try it using my pjsip.conf. it will work
i am using odbc and all these configurations are done in database
ok then change it over there it work same
can you share your endpoint details using cli command if possible
pjsip show endpoint <endpoint_name>
just as i have shared above
i just used sdes but facing same problem there also
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: whatsapp-279291-91XXXXXXX Not In Use 0 of inf
Aor: whatsapp-279291-91XXXXXXXX 1
Transport: transport-tls tls 0 0 0.0.0.0:5061
Identify: whatsapp-279291-91XXXXXX/whatsapp-279291-91XXXXXXXXX
Match: 0.0.0.0/0
ParameterName : ParameterValue
===================================================================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (opus)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : whatsapp-279291-91XXXXXXXX
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 : from-meta
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_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
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 : sdes
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
overlap_context :
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 : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
security_mechanisms :
security_negotiation : no
send_aoc : false
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 : no
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
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 : transport-tls
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
tried with this config also but issue not resolved
Transport: transport-tls tls 0 0 0.0.0.0:5061
ParameterName : ParameterValue
========================================================
allow_reload : true
allow_wildcard_certs : No
async_operations : 1
bind : 0.0.0.0:5061
ca_list_file : /etc/certs/asterisk_ca.pem
ca_list_path :
cert_file : /etc/certs/fullchain.pem
cipher :
cos : 0
domain :
external_media_address : 20.196.26.154
external_signaling_address : 20.196.26.154
external_signaling_port : 0
local_net : 10.8.0.0/255.255.255.0
method : tlsv1_2
password :
priv_key_file : /etc/certs/privkey.pem
protocol : tls
require_client_cert : No
symmetric_transport : false
tcp_keepalive_enable : false
tcp_keepalive_idle_time : 30
tcp_keepalive_interval_time : 1
tcp_keepalive_probe_count : 5
tos : 0
verify_client : No
verify_server : No
websocket_write_timeout : 100
Although i have set the signalling and media address here but still contact header is still showing private ip
Call disconnected after 30 seconds
This doesn’t match your own local address!
i have changed it before sharing
just wanted to share that it doesnt have any effect
whenever i specify media and signaling address and local_net asterisk crashes when call comes after the dialplan execution
actually i am using external media for streaming rtp
also contact header is not taking my public ip it shows private ip only
Not on every call but crash happens frequently
Also the 30 second call termination issue is still there
If you really mean crash, you need to provide the backtrace, although, for a real crash, you should report it as a github issue, with sufficient supporting information.
If you are using “crash” loosely, you need to be more precise.
Hi,
This problem is solved .There was some problem with azure vm where we got private ip in eth0 which was causing problem if nat not resolved properly.
Now we are using digitalocean vm which do not have such issues so that problem of ACK lost and call cut after 32 seconds is resolved.
Now we are facing a problem that whatsapp sending invite during incoming call
then sometimes it reaches to correct endpoint (i.e endpoint is identified) and sometimes it doesn’t and throws 401 unauthorised
Although i have tried different methods setting the order to prefer auth_username.. in global config of pjsip.conf
..
Also i tried using match header in ps_identify_id_ips matching Authorization header but it is not working
auth_username won’t work unless they respond correctly to 401, as the only way to get them to offer it is to challenge them for authentication; it is part of the authentication details.
401 Challenge is sent in both cases and they reply with correct auth username and the invite is somewhat same in both cases but in one case it correctly reaches to execute dialplan while in other case asterisk throws another 401 unauthorised
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