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VoIP Phones - FreePBX - chan sip: Difference between revisions
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==Setting up your A&A trunk==
===Configuration in FreePBX Web UI===
Log into your FreePBX administration interface and go to Connectivity, Trunks - then select "Add SIP Trunk".
====General Settings====
* '''Trunk Name:''' A descriptive name for the trunk - enter whatever you wish.
* '''Outbound CallerID:''' The caller ID you will use for calls made on this trunk. I entered my phone number in the format: 01234567890.
* '''CID Options:''' Set this as you prefer. I left it set to "Allow Any CID".
* '''Maximum Channels:''' Leave this blank.
* '''Asterisk Trunk Dial Options:''' Leave this blank and make sure "Override" isn't ticked.
* '''Continue if Busy:''' Set this as you prefer.
* '''Disable Trunk:''' Make sure "Disable" is not ticked.
====Dialed Number Manipulation Rules====
This can be fixed!
# Log into the FreePBX administration UI.
# Go to '''Settings, Asterisk SIP Settings'''.
# Scroll down to '''Advanced General Settings'''.
# Change '''Bind Address''' to an [[IPv6]] Address. <code>::</code> can be used for binding to all available [[IPv6]] addresses.
# Scroll to the bottom and click '''Submit Changes'''.
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