Call disconnect for asterisk

自闭症网瘾萝莉.ら 提交于 2020-01-07 05:04:19
问题 How to capture call disconnect for asterisk using PHPAGI ? For e.g. if user disconnects the call which event is invoked ? How to capture it ? 回答1: You can check the Return Results of your PHP-AGI API calls, for example stream_file returns -1 on hangup. You could also invoke another AGI Script on the h Extension in the Dialplan. If you have to clean up something, you could also register a Shutdown function. Another Approach is to register a Signal Handler which edmund long described in his

What's your Interactive Voice Response Platform? [closed]

久未见 提交于 2020-01-01 12:32:13
问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 3 years ago . For those of you working in the voice space, what are you using as your IVR platform? I am using Microsoft Speech Server 2007. What are some equivalent packages? Is anyone using open source software for handling inbound or outbound calls? Note that I'm not just talking about speech recognition, which is one


江枫思渺然 提交于 2019-12-25 16:04:39
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 1) 总所周知,Asterisk 支持播放的语音文件格式.wav 为: PCM非压缩编码,8Khz,16位,单声道,下面介绍几款转换工具 2)语音编码转换工具 1. GoldWave 【文件】——>【批量处理】,剩下的操作如下图了: 2,英文界面的WavePad Sound Editor 如下图 3. Audacity简体中文版,如下图: 修改为 单声道 ,16 bit ,如下图: 【文件】——>【导出】 来源: oschina 链接: https://my.oschina.net/u/939089/blog/145757

Seek Help concerning IVR Menu in Asterisk

白昼怎懂夜的黑 提交于 2019-12-24 20:03:03
问题 I am writing an IVR menu and I need to allow my users to press 0 anytime during the communication to exit. The following is how I do it: exten => 0,1,Playback(good-bye) exten => 0,2,Playback(beep) exten => 0,3,Hangup However, by doing so, when the user presses zero while some file is being played back or some other operation is taking place, he/she cannot exit, it is like if he/she didn't press zero. I hope I am clear enough and that you can help me out with this. cheers 回答1: For an IVR, you

Asterisk TDM410

为君一笑 提交于 2019-12-24 10:51:01
问题 This is not a programming question per se. I am trying to build a system which consists of the following: User calls system using regular land line Some processing is done by asterisk Call is forwarded to an external number (another landline/mobile phone) Now I would like atleast 2 simultaneous lines on which the user can call. I would like to know the following: Will the TDM410 work for what I am trying to achieve? Since I want call forwarding, do I need an extra line for that? Or can I do

Kick all user from confbridge when one user left

北战南征 提交于 2019-12-23 05:19:06
问题 I have a problem,if a single user left the confbridge or disconnect his call... I want to hangup calls of all other users who are in that particular conference room...Any idea regarding this??? Basically I want to disconnect all channels if any of the channel hangup the call.Any guidance? Many thanks. 回答1: There are no simple way do that. Reason is simple. Anyway at some moment in conference will be single user(at start) You can use marked user(and close on marked user exist) or you can use

ZPK Encryption ISO format 9594-1 Format 0

人盡茶涼 提交于 2019-12-23 05:01:53
问题 I need to integrate our IVR with ATM switch. In this case IVR needs to send pin block formed in ISO format 9594-1 Format 0 only (Zone pin key – Pin encryption). WE have Clear component - 1 ,2, & 3 & ZMK - Key check value. I need to know steps to generate PIN Block format 0 using ZPK encryption. Also as I am beginner need to know the role played here of clear component & ZMK . 回答1: Here is some javascript that will encrypt a pin into a Format-0 (PAN-free) pinblock. In this example it is

Nexmo: How to transfer call route to the same function again and again to form a loop

╄→гoц情女王★ 提交于 2019-12-23 04:34:07
问题 I am trying to make a simple voice IVR for my project using nexmo API. Please refer to the image for exact clearance. basic idea by flow diagram of what I am trying to do as ivr. Now the problem occurs that I can't able to make a loop to return to the mainMenu if digit pressed was wrong. The problem till I get understood is in GET and POST method of my function. from flask import Flask, request, jsonify, Response, render_template import nexmo from pprint import pprint from random import

Voiceglue Logger says Maximum loop count exceeded. There is probably an infinite loop of in your VXML document

ε祈祈猫儿з 提交于 2019-12-23 02:24:06
问题 Can Any please explain why this is happening. what are the possibilities of errors that are been counted as I have set maxerrorcount = 3 EROR OPEN_VXI luke---- callid=[68] |1098905920|68|CRITICAL|com.vocalocity.vxi|216|VXIinterpreterRun: Maximum loop count exceeded. There is probably an infinite loop of in your VXML document.|URL Please let me know if any further details are required. 回答1: Perhaps, "infinite loop" means to call same form again and again, And it was not inserted caller input

What's a good open source VoiceXML implementation?

天涯浪子 提交于 2019-12-21 03:38:09
问题 I am trying to find out if it's possible to build a complete IVR application by cobbling together parts from open source projects. Is anyone using a non-commercial VoiceXML implementation to build speech-enabled systems? 回答1: I've tried JVoiceXML in the past and had some luck with it. http://jvoicexml.sourceforge.net/ It's java of course, but that wasn't a problem for my situation. 回答2: Voiceglue (http://www.voiceglue.org/) is an implementation of voicexml using openvxi and asterisk. It may