Incoming calls errors using Grandstream HT813 with pstn in FXO port

Asterisk Asterisk Endpoints
1

Issue to setup a HT813 ATA in a pstn line and an Asterisk PBX 13 with PJSIP and Realtime behind NAT, when I call to pstn lines the call is not forwarded to the extension that should

Invites arriving in Asterisk CLI console:

[Jan 16 12:05:53] NOTICE[32270]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:019976401569@54.236.1.32>' failed for '201.75.25.1:28140' (callid: 851190526-28140-9@BJC.BGI.A.BG) - No matching endpoint found
[Jan 16 12:05:53] NOTICE[31698]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:019976401569@54.236.1.32>' failed for '201.75.25.1:28140' (callid: 851190526-28140-9@BJC.BGI.A.BG) - No matching endpoint found
[Jan 16 12:05:53] NOTICE[31698]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:019976401569@54.236.1.32>' failed for '201.75.25.1:28140' (callid: 851190526-28140-9@BJC.BGI.A.BG) - Failed to authenticate
[Jan 16 12:05:53] NOTICE[25921]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:019976401569@54.236.1.32>' failed for '201.75.25.1:28140' (callid: 851190526-28140-9@BJC.BGI.A.BG) - No matching endpoint found

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Endpoint: 19028020/19028020 Not in use 0 of inf
InAuth: auth-19028020/19028020
Aor: 19028020 3
Contact: 19028020/sip:19028020@201.75.x.y:29313 88af395db8 Avail 19.300
Transport: simpletrans udp 0 0 0.0.0.0:5060

ParameterName : ParameterValue (ENDPOINT)

100rel : yes
accountcode :
acl :
aggregate_mwi : true
allow : (g729|gsm|ulaw|alaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 19028020
asymmetric_rtp_codec : false
auth : auth-19028020
bind_rtp_to_media_address : false
call_group :
callerid : 19028020
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : incoming-813a
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_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
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : false
identify_by : username,ip
inband_progress : false
incoming_mwi_mailbox :
language :
mailboxes :
media_address : 54.x.y.z
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : false
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth :
outbound_proxy :
pickup_group :
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_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
subscribe_context :
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 : simpletrans
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :

ParameterName : ParameterValue (AOR)

authenticate_qualify : false
contact : sip:19028020@201.75.x.y:29313
default_expiration : 120
mailboxes :
max_contacts : 3
maximum_expiration : 600
minimum_expiration : 60
outbound_proxy :
qualify_frequency : 60
qualify_timeout : 10.000000
remove_existing : true
support_path : false
voicemail_extension :

I noted that incoming calls works well if I remove the user and password information and create the authentication only by IP, but I can’t use with that because the ATA will be installed in a place that doesn’t have fixed IP.

2

Mis-configuration, but you haven’t provided the configuration, in particular, your endpoints, identifies, and authentication.

3

@david551 I updated the post, sorry… If necessary I add the ATA informations print screens.

I noted that incoming calls works well if I remove the user and password information and create the authentication only by IP, but I can’t use with that because the ATA will be installed in a place that doesn’t have fixed IP.

4

I expected to see this as the contents of pjsp.conf, and therefore to be much shorter. However, I suspect the reason is that the above doesn’t match 19976401569

Are you, by any chance, using a GUI?

5

No I don’t use GUI, just Asterisk Realtime

6

I solved the situation enabling the chan_sip of my Asterisk PBX and using it to register this equipment because PJSIP give me many troubles to use that.

Was not the better solution but it works to me.

7

The problem, here, with using ARA is that one cannot easily see what options are non-default,

8

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