PJSIP Encryption not working with audio

Hi. I recently installed and configured Asterisk 17.9.0. Once I got the basics down, I moved on to TLS encryption. This was a bit more difficult though.

I got the TLS encryption working for the most part. I was able to log in with my SIP Client (MicroSIP) and place calls. I tested it out by calling the extension 1000 (the demo extension). Without the TLS encryption enabled, I was able to hear what the demo extension was saying. When I enabled TLS, I heard nothing at all. I was connected to the call, but I heard nothing.

I’m still fairly new to Asterisk so I’m not sure which logs I’m supposed to put here, so please ask if you would like to see any specific logs from the console.

pjsip.conf:

[tls-transport]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
method=tlsv1
verify_client=no
verify_server=no

[endpoint_internal](!)
type=endpoint
context=from-internal
allow=all
transport=tls-transport
media_encryption=sdes
dtmf_mode=rfc4733

Hey, in case someone wanted to see the full configuration I had on the pjsip.conf file, here’s it.

[tls-transport]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
method=tlsv1
verify_server=no
verify_client=no

[endpoint_internal](!)
type=endpoint
context=from-internal
allow=g722
media_encryption=no

[auth_userpass](!)
type=auth
auth_type=userpass

[aor_dynamic](!)
type=aor
max_contacts=5

;Definitions for our phones, using the templates above

[john](endpoint_internal)
auth=john
aors=john
[john](auth_userpass)
password=realpasswordgoeshere
username=john
[john](aor_dynamic)

[joe](endpoint_internal)
auth=joe
aors=joe
[joe](auth_userpass)
password=realpasswordgoeshere
username=joe
[joe](aor_dynamic)

I changed up some things from earlier. Keep in mind that enabling media_encryption and disabling it changed nothing.

In other posts, I see people usually asking for a debug log.
This is the debug log for when I called the 1000 extension. I called, I received no audio, and then I hung up a few seconds later. Keep in mind that hanging up isn’t the issue; receiving audio is.

[Nov 22 22:51:40] DEBUG[29218]: res_pjsip/pjsip_distributor.c:394 find_dialog: Could not find matching transaction for Request msg BYE/cseq=4912 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[29218]: res_pjsip/pjsip_distributor.c:394 find_dialog: Could not find matching transaction for Request msg INVITE/cseq=10622 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[29218]: res_pjsip/pjsip_distributor.c:472 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000027 to use for Request msg INVITE/cseq=10622 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[29215]: threadpool.c:536 grow: Increasing threadpool pjsip/pool's size by 5

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username 'john' domain '[REDACTED DOMAIN]'

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username 'john' domain '[REDACTED DOMAIN]'

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_authenticator_digest.c:454 digest_check_auth: Using default realm 'asterisk' on incoming auth 'john'.

[Nov 22 22:51:42] DEBUG[29218]: res_pjsip/pjsip_distributor.c:394 find_dialog: Could not find matching transaction for Request msg ACK/cseq=10622 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[29218]: res_pjsip/pjsip_distributor.c:472 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000027 to use for Request msg ACK/cseq=10622 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username 'john' domain '[REDACTED DOMAIN]'

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username 'john' domain '[REDACTED DOMAIN]'

[Nov 22 22:51:42] DEBUG[29218]: res_pjsip/pjsip_distributor.c:394 find_dialog: Could not find matching transaction for Request msg INVITE/cseq=10623 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[29218]: res_pjsip/pjsip_distributor.c:472 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000027 to use for Request msg INVITE/cseq=10623 (rdata0x7f66e0047438)

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username 'john' domain '[REDACTED DOMAIN]'

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username 'john' domain '[REDACTED DOMAIN]'

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_authenticator_digest.c:454 digest_check_auth: Using default realm 'asterisk' on incoming auth 'john'.

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_authenticator_digest.c:259 check_nonce: Calculated nonce 1606099902/eed686eb0d7908bb8e83014d966cbbfe. Actual nonce is 1606099902/eed686eb0d7908bb8e83014d966cbbfe

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip/pjsip_distributor.c:472 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000027 to use for Request msg INVITE/cseq=10623 (rdata0x7f66e0057958)

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:3953 new_invite:  john

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4040 new_invite:  john: Call (TLS:[REDACTED IP ADDRESS OF USER]:63709) to extension '1000' sending 100 Trying

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4481 handle_outgoing_response:  john: Method is INVITE, Response is 100 Trying

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4497 handle_outgoing_response:  john

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4619 session_inv_on_state_changed: john: Source of transaction state change is TX_MSG

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:765 handle_incoming_sdp:  john: Media count: 1

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:493 ast_sip_session_media_state_add:  john Adding position 0

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:539 ast_sip_session_media_state_add:  Creating new media session

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:580 ast_sip_session_media_state_add:  Setting media session as default for audio

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:585 ast_sip_session_media_state_add:  Done

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:526 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7f66d001c770'

[Nov 22 22:51:42] DEBUG[32553]: res_rtp_asterisk.c:3785 rtp_allocate_transport: Allocated port 18422 for RTP instance '0x7f66d001c770'

[Nov 22 22:51:42] DEBUG[32553]: res_rtp_asterisk.c:3815 rtp_allocate_transport: Creating ICE session [::]:18422 (18422) for RTP instance '0x7f66d001c770'

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:543 ast_rtp_instance_new: RTP instance '0x7f66d001c770' is setup and ready to go

[Nov 22 22:51:42] DEBUG[32553]: res_rtp_asterisk.c:8190 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7f66d001c770'

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1319 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 9 based on m type on 0x7f667e1b8320

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1283 ast_rtp_codecs_payloads_xover: Crossover copying tx to rx payload mapping 9 (0x7f66d0084df8) from 0x7f667e1b8320 to 0x7f667e1b8320

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1283 ast_rtp_codecs_payloads_xover: Crossover copying tx to rx payload mapping 101 (0x7f66d0084f58) from 0x7f667e1b8320 to 0x7f667e1b8320

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1120 rtp_codecs_payloads_copy_rx: Copying rx payload mapping 9 (0x7f66d0084df8) from 0x7f667e1b8320 to 0x7f66d001c948

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1120 rtp_codecs_payloads_copy_rx: Copying rx payload mapping 101 (0x7f66d0084f58) from 0x7f667e1b8320 to 0x7f66d001c948

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1205 rtp_codecs_payloads_copy_tx: Copying tx payload mapping 9 (0x7f66d0084df8) from 0x7f667e1b8320 to 0x7f66d001c948

[Nov 22 22:51:42] DEBUG[32553]: rtp_engine.c:1205 rtp_codecs_payloads_copy_tx: Copying tx payload mapping 101 (0x7f66d0084f58) from 0x7f667e1b8320 to 0x7f66d001c948

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:957 handle_incoming_sdp:  john: Handled? yes

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:5053 create_local_sdp:  john

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:5090 create_local_sdp:  john: Processing streams

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:493 ast_sip_session_media_state_add:  john Adding position 0

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:501 ast_sip_session_media_state_add:  Using existing media_session

[Nov 22 22:51:42] DEBUG[32553]: res_rtp_asterisk.c:8089 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f66d001c770'

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:5147 create_local_sdp:  john: Adding bundle groups (if available)

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:5153 create_local_sdp:  john: Copying connection details

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:5197 create_local_sdp:  john

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4377 handle_incoming_request:  john: Method is INVITE

[Nov 22 22:51:42] DEBUG[32553]: chan_pjsip.c:3055 chan_pjsip_incoming_request:  john

[Nov 22 22:51:42] DEBUG[32553]: stasis.c:575 stasis_topic_create_with_detail: Creating topic. name: channel:1606099902.16, detail:

[Nov 22 22:51:42] DEBUG[32553]: stasis.c:609 stasis_topic_create_with_detail: Topic 'channel:1606099902.16': 0x7f66d0029c60 created

[Nov 22 22:51:42] DEBUG[29193]: threadpool.c:536 grow: Increasing threadpool stasis/pool's size by 1

[Nov 22 22:51:42] DEBUG[32553]: channel.c:951 __ast_channel_alloc_ap: Channel 0x7f66d00276a0 'PJSIP/john-00000008' allocated

[Nov 22 22:51:42] DEBUG[32553]: chan_pjsip.c:3102 chan_pjsip_incoming_request:  PJSIP/john-00000008

[Nov 22 22:51:42] DEBUG[32553]: chan_pjsip.c:3183 pbx_start_incoming_request: Started PBX on new PJSIP channel PJSIP/john-00000008

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4387 handle_incoming_request:  PJSIP/john-00000008

[Nov 22 22:51:42] DEBUG[32553]: res_pjsip_session.c:4088 new_invite:  PJSIP/john-00000008

[Nov 22 22:51:42] DEBUG[32559][C-00000009]: pbx.c:2938 pbx_extension_helper: Launching 'Goto'

[Nov 22 22:51:42] DEBUG[32559][C-00000009]: pbx_lua.c:1497 lua_find_extension: Looking up s@default:1

[Nov 22 22:51:42] DEBUG[32559][C-00000009]: pbx_lua.c:1497 lua_find_extension: Looking up s@demo:1

[Nov 22 22:51:43] DEBUG[32559][C-00000009]: pbx_lua.c:1497 lua_find_extension: Looking up s@default:1

[Nov 22 22:51:43] DEBUG[32559][C-00000009]: pbx_lua.c:1497 lua_find_extension: Looking up s@demo:1

[Nov 22 22:51:44] DEBUG[29205]: devicestate.c:466 do_state_change: Changing state for PJSIP/john - state 2 (In use)

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:5234 session_inv_on_media_update:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32553]: res_rtp_asterisk.c:8089 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7f66d001c770'

[Nov 22 22:51:44] DEBUG[29272]: app_queue.c:2589 device_state_cb: Device 'PJSIP/john' changed to state '2' (In use) but we don't care because they're not a member of any queue.

[Nov 22 22:51:44] DEBUG[32553]: acl.c:1047 ast_ouraddrfor: For destination '192.168.0.20', our source address is '[REDACTED IP ADDRESS OF SERVER]'.

[Nov 22 22:51:44] DEBUG[32553]: res_rtp_asterisk.c:8292 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7f66d001c770'

[Nov 22 22:51:44] DEBUG[32553]: rtp_engine.c:1319 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 9 based on m type on 0x7f667e1b7fc0

[Nov 22 22:51:44] DEBUG[32553]: rtp_engine.c:1205 rtp_codecs_payloads_copy_tx: Copying tx payload mapping 9 (0x7f66d002b5e8) from 0x7f667e1b7fc0 to 0x7f66d001c948

[Nov 22 22:51:44] DEBUG[32553]: rtp_engine.c:1205 rtp_codecs_payloads_copy_tx: Copying tx payload mapping 101 (0x7f66d002a6c8) from 0x7f667e1b7fc0 to 0x7f66d001c948

[Nov 22 22:51:44] DEBUG[32553]: channel.c:5711 set_format: Channel PJSIP/john-00000008 setting read format path: g722 -> g722

[Nov 22 22:51:44] DEBUG[32553]: channel.c:5711 set_format: Channel PJSIP/john-00000008 setting write format path: g722 -> g722

[Nov 22 22:51:44] DEBUG[32553]: res_rtp_asterisk.c:8788 ast_rtp_activate: ast_rtp_activate (0x7f66d001d310) - setup and perform DTLS'

[Nov 22 22:51:44] DEBUG[32553]: res_rtp_asterisk.c:2473 dtls_perform_handshake: dtls_perform_handshake (0x7f66d001d310) - ssl = (nil), setup = 0

[Nov 22 22:51:44] DEBUG[32553]: res_rtp_asterisk.c:2473 dtls_perform_handshake: dtls_perform_handshake (0x7f66d001d310) - ssl = (nil), setup = 0

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:5298 session_inv_on_media_update:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:4481 handle_outgoing_response:  PJSIP/john-00000008: Method is INVITE, Response is 200 OK

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:4497 handle_outgoing_response:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:4619 session_inv_on_state_changed: PJSIP/john-00000008: Source of transaction state change is TX_MSG

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: chan_pjsip.c:1645 chan_pjsip_indicate:  PJSIP/john-00000008: Indicated Stop generators

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: chan_pjsip.c:1882 chan_pjsip_indicate:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:6032 ast_rtcp_interpret: Got RTCP report of 60 bytes from [REDACTED IP ADDRESS OF USER]:4007

[Nov 22 22:51:44] DEBUG[30832][C-00000009]: res_rtp_asterisk.c:6032 ast_rtcp_interpret: Got RTCP report of 60 bytes from [REDACTED IP ADDRESS OF USER]:4007

[Nov 22 22:51:44] DEBUG[30832][C-00000009]: res_rtp_asterisk.c:7624 ast_rtp_read: 0x7f66d001d310 -- Received RTP packet from [REDACTED IP ADDRESS OF USER]:4006, dropping due to strict RTP protection. Qualifying new stream.

[Nov 22 22:51:44] DEBUG[29218]: res_pjsip/pjsip_distributor.c:503 distributor: Searching for serializer associated with dialog dlg0x7f66bc0264e8 for Request msg ACK/cseq=10623 (rdata0x7f66e0047438)

[Nov 22 22:51:44] DEBUG[29218]: res_pjsip/pjsip_distributor.c:511 distributor: Found serializer pjsip/distributor-00000027 associated with dialog dlg0x7f66bc0264e8

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:4563 handle_incoming_before_media: PJSIP/john-00000008: Received request

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:4377 handle_incoming_request:  PJSIP/john-00000008: Method is ACK

[Nov 22 22:51:44] DEBUG[32553]: chan_pjsip.c:3256 chan_pjsip_incoming_ack:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32553]: chan_pjsip.c:3260 chan_pjsip_incoming_ack:  PJSIP/john-00000008: Queueing SRCCHANGE

[Nov 22 22:51:44] DEBUG[32553]: chan_pjsip.c:3264 chan_pjsip_incoming_ack:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32553]: res_pjsip_session.c:4387 handle_incoming_request:  PJSIP/john-00000008

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: channel.c:5711 set_format: Channel PJSIP/john-00000008 setting write format path: gsm -> g722

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:5277 ast_rtp_write: Ooh, format changed from none to g722

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:5017 rtp_raw_write: Starting RTCP transmission on RTP instance '0x7f66d001c770'

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: channel.c:3178 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:7615 ast_rtp_read: 0x7f66d001d310 -- Received RTP packet from [REDACTED IP ADDRESS OF USER]:4006, dropping due to strict RTP protection. Will switch to it in 3 packets.

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:7615 ast_rtp_read: 0x7f66d001d310 -- Received RTP packet from [REDACTED IP ADDRESS OF USER]:4006, dropping due to strict RTP protection. Will switch to it in 2 packets.

[Nov 22 22:51:44] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:7615 ast_rtp_read: 0x7f66d001d310 -- Received RTP packet from [REDACTED IP ADDRESS OF USER]:4006, dropping due to strict RTP protection. Will switch to it in 1 packets.

[Nov 22 22:51:55] DEBUG[32559][C-00000009]: res_rtp_asterisk.c:6032 ast_rtcp_interpret: Got RTCP report of 80 bytes from [REDACTED IP ADDRESS OF USER]:4007

[Nov 22 22:51:56] DEBUG[29230]: res_pjsip_registrar.c:1293 check_expiration_thread: Expiring 0 contacts

[Nov 22 22:52:04] DEBUG[32558]: threadpool.c:1169 worker_idle: Worker thread idle timeout reached. Dying.

[Nov 22 22:52:04] DEBUG[29193]: threadpool.c:1028 worker_thread_destroy: Destroying worker thread 87

[Nov 22 22:52:05] DEBUG[29218]: res_pjsip/pjsip_distributor.c:503 distributor: Searching for serializer associated with dialog dlg0x7f66bc0264e8 for Request msg BYE/cseq=10624 (rdata0x7f66e0047438)

[Nov 22 22:52:05] DEBUG[29218]: res_pjsip/pjsip_distributor.c:511 distributor: Found serializer pjsip/distributor-00000027 associated with dialog dlg0x7f66bc0264e8

[Nov 22 22:52:05] DEBUG[32553]: res_pjsip_session.c:4619 session_inv_on_state_changed: PJSIP/john-00000008: Source of transaction state change is RX_MSG

[Nov 22 22:52:05] DEBUG[32553]: res_pjsip_session.c:4563 handle_incoming_before_media: PJSIP/john-00000008: Received request

[Nov 22 22:52:05] DEBUG[32553]: res_pjsip_session.c:4377 handle_incoming_request:  PJSIP/john-00000008: Method is BYE

[Nov 22 22:52:05] DEBUG[32553]: res_pjsip_session.c:4387 handle_incoming_request:  PJSIP/john-00000008

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: channel.c:3178 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: channel.c:3178 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: channel.c:5711 set_format: Channel PJSIP/john-00000008 setting write format path: g722 -> g722

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: pbx.c:4441 __ast_pbx_run: Spawn extension (default,s,1) exited non-zero on 'PJSIP/john-00000008'

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: channel.c:2440 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'PJSIP/john-00000008'

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: pbx_lua.c:1497 lua_find_extension: Looking up h@default:1

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: pbx_lua.c:1497 lua_find_extension: Looking up h@demo:1

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: channel.c:2531 ast_hangup: Channel 0x7f66d00276a0 'PJSIP/john-00000008' hanging up.  Refs: 2

[Nov 22 22:52:05] DEBUG[32559][C-00000009]: chan_pjsip.c:2493 hangup_cause2sip: AST hangup cause 16 (no match found in PJSIP)

[Nov 22 22:52:05] DEBUG[32553]: rtp_engine.c:455 instance_destructor: Destroyed RTP instance '0x7f66d001c770'

[Nov 22 22:52:05] DEBUG[32553]: res_pjsip_session.c:2943 session_destructor: john: Destroying SIP session

[Nov 22 22:52:05] DEBUG[32553]: channel.c:2177 ast_channel_destructor: Channel 0x7f66d00276a0 'PJSIP/john-00000008' destroying

[Nov 22 22:52:05] DEBUG[29213]: cdr.c:1470 cdr_object_finalize: Finalized CDR for PJSIP/john-00000008 - start 1606099902.992619 answer 1606099904.006828 end 1606099925.336573 dur 22.343 bill 21.329 dispo ANSWERED

[Nov 22 22:52:05] DEBUG[32553]: stasis.c:438 topic_dtor: Destroying topic. name: channel:1606099902.16, detail:

[Nov 22 22:52:05] DEBUG[32553]: stasis.c:446 topic_dtor: Topic 'channel:1606099902.16': 0x7f66d0029c60 destroyed

[Nov 22 22:52:05] DEBUG[29205]: devicestate.c:466 do_state_change: Changing state for PJSIP/john - state 1 (Not in use)

[Nov 22 22:52:05] DEBUG[29272]: app_queue.c:2589 device_state_cb: Device 'PJSIP/john' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.

[Nov 22 22:52:05] DEBUG[29213]: stasis.c:575 stasis_topic_create_with_detail: Creating topic. name: channel:1606099925.17, detail:

[Nov 22 22:52:05] DEBUG[29213]: stasis.c:609 stasis_topic_create_with_detail: Topic 'channel:1606099925.17': 0x7f66c400ac90 created

[Nov 22 22:52:05] DEBUG[29213]: stasis.c:438 topic_dtor: Destroying topic. name: channel:1606099925.17, detail:

[Nov 22 22:52:05] DEBUG[29213]: stasis.c:446 topic_dtor: Topic 'channel:1606099925.17': 0x7f66c400ac90 destroyed

[Nov 22 22:52:05] DEBUG[29193]: threadpool.c:536 grow: Increasing threadpool stasis/pool's size by 1

People would normally call for the full log (with warnings,verbose, add notice as well, or for the channel technology protocol log, which is actually output to the verbose log channel.

Hi David. Thanks for the response. I fixed the issue now. I realised that it was an issue with my SIP client after debugging for a few hours.