[2020-11-12 22:16:01] NOTICE res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/100-000001aa' for lack of RTP activity in 30 seconds
Normally, extension 100 would have signaled ‘hold’ on the incoming call by sending a re-invite telling Asterisk not to send it RTP. Asterisk knows that this means hold and would then allow 300 seconds of no RTP before disconnecting that leg. I don’t know why this didn’t happen, but posting a new log that includes a SIP trace will likely show the problem.
To do that, at the Asterisk command prompt, type
pjsip set logger on
then make your test call.
However, I assume that in this case the conversation between 100 and 107 lasted less than 30 seconds so the disconnect for lack of RTP should not have happened. Can you please confirm that? However, there is no evidence of 100 even attempting to complete the transfer so that’s a separate problem (though it may have the same cause).
Along with the new log, please post make, model and version of the device at extension 100, as well as what buttons were pressed to call 107 and what was pressed to attempt to complete the transfer.