similar to: Playback noanswer & SIP

We have working in a rural project and I have selected Asterisk. Because * will work with a prepay system we need a precise answer supervision for outgoing calls. The pilot system will have 8 trunks to PSTN. The Telephone Company provides EURO BRI and analog CO trunks. On CO trunks the answer supervision is reversal polarity. The village to be served will be connected through a WLAN. Thus,

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

Hi! Our telco routes multiple numbers through PRI to our Asterisk. Not all of these numbers are in use. I have noticed recently that someone keeps calling unused phone number from outside world. I called them and asked why do they call dead number. The person on the far end explained that she keeps calling this number because she hears "busy" tone every time... Most telcos these

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 List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hello, I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not because the dialed party does not pickup the phone, it is because the call does not go through the provider. I've written a dial macro which route the call via second provider if the first returns

Hello List, I am an Asterisk newbie, and I got a question about Asterisk Background command's option "noanswer": What is required from the user agent, such as a SIP phone, to be able to hear the playback without Answer()? I'm asking this because when I used X-Lite, I could hear the the audio file but when I used a hardware phone (an ATA in fact) I couldn't hear it. The

Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be?

Hi Guys.. I am making an anoucement machine that is not allowed to "answer" the call due to a billing issue. I found that Playback with "noanwser" is usefull in this case. $AGI->exec('Playback',"$message","noanswer")} But when i request some values to the user with get_data, i think there is an answer anywere. Is there a way to get_data

Gang, I'm wondering if anyone has run into this problem and found a solution. When I use the manager interface to generate a call, I don't get very much information in my CDR records when the dial status is BUSY, FAILED, NOANSWER, etc. I am putting the dialed number into the CDR Userfield in my dialplan, but the field doesn't populate the CDR record unless the Originate action is

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

How can I get external (telecom local exchange) dialtone on HFC ISDN BRI with bristuff/zaphfc driver? with capi, voip-info say that it should be something like: Dial(CAPI/MSN:b) But with zaphfc, if I try: Dial(ZAP/1/), I just get NOANSWER.

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="Content-Type" content="text/html;charset=utf-8"> <title></title> </head> <body> <br> <!--+GtkHTML:<DATA class="ClueFlow" key="orig" value="0">--> I connected my * box to two

Some days ago there was a subject regarding BLF (SIP Phone-receptionist Setup). We are the developers of a Price Verify Terminal for a French company. We have developed the hardware (small board based on a PPC 823e), working with Linux embedded (based on Wolfgang Denk's work). I think that it can be a good BLF. Probably it is possible to integrate the Nicolas's FOP or a new application.

hi,all i find in asterisk 1.6.2.1, before play a sound file use playback or background, it will answer the channel first. but i want to answer the channel when dial someone and pick up the phone.not play a file. i know there are some params such as 'noanswer' for playback or 'n' for background can not answer before play a file. but it is not always take effect on my tests.as it

Gracias a todos, finalmente instalando primero el libxml2-dev y luego el dplR, funciono ----- Mensaje original ----- De: r-help-es-request en r-project.org Para: r-help-es en r-project.org Enviados: Miércoles, 17 de Julio 2013 12:59:47 Asunto: Resumen de R-help-es, Vol 53, Envío 26 Envíe los mensajes para la lista R-help-es a r-help-es en r-project.org Para subscribirse o anular su

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

Bueno, creo que no contesté tu pregunta. Con training <- data[-i, ] crea una df llamada training, sin la muestra i, que después utiliza para entrenar el algoritmo. Quoting Javier Marcuzzi <javier.ruben.marcuzzi en gmail.com>: > Estimado Manuel Mendoza > > Con sus datos y a modo de curiosidad, ¿que pasa en training <- data[-i, ]? > > Javier Rubén Marcuzzi >

Hello, I have an ISDN PRI E1. I want to send an audio before answering, I am using noanswer option in playback app but all the audio is muted before the answer. I would like to play this audio. Regards, ia