Does ENUM trunks in FreePBX relies on the availability of chan_sip ?
I ask this because I had a FreePBX 15 system were I use an ENUM trunk to connect to a remote institute.
After migrated that system to FreePBX 17 with Asterisk 20 and chan_sip disabled, I was unable to communicate with that trunk! Looking at Asterisk logs I’ve seen this:
channel.c: No channel type registered for 'sip'
app_dial.c: Unable to create channel of type 'sip' (cause 66 - Channel not implemented)
Only after enabling chan_sip I was able to restore the calls with that trunk.
As this can not be a long term solution, because chan_sip was removed in Asterisk 21, I was wondering if anyone was able to use ENUM trunks with chan_pjsip?
Log lines do indicate that enum module relies on sip. I haven’t looked at an enum trunk for over a decade, so going on old memories.
If I recall correctly, if the enum lookup returns a string in the form of a sip uri, there is dialplan to extract the dial portion, so this would have to be updated to use pjsip. You can create a ticket with a full call trace of an enum call that uses sip, but I suspect it will get a low priority. This is like the third time I’ve encountered it in 15 years.
