12,270
edits
VoIP Phones - FreePBX: Difference between revisions
m
no edit summary
(Created page with "<indicator name="VoIPConfiguring">link=:Category:VoIP Phones|30px|Back up to the VoIP Configuring page</indicator> File:Freepbx_logo.png =Setting u...") |
mNo edit summary |
||
| (8 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 - FreePBX - 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
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==
/etc/asterisk/pjsip.transports_custom.conf
<syntaxhighlight lang=bash>
; Set the bind parameter to the address (and port)
; you want the trunk to listen on. I tend to use
type=transport
protocol=udp
bind=[2001:8b0:
allow_reload=no
tos=cs3
cos=3
</syntaxhighlight>
▲ Adding an IPV6 trunk via the Freepbx GUI
Create the new trunk as a normal ipv4 udp trunk using pjsip. On the
to contact you.
▲ And finally
Do to some horrendous interactions between the Freepbx dialplan
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]]
| |||