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 contact tell 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.
Any help regarding this would be greatly appreciated. Thanks in advance.