Jump to content

This is the support site for Andrews & Arnold Ltd, a UK Internet provider. Information on these pages is generally for our customers but may be useful to others, enjoy!

VoIP Phones - FreePBX: Difference between revisions

Content deleted Content added
AA-Andrew (talk | contribs)
AA-Andrew (talk | contribs)
mNo edit summary
 
(3 intermediate revisions by 2 users not shown)
Line 1: Line 1:
<indicator name="VoIPConfiguring">[[File:menu-voip.svg|link=:Category:VoIP Phones|30px|Back up to the VoIP Configuring page]]</indicator>
<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]]
[[File:Freepbx_logo.png]]


*There is an older guide if you are using chan_sip: [[VoIP_Phones_-_FreePBX_-_chan_sip]]
*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=
=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.
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==
==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 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
the incoming and outgoing directions, and for the "To your server by
sip" entry on the incoming page. If you do not do this you will find
sip" entry on the incoming page. If you do not do this you will find
yourself in a bottomless pit of pain.
yourself in a bottomless pit of pain.



==Adding the neccessary additional conf files==
==Adding the neccessary additional conf files==
Line 25: Line 24:


/etc/asterisk/pjsip.endpoint_custom_post.conf
/etc/asterisk/pjsip.endpoint_custom_post.conf
<syntaxhighlight lang=bash line>
<syntaxhighlight lang=bash>
[anonymous](+)
[anonymous](+)
;
;
Line 74: Line 73:
[2001:8b0:0:30::5060:0]/125 this should allow all of the current servers
[2001:8b0:0:30::5060:0]/125 this should allow all of the current servers
to contact you.
to contact you.



==And finally==
==And finally==
Line 82: Line 80:
used by pjsip, it is essential that you fully restart the asterisk
used by pjsip, it is essential that you fully restart the asterisk
server, either by rebooting your box or by using systemd etc.
server, either by rebooting your box or by using systemd etc.




[[Category:VoIP Phones|FreePBX]]