Connecting the 25-Line account to Asterisk or FreePBX

The 25-Line account registers cleanly as a SIP trunk on Asterisk-based systems, including FreePBX.

In FreePBX you create a new chan_pjsip (or chan_sip) trunk with: - SIP server / proxy: sip.transcom.net - Username / Auth: your account user ID - Password: your account password - Register / signalling port: 5060 (UDP) or 5061 (TLS) - Codecs: allow G.711u (PCMU), G.729, G.722

Your dial plan then sends outbound calls over the trunk and routes inbound DIDs to extensions, IVRs or queues on your PBX. Because all twenty-five channels ride the one trunk, there is nothing to license per user on the FreeVoIP side — your PBX handles the extensions and the account handles the concurrency.