After applying the Usecallmanager patch to Asterisk v20, there are some settings need to be setup in FreePBX v17.
First, login to the GUI of the FreePBX box and go to Settings -> Advanced Settings, at the top most section, you will see this:
Chanage both "Display Readonly Settings" and "Override Readonly Settings" to YES.
Down below this screen, look for "SIP Channel Driver" and change to Both like the screen shot below:
Click "Submit" and then the red button at top right corner to "Apply".
Now, the Chan_Sip settings will be available. Then, under FreePBX v17 GUI, go to Settings -> Asterisk SIP Settings, you will see three tabs: "General SIP Settings" ; "SIP Settings [chan_pjsip]" ; "SIP Legacy Settings [chan_sip]". The one we concern most is "SIP Legacy Settings [chan_sip]". Thus, click on this tab.
We can use this page to input some settings that is required by Usecallmanger patch. Go to the bottom of the screen and set "Enable TCP" to YES; "Call Events" to YES. Also, input the extra settings like the picture below:
Settings in the above screen capture is actually setting of : https://usecallmanager.nz/sip-conf.html#Extension-Template
Not all the extra settings shown above are related to Usecallmanger Patch, which
are applicable if you need to support sending text messages between Chan_Sip extensions. That is useful for SoftPhones I guess. Thus, it is up to you whether you need these three settings or not.
All the settings we set are actually written in the "sip_general_additional.conf" file. After you Sumbit and Apply the changes, you can check the updated .conf file from the FreePBX GUI: Admin -> Config Edit; under the section "Asterisk System Configuration Files" which should be at the lower part of the file list. Searh the file name "sip_general_additional.conf", you can check the settings there. Remember that this file is not editable here as it is generated by the FreePBX GUI each time you make changes on the GUI.
For the explanation of the settings, you can refer to https://usecallmanager.nz .
It is pretty much for the general chan_sip setup for the Cisco Phones.
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