Honestly, Asterisk is not very difficult to set up and get running for something like this. Get an inexpensive and unlimited inbound SIP channel from Callcentric or another trunk provider, and write your IVR as the default extension in the dialplan.
Don't let yourself get overwhelmed. You only need to define your trunk as a SIP peer in sip.conf to get the box talking to your provider, and the dialplan syntax is very, very simple. You don't really need any other configuration for Asterisk itself, unless you have to use a land line, which would require a digium or sangoma board with fxo ports on it, and configuration of the zaptel drivers within asterisk.
Text to speech isn't likely to be as great as you'd like, but it may not be feasible to record all of the audio files that you'd need. The problem is that cloud IVR providers may not be cost effective for your application.
Any other questions, please don't hesitate to ask.
如果您不想使用 Asterisk，还有其他基于网络的选项。我最了解的（因为我在那里工作）是 Twilio，它可以让您像构建 Web 应用程序一样构建电话应用程序。它比设置和托管您自己的 Asterisk 实例以及处理运营商连接要简单得多。还有其他类似的服务。
If you don't want to deal with Asterisk, there are other web-based options as well. The one I know the best (because I work there) is Twilio which lets you build phone applications the same way you build web applications. It's much simpler than setting up and hosting your own Asterisk instance and dealing with carrier connections. There are other similar services out there as well.
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您正在寻找的是交互式语音应答系统（IVR）。现有的系统可以针对单独的应用程序进行编程。
W3C甚至有一个名为VoiceXML的标准，它是语音系统和计算机之间交互的标准。
不幸的是，这就是我对该主题的了解程度。
What you are looking for is an Interactive Voice Response system (IVR). There are existing systems that can be programmed for individual applications.
The W3C even has a standard called VoiceXML, which is a standard for interaction between voice systems and computers.
Unfortionately, that is the extent of my knowledge on the topic.
老实说，Asterisk 的设置和运行并不是很难。从 Callcentric 或其他中继提供商处获取廉价且无限制的入站 SIP 通道，并将您的 IVR 写入拨号方案中作为默认分机。
不要让自己不知所措。您只需在 sip.conf 中将您的中继定义为 SIP 对等点即可让盒子与您的提供商通话，并且拨号方案语法非常非常简单。 Asterisk 本身并不需要任何其他配置，除非您必须使用陆线，这需要带有 fxo 端口的 digium 或 sangoma 板，并在 asterisk 内配置 zaptel 驱动程序。
文本转语音功能可能不会像您希望的那么好，但录制您需要的所有音频文件可能并不可行。问题是云 IVR 提供商对于您的应用程序来说可能不具有成本效益。
如有任何其他问题，请随时询问。
Honestly, Asterisk is not very difficult to set up and get running for something like this. Get an inexpensive and unlimited inbound SIP channel from Callcentric or another trunk provider, and write your IVR as the default extension in the dialplan.
Don't let yourself get overwhelmed. You only need to define your trunk as a SIP peer in sip.conf to get the box talking to your provider, and the dialplan syntax is very, very simple. You don't really need any other configuration for Asterisk itself, unless you have to use a land line, which would require a digium or sangoma board with fxo ports on it, and configuration of the zaptel drivers within asterisk.
Text to speech isn't likely to be as great as you'd like, but it may not be feasible to record all of the audio files that you'd need. The problem is that cloud IVR providers may not be cost effective for your application.
Any other questions, please don't hesitate to ask.
如果您不想使用 Asterisk，还有其他基于网络的选项。我最了解的（因为我在那里工作）是 Twilio，它可以让您像构建 Web 应用程序一样构建电话应用程序。它比设置和托管您自己的 Asterisk 实例以及处理运营商连接要简单得多。还有其他类似的服务。
If you don't want to deal with Asterisk, there are other web-based options as well. The one I know the best (because I work there) is Twilio which lets you build phone applications the same way you build web applications. It's much simpler than setting up and hosting your own Asterisk instance and dealing with carrier connections. There are other similar services out there as well.