the use of Cox regression as a substitute for Logistic regression (<a href="http://www.biomedcentral.com/1471-2288/3/21.pdf"> Barros AJD, Hirakata VN. BMCMedical Research Methodology, 2003;

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on

Hi all, I am trying to get a Menu system to work, and having probs with the internal extensions from the prompts. Below is the extensions.conf section. [mainmenu] ; ;"main menu" context with submenu ; include => default exten => s,1,Answer exten => s,2,Background(hello) exten => s,3,Background(thank_you) exten => s,4,Background(if_you_know_extension) exten =>

Hi List, I'm working on making one of my applications multi-lingual and find that I have this problem. The SayDigits and SayNumber functions in 1.4.26.2 recognize but don't process the CHANNEL(language) function. Here's a snippet to verify. exten => 317,1,Answer exten => 317,n,playback(tt-monkeysintro) exten => 317,n,Set(CHANNEL(language)=es) exten =>