How to connect the trunk
- Get the server address, username and password from the provider (or agree on IP authentication and give them the PBX public address).
- Asterisk: add the blocks to /etc/asterisk/pjsip.conf and extensions.conf, run pjsip reload and dialplan reload, check pjsip show registrations.
- FreeSWITCH: put the gateway into sip_profiles/external, run sofia profile external rescan and sofia status gateway provider.
- Check the provider server with our SIP check, and if something fails paste the log into the SIP log analyzer.
Common problems
- 401/403 on registration: wrong auth username or From domain. Many providers need From user to match the number.
- No audio or one-way audio: the PBX is behind NAT without external_media_address, or the router runs SIP ALG. Turn ALG off (on MikroTik use our generator) and open the RTP ports.
- Inbound calls do not arrive: the call came from an IP missing in identify, or the number in the Request-URI has another format (with “+”, without the country code).
- 488 Not Acceptable: no common codec. Providers usually accept G.711 A-law (PCMA).
How many channels you need
The number of lines for a given load comes from the Erlang calculator, bandwidth per call from the bandwidth calculator.