488 Not Acceptable Here - Not Codecs or Encryption

I have 2 Asterisk servers, both use Bandwidth as the provider, and the pjsip.conf files are virtually identical other than the from_domain

Here is config:

[bandwidth]
type = aor
contact = sip:801XXXXXXX@67.231.12.81

[bandwidth]
type = identify
endpoint = bandwidth
match = 67.231.12.81

[bandwidth]
type = endpoint
context = Dial-Users
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = g729
allow = ilbc
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
timers = no
direct_media = no
from_domain = 23.0.56.6
aors = bandwidth
send_rpid = yes
trust_id_outbound = yes

The two servers are running versions 13.17.0 (working) and 18.12.1 (not working)

Everything was working just fine on both servers until last week. The only thing that was different was I adjusted the rules for fail2ban. Also, at one point the hard drive filled up with log files. I freed up the space, rebooted the server, shutdown fail2ban and flushed all iptables rules. I added back the signalling servers for Bandwidth and desk phones.

Here is the pjsip trace when I try placing an inbound call from my cell into the office.


*CLI> <--- Received SIP request (1366 bytes) from UDP:67.231.12.81:5060 --->
INVITE sip:**DIALED_NUMBER**@23.0.56.6:5060 SIP/2.0
Via: SIP/2.0/UDP 67.231.12.81:5060;branch=z9hG4bK00B3befcadd7acca053
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>
Call-ID: 86040486_131062892@67.231.12.81
CSeq: 822948 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Accept: application/sdp
Contact: <sip:**INBOUND_NUMBER**@67.231.12.81:5060>
P-Asserted-Identity: <sip:**INBOUND_NUMBER**@67.231.12.81:5060>
Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9zdGkudmVyaXpvbi5jb20vdnp3Y2VydC92enNoYWtlbi0wMi0yMDI0LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxODAxNTE1Nzc3MSJdfSwiaWF0IjoxNjYwMTU4ODUyLCJvcmlnIjp7InRuIjoiMTQ4MDU5MzQwMjcifSwib3JpZ2lkIjoiRkZGRkZGRkYtRkZGRi1GRkZGLUZGRkYtMDEwMTAwMDAwMDAwIn0.trn2OI4OZubqZ0FSQXPu5x-pXBVtshG3tFunYEysYtV2kJSFalagB_BYsZ9KYbl132uFUolFVEi-PlyORw22Zg;info=<https://sti.verizon.com/vzwcert/vzshaken-02-2024.crt>;alg=ES256;ppt=shaken
Supported: replaces
Content-Length:   282
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 676109 944994 IN IP4 67.231.12.81
s=SIP Media Capabilities
c=IN IP4 67.231.13.29
t=0 0
m=audio 17044 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

<--- Transmitting SIP response (324 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK00B3befcadd7acca053
Call-ID: 86040486_131062892@67.231.12.81
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>
CSeq: 822948 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


<--- Transmitting SIP response (378 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK00B3befcadd7acca053
Call-ID: 86040486_131062892@67.231.12.81
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>;tag=bdc1c5df-a308-44e2-b285-a57e18c1d7cd
CSeq: 822948 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


<--- Received SIP request (403 bytes) from UDP:67.231.12.81:5060 --->
ACK sip:**DIALED_NUMBER**@23.0.56.6:5060 SIP/2.0
Via: SIP/2.0/UDP 67.231.12.81:5060;branch=z9hG4bK00B3befcadd7acca053
From: <sip:**INBOUND_NUMBER**@67.231.12.81>;tag=gK007de122
To: <sip:**DIALED_NUMBER**@23.0.56.6>;tag=bdc1c5df-a308-44e2-b285-a57e18c1d7cd
Call-ID: 86040486_131062892@67.231.12.81
CSeq: 822948 ACK
Max-Forwards: 70
Route: <sip:23.0.56.6:5060;lr;dtg=INNOSG6KQRO_TG>
Content-Length: 0

The only thing different I see in the INVITE section on the server where it works, is the Identity: section is missing.

I have tried screwing around with the codecs, verifying which are supported by Bandwidth ( https://support.bandwidth.com/hc/en-us/articles/360021075973-PBX-integration-guide).

I have seen other topics refer to encryption, which isn’t configured on the server.

It doesn’t seem to be firewall related since the trace is working. It isn’t the provider, because the other server is working. The signalling settings are right, or we wouldn’t get a trace.

Not sure what to be looking at next.

Anyone have any ideas at all?

You say you have eliminated all the possible reasons that I know of but haven’t provide enough information to verify that.

Is it even matching the correct endpoint?

Here is my endpoint configuration. It seems that if the endpoint was wrong, the trace wouldn’t even happen on an inbound call.


pjsip show endpoint bandwidth

 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
    Channel:  <ChannelId......................................>  <State.....>  <Time.....>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
==========================================================================================

 Endpoint:  bandwidth                                            Not in use    0 of inf
        Aor:  bandwidth                                          0
      Contact:  bandwidth/sip:8015555555@67.231.12.81      ef174f10a6 NonQual         nan
   Identify:  bandwidth/bandwidth
        Match: 67.231.12.81/32


 ParameterName                      : ParameterValue
 ===================================================================================================
 100rel                             : yes
 accept_multiple_sdp_answers        : false
 accountcode                        : 
 acl                                : 
 aggregate_mwi                      : true
 allow                              : (ulaw|g729|ilbc)
 allow_overlap                      : true
 allow_subscribe                    : true
 allow_transfer                     : true
 allow_unauthenticated_options      : false
 aors                               : bandwidth
 asymmetric_rtp_codec               : false
 auth                               : 
 bind_rtp_to_media_address          : false
 bundle                             : false
 call_group                         : 
 callerid                           : <unknown>
 callerid_privacy                   : allowed_not_screened
 callerid_tag                       : 
 codec_prefs_incoming_answer        : prefer:pending, operation:intersect, keep:all, transcode:allow
 codec_prefs_incoming_offer         : prefer:pending, operation:intersect, keep:all, transcode:allow
 codec_prefs_outgoing_answer        : prefer:pending, operation:intersect, keep:all, transcode:allow
 codec_prefs_outgoing_offer         : prefer:pending, operation:union, keep:all, transcode:allow
 connected_line_method              : invite
 contact_acl                        : 
 context                            : Dial-Users
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : true
 direct_media_glare_mitigation      : none
 direct_media_method                : invite
 disable_direct_media_on_nat        : false
 dtls_auto_generate_cert            : No
 dtls_ca_file                       : 
 dtls_ca_path                       : 
 dtls_cert_file                     : 
 dtls_cipher                        : 
 dtls_fingerprint                   : SHA-256
 dtls_private_key                   : 
 dtls_rekey                         : 0
 dtls_setup                         : active
 dtls_verify                        : No
 dtmf_mode                          : rfc4733
 fax_detect                         : false
 fax_detect_timeout                 : 0
 follow_early_media_fork            : true
 force_avp                          : false
 force_rport                        : true
 from_domain                        : 23.30.xx.xx (Obfuscated)
 from_user                          : 
 g726_non_standard                  : false
 ice_support                        : false
 identify_by                        : username,ip
 ignore_183_without_sdp             : false
 inband_progress                    : false
 incoming_call_offer_pref           : local
 incoming_mwi_mailbox               : 
 language                           : 
 mailboxes                          : 
 max_audio_streams                  : 1
 max_video_streams                  : 1
 media_address                      : 
 media_encryption                   : no
 media_encryption_optimistic        : false
 media_use_received_transport       : false
 message_context                    : 
 moh_passthrough                    : false
 moh_suggest                        : default
 mwi_from_user                      : 
 mwi_subscribe_replaces_unsolicited : no
 named_call_group                   : 
 named_pickup_group                 : 
 notify_early_inuse_ringing         : false
 one_touch_recording                : false
 outbound_auth                      : 
 outbound_proxy                     : 
 outgoing_call_offer_pref           : remote_merge
 pickup_group                       : 
 preferred_codec_only               : false
 record_off_feature                 : automixmon
 record_on_feature                  : automixmon
 refer_blind_progress               : true
 rewrite_contact                    : true
 rpid_immediate                     : false
 rtcp_mux                           : false
 rtp_engine                         : asterisk
 rtp_ipv6                           : false
 rtp_keepalive                      : 0
 rtp_symmetric                      : true
 rtp_timeout                        : 0
 rtp_timeout_hold                   : 0
 sdp_owner                          : -
 sdp_session                        : Asterisk
 send_connected_line                : yes
 send_diversion                     : true
 send_history_info                  : false
 send_pai                           : false
 send_rpid                          : true
 set_var                            : 
 srtp_tag_32                        : false
 stir_shaken                        : off
 stir_shaken_profile                : 
 sub_min_expiry                     : 0
 subscribe_context                  : 
 suppress_q850_reason_headers       : false
 t38_bind_udptl_to_media_address    : false
 t38_udptl                          : false
 t38_udptl_ec                       : none
 t38_udptl_ipv6                     : false
 t38_udptl_maxdatagram              : 0
 t38_udptl_nat                      : false
 timers                             : no
 timers_min_se                      : 90
 timers_sess_expires                : 1800
 tone_zone                          : 
 tos_audio                          : 0
 tos_video                          : 0
 transport                          : 
 trust_connected_line               : yes
 trust_id_inbound                   : false
 trust_id_outbound                  : true
 use_avpf                           : false
 use_ptime                          : false
 user_eq_phone                      : false
 voicemail_extension                : 
 webrtc                             : no

What clues does the full log give you with debugging and verbosity turned up.

This is what I see, but I am not sure what the issue is. I do see that it is getting a null topology and a media count of 0, but not sure why, if that is an issue with the endpoint provider not sending the right things, firewall stripping things out (shouldn’t be), or missing module that isn’t loaded or something else entirely.


<--- Transmitting SIP response (313 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.231.12.81:5060;received=67.231.12.81;branch=z9hG4bK00Ba0bab99e970c438c
Call-ID: 107013851_83830081@67.231.12.81
From: <sip:480XXXXXXX@67.231.12.81>;tag=gK001cfcdf
To: <sip:8015157771@23.XX.XX.XX9>
CSeq: 552868 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4630 session_inv_on_state_changed:  bandwidth Event: TSX_STATE  Inv State: INCOMING
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_state_changed called on event TSX_STATE
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4382 __print_debug_details: The state change pertains to the endpoint 'bandwidth()'
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa4bc0516a8)
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4402 __print_debug_details: There is no transaction involved in this state change
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is INCOMING
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4656 session_inv_on_state_changed: bandwidth: Source of transaction state change is TX_MSG
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4704 session_inv_on_state_changed:  
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4748 session_inv_on_tsx_state_changed:  bandwidth TSX State: Proceeding  Inv State: INCOMING
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4382 __print_debug_details: The state change pertains to the endpoint 'bandwidth()'
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa4bc0516a8)
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4393 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7fa4bc0516a8
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4397 __print_debug_details: The current transaction state is Proceeding
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4399 __print_debug_details: The transaction state change event is TX_MSG
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is INCOMING
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4936 session_inv_on_tsx_state_changed:  Nothing delayed
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4180 session_on_tsx_state:  bandwidth TSX State: Proceeding  Inv State: INCOMING
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4184 session_on_tsx_state:  Topology: Pending: (null topology)  Active: (null topology)
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4189 session_on_tsx_state:  
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:769 handle_incoming_sdp:  bandwidth: Media count: 0
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:960 handle_incoming_sdp:  bandwidth: Handled? no
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4514 handle_outgoing_response:  bandwidth: Method is INVITE, Response is 488 Not Acceptable Here
[Aug 19 15:46:07] DEBUG[55348]: res_pjsip_session.c:4533 handle_outgoing_response:  bandwidth
[Aug 19 15:46:07] DEBUG[55348]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 15:46:07] DEBUG[55348]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 15:46:07] DEBUG[55348]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 15:46:07] DEBUG[55348]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.

Here are my loaded modules:


Module                         Description                              Use Count  Status      Support Level
acl                            Named ACL system                         2          Running              core
app_bridgewait.so              Place the channel into a holding bridge  0          Running              core
app_confbridge.so              Conference Bridge Application            0          Running              core
app_dial.so                    Dialing Application                      0          Running              core
app_directory.so               Extension Directory                      0          Running              core
app_playback.so                Sound File Playback Application          0          Running              core
app_queue.so                   True Call Queueing                       0          Running              core
app_stack.so                   Dialplan subroutines (Gosub, Return, etc 0          Running              core
app_verbose.so                 Send verbose output                      0          Running              core
app_voicemail.so               Comedian Mail (Voicemail System)         0          Running              core
bridge_builtin_features.so     Built in bridging features               1          Running              core
bridge_builtin_interval_features.so Built in bridging interval features      0          Running              core
bridge_holding.so              Holding bridge module                    0          Running              core
bridge_native_rtp.so           Native RTP bridging module               0          Running              core
bridge_simple.so               Simple two channel bridging module       0          Running              core
bridge_softmix.so              Multi-party software based channel mixin 0          Running              core
ccss                           Call Completion Supplementary Services   2          Running              core
cdr                            CDR Engine                               3          Running              core
cdr_custom.so                  Customizable Comma Separated Values CDR  0          Running              core
cel                            CEL Engine                               1          Running              core
chan_bridge_media.so           Bridge Media Channel Driver              0          Running              core
chan_pjsip.so                  PJSIP Channel Driver                     0          Running              core
codec_g722.so                  ITU G.722-64kbps G722 Transcoder         0          Running              core
codec_gsm.so                   GSM Coder/Decoder                        0          Running              core
codec_resample.so              SLIN Resampling Codec                    0          Running              core
codec_ulaw.so                  mu-Law Coder/Decoder                     0          Running              core
dnsmgr                         DNS Manager                              2          Running              core
dsp                            DSP                                      1          Running              core
enum                           ENUM Support                             1          Running              core
extconfig                      Configuration                            13         Running              core
features                       Call Features                            1          Running              core
format_gsm.so                  Raw GSM data                             0          Running              core
format_pcm.so                  Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0          Running              core
format_wav.so                  Microsoft WAV/WAV16 format (8kHz/16kHz S 0          Running              core
format_wav_gsm.so              Microsoft WAV format (Proprietary GSM)   0          Running              core
func_callerid.so               Party ID related dialplan functions (Cal 0          Running              core
func_cdr.so                    Call Detail Record (CDR) dialplan functi 0          Running              core
func_devstate.so               Gets or sets a device state in the dialp 0          Running              core
func_pjsip_endpoint.so         Get information about a PJSIP endpoint   0          Running              core
func_sorcery.so                Get a field from a sorcery object        0          Running              core
func_strings.so                String handling dialplan functions       0          Running              core
http                           Built-in HTTP Server                     3          Running              core
indications                    Indication Tone Handling                 1          Running              core
logger                         Logger                                   1          Running              core
manager                        Asterisk Manager Interface               1          Running              core
pbx_config.so                  Text Extension Configuration             0          Running              core
plc                            PLC                                      1          Running              core
res_http_websocket.so          HTTP WebSocket Support                   1          Running              core
res_musiconhold.so             Music On Hold Resource                   0          Running              core
res_pjproject.so               PJPROJECT Log and Utility Support        4          Running              core
res_pjsip.so                   Basic SIP resource                       38         Running              core
res_pjsip_acl.so               PJSIP ACL Resource                       0          Running              core
res_pjsip_authenticator_digest.so PJSIP authentication resource            0          Running              core
res_pjsip_caller_id.so         PJSIP Caller ID Support                  1          Running              core
res_pjsip_dialog_info_body_generator.so PJSIP Extension State Dialog Info+XML Pr 0          Running              core
res_pjsip_diversion.so         PJSIP Add Diversion Header Support       1          Running              core
res_pjsip_dtmf_info.so         PJSIP DTMF INFO Support                  0          Running              core
res_pjsip_endpoint_identifier_anonymous.so PJSIP Anonymous endpoint identifier      0          Running              core
res_pjsip_endpoint_identifier_ip.so PJSIP IP endpoint identifier             0          Running              core
res_pjsip_endpoint_identifier_user.so PJSIP username endpoint identifier       0          Running              core
res_pjsip_exten_state.so       PJSIP Extension State Notifications      0          Running              core
res_pjsip_header_funcs.so      PJSIP Header Functions                   0          Running              core
res_pjsip_logger.so            PJSIP Packet Logger                      0          Running              core
res_pjsip_messaging.so         PJSIP Messaging Support                  0          Running              core
res_pjsip_mwi.so               PJSIP MWI resource                       0          Running              core
res_pjsip_mwi_body_generator.so PJSIP MWI resource                       0          Running              core
res_pjsip_nat.so               PJSIP NAT Support                        0          Running              core
res_pjsip_notify.so            CLI/AMI PJSIP NOTIFY Support             0          Running              core
res_pjsip_one_touch_record_info.so PJSIP INFO One Touch Recording Support   0          Running              core
res_pjsip_outbound_authenticator_digest.so PJSIP authentication resource            0          Running              core
res_pjsip_outbound_publish.so  PJSIP Outbound Publish Support           2          Running              core
res_pjsip_outbound_registration.so PJSIP Outbound Registration Support      0          Running              core
res_pjsip_path.so              PJSIP Path Header Support                0          Running              core
res_pjsip_pidf_body_generator.so PJSIP Extension State PIDF Provider      0          Running              core
res_pjsip_pidf_digium_body_supplement.so PJSIP PIDF Sangoma presence supplement   0          Running              core
res_pjsip_pidf_eyebeam_body_supplement.so PJSIP PIDF Eyebeam supplement            0          Running              core
res_pjsip_publish_asterisk.so  PJSIP Asterisk Event PUBLISH Support     0          Running              core
res_pjsip_pubsub.so            PJSIP event resource                     12         Running              core
res_pjsip_refer.so             PJSIP Blind and Attended Transfer Suppor 1          Running              core
res_pjsip_registrar.so         PJSIP Registrar Support                  0          Running              core
res_pjsip_rfc3326.so           PJSIP RFC3326 Support                    0          Running              core
res_pjsip_sdp_rtp.so           PJSIP SDP RTP/AVP stream handler         0          Running              core
res_pjsip_send_to_voicemail.so PJSIP REFER Send to Voicemail Support    0          Running              core
res_pjsip_session.so           PJSIP Session resource                   15         Running              core
res_pjsip_t38.so               PJSIP T.38 UDPTL Support                 0          Running              core
res_pjsip_transport_websocket.so PJSIP WebSocket Transport Support        0          Running              core
res_pjsip_xpidf_body_generator.so PJSIP Extension State PIDF Provider      0          Running              core
res_rtp_asterisk.so            Asterisk RTP Stack                       0          Running              core
res_sorcery_astdb.so           Sorcery Astdb Object Wizard              3          Running              core
res_sorcery_config.so          Sorcery Configuration File Object Wizard 17         Running              core
res_sorcery_memory.so          Sorcery In-Memory Object Wizard          2          Running              core
res_sorcery_realtime.so        Sorcery Realtime Object Wizard           0          Running              core
res_srtp.so                    Secure RTP (SRTP)                        0          Running              core
res_timing_timerfd.so          Timerfd Timing Interface                 0          Running              core
sounds                         Sounds Index                             1          Running              core
udptl                          UDPTL                                    2          Running              core
96 modules loaded

Load everything and see if it works. If it does, then it’s a missing module and you’d need to narrow it down further.

It did not work. Here are all the messages that aren’t a straight up debug message from launching in console mode.


[Aug 19 19:17:39] NOTICE[56740]: loader.c:2389 load_modules: 357 modules will be loaded.
[Aug 19 19:17:40] ERROR[56740]: res_config_ldap.c:1720 load_module: Cannot load LDAP RealTime driver.
[Aug 19 19:17:40] ERROR[56740]: res_config_ldap.c:1858 parse_config: No directory URL or host found.
[Aug 19 19:17:40] NOTICE[56740]: res_config_ldap.c:1832 parse_config: No directory user found, anonymous binding as default.
[Aug 19 19:17:40] WARNING[56740]: res_config_mysql.c:1247 load_mysql_config: MySQL RealTime: No database user found, using 'asterisk' as default.
[Aug 19 19:17:40] WARNING[56740]: res_config_mysql.c:1253 load_mysql_config: MySQL RealTime: No database password found, using 'asterisk' as default.
[Aug 19 19:17:40] WARNING[56740]: res_config_mysql.c:1259 load_mysql_config: MySQL RealTime: No database host found, using localhost via socket.
[Aug 19 19:17:40] WARNING[56740]: res_config_mysql.c:1265 load_mysql_config: MySQL RealTime: No database name found, using 'asterisk' as default.
[Aug 19 19:17:40] WARNING[56740]: res_config_mysql.c:1271 load_mysql_config: MySQL RealTime: No database port found, using 3306 as default.
[Aug 19 19:17:40] WARNING[56740]: res_config_mysql.c:1288 load_mysql_config: MySQL RealTime: No database socket found (and unable to detect a suitable path).

I haven’t been using ldap or mysql, so I don’t expect it to be an issue.

Here is the full trace with all the modules loaded:


<--- Received SIP request (1366 bytes) from UDP:67.231.12.81:5060 --->
INVITE sip:801YYYYYYY@23.XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 67.231.12.81:5060;branch=z9hG4bK04B66bfeb4019032e43
From: <sip:480XXXXXX@67.231.12.81>;tag=gK0426d0dc
To: <sip:801YYYYYYY@23.XX.XX.XX>
Call-ID: 105177797_41407786@67.231.12.81
CSeq: 334583 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Accept: application/sdp
Contact: <sip:480XXXXXX@67.231.12.81:5060>
P-Asserted-Identity: <sip:480XXXXXX@67.231.12.81:5060>
Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9zdGkudmVyaXpvbi5jb20vdnp3Y2VydC92enNoYWtlbi0wMi0yMDI0LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxODAxNTE1Nzc3MSJdfSwiaWF0IjoxNjYwOTM3MDEwLCJvcmlnIjp7InRuIjoiMTQ4MDU5MzQwMjcifSwib3JpZ2lkIjoiRkZGRkZGRkYtRkZGRi1GRkZGLUZGRkYtMDEwMTAwMDAwMDAwIn0.E8g9FP-nLewGZqVJJ04hgZIrLnmTn4tHmSGmCPz4qUPlO3XbtnsiH34oav6_P3y0RznucV6OoYURpIAOVIoa3g;info=<https://sti.verizon.com/vzwcert/vzshaken-02-2024.crt>;alg=ES256;ppt=shaken
Supported: replaces
Content-Length:   282
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 552470 420717 IN IP4 67.231.12.81
s=SIP Media Capabilities
c=IN IP4 67.231.13.29
t=0 0
m=audio 23410 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

[Aug 19 19:23:30] DEBUG[56770]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56770]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 19:23:30] DEBUG[56770]: res_pjsip/pjsip_distributor.c:393 find_dialog: Could not find matching transaction for Request msg INVITE/cseq=334583 (rdata0x7fa9f8005b28)
[Aug 19 19:23:30] DEBUG[56770]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000029 to use for Request msg INVITE/cseq=334583 (rdata0x7fa9f8005b28)
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_endpoint_identifier_ip.c:253 ip_identify_match_check: Source address 67.231.12.81:5060 matches identify 'bandwidth'
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_endpoint_identifier_ip.c:287 common_identify: Identify 'bandwidth' SIP message matched to endpoint bandwidth
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4235 session_on_rx_request:  (null session) Request: INVITE 
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4062 handle_new_invite_request:  Request: 
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000029 to use for Request msg INVITE/cseq=334583 (rdata0x7fa9f8019248)
[Aug 19 19:23:30] DEBUG[56859]: chan_pjsip.c:2974 chan_pjsip_session_begin:  bandwidth
[Aug 19 19:23:30] DEBUG[56859]: chan_pjsip.c:2978 chan_pjsip_session_begin:  Direct media no glare mitigation
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:3916 new_invite:  bandwidth
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:3996 new_invite:  bandwidth: Call (UDP:67.231.12.81:5060) to extension '801YYYYYYY' sending 100 Trying
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4514 handle_outgoing_response:  bandwidth: Method is INVITE, Response is 100 Trying
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4533 handle_outgoing_response:  bandwidth
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
<--- Transmitting SIP response (324 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK04B66bfeb4019032e43
Call-ID: 105177797_41407786@67.231.12.81
From: <sip:480XXXXXX@67.231.12.81>;tag=gK0426d0dc
To: <sip:801YYYYYYY@23.XX.XX.XX>
CSeq: 334583 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4630 session_inv_on_state_changed:  bandwidth Event: TSX_STATE  Inv State: INCOMING
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_state_changed called on event TSX_STATE
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4382 __print_debug_details: The state change pertains to the endpoint 'bandwidth()'
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa9a401ea08)
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4402 __print_debug_details: There is no transaction involved in this state change
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is INCOMING
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4656 session_inv_on_state_changed: bandwidth: Source of transaction state change is TX_MSG
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4704 session_inv_on_state_changed:  
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4748 session_inv_on_tsx_state_changed:  bandwidth TSX State: Proceeding  Inv State: INCOMING
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4382 __print_debug_details: The state change pertains to the endpoint 'bandwidth()'
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa9a401ea08)
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4393 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7fa9a401ea08
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4397 __print_debug_details: The current transaction state is Proceeding
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4399 __print_debug_details: The transaction state change event is TX_MSG
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is INCOMING
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4936 session_inv_on_tsx_state_changed:  Nothing delayed
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4180 session_on_tsx_state:  bandwidth TSX State: Proceeding  Inv State: INCOMING
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4184 session_on_tsx_state:  Topology: Pending: (null topology)  Active: (null topology)
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4189 session_on_tsx_state:  
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:769 handle_incoming_sdp:  bandwidth: Media count: 0
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:960 handle_incoming_sdp:  bandwidth: Handled? no
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4514 handle_outgoing_response:  bandwidth: Method is INVITE, Response is 488 Not Acceptable Here
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4533 handle_outgoing_response:  bandwidth
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
<--- Transmitting SIP response (378 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK04B66bfeb4019032e43
Call-ID: 105177797_41407786@67.231.12.81
From: <sip:480XXXXXX@67.231.12.81>;tag=gK0426d0dc
To: <sip:801YYYYYYY@23.XX.XX.XX>;tag=b51d31bd-0183-4b54-ac62-78359c603636
CSeq: 334583 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0


[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4630 session_inv_on_state_changed:  bandwidth Event: TSX_STATE  Inv State: DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_state_changed called on event TSX_STATE
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4382 __print_debug_details: The state change pertains to the endpoint 'bandwidth()'
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa9a401ea08)
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4402 __print_debug_details: There is no transaction involved in this state change
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4656 session_inv_on_state_changed: bandwidth: Source of transaction state change is TX_MSG
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4704 session_inv_on_state_changed:  
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4748 session_inv_on_tsx_state_changed:  bandwidth TSX State: Completed  Inv State: DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4382 __print_debug_details: The state change pertains to the endpoint 'bandwidth()'
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa9a401ea08)
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4393 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7fa9a401ea08
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4397 __print_debug_details: The current transaction state is Completed
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4399 __print_debug_details: The transaction state change event is TX_MSG
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4770 session_inv_on_tsx_state_changed:  Disconnected
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4180 session_on_tsx_state:  (null session) TSX State: Completed  Inv State: DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4189 session_on_tsx_state:  
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4045 new_invite:  bandwidth
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4136 handle_new_invite_request:  Request:  Session: bandwidth
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4254 session_on_rx_request:  (null session) Handled request INVITE  ? yes
[Aug 19 19:23:30] DEBUG[56859]: chan_pjsip.c:2995 chan_pjsip_session_end:  bandwidth
[Aug 19 19:23:30] DEBUG[56859]: chan_pjsip.c:2998 chan_pjsip_session_end:  No channel
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:2912 session_destructor: bandwidth: Destroying SIP session
[Aug 19 19:23:30] DEBUG[56770]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56770]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
<--- Received SIP request (403 bytes) from UDP:67.231.12.81:5060 --->
ACK sip:801YYYYYYY@23.XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 67.231.12.81:5060;branch=z9hG4bK04B66bfeb4019032e43
From: <sip:480XXXXXX@67.231.12.81>;tag=gK0426d0dc
To: <sip:801YYYYYYY@23.XX.XX.XX>;tag=b51d31bd-0183-4b54-ac62-78359c603636
Call-ID: 105177797_41407786@67.231.12.81
CSeq: 334583 ACK
Max-Forwards: 70
Route: <sip:23.XX.XX.XX:5060;lr;dtg=INNOVATIONGUYSG6KQRO_TG>
Content-Length: 0


[Aug 19 19:23:30] DEBUG[56770]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56770]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 19:23:30] DEBUG[56770]: res_pjsip/pjsip_distributor.c:502 distributor: Searching for serializer associated with dialog dlg0x7fa9a8007998 for Request msg ACK/cseq=334583 (rdata0x7fa9f8005b28)
[Aug 19 19:23:30] DEBUG[56770]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000029 to use for Request msg ACK/cseq=334583 (rdata0x7fa9f8005b28)
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 19:23:30] DEBUG[56859]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_endpoint_identifier_ip.c:253 ip_identify_match_check: Source address 67.231.12.81:5060 matches identify 'bandwidth'
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_endpoint_identifier_ip.c:287 common_identify: Identify 'bandwidth' SIP message matched to endpoint bandwidth
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4748 session_inv_on_tsx_state_changed:  (null session) TSX State: Confirmed  Inv State: DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4380 __print_debug_details: inv_session 0x7fa9f4020068 has no ast session
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4387 __print_debug_details: The inv session still has an invite_tsx (0x7fa9a401ea08)
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4393 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7fa9a401ea08
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4397 __print_debug_details: The current transaction state is Confirmed
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4399 __print_debug_details: The transaction state change event is RX_MSG
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4760 session_inv_on_tsx_state_changed:  Session ended
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4180 session_on_tsx_state:  (null session) TSX State: Confirmed  Inv State: DISCONNCTD
[Aug 19 19:23:30] DEBUG[56859]: res_pjsip_session.c:4189 session_on_tsx_state:  
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4748 session_inv_on_tsx_state_changed:  (null session) TSX State: Terminated  Inv State: DISCONNCTD
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4368 __print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4380 __print_debug_details: inv_session 0x7fa9f4020068 has no ast session
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4390 __print_debug_details: The inv session does NOT have an invite_tsx
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4393 __print_debug_details: The UAS INVITE transaction involved in this state change is 0x7fa9a401ea08
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4397 __print_debug_details: The current transaction state is Terminated
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4399 __print_debug_details: The transaction state change event is TIMER
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4404 __print_debug_details: The current inv state is DISCONNCTD
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4760 session_inv_on_tsx_state_changed:  Session ended
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4180 session_on_tsx_state:  (null session) TSX State: Terminated  Inv State: DISCONNCTD
[Aug 19 19:23:35] DEBUG[56770]: res_pjsip_session.c:4189 session_on_tsx_state:

For what it is worth, this is my firewall config:

iptables -L
Chain INPUT (policy DROP)
target     prot opt source               destination         
ACCEPT     all  --  67.231.12.81         anywhere            
ACCEPT     all  --  192.168.0.0/24       anywhere            
ACCEPT     all  --  23.XX.XX.XX/28       anywhere            
ACCEPT     all  --  67.231.13.11         anywhere            
ACCEPT     all  --  67.231.13.12         anywhere            
ACCEPT     all  --  67.231.13.13         anywhere            
ACCEPT     all  --  67.231.13.14         anywhere            
ACCEPT     all  --  67.231.9.91          anywhere            
ACCEPT     all  --  67.231.9.92          anywhere            
ACCEPT     all  --  67.231.9.93          anywhere            
ACCEPT     all  --  67.231.9.94          anywhere            
ACCEPT     all  --  67.231.0.7           anywhere            
ACCEPT     all  --  67.231.0.8           anywhere            
ACCEPT     all  --  67.231.0.9           anywhere            
ACCEPT     all  --  67.231.0.10          anywhere            
ACCEPT     all  --  67.231.4.6           anywhere            
ACCEPT     all  --  67.231.4.7           anywhere            
ACCEPT     all  --  67.231.4.8           anywhere            
ACCEPT     all  --  67.231.4.9           anywhere            

Chain FORWARD (policy ACCEPT)
target     prot opt source               destination         

Chain OUTPUT (policy DROP)
target     prot opt source               destination         
ACCEPT     all  --  anywhere             67.231.12.81        
ACCEPT     all  --  anywhere             192.168.0.0/24      
ACCEPT     all  --  anywhere             23.XX.XX.XX/28      
ACCEPT     all  --  anywhere             67.231.13.11        
ACCEPT     all  --  anywhere             67.231.13.12        
ACCEPT     all  --  anywhere             67.231.13.13        
ACCEPT     all  --  anywhere             67.231.13.14        
ACCEPT     all  --  anywhere             67.231.9.91         
ACCEPT     all  --  anywhere             67.231.9.92         
ACCEPT     all  --  anywhere             67.231.9.93         
ACCEPT     all  --  anywhere             67.231.9.94         
ACCEPT     all  --  anywhere             67.231.0.7          
ACCEPT     all  --  anywhere             67.231.0.8          
ACCEPT     all  --  anywhere             67.231.0.9          
ACCEPT     all  --  anywhere             67.231.0.10         
ACCEPT     all  --  anywhere             67.231.4.6          
ACCEPT     all  --  anywhere             67.231.4.7          
ACCEPT     all  --  anywhere             67.231.4.8          
ACCEPT     all  --  anywhere             67.231.4.9

Just to be sure, I cleared the rules from the firewall:


 # iptables -L
Chain INPUT (policy DROP)
target     prot opt source               destination         
ACCEPT     all  --  anywhere             anywhere            

Chain FORWARD (policy ACCEPT)
target     prot opt source               destination         

Chain OUTPUT (policy DROP)
target     prot opt source               destination         
ACCEPT     all  --  anywhere             anywhere

According to this, PJMEDIA-SDP has parsed the SDP and come up with no media streams, which is odd, since according to your trace there is a media stream.

I do wonder if the size of the packet is the cause in some way, due to the Identity header. For example the Content-Length says 282 but the size of the SDP doesn’t seem to be that.

Great catch on the packet size. BTW, I realized earlier that I had not thanked you for coming to my rescue. I am grateful for your help.

So, I did notice the identity thing, it doesn’t show up at all on the packet trace on the other Asterisk server. I had assumed it was because of the version, that it wasn’t parsing it, but now I am wondering if there is a configuration difference in my account at the provider level.

The content length looks good to me, as long was what is on the wire has CR as well as LF. Copying and pasting and feeding to wc, on Linux, I get:

13 33 269

and 13 + 269 is 282.

So, I was able to reconfigure the provider to stop sending that header, and then as I was cutting and pasting from the server that worked, just to reset it, I forgot to update the context to what it is on this server for the inbound calls. That left me with a whole new error, not a 488. So I cleared everything out of the dialplan except this:


[Aug 19 21:32:22] DEBUG[58514]: chan_pjsip.c:2974 chan_pjsip_session_begin:  bandwidth
[Aug 19 21:32:22] DEBUG[58514]: chan_pjsip.c:2978 chan_pjsip_session_begin:  Direct media no glare mitigation
[Aug 19 21:32:22] DEBUG[58514]: res_pjsip_session.c:3916 new_invite:  bandwidth
[Aug 19 21:32:22] NOTICE[58514]: res_pjsip_session.c:3962 new_invite:  bandwidth: Call (UDP:67.231.12.81:5060) to extension '8015157771' rejected because extension not found in context 'inbound-sip'.
[Aug 19 21:32:22] DEBUG[58514]: res_pjsip_session.c:3962 new_invite:   bandwidth: Call (UDP:67.231.12.81:5060) to extension '8015157771' rejected because extension not found in context 'inbound-sip'.
[Aug 19 21:32:22] DEBUG[58514]: res_pjsip_session.c:4514 handle_outgoing_response:  bandwidth: Method is INVITE, Response is 404 Not Found
[Aug 19 21:32:22] DEBUG[58514]: res_pjsip_session.c:4533 handle_outgoing_response:  bandwidth
[Aug 19 21:32:22] DEBUG[58514]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 21:32:22] DEBUG[58514]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
[Aug 19 21:32:22] DEBUG[58514]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '67.231.12.81' into...
[Aug 19 21:32:22] DEBUG[58514]: netsock2.c:224 ast_sockaddr_split_hostport: ...host '67.231.12.81' and port ''.
<--- Transmitting SIP response (369 bytes) to UDP:67.231.12.81:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 67.231.12.81:5060;rport=5060;received=67.231.12.81;branch=z9hG4bK0cBbc59225f875a7070
Call-ID: 101492367_111098111@67.231.12.81
From: <sip:480XXXXXXX@67.231.12.81>;tag=gK0c7aec8c
To: <sip:801YYYYYYY@23.XX.XX.XX>;tag=2feecbb4-6156-4f21-b213-d26b34e9b48b
CSeq: 870915 INVITE
Server: Asterisk PBX 18.12.1
Content-Length:  0

At least it was something different. Anyway, I tried reconfiguring the dialplan to be ultra simple and call my cell, but I got the same 488.

[Dial-Users]
    exten => _.,1,Dial(PJSIP/480XXXXXXX@bandwidth)

That comes back to matching the wrong endpoint. Which endpoints have context inbound-sip?

Is there any way you can eliminate the machine, itself, as the source of the problem.

Doing a full OS reinstall and starting over fixed it. I don’t know if a system file got corrupted when the HDD got filled up, or what, but it worked out of the gate.