We do not require you to send any 00 nor 011 code for international calls, we just need to receive the country code followed by the number, please check the examples below
To call AVOXI USA Support number
To Call AVOXI Costa Rica Support number (Costa Rica country code is 506)
Add PJ_SIP Trunk
To do this, go to Connectivity > Trunks. Click Add Trunk and select "Add SIP (chan_pjsip) Trunk". In the General tab, define the Trunk name (can be anything you want). In the pjsip settings, change the Authentication to None (or change it to Outbound and define a Username + Secret to use with the SIP Trunking section in Genius), change Registration to None, and define the SIP Server + Port shown in the screenshot. Also navigate to the advanced section of the PJSIP Trunk and add in the Match (Permit) section: 126.96.36.199/32, 188.8.131.52/32
The default transport set is UDP. If you prefer TCP connection, you will need to go to Settings > Asterisk SIP Settings > SIP Settings (chan_pjsip). Under Transports, you can enable TCP and TLS (we do not use ws or wss).
If you are setting up TLS with SRTP on FreePBX, please use this guide to do so: https://wiki.freepbx.org/display/phon/tls+and+srtp. Also refer to the Genius SIP Trunking for Outbound Termination guide for the files necessary from us: https://support.avoxi.com/153319-sip-configuration-guides/configuring-sip-trunks-in-genius#downloading-sip-uri-tls-certificate-4
Add Inbound Routes
Here is where you define what you want calls inbound to your PBX to do. This can be found via Connectivity > Inbound Routes. If you have multiple DIDs and need them to do specific things, you'd define a single DID in the DID Number section per Inbound Route.
Add Outbound Routes
Here is where you define what you want calls initiated by your extensions to do. This is useful for if you have multiple trunks going to different providers or you want certain calls recorded locally (Additional Settings > Call Recording) or if you want calls placed intra-company to use different Caller ID settings.
For info on setting up SIP URIs and SIP Trunks in Genius, refer to the below section: