We are working on a web phone application that can make calls to other devices and make calls using PSTN. We use Asterisk 1.8 as our sip server. SIP calls from a web phone work fine.
We want to be able to provide SIP services for PSTN calls to our customers, and therefore you need to connect to the PSTN VOIP gateway. Our system requires only outgoing (SIP for PSTN) calls. My question is: are there companies providing such services that can provide us with a gateway connection for routing calls to PSTN without any restrictions on the number of simultaneous calls. All the companies I contacted told me about SIP Trunking with a fixed number of ports. We plan to have several clients registered in our system and cannot be sure of the number of simultaneous calls required.
source
share