currently I`m trying to setup a new SIP-Trunk for provider fonial.de.
They have posted general Asterisk-Settings on their page:
Register-String: SIP-User:[email protected]/SIP-User
For sip.conf the suggest the following settings:
; SIP-User for Registration at fonial-Trunk
[fo279XXXtr24XXXtr_01] type=friend qualify=yes nat=force_rport,comedia dtmfmode=RFC2833 insecure=port,invite canreinvite=no secret=passwort username= fo279XXXtr24XXXtr_01 fromdomain=sip.solucon.com outboundproxy=proxy01.sip.solucon.com host=sip.solucon.com disallow=all allow=ulaw allow=alaw
For the phone they suggest:
; For any phone that is registred on your Asterisk:
username=phone ; User, set in the phone.
secret=123456 ; Password should be choosen by yourself
context=dialout ; Change context according to extensions.conf
I have set-up in FreePBX-GUI:
Unfortunately it didn`t work with the settings I have put in the GUI.
Could someone maybe help me converting the general Asterisk-Setting to a FreePBX-Version?