VoIP Phones - FreePBX - chan sip: Difference between revisions

These instructions were last tested on FreePBX 2.11.0.23.

 

Log into your FreePBX administration interface and go to Connectivity, Trunks - then select "Add SIP Trunk".

 

Fill out the fields as below:

 

* '''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.

* '''Disable Trunk:''' Make sure "Disable" is not ticked.

 

I left this entire section alone.

 

These are your outgoing call settings - for calls you make from your phone through voiceless. These settings can be found under the "SIP Phone" heading in clueless for your number.

 

insecure=invite</pre>

 

These are your incoming call settings - for calls you receive from voiceless. These settings can be found under the "To your server via SIP" heading in clueless for your number.

 

insecure=invite</pre>

 

If your FreePBX is behind a NAT you may need to enter a registration string here. More details can be found on the [[VoIP_Phones_-_Asterisk]] article.

 

After using the above for a while, you may well find that your incoming calls randomly stop working - this is because they are being rejected by Asterisk as it does not recognise the incoming calls properly.

 

You will need to force Asterisk to reload the configuration files for this to take effect - this can be done by restarting Asterisk or by running <code>asterisk -rx 'reload'</code>.

 

You may notice that out of the box your FreePBX install will not talk [[IPv6]]. This is because by default it configures Asterisk to only listen on IPv4.

 

# Scroll to the bottom and click '''Submit Changes'''.

 

For best results, your FreePBX install should be set up to use only alaw audio.

 

# In the list of codecs, untick every codec except alaw.

# Scroll to the bottom and click '''Submit Changes'''.