VoIP Phones - FreePBX: Difference between revisions

← Older edit

VoIP Phones - FreePBX (view source)

Revision as of 19:19, 18 September 2022

11 bytes removed , 18 September 2022

m

no edit summary

AA-Andrew

autoreview, Bureaucrats, editor, Interface administrators, reviewer, Administrators

12,270

edits

Revision as of 16:13, 20 March 2018 (view source) AA-Andrew (talk \| contribs) mNo edit summary ← Older edit		Latest revision as of 19:19, 18 September 2022 (view source) AA-Andrew (talk \| contribs) mNo edit summary
(5 intermediate revisions by 2 users not shown)
<indicator name="VoIPConfiguring">[[File:menu-voip.svg\|link=:Category:VoIP Phones\|30px\|Back up to the VoIP Configuring page]]</indicator> [[Category:VoIP Phones\|FreePBX]]▼ [[File:Freepbx_logo.png]] There is an older guide if you are using chan_sip: [[~~VoIP_Phones_~~VoIP Phones -~~_FreePBX_~~ FreePBX -~~_chan_sip~~ chan sip]] =Setting up a A&A ipv6 trunk using pjsip via the Freepbx GUI= This guide works with Freepbx 14.0.1.36. Earlier versions may not work. Version 13 will certainly not work. ==Setting up the trunk on the A&A control pages== The key to success is that you _MUST*_ use the same password for both the incoming and outgoing directions, and for ~~the~~ the "To your server by sip" entry on the incoming page. If you do not do this you will find yourself in a bottomless pit of pain. ==Adding the neccessary additional conf files== type=transport protocol=udp bind=[2001:8b0:~~1418~~xxxx:xxxx::x]:xxxx allow_reload=no tos=cs3 [2001:8b0:0:30::5060:0]/125 this should allow all of the current servers to contact you. ==And finally== used by pjsip, it is essential that you fully restart the asterisk server, either by rebooting your box or by using systemd etc. ▲[[Category:VoIP Phones\|FreePBX]]