[Freeswitch-users] Modems

[Tim St. Pierre]

The problem mainly is as I said, I cannot figure how VOIP -> PSTN bridging can be achieved whatever
the gateway device.
> 
> A SIP client (VOIP phone, softphone etc) uses only the SIP URI to make the call. From that piece of data it gets the SIP domain and the ID of the user. So how is the gateway able to be inserted in this process. The SIP client has no knowledge of this device and no way to include that third piece of data into its process.
> 
> Obviously such gateways do exist, but I do not yet understand how the process works. Some advice on the basic process flow would therefore assist me to set up what I need irrespective of what devices I am using. Sorry to be so ignorant about this, but anyone able and willing to help with an explanation?
>

Here's a simple answer:

Usually there is some sort of intermediary to help with routing, but purely from the gateway's
perspective:

The device listens for SIP signalling

A call comes in to the domain or IP address of the gateway with the destination number as the user
part of the SIP uri (the part before the @ sign).

The gateway interprets the user part as the PSTN number, subject to any internal rules it has about
adding or removing prefixes, etc.

The gateway then picks up the line and places the call, sending back SIP messages as to the status.

Here's a link that brings up a PDF document explaining SIP to PSTN call flow.

http://www.eventhelix.com/EventStudio/telecom_message_sequence_charts/SIP_PSTN_Call_Flow.pdf