Hi,
I have a pjsip endpoint configured to be identified by header only. The identify entity is configured in this way:
01fdd3b4dca3*CLI> pjsip show identify by_username_6436d017cb2c40001fe4fee6
endpoint : 6436d017cb2c40001fe4fee6
match :
match_header : Authorization: /Digest username="6436d017cb2c40001fe4fee6_9529",/
srv_lookups : false
The problem: Sometimes the INVITE is received, 401 Unautourized is replied and Voip provider send the Authorized header and the endpoint matches successfully, but sometimes (no configuration changes, just calling again and again), the Voip provider send the INVITE with no Authorization header and endpoint is identified even if the header don’t exists in this INVITE. So why the first INVITE is matched sometimes, if no Authorization header exists?
Invite coming_
Dialplan incoming, using the identified endpoint_
Endpoint configuration:
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl : deny/permit
aggregate_mwi : true
allow : (alaw|ulaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : 6436d017cb2c40001fe4fee6
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 : trunkin
cos_audio : 0
cos_video : 0
device_state_busy_at : 1
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 : actpass
dtls_verify : Yes
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 : 9529
g726_non_standard : false
ice_support : false
identify_by : header
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language : pt-br
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : true
message_context : textmessages
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 : 6436d017cb2c40001fe4fee6
outbound_proxy : sip:sip_proxy_udp:5060;lr
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 : false
rpid_immediate : false
rtcp_mux : true
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 120
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
set_var :
srtp_tag_32 : false
stir_shaken : off
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context : subscriptions
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 : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : udp-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
Asterisk: 18.11.2
S.O: Centos 7
PJSIP version: 2.10