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The default behavior of FreePBX, starting at version 12, is to use chan_pjsip for endpoints and trunks. Sections of this article will cover installations of FreePBX configured with either chan_pjsip or chan_sip. Selection of either chan_pjsip or can_sip from within your distribution can be found in the Admin Web tool under Settings -> Advanced Settings -> Dialplan and Operational -> SIP Channel Driver. The Admin Web tool may be viewed by visiting: http://[ip of your freepbx]/admin
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General Settings
Trunk Name: digium-siptrunk
Outbound CallerID: your_digium_number, e.g. 2565551234
CID Options: Force Trunk CID
Dialed Number Manipulation Rules (This entire section can be left at defaults)
Outgoing Settings
Trunk Name: digium-siptrunk
PEER Details:
type=peer
directmedia=no
host=sip.digiumcloud.net
defaultuser=your_digium_username
fromuser=your_digium_username
secret=your_digium_password
insecure=invite
trustrpid=yes
sendrpid=pai
disallow=all
allow=g722
allow=ulaw
allow=g729
session-timers=refuse
G729 should typically only be allowed if you've installed Digium's G.729 Add-on for Asterisk. G.729 is a licensed algorithm that cannot be distributed or used freely without this add-on.G.729 should be used on
Incoming Settings (USER Details: blank out this section)
Registration
Register String: your_digum_username:your_digium_password@sip.digiumcloud.net:5060/your_digium_username
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