similar to: Detection of Answering Machine

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

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

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, 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

I've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no

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 =>

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as "Channel: DAHDI/1/8775203463" When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to

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 *

I'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel

Nik, Looks like you're making some progress. When I first started using A@H I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten =>

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

Gang, I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I'm wondering if I was hallucinating... Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

> you should mv the file (and in the same filesystem, so 'rename' is used) > You might want to chmod or even chown the file first as well. I wrote a little script that does all of this before the .call file is mv'd into the outgoing directory: cp /tmp/test3.call /tmp/test1.call chmod 666 /tmp/test1.call chgrp asterisk /tmp/test1.call chown asterisk /tmp/test1.call mv

Greetings all, This is my first post to this forum, so please be kind. I am looking to integrate Asterisk as our primary voice mail server (IVS) for our NEC NEAX 2000 IVS. We currently use a 6-port Mitel system. The two communicate via analog TDM ports and a MCI serial connection. Is anyone familiar with setting up a NEAX 2000 with Asterisk as an IVR server? I know I would need six FXO and

Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good).

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,