从 asterisk 到 PSTN 网关的 SIP 连接？
我们正在开发一款网络电话应用程序，它可以向其他设备拨打 sip 电话，也可以拨打 PSTN 电话。我们使用 Asterisk 1.8 作为我们的 sip 服务器。来自网络电话的 SIP 呼叫工作正常。
我们希望能够向客户提供 SIP 到 PSTN 的呼叫服务，因此需要连接到 PSTN VOIP 网关。我们的系统仅需要拨出（SIP 到 PSTN）呼叫。我的问题是，是否有提供此类服务的公司可以为我们提供与网关的连接，将呼叫路由到 PSTN，而没有同时呼叫数量的限制。我联系过的所有公司都告诉我有关具有固定端口数量的 SIP 中继。我们计划让多个客户注册到我们的系统，但无法确定所需的同时呼叫数量。
We are working on a web phone application that can make sip calls to other devices and make PSTN calls as well. We use Asterisk 1.8 as our sip server. The SIP calls from the web phone is working fine.
We want to be able to provide SIP to PSTN calling service to our clients and thus require to connect to a PSTN VOIP Gateway. Only outgoing (SIP to PSTN) calls are required for our system. My question is, are there companies that provide such services which can provide us with connection to the gateway to route calls to PSTN without any limit for the number of simultaneous calls. All the companies I had contacted told me about SIP Trunking with a fixed number of ports. We plan to have multiple clients registered to our system and cannot be sure of the number of simultaneous calls required.
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我知道这是一个老问题，但我在寻找一些东西，发现这个问题没有得到正确的回答。
所有告诉您获取 SIP 中继的供应商都是正确的。即使您不知道需要多少个同时通话，但这也没关系。端口数（如果您可能最终使用的并发呼叫数）。您需要向他们提供这个号码，以确保他们可以为您提供这些号码。
Sip 中继基本上是可以通过 PSTN 网络进行呼叫的 SIP 线路。
I know this is an old question, but I was searching for something and saw this hadnt been answered properly.
All the vendors telling you to get a SIP trunk are correct. Even though you dont know how many simultaneous calls you need, that doesnt matter. The number of ports if the number of simultaneous calls you May end up using. This is a number you need to provide them for ensuring they can dedicate those for you.
Sip Trunks are basically SIP lines that can call over the PSTN network.