Hi,
I’m installing a new test FreePBX instance for demo purposes. I was a little bit too enthusiastic and rolled on the “old” configuration., which is not working .
System : Ubuntu 20 - Asterisk 18.7.1 - FreePBX 15.0.17.55
I’m then back to basics to first have a very light config working.
Here is where am I so far
I need 2 clients / extensions connect to each other , so
created 2 extentions :
Configured my SIP settings :
blured is my public Server IP address
Please note I’m using custom ports :
I can get my SIP client registered, but trying to get as exemple the extension number told ( with * 65 ) , no sound and 503 Error.
Here is the debug from the Asterisk console, I see a lot of warnings but nothing clear to me. any insights ? ( W.X.Y.Z being client public IP and A.B.C.D being Server public IP )
Thanks !
DEBUG[24381]: res_pjsip/pjsip_distributor.c:393 find_dialog: Could not find matching transaction for Request msg INVITE/cseq=1 (rdata0x7ff468002928)
DEBUG[24381]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000034 to use for Request msg INVITE/cseq=1 (rdata0x7ff468002928)
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_authenticator_digest.c:470 digest_check_auth: Using default realm 'asterisk' on incoming auth '1234-auth'.
DEBUG[49340]: res_pjsip_authenticator_digest.c:357 verify: Realm: asterisk Username: 1234 Result: NOAUTH
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '172.31.17.207' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '172.31.17.207' and port ''.
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[24381]: res_pjsip/pjsip_distributor.c:393 find_dialog: Could not find matching transaction for Request msg INVITE/cseq=1 (rdata0x7ff468002928)
DEBUG[24381]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000034 to use for Request msg INVITE/cseq=1 (rdata0x7ff468002928)
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_authenticator_digest.c:470 digest_check_auth: Using default realm 'asterisk' on incoming auth '1234-auth'.
DEBUG[49340]: res_pjsip_authenticator_digest.c:357 verify: Realm: asterisk Username: 1234 Result: NOAUTH
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '172.31.17.207' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '172.31.17.207' and port ''.
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[24381]: res_pjsip/pjsip_distributor.c:393 find_dialog: Could not find matching transaction for Request msg ACK/cseq=1 (rdata0x7ff468002928)
DEBUG[24381]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000034 to use for Request msg ACK/cseq=1 (rdata0x7ff468002928)
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username '1234' domain 'A.B.C.D'
DEBUG[24381]: res_pjsip/pjsip_distributor.c:393 find_dialog: Could not find matching transaction for Request msg INVITE/cseq=2 (rdata0x7ff468002928)
DEBUG[24381]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000034 to use for Request msg INVITE/cseq=2 (rdata0x7ff468002928)
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: res_pjsip_endpoint_identifier_ip.c:275 common_identify: No identify sections to match against
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username '1234' domain 'A.B.C.D'
DEBUG[49340]: res_pjsip_authenticator_digest.c:470 digest_check_auth: Using default realm 'asterisk' on incoming auth '1234-auth'.
DEBUG[49340]: res_pjsip_authenticator_digest.c:260 check_nonce: Calculated nonce 1634298640/90bc44322ee42b65f3c0c9fb55494fdf. Actual nonce is 1634298640/90bc44322ee42b65f3c0c9fb55494fdf
DEBUG[49340]: res_pjsip_authenticator_digest.c:357 verify: Realm: asterisk Username: 1234 Result: SUCCESS
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '172.31.17.207' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '172.31.17.207' and port ''.
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: res_pjsip_session.c:4224 session_on_rx_request: (null session) Request: INVITE
DEBUG[49340]: res_pjsip_session.c:4059 handle_new_invite_request: Request:
DEBUG[49340]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000034 to use for Request msg INVITE/cseq=2 (rdata0x7ff46803f818)
DEBUG[49340]: chan_pjsip.c:2954 chan_pjsip_session_begin: 1234
DEBUG[49340]: chan_pjsip.c:2958 chan_pjsip_session_begin: Direct media no glare mitigation
DEBUG[49340]: res_pjsip_session.c:3918 new_invite: 1234
DEBUG[49340]: res_pjsip_session.c:3998 new_invite: 1234: Call (UDP:W.X.Y.Z:24960) to extension '*65' sending 100 Trying
DEBUG[49340]: res_pjsip_session.c:4503 handle_outgoing_response: 1234: Method is INVITE, Response is 100 Trying
DEBUG[49340]: res_pjsip_session.c:4522 handle_outgoing_response: 1234
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'W.X.Y.Z' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'W.X.Y.Z' and port ''.
DEBUG[49340]: res_pjsip_session.c:4619 session_inv_on_state_changed: 1234 Event: TSX_STATE Inv State: INCOMING
DEBUG[49340]: res_pjsip_session.c:4357 __print_debug_details: Function session_inv_on_state_changed called on event TSX_STATE
DEBUG[49340]: res_pjsip_session.c:4371 __print_debug_details: The state change pertains to the endpoint '1234()'
DEBUG[49340]: res_pjsip_session.c:4376 __print_debug_details: The inv session still has an invite_tsx (0x7ff474033968)
DEBUG[49340]: res_pjsip_session.c:4391 __print_debug_details: There is no transaction involved in this state change
DEBUG[49340]: res_pjsip_session.c:4393 __print_debug_details: The current inv state is INCOMING
DEBUG[49340]: res_pjsip_session.c:4645 session_inv_on_state_changed: 1234: Source of transaction state change is TX_MSG
DEBUG[49340]: res_pjsip_session.c:4693 session_inv_on_state_changed:
DEBUG[49340]: res_pjsip_session.c:4737 session_inv_on_tsx_state_changed: 1234 TSX State: Proceeding Inv State: INCOMING
DEBUG[49340]: res_pjsip_session.c:4357 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
DEBUG[49340]: res_pjsip_session.c:4371 __print_debug_details: The state change pertains to the endpoint '1234()'
DEBUG[49340]: res_pjsip_session.c:4376 __print_debug_details: The inv session still has an invite_tsx (0x7ff474033968)
DEBUG[49340]: res_pjsip_session.c:4382 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7ff474033968
DEBUG[49340]: res_pjsip_session.c:4386 __print_debug_details: The current transaction state is Proceeding
DEBUG[49340]: res_pjsip_session.c:4388 __print_debug_details: The transaction state change event is TX_MSG
DEBUG[49340]: res_pjsip_session.c:4393 __print_debug_details: The current inv state is INCOMING
DEBUG[49340]: res_pjsip_session.c:4925 session_inv_on_tsx_state_changed: Nothing delayed
DEBUG[49340]: res_pjsip_session.c:4169 session_on_tsx_state: 1234 TSX State: Proceeding Inv State: INCOMING
DEBUG[49340]: res_pjsip_session.c:4173 session_on_tsx_state: Topology: Pending: (null topology) Active: (null topology)
DEBUG[49340]: res_pjsip_session.c:4178 session_on_tsx_state:
DEBUG[49340]: res_pjsip_session.c:769 handle_incoming_sdp: 1234: Media count: 1
DEBUG[49340]: res_pjsip_session.c:795 handle_incoming_sdp: 1234: Processing stream 0
DEBUG[49340]: res_pjsip_session.c:832 handle_incoming_sdp: 1234: Using audio-0 for new stream name
DEBUG[49340]: res_pjsip_session.c:876 handle_incoming_sdp: 1234: Using new stream 0:audio-0:audio:sendrecv (nothing)
DEBUG[49340]: res_pjsip_session.c:495 ast_sip_session_media_state_add: 1234 Adding position 0
DEBUG[49340]: res_pjsip_session.c:541 ast_sip_session_media_state_add: Creating new media session
DEBUG[49340]: res_pjsip_session.c:584 ast_sip_session_media_state_add: Setting media session as default for audio
DEBUG[49340]: res_pjsip_session.c:589 ast_sip_session_media_state_add: Done
DEBUG[49340]: res_pjsip_session.c:929 handle_incoming_sdp: 1234: Negotiating incoming SDP media stream 0:audio-0:audio:sendrecv (nothing) using audio SDP handler
DEBUG[49340]: res_pjsip_sdp_rtp.c:1497 negotiate_incoming_sdp_stream: 1234
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '192.168.1.35' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '192.168.1.35' and port ''.
DEBUG[49340]: rtp_engine.c:526 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7ff44c022d10'
DEBUG[49340]: res_rtp_asterisk.c:3872 rtp_allocate_transport: (0x7ff44c022d10) RTP allocated port 26000
DEBUG[49340]: res_rtp_asterisk.c:3902 rtp_allocate_transport: (0x7ff44c022d10) ICE creating session [::]:26000 (26000)
DEBUG[49340]: res_rtp_asterisk.c:3784 ice_create: (0x7ff44c022d10) ICE create
DEBUG[49340]: res_rtp_asterisk.c:3561 rtp_add_candidates_to_ice: (0x7ff44c022d10) ICE add system candidates
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '172.31.17.207' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '172.31.17.207' and port ''.
DEBUG[49340]: res_rtp_asterisk.c:1329 ast_rtp_ice_add_cand: (0x7ff44c022d10) ICE add candidate: 172.31.17.207:26000, 2130706431
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'fe80::b7:dbff:fe82:5f80' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'fe80::b7:dbff:fe82:5f80' and port ''.
DEBUG[49340]: res_rtp_asterisk.c:1329 ast_rtp_ice_add_cand: (0x7ff44c022d10) ICE add candidate: [fe80::b7:dbff:fe82:5f80]:26000, 2130706431
DEBUG[49340]: rtp_engine.c:543 ast_rtp_instance_new: RTP instance '0x7ff44c022d10' is setup and ready to go
DEBUG[49340]: res_rtp_asterisk.c:918 ast_rtp_ice_stop: (0x7ff44c022d10) ICE stopped
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting 'ip-172-31-17-207' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host 'ip-172-31-17-207' and port ''.
DEBUG[49340]: res_rtp_asterisk.c:8433 ast_rtp_prop_set: (0x7ff44c022d10) RTCP setup on RTP instance
DEBUG[49340]: res_pjsip_sdp_rtp.c:446 set_incoming_call_offer_cap: 1234
DEBUG[49340]: res_pjsip_sdp_rtp.c:328 get_codecs: 1234
DEBUG[49340]: rtp_engine.c:1318 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 9 based on m type on 0x7ff4398002a0
DEBUG[49340]: rtp_engine.c:1318 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 0 based on m type on 0x7ff4398002a0
DEBUG[49340]: rtp_engine.c:1318 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 8 based on m type on 0x7ff4398002a0
DEBUG[49340]: rtp_engine.c:1318 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 3 based on m type on 0x7ff4398002a0
DEBUG[49340]: res_pjsip_sdp_rtp.c:404 get_codecs:
DEBUG[49340]: res_pjsip_session/pjsip_session_caps.c:161 ast_sip_session_create_joint_call_cap: '1234' Caps for incoming audio call with pref 'local' - remote: (ulaw|gsm|alaw|g722|ilbc|opus) local: (ulaw|alaw|gsm|g726|g722|testlaw) joint: (ulaw|alaw|gsm|g722)
DEBUG[49340]: rtp_engine.c:1282 ast_rtp_codecs_payloads_xover: Crossover copying tx to rx payload mapping 0 (0x7ff44c015d58) from 0x7ff4398002a0 to 0x7ff4398002a0
DEBUG[49340]: rtp_engine.c:1282 ast_rtp_codecs_payloads_xover: Crossover copying tx to rx payload mapping 3 (0x7ff44c0258e8) from 0x7ff4398002a0 to 0x7ff4398002a0
[ CUT ]
DEBUG[49340]: res_pjsip_sdp_rtp.c:482 set_incoming_call_offer_cap:
DEBUG[49340]: res_pjsip_sdp_rtp.c:1587 negotiate_incoming_sdp_stream:
DEBUG[49340]: res_pjsip_session.c:948 handle_incoming_sdp: 1234: Media stream 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722) handled by audio
DEBUG[49340]: res_pjsip_session.c:955 handle_incoming_sdp: 1234: Done with stream 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)
DEBUG[49340]: res_pjsip_session.c:960 handle_incoming_sdp: 1234: Handled? yes
DEBUG[49340]: res_pjsip_session.c:5089 create_local_sdp: 1234
DEBUG[49340]: res_pjsip_session.c:5126 create_local_sdp: 1234: Processing streams
DEBUG[49340]: res_pjsip_session.c:5132 create_local_sdp: 1234: Processing stream 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)
DEBUG[49340]: res_pjsip_session.c:495 ast_sip_session_media_state_add: 1234 Adding position 0
DEBUG[49340]: res_pjsip_session.c:503 ast_sip_session_media_state_add: Using existing media_session
DEBUG[49340]: res_pjsip_session.c:4965 add_sdp_streams: 1234 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)
DEBUG[49340]: res_pjsip_sdp_rtp.c:1742 create_outgoing_sdp_stream: 1234 Type: audio 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)
DEBUG[49340]: res_rtp_asterisk.c:8332 ast_rtp_prop_set: (0x7ff44c022d10) RTCP ignoring duplicate property
DEBUG[49340]: res_pjsip_sdp_rtp.c:2027 create_outgoing_sdp_stream: RC: 1
DEBUG[49340]: res_pjsip_session.c:4974 add_sdp_streams: Had handler
DEBUG[49340]: res_pjsip_session.c:5164 create_local_sdp: 1234: Stream 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722) added
DEBUG[49340]: res_pjsip_session.c:5177 create_local_sdp: 1234: Done with 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)
DEBUG[49340]: res_pjsip_session.c:5183 create_local_sdp: 1234: Adding bundle groups (if available)
DEBUG[49340]: res_pjsip_session.c:5189 create_local_sdp: 1234: Copying connection details
DEBUG[49340]: res_pjsip_session.c:5192 create_local_sdp: 1234: Processing media 0
DEBUG[49340]: res_pjsip_session.c:5210 create_local_sdp: 1234: Media 0 reset
DEBUG[49340]: res_pjsip_session.c:5233 create_local_sdp: 1234
DEBUG[49340]: res_pjsip_session.c:4402 handle_incoming_request: 1234: Method is INVITE
DEBUG[49340]: chan_pjsip.c:3013 chan_pjsip_incoming_request: 1234
DEBUG[49340]: chan_pjsip.c:555 chan_pjsip_new: 1234
DEBUG[49340]: channel_internal_api.c:680 ast_channel_nativeformats_set: <initializing>: Formats: (none)
DEBUG[49340]: channel_internal_api.c:692 ast_channel_nativeformats_set: Channel is being initialized or destroyed
DEBUG[49340]: stasis.c:579 stasis_topic_create_with_detail: Creating topic. name: channel:1634298640.17, detail:
DEBUG[49340]: stasis.c:613 stasis_topic_create_with_detail: Topic 'channel:1634298640.17': 0x7ff44c0256c0 created
DEBUG[49340]: channel.c:951 __ast_channel_alloc_ap: Channel 0x7ff44c059230 'PJSIP/1234-00000011' allocated
DEBUG[49340]: chan_pjsip.c:530 compatible_formats_exist: Topology: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)> Formats: (ulaw|alaw|gsm|g726|g722|testlaw)
DEBUG[49340]: chan_pjsip.c:543 compatible_formats_exist: Compatible? yes
DEBUG[49340]: channel_internal_api.c:680 ast_channel_nativeformats_set: PJSIP/1234-00000011: MultistreamFormats: (ulaw|alaw|gsm|g722)
DEBUG[49340]: channel_internal_api.c:702 ast_channel_nativeformats_set: Set native formats but not topology
DEBUG[49340]: channel_internal_api.c:1595 ast_channel_set_stream_topology: PJSIP/1234-00000011: <0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)>
DEBUG[49340]: channel_internal_api.c:1616 ast_channel_set_stream_topology: Used provided topology
DEBUG[49340]: chan_pjsip.c:675 chan_pjsip_new:
DEBUG[49340]: chan_pjsip.c:3063 chan_pjsip_incoming_request: PJSIP/1234-00000011
DEBUG[49340]: chan_pjsip.c:3119 pbx_start_incoming_request: PJSIP/1234-00000011
DEBUG[49340]: chan_pjsip.c:3145 pbx_start_incoming_request: Started PBX on new PJSIP channel PJSIP/1234-00000011
DEBUG[49340]: chan_pjsip.c:3147 pbx_start_incoming_request: RC: 0
DEBUG[49340]: res_pjsip_session.c:4412 handle_incoming_request: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:4047 new_invite: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:4125 handle_new_invite_request: Request: Session: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:4243 session_on_rx_request: (null session) Handled request INVITE ? yes
DEBUG[24351]: threadpool.c:535 grow: Increasing threadpool stasis/pool's size by 1
DEBUG[73441][C-00000012]: pbx.c:2938 pbx_extension_helper: Launching 'Set'
DEBUG[73441][C-00000012]: pbx.c:2938 pbx_extension_helper: Launching 'Set'
DEBUG[73441][C-00000012]: pbx.c:2938 pbx_extension_helper: Launching 'Set'
DEBUG[73441][C-00000012]: pbx.c:2938 pbx_extension_helper: Launching 'Answer'
DEBUG[73441][C-00000012]: chan_pjsip.c:733 chan_pjsip_answer: PJSIP/1234-00000011
DEBUG[24363]: devicestate.c:361 _ast_device_state: No provider found, checking channel drivers for PJSIP - 1234
DEBUG[24363]: devicestate.c:466 do_state_change: Changing state for PJSIP/1234 - state 2 (In use)
DEBUG[24443]: app_queue.c:2586 device_state_cb: Device 'PJSIP/1234' changed to state '2' (In use) but we don't care because they're not a member of any queue.
DEBUG[49340]: chan_pjsip.c:689 answer: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:5338 session_inv_on_media_update: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:1068 handle_negotiated_sdp: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:974 handle_negotiated_sdp_session_media: PJSIP/1234-00000011
DEBUG[49340]: res_pjsip_session.c:1005 handle_negotiated_sdp_session_media: PJSIP/1234-00000011: Applying negotiated SDP media stream 'audio' using audio SDP handler
DEBUG[49340]: res_pjsip_sdp_rtp.c:2085 apply_negotiated_sdp_stream: PJSIP/1234-00000011 Stream: 0:audio-0:audio:sendrecv (ulaw|alaw|gsm|g722)
DEBUG[49340]: res_rtp_asterisk.c:8332 ast_rtp_prop_set: (0x7ff44c022d10) RTCP ignoring duplicate property
DEBUG[49340]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '192.168.1.35' into...
DEBUG[49340]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '192.168.1.35' and port ''.
DEBUG[49340]: acl.c:1045 ast_ouraddrfor: For destination '192.168.1.35', our source address is '172.31.17.207'.
DEBUG[49340]: res_rtp_asterisk.c:8535 ast_rtp_remote_address_set: (0x7ff44c022d10) RTCP setting address on RTP instance
DEBUG[49340]: res_pjsip_sdp_rtp.c:500 set_caps: PJSIP/1234-00000011 ANSWER
DEBUG[49340]: res_pjsip_sdp_rtp.c:328 get_codecs: PJSIP/1234-00000011
DEBUG[49340]: rtp_engine.c:1318 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 9 based on m type on 0x7ff439800160
[CUT]