ivr

问题 I want to create a custom element that sends DTMF according to my logic . I can file play .wav files of dtmf tones but there is a better solution i believe. How can i use vxml feature below to create an element for on call studio.Can you share a document or code block on how to do it? 来源： https://stackoverflow.com/questions/65658565/how-to-create-custom-send-dtmf-element-on-cvp-cisco

来源： https://stackoverflow.com/questions/37114632/can-arduino-tell-room-temperature-by-voice-call-using-sim900

来源： https://stackoverflow.com/questions/37114632/can-arduino-tell-room-temperature-by-voice-call-using-sim900

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

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

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

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

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

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