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Is there a way to not have to path and build whole Asterisk 13?
I've tried proposed procedure in an already installed Asterisk 13 on Debian 9 (from repositories); it builds and installs a new Asterisk 13 and it's not starting

Jul 31 09:56:50 hostname systemd[1]: asterisk.service: Start operation timed out. Terminating.
Jul 31 09:56:50 hostname asterisk[22230]: 
Jul 31 09:56:50 hostname systemd[1]: Failed to start Asterisk PBX.
Jul 31 09:56:50 hostname systemd[1]: asterisk.service: Unit entered failed state.
Jul 31 09:56:50 hostname systemd[1]: asterisk.service: Failed with result 'timeout'.

When Asterisk started the call but the remote party sends a re-INVITE on the SIP layer, the call might face one-way audio. One such a call scenario is call hold, for example. One-way audio happens because this codec here uses a dynamic RTP payload type in SDP negotiation, which Asterisk 13 LTS does not support correctly. Complete details are documented in ASTERISK-27056. Even if you are not affected, applying the patch provided there is recommended for any user. If you face an issues with that patch, report in ASTERISK-27056, please. If possible, consider upgrading to a newer Asterisk branch.