No PeerStatus events when using realtime

Hello,

I’m having some problems to receive AMI events (PeerStatus for example) when using realtime to register users.

When configuring endpoints on pjsip.conf everything is ok, when using realtime, I only receive ChallengeSent and SuccessfulAuth events.

Here is my sorcery.conf :

[res_pjsip]
endpoint/cache=memory_cache,expire_on_reload=yes,object_lifetime_maximum=1
endpoint=realtime,ps_endpoints
endpoint=config,pjsip.conf,criteria=type=endpoint
auth/cache=memory_cache,expire_on_reload=yes,object_lifetime_maximum=1
auth=realtime,ps_auths
auth=config,pjsip.conf,criteria=type=auth
aor/cache=memory_cache,expire_on_reload=yes,object_lifetime_maximum=1
aor=realtime,ps_aors
aor=config,pjsip.conf,criteria=type=aor

Here my extconfig.conf:

[settings]
ps_endpoints => curl,localhost:8083/box-services/openapi/telephony/sip/account/info/endpoints
ps_aors => curl,localhost:8083/box-services/openapi/telephony/sip/account/info/aors
ps_auths => curl,localhost:8083/box-services/openapi/telephony/sip/account/info/auths

User is registered on my Asterisk server

box*CLI> pjsip show contacts

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  sipphone1/sip:sipphone1@172.20.100.1:36141;rin 55eb273a96 Avail        34.247

Objects found: 1

box*CLI> 
box*CLI> pjsip show aor sipphone1

      Aor:  <Aor..............................................>  <MaxContact>
    Contact:  <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

      Aor:  sipphone1                                            1
    Contact:  sipphone1/sip:sipphone1@172.20.100.1:36141;r 55eb273a96 Avail        32.123


 ParameterName        : ParameterValue
 ==================================================================================
 authenticate_qualify : false
 contact              : sip:sipphone1@172.20.100.1:36141;rinstance=e3c36db5f8a64cc1
 default_expiration   : 3600
 mailboxes            : 
 max_contacts         : 1
 maximum_expiration   : 7200
 minimum_expiration   : 60
 outbound_proxy       : 
 qualify_frequency    : 10
 qualify_timeout      : 15.000000
 remove_existing      : false
 support_path         : false
 voicemail_extension  : 

box*CLI> 
box*CLI> pjsip show auth sipphone1

  I/OAuth:  <AuthId/UserName.............................................................>
==========================================================================================

     Auth:  sipphone1/sipphone1

 ParameterName  : ParameterValue
 =================================================
 auth_type      : md5
 md5_cred       : xxxxxxxxxxxxxxxxxxxxxxxx
 nonce_lifetime : 32
 oauth_clientid : 
 oauth_secret   : 
 password       : 
 realm          : box
 refresh_token  : 
 username       : sipphone1

box*CLI> 
box*CLI> pjsip show endpoint sipphone1

 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:  sipphone1                                            Unavailable   0 of inf
     InAuth:  sipphone1/sipphone1
        Aor:  sipphone1                                          1
      Contact:  sipphone1/sip:sipphone1@172.20.100.1:36141 55eb273a96 Avail        36.836


 ParameterName                      : ParameterValue
 ===================================================================================================
 100rel                             : yes
 PHONENAME                          : sipphone1
 PHONETYPE                          : SIP
 accept_multiple_sdp_answers        : false
 accountcode                        : 
 acl                                : 
 aggregate_mwi                      : true
 allow                              : (alaw|ilbc|g729|gsm|g723|ulaw)
 allow_overlap                      : true
 allow_subscribe                    : true
 allow_transfer                     : true
 allow_unauthenticated_options      : false
 aors                               : sipphone1
 asymmetric_rtp_codec               : false
 auth                               : sipphone1
 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                            : from-phone
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : false
 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                        : 
 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                          : false
 srtp_tag_32                        : false
 stir_shaken                        : false
 sub_min_expiry                     : 0
 subscribe_context                  : 
 suppress_q850_reason_headers       : false
 t38_udptl                          : false
 t38_udptl_ec                       : none
 t38_udptl_ipv6                     : false
 t38_udptl_maxdatagram              : 0
 t38_udptl_nat                      : false
 timers                             : yes
 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                  : false
 use_avpf                           : false
 use_ptime                          : false
 user_eq_phone                      : false
 voicemail_extension                : 
 webrtc                             : no

My manager.conf file :

[root@box asterisk]# cat manager.conf 
;
; Asterisk Call Management support
;

; By default asterisk will listen on localhost only. 
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[manager]
secret=xxxxxxxxx
permit=0.0.0.0/0.0.0.0
;read=system,call,log,verbose,agent,command,user,all
;write=system,call,log,verbose,agent,command,user,all
read=all
write=all
[root@box asterisk]# 

When I register my soft phone, I only receive following events:

[2021-07-06 09:11:00] VERBOSE[23731] manager.c: <-- Examining AMI event: -->
Event: ChallengeSent
Privilege: security,all
SequenceNumber: 61
File: manager.c
Line: 1864
Func: manager_default_msg_cb
EventTV: 2021-07-06T09:11:00.765+0000
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: sipphone1
SessionID: _LiZG85XUZzqO3gcQT9ySA..
LocalAddress: IPV4/UDP/172.20.100.65/5060
RemoteAddress: IPV4/UDP/172.20.100.1/36141
Challenge: 


[2021-07-06 09:11:00] VERBOSE[23735] manager.c: <-- Examining AMI event: -->
Event: ChallengeSent
Privilege: security,all
SequenceNumber: 61
File: manager.c
Line: 1864
Func: manager_default_msg_cb
EventTV: 2021-07-06T09:11:00.765+0000
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: sipphone1
SessionID: _LiZG85XUZzqO3gcQT9ySA..
LocalAddress: IPV4/UDP/172.20.100.65/5060
RemoteAddress: IPV4/UDP/172.20.100.1/36141
Challenge: 


[2021-07-06 09:11:00] VERBOSE[23731] manager.c: <-- Examining AMI event: -->
Event: SuccessfulAuth
Privilege: security,all
SequenceNumber: 62
File: manager.c
Line: 1864
Func: manager_default_msg_cb
EventTV: 2021-07-06T09:11:00.799+0000
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: sipphone1
SessionID: _LiZG85XUZzqO3gcQT9ySA..
LocalAddress: IPV4/UDP/172.20.100.65/5060
RemoteAddress: IPV4/UDP/172.20.100.1/36141
UsingPassword: 1


[2021-07-06 09:11:00] VERBOSE[23735] manager.c: <-- Examining AMI event: -->
Event: SuccessfulAuth
Privilege: security,all
SequenceNumber: 62
File: manager.c
Line: 1864
Func: manager_default_msg_cb
EventTV: 2021-07-06T09:11:00.799+0000
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: sipphone1
SessionID: _LiZG85XUZzqO3gcQT9ySA..
LocalAddress: IPV4/UDP/172.20.100.65/5060
RemoteAddress: IPV4/UDP/172.20.100.1/36141
UsingPassword: 1

I’m running Asterisk 18.4.0

Any help would be nice on my problem, I’ve read a lot of discussions about it but didn’t find any answer …

Thank you.
Florian