Using FreePBX 17 with chan_sip
- Lorne Gaetz (Unlicensed)
The Asterisk has long marked the deprecation of the chan_sip channel driver and Asterisk 21 does not support chan_sip at all. It’s also deprecated for FreePBX.
Nobody should be using chan_sip at this point, there are tools available to convert your chan_sip trunks and extensions to pjsip.
FreePBX 17.0 still has the legacy code to support chan_sip, and the project will take steps going forward to not break anything, but the project is no longer testing and are certainly not writing or accepting new contributions to support chan_sip
Using chan_sip is strongly discouraged.
Steps to enable chan_sip support in FreePBX 17
Start with a new FreePBX 17 install FreePBX 17 Installation
Post install downgrade asterisk to version 20 or earlier using the asterisk-version-switch script Changing Major Asterisk Versions on the Fly. Allow the downgrade process to complete.
In the FreePBX admin GUI, browse to Settings, Advanced Settings and locate the “SIP Channel Driver” option. In the drop down menu select ‘both’ and submit the change.
Browse to Settings, Asterisk SIP settings and review the pjsip and chan_sip settings. Ensure that bind port settings between the two drivers and transports are correct and, if making changes, ensure there are no port conflicts. Review and update any other SIP config changes at this point as well.
Press the red Apply Config button and wait for it to complete.
Restart Asterisk (or restart the system).
You can confirm that chan_sip has been enabled from the asterisk CLI:
*CLI> sip show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5160 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled RTP Bindaddress: Disabled * snip *At this point, you can create/edit chan_sip extensions and trunks as has historically been done in previous versions. Restoring a backup with chan_sip settings should result in those chan_sip extensions and/or trunks.