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VoIP Phones - FreePBX - chan sip: Difference between revisions

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<indicator name="VoIPConfiguring">[[File:menu-voip.svg|link=:Category:VoIP Phones|30px|Back up to the VoIP Configuring page]]</indicator>
[[File:Freepbx_logo.png]]
 
'''2016-11 The information below is for an older version of FreePBX - newer versions use 'pjsip' rather than 'chan_sip', see: [[VoIP Phones - FreePBX]] '''
 
*[[IPv6]] Works!
FreePBX is based on [[VoIP Phones - Asterisk|Asterisk]] - you may wish to read this page for more background information.
 
These instructions were last tested on FreePBX 212.0.76.2 - from the FreePBX Distro installer with Asterisk 11.019.230.
 
=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'''.
 
Fill out the fields as below:
 
===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.
* '''Disable Trunk:''' Make sure "Disable" is not ticked.
 
===Dialed Number Manipulation Rules===
I left this entire section alone.
 
===Outgoing Settings===
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.
 
'''Trunk Name:''' <code>out-01234567890</code><br />
'''PEER Details:''' <pre>host=voiceless.aa.net.uk
username=+441234567890▼
secret=YOUR-OUTGOING-PASSWORD-HERE▼
type=peer
▲username=+441234567890
insecure=invite</pre>
▲secretremotesecret=YOUR-OUTGOING-PASSWORD-HERE
transport=udp
disallow=all
allow=alaw
qualify=yes
</pre>▼
 
===Incoming Settings===
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.
 
 
'''USER Context:''' <code>in-01234567890</code>
'''USER Details:''' <pre>secrettype=YOUR-INCOMING-PASSWORD-HEREuser
type=peer
context=from-trunk
username=in-01234567890
insecure=invite</pre>
remotesecret=YOUR-INCOMING-PASSWORD-HERE
transport=udp
disallow=all
allow=alaw
trustrpid=yes</pre>
 
===Registration===
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.
 
If there is no NAT between your FreePBX and voiceless, you should not use registration if possible.
==Fix for incoming calls==▼
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.
 
More details can be found on the [[VoIP Phones - Asterisk]] article.
This can be fixed by manually adding some fixes to <code>/etc/asterisk/sip_custom_post.conf</code>:
<pre>
<nowiki>
; IPv4 voiceless addresses for 01234567890
[in-01234567890-a4](in-01234567890)
host=a4.voiceless.aa.net.uk
[in-01234567890-b4](in-01234567890)
host=b4.voiceless.aa.net.uk
[in-01234567890-c4](in-01234567890)
host=c4.voiceless.aa.net.uk
[in-01234567890-d4](in-01234567890)
host=d4.voiceless.aa.net.uk
[in-01234567890-e4](in-01234567890)
host=e4.voiceless.aa.net.uk
[in-01234567890-f4](in-01234567890)
host=f4.voiceless.aa.net.uk
[in-01234567890-g4](in-01234567890)
host=g4.voiceless.aa.net.uk
[in-01234567890-h4](in-01234567890)
host=h4.voiceless.aa.net.uk
[in-01234567890-i4](in-01234567890)
host=i4.voiceless.aa.net.uk
[in-01234567890-j4](in-01234567890)
host=j4.voiceless.aa.net.uk
 
▲==Fix forFixing incoming calls==
; IPv6 voiceless addresses for 01234567890
In order for FreePBX to recognise incoming calls from the voiceless.aa.net.uk platform, we must change a FreePBX setting:
[in-01234567890-a6](in-01234567890)
 
host=a6.voiceless.aa.net.uk
Go to '''Settings, Asterisk SIP Settings''', then click on '''Chan SIP'''.
[in-01234567890-b6](in-01234567890)
 
host=b6.voiceless.aa.net.uk
Scroll to the bottom of the page and find the '''Other SIP Settings''' entry.
[in-01234567890-c6](in-01234567890)
 
host=c6.voiceless.aa.net.uk
In the first field, enter <code>match_auth_username</code>, and in the second field enter <code>yes</code>.
[in-01234567890-d6](in-01234567890)
host=d6.voiceless.aa.net.uk
[in-01234567890-e6](in-01234567890)
host=e6.voiceless.aa.net.uk
[in-01234567890-f6](in-01234567890)
host=f6.voiceless.aa.net.uk
[in-01234567890-g6](in-01234567890)
host=g6.voiceless.aa.net.uk
[in-01234567890-h6](in-01234567890)
host=h6.voiceless.aa.net.uk
[in-01234567890-i6](in-01234567890)
host=i6.voiceless.aa.net.uk
[in-01234567890-j6](in-01234567890)
host=j6.voiceless.aa.net.uk
▲</pre>
 
[[File:Freepbx_other_sip_settings.png]]
This seems to keep incoming calls working OK.
 
Then click '''Submit Changes'''!
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>.
 
=Making IPv6 work=
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.
 
This can be fixed!
# 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]] and IPv4 addresses.
# Scroll to the bottom and click '''Submit Changes'''.
 
*Also see the [[VoIP Security]] page for information about securing your VoIP service.
 
 
[[Category:VoIP]]
[[Category:VoIP Phones|FreePBX]]
[[Category:VoIP IPv6]]
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