SIP to PSTN gateway connection from asterisk?

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 hav开发者_开发百科e multiple clients registered to our system and cannot be sure of the number of simultaneous calls required.

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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.