I am wanting to convert over to Asterisk 13 and PJSIP but I can’t seem to translate the SIP Trunk settings to a PJSIP Trunk that would actually register and take and make calls - Here is what I currently use in SIP:
Trunk Name: BluIP-Out
I have read the Digium instructions but the translation into FreePBX just never registers - Has anyone else got this working? I don’t think our SIP trunks are weird but maybe they are.
And no, I can’t use SIP for the trunks in 13 - we get audio drops (they are using a Broadsoft - has anyone else seen that problem also?).