Registration trunk pjsip realtime

Asterisk Asterisk SIP
1

hello to the whole list

i have a problem and i ask for help to all of you.

i installed asterisk 22.2 and configured it in REALTIME

the trunk-provider registration is done correctly and is registered.

the problem is that if i call, the call is rejected by the voip-provider.

the problem is the CONTACT and the FORM.

both of them show up with incorrect values ​​to the provider.

i can’t figure out in which table to insert them.

The tables I used for recording are:

PS_AORS
PS_AUTHS
PS_ENDPOINTS_ID_IPS
PS_ENDPOINTS
PS_REGISTRATIONS

=====================================
PS_AUTHS
ParameterName : ParameterValue

auth_type : userpass
md5_cred :
nonce_lifetime : 32
oauth_clientid :
oauth_secret :
password: xxxxxxxx
realm :
refresh_token :
username: 6xxxxxxx

PS_AOR:

ParameterName : ParameterValue

authenticate_qualify : false
contact : sip:xxxxxx@provider.sip:5060
default_expiration : 3600
mailboxes:
max_contacts : 10
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy :
qualify_frequency : 20
qualify_timeout : 3.000000
remove_existing : false
remove_unavailable : false
support_path : false
voicemail_extension :

ParameterName : ParameterValue

100rel : yes
accept_multiple_sdp_answers : false
accountcode:
acl :
aggregate_mwi : true
allow : (ulaw|alaw|gsm|g726|g722|h264|mpeg4)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : xxxxxxxx
asymmetric_rtp_codec : false
auth :
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid :
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 :
contact_user : xxxxxxxxxxxx
context : from-voip
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 : xxxxxxxxxx
g726_non_standard : false
geoloc_incoming_call_profile :
geoloc_outgoing_call_profile :
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

=============================================================

Is there anyone who can help me to register the sip trunk correctly using pjsip realtime?

Thank you all

2

It would be on the endpoint for outgoing calls.

3

yes yes,
the outgoing call arrives at the voip provider.
only they tell me that the call cannot continue because the contact that arrives and the from that arrives at the voip provider is wrong.

basically they receive my internal number and not my public number

4

You haven’t shown an actual SIP trace, or described what is incorrect with it/what it should be.

5

I solved it, I had a problem in the dialplan.
and when it made the call it added a dirty character on the number.

so I heard the voice saying error