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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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Asterisk SIP Channels. The SIP Channel Module enables Asterisk to communicate via VoIP with SIP telephones and exchanges. Asterisk is able to act as: a SIP client: This means that Asterisk registers as a client to another SIP server and receives and places calls to this server.
A SIP VoIP gateway is a device that enables a user to place VoIP calls using session initiation protocol (SIP). SIP is a telephony technology developed as a fast, easy standard for delivering IP-based voice, data, and video communications.
Asterisk supports several standard voice over IP protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), and H. 323. Asterisk supports most SIP telephones, acting both as registrar and back-to-back user agent.
The gateway connects to the legacy system through either analog or digital trunk ports. The PBX sees the gateway as either the phone company or as another networked PBX. Calls from the PBX to the outside world are converted into VoIP calls and sent over the Internet to a VoIP service provider or other VoIP peers.
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.
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