i know sip was removed in asterisk 21 use call manager said it uses a modern version of the driver, i dont know how to make that work for my cisco phones
The USECALLMANAGER patch for Asterisk 22 adds that channel driver back in, modified for Cisco phones, see:
Install Debian 12, then download and build Asterisk with the patch, then install FreePBX 17 with the option to use your own Asterisk
ok will do thanks much!!
So the driver option should just show up for me to add my phones just like version 20?
ok i tried it and i still cant connect to SIP
When you use chan_sip instead of pjsip, you have to check the port in asterisk-sip-settings.
FreePBX 17 auto-detects the Asterisk version in use on the system and will not display the Add Extension button for chan_sip by default. If you go into the Settings and enable Advanced Settings, then go into Advanced Settings, you can set the Asterisk version to 20, and then FreePBX will display the additional buttons to allow you to add chan_sip, as well as the additional buttons that allow you to configure ports and so on in chan_sip
Upgrading to Asterisk 22 is mostly smooth, but SIP changes can catch people off guard. I’ve seen setups break simply because legacy chan_sip configs were still assumed to work. Moving fully to PJSIP and double-checking endpoint, auth, and transport sections usually fixes weird behavior. Testing extensions one by one after the patch saved me a lot of guessing and rollback stress.
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