similar to: ADM v. homemade code

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 =>

Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the

Hi, I am using a call file formated like this: Channel: local/12125557891 at outbound/n Callerid: 12125551212 Context: detect Extension: s Priority: 1 This sends the call into the dialplan at the [outbound] context. In [outbound], I have: [outbound] exten => _1.,1,Dial(SIP/${EXTEN}@flowroute,43) If the call is answered, it move on to the [detect] context. When using this method, it appears

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

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

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

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- this seems to be simple to figure out but since im new to writing functions I dont know what is happening. Here is my code along with the error I am receiving: semivario=function(data,ids,times,resids){ id=unique(data$ids) index=combinations(length(data$times[data$ids==id[1]]),2) time=gamma=numeric(dim(index)[1]) for (j in 1:dim(index)[1]){

I've been given the task to hunt down a codec that out solar car team can use to transmit and receive data over a radio link between a chase car and the solar car its self. As of now I do not know the exact specifications of the internal hardware of the car, what I do know is they are looking at transmitting across a 2400baud connection, the unit in the car is homemade from our labs here.

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 I was interested to see how Mongrel uses Lex/Yacc to parse the HTTP requests using a Regal generated parser. I downloaded the source but do not see the lex and yacc files...

> but it isn't going to be quite so low-bitrate when you > have IP (or some other) packets encapsulating it. Most packet systems, such as PPP and SLIP, want at least 9600 baud to play ball. Traditionally, low bandwidth systems like those are just open streams. The protocols built on top of those - old stalwarts like YModem/G, Kermit, and so forth - do admittedly generally chunk

so this is a weird one: http://babelmark.bobtfish.net/?markdown=%3Csup%3E*%3C%2Fsup%3EThe+[distinction]+between+certification+and+qualification%2C+although+vague%2C+appears+to+be+that+under+the+NASED+system%2C+states+did+the+ultimate+*certification*+of+a+voting+system+for+fitness+in+future+elections. Note that the `<sup>*</sup>` is being turned into

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?

I've recently installed R-1.4.0 and I'm having the devil's own time trying to install a home-made package. As far as I can tell, I'm doing the same things that I did previously, under 1.3.1, but somehow everything goes haywire. (Quite possibly I'm simply mis-remembering what I did before, but I can't for the life of me figure out what to change). A toy example of what I

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

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