similar to: analog call progress - can I use backgrounddetect

I'm looking for an application that can monitor a channel for voice input and then proceed on. The closest thing I've found is BackgroundDetect, which expects DTMF. Here's what I'm doing: -Call file generated which calls someone and connects them to an extension. -Extension plays stuff, etc. etc. etc (not important) With digital or VoIP termination, this works fine, because *

Hi all, I hate to ask such a simple question, but it has stumped me over the past couple of days. I have 2 asterisk servers connected to the house lan and also via a crossover ethernet cable. The original purpose of the crossover was to create a private lan for TDMoE. I have a TE410P in each machine using PRI. I also have setup SIP and IAX2 between both machines. What I'm doing is

Hi, I am attempting to implement Answering Machine Detect and have also played with using BackgroundDetect instead. Does anyone recommend one over the other? Here is the code I am using for the BackgroundDetect method (from voip-info.org). Thanks. [detect] exten => s,1,Set(MACHINE=0) exten => s,2,Answer exten => s,3,BackgroundDetect(silence/5, 1000, 50) exten =>

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

Hello, I am working on an outdial project and the Asterisk box is connected behind a PBX via SIP. When a call from the Asterisk box is routed out over the PRI attached to the PBX I am not getting proper call progress. The PBX is indicating that the call is answered while it is still ringing at the far end. Does anyone have any suggestions on how I should go about waiting for a variable number

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

Asterisk gurus, To have analog call progress (which as far as I know asterisk does not have right now) does it not come down to 4 states to detect - and I hope that some asterisk gurus can implement those 4 states easily. I see this as 4 states: Busy - a 50% duty cycle Ringing - a % on duty and MORE of a bigger % off duty cycle - or basically not BUSY and not TALK detect Talk - something

Guys, is there any way to detect voice when calling a zap channel? For example, if you want to send out or playback a recorded message, you need to wait for somebody to actually answer the phone before playing starts.. Anyway to detect this?

Hi: when add a new voicemailbox, asterisk will create a new directory to it. since linux has limitation for the number of subdirectory. i wonder how many voicemailbox can asterisk support? thanks. yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030430/bd36cdaa/attachment.htm

Does Asterisk know when it hits a voicemailbox ? When calling to a cell-phone or GSM, after some rings and no pickup you arrive at a voicemailbox. If Asterisk does not know it's a voicemailbox that has answered the call, the voicemailbox will contain 60minutes of 'silence'. This is very expensive 'silence'. How to avoid this ? Jonas -------------- next part --------------

Hi, I wonder whether similarly to the very quick rowSums and colSums functions in the base package, one could add quick functions that calculate the min or max over rows / cols in a matrix. While apply(x,1,min) works, I found out by profiling a program of mine that it is rather slow for matrices with a very large number of rows. A quick functionality seems to be already there in the

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

I really want to do this: abline( a=tan(-kT*pi/180), b=kY-tan(-kT*pi/180)*kX ) where kX,kY and kT are vectors of equal length. But I can't do that with abline unless I use a loop, and I haven't figured out the least unelegant way of writing the loop yet. So is there a way to do this without a loop? Or if I am to resort to the loop, what's the best way of doing it considering that I

Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO

I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a

Ok so a while back I found an example for having a number dial multiple numbers and then whoever answers and confirms gets the call. (don't recall who the example was from, but thank you!) But Now today I've been playing with TTS and STT and came across the BackgroundDetect command. Now If I use this allow it works fine. But when I try and use it with this it never actually detects me

Hi - I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to

I've installed a TDM400. Having a go with AMP. I would like incoming calls to be put throuhg to an extension (at my desk) and a mobile (cell phone) at the same time. Whichever picks up, gets the call.. This should be possible with AMP (call groups, 200|201|0*0408xxxxxx), but it didn't work, so I have created a custom-incoming in extensions-custom.conf What is happening is, The extension

Hello all, I've been invited to do a presentation on Asterisk for the Ohio Linuxfest in Columbus this weekend (http://www.ohiolinux.org). Rough estimates are that nearly 500 people will be attending. I've been working on an outline for a couple of weeks and I would like to have some peer review of the information presented. I am going to have to cut down the content to make it fit in