This is definitely doable, however, it seems that your specifications need to be modified a bit. Here are some things to consider:
What type of PSTN connectivity will your remote Asterisk server have? (SIP / POTS / T1 / PRI / etc.)
If your remote Asterisk server is going to be using a physical medium, do you have the connectors and hardware in place? EG: If you are using a T1 line, do you have a channel bank or T1 card?
Are you comfortable with Asterisk dialplan / AGI / AMI, or are you going to use an Asterisk distribution like trixbox, AsteriskNOW, Elastix, etc?
Will your client location (with the POTS line you wish to ring) have a PBX, or will it just be a typical POTS line hooked up to an analog handset?
My recommendation to you:
Get a cheap server (any 1U with a dual core processor and 512MB of RAM will do), and put it at your remote location.
Load Asterisk 1.6+ onto your server. I recommend 1.6+ as it can use the dahdi_dummy driver as a reliable software timing source (it will ensure that your audio quality is not choppy and broken).
Get a SIP account with a reliable SIP provider. My personal favorites are: flowroute and voipms.
Set up your new SIP account in Asterisk, and purchase a DID (phone number). This phone number will be your business phone number, the one that you give out to clients and put on business cards.
Configure your Asterisk dialplan to receive calls from your SIP account to your IVR menu.
Your IVR menu logic should be something like:
a. Play the IVR menu.
b. Wait for a keypress.
c. If the user dialed '1', then make an outgoing SIP call to the POTS line phone number you want to reach. If the user dialed '2', then playback the recorded message.
Now, if you are looking to save money, and have the most cost-effective setup for your remote IVR, I would recommend throwing up a second Asterisk server on site at your client location (where the POTS line comes in), and throw away the pots line and just setup an IAX2 trunk between your client location and your hosted server location. This way, when calls come in to your remote Asterisk server via your SIP provider, you can route the calls (when the user hits option 1) over your IAX2 trunk, directly to the client location for free!
Depending on your skill level, and comfort with Asterisk, this could be either a really fun learning experience or a confusing nightmare. If you would like to learn more about telephony and Asterisk, especially if you are going to use it for your business, you may want to use a simple (free) Asterisk distribution like: trixbox CE, Elastix, or AsteriskNOW.
You could code up a simple IVR in less time than it takes to install Asterisk if you use a hosted service like Twilio or Tropo. I'm partial to Twilio myself.
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这绝对是可行的，但是，似乎您的规格需要稍微修改一下。以下是需要考虑的一些事项：
我给您的建议是：
配置您的 Asterisk 拨号方案以接收从您的 SIP 帐户到 IVR 菜单的呼叫。
您的 IVR 菜单逻辑应该类似于：
a.播放 IVR 菜单。
b.等待按键。
c.如果用户拨打“1”，则向您想要联系的 POTS 线路电话号码拨打出局 SIP 电话。如果用户拨打“2”，则播放录制的消息。
现在，如果您想省钱，并为您的远程 IVR 设置最具成本效益的设备，我建议您在您的客户端位置（POTS 线路进入的地方）现场安装第二台 Asterisk 服务器，然后扔掉pots 线路，只需在您的客户端位置和托管服务器位置之间设置一条 IAX2 中继。这样，当呼叫通过 SIP 提供商进入远程 Asterisk 服务器时，您可以通过 IAX2 中继将呼叫（当用户点击选项 1 时）直接免费路由到客户端位置！
根据您的技能水平以及对 Asterisk 的熟悉程度，这可能是一次非常有趣的学习体验，也可能是一场令人困惑的噩梦。如果您想了解有关电话和 Asterisk 的更多信息，特别是如果您要将其用于您的业务，您可能需要使用简单（免费）的 Asterisk 发行版，例如： trixbox CE、Elastix 或 AsteriskNOW。
This is definitely doable, however, it seems that your specifications need to be modified a bit. Here are some things to consider:
My recommendation to you:
Configure your Asterisk dialplan to receive calls from your SIP account to your IVR menu.
Your IVR menu logic should be something like:
a. Play the IVR menu.
b. Wait for a keypress.
c. If the user dialed '1', then make an outgoing SIP call to the POTS line phone number you want to reach. If the user dialed '2', then playback the recorded message.
Now, if you are looking to save money, and have the most cost-effective setup for your remote IVR, I would recommend throwing up a second Asterisk server on site at your client location (where the POTS line comes in), and throw away the pots line and just setup an IAX2 trunk between your client location and your hosted server location. This way, when calls come in to your remote Asterisk server via your SIP provider, you can route the calls (when the user hits option 1) over your IAX2 trunk, directly to the client location for free!
Depending on your skill level, and comfort with Asterisk, this could be either a really fun learning experience or a confusing nightmare. If you would like to learn more about telephony and Asterisk, especially if you are going to use it for your business, you may want to use a simple (free) Asterisk distribution like: trixbox CE, Elastix, or AsteriskNOW.
如果您使用像 Twilio 或 Tropo。我自己就偏爱 Twilio。
编辑：这是一个简单的电话菜单示例。
You could code up a simple IVR in less time than it takes to install Asterisk if you use a hosted service like Twilio or Tropo. I'm partial to Twilio myself.
Edit: here's an example of a simple phone menu.
快速回答您的简单问题：
我认为你应该从 wiki 和 免费的 Asterisk 书
Quick answers to your simple questions:
I think you should start with wiki and free Asterisk book