Hi,
We have a problem where pjsip endpoints become unavailable. In fact, we have a Kamailio in front of our Asterisk and client are registered with Kamailio. Asterisk is configured with realtime to fetch endpoints and aors from the Kamailio DB.
I have seen other posts of people having the same problem, the solution was to set qualify_frequency > 0 but we don’t want to qualify since Kamailio is handling this part of the job.
Does anyone have any idea how to solve this problem?
Please find some more information below.
asterisk1*CLI> core show version
Asterisk 15.2.2 built by root @ asterisk1 on a x86_64 running Linux on 2018-02-23 09:35:39 UTC
asterisk1*CLI> pjsip show endpoint 200@demo4.mydomain.com
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: 200@demo4.mydomain.com Unavailable 0 of inf
Aor: 200-demo4.mydomain.com 1
Contact: 200-demo4.mydomain.com/sip:200@demo4.mydomain. 9cc82bcb6d Unknown nan
Transport: transport-udp udp 0 0 0.0.0.0:5080
ParameterName : ParameterValue
=========================================================
100rel : yes
accountcode :
acl :
aggregate_mwi : true
allow : (alaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 200-demo4.mydomain.com
asymmetric_rtp_codec : false
auth :
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : <unknown>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : default
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 : auto
fax_detect : false
fax_detect_timeout : 0
force_avp : false
force_rport : true
from_domain : demo4.mydomain.com
from_user :
g726_non_standard : false
ice_support : false
identify_by : username,ip
inband_progress : false
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 : false
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth :
outbound_proxy : sip:192.168.102.10
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 : 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 : true
send_rpid : true
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 : transport-udp
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
asterisk1*CLI> pjsip show contact 200-demo4.mydomain.com/sip:200@demo4.mydomain.com
Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================
Contact: 200-demo4.mydomain.com/sip:200@demo4.mydomain.com 9cc82bcb6d Unknown nan
asterisk1*CLI> pjsip show aor 200-demo4.mydomain.com
Aor: <Aor..............................................> <MaxContact>
Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================
Aor: 200-demo4.mydomain.com 1
Contact: 200-demo4.mydomain.com/sip:200@demo4.mydomain.co 9cc82bcb6d Unknown nan
ParameterName : ParameterValue
===============================================
authenticate_qualify : false
contact : sip:200@demo4.mydomain.com
default_expiration : 3600
mailboxes :
max_contacts : 1
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : sip:192.168.102.10
qualify_frequency : 0
qualify_timeout : 3.000000
remove_existing : false
support_path : false
voicemail_extension :
Thank you,
Cyrille