similar to: wait for rings, answer on outdial via SIP

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 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

Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. [callprogress] 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

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 *

Hello, I have seen the Mitel 5055 SIP phone mentioned a few times on the list, does anyone have any wonderful or horrible things to say about it? We are thinking about using them because they have many more programmable buttons than the Snom200 phones and are about $70 cheaper. Thanks, MATT---

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

I would like to use * as a standalone voicemail system. As such I need it to be able to outdial a certain extension for MWI-ON and another extension for MWI-OFF Is there anyway to get * to automatically dial an extension when a voicemail is left and another extension when the mailbox is cleared? Thanks -------------- next part -------------- An HTML attachment was

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

Hi all, Does anyone have an application/script or extensions.conf file which will do the following? "When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM?" Anyone have any examples of it

Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0. We're placing outbound calls over an analog line. Some of these calls are going to cell phones that play music rather than providing a standard ring. As a result, the Dial command sometimes returns a DIALSTATUS of CHANUNAVAIL and sometimes it returns BUSY. The problem is that this is happening on calls that are being answered. Has anyone else

I have successfully built and made asterisk talk SIP extension to SIP extension, read all the docs, and about 1000 emails from the archive. The trunk side of Asterisk, from the docs perspective, is a smidgin TDM-centric, Analogue, T1, zaptel.conf etc..... Asterisk cares not about the externally presented digits as the telco KNOWS which time-slot or analogue line the call came from I live in an

Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. I've been digging google, wikis but have not found what might be causing this. SIP works fine, this problem seems to be only zaptel specific. I could use the NVLineDetect application

I managed to get Asterisk working with my PBX using T1, now I am moving on to trying to make PRI work. I have my zaptel.conf and zapata.conf configured as follows: Zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us Zapata.conf: [channels] transfer=yes immediate=yes callprogress=yes language=en context=default switchtype=national signalling=pri_net group=1

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

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?

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

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

> From: "Patrick Healy" <pjhealy@healyville.com> > Subject: [Asterisk-Users] Using call.sample on Zap hardware - > Answering problem > I've got a X100P connected to a POTS line and am using it to call out to > play a recorded message. I drop a copy of sample.call into > /var/spool/asterisk/outgoing and Asterisk picks up the line and initiates > the call.