Avaya 1120 IP Phone Registering Problem

Hi all,

I upgrade my asterisk 1.6.2.20 to 13.8 with pjsip.
When I registered Avaya 1120 asterisk 13.8, I have recived :

“[May 30 16:40:18] NOTICE[5372]: res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from ‘sip:katmer’ failed for ‘10.22.16.42:5060’ (callid: 13ab1487b31fb6c0) - No matching endpoint found
” but I can call this number olso it can call me . İs this log importand? When I have used 1.6 ı didn’t have this problem.

pjsip show endpoint 5346 output below

Katmer*CLI> pjsip show endpoint 5346

Endpoint: <Endpoint/CID…> <State…> <Channels.>
I/OAuth: <AuthId/UserName…>
Aor: <Aor…>
Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>
Transport: <TransportId…> <BindAddress…>
Identify: <Identify/Endpoint…>
Match: <ip/cidr…>
Channel: <ChannelId…> <State…> <Time(sec)>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>

Endpoint: 5346/5346 Not in use 0 of inf
InAuth: 5346/5346
Aor: 5346 1
Contact: 5346/sip:5346@10.15.161.226 5e3d8d00b2 Avail 15.821
Transport: transport-udp udp 0 0 10.20.6.234:5060

ParameterName : ParameterValue

100rel : yes
accountcode :
aggregate_mwi : true
allow : (g729|ulaw)
allow_subscribe : true
allow_transfer : true
aors : 5346
auth : 5346
bind_rtp_to_media_address : false
call_group : 10
callerid : “Avaya 1120” <5346>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
context : custom-uluslararasi
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 : true
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
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username
inband_progress : false
language :
mailboxes :
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context : mesajlar
moh_suggest : moh-ise
mwi_from_user :
named_call_group :
named_pickup_group :
one_touch_recording : false
outbound_auth :
outbound_proxy :
pickup_group : 10
record_off_feature : automixmon
record_on_feature : automixmon
rewrite_contact : false
rpid_immediate : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 30
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 : transport-udp
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false