We have all our extension set to PJSIP in our FreePBX 13 with Asterisk 13. Some of the extensions ring multiple endpoints as configured in their extension while other will only ring one endpoint. In SIP it was the last registered device but with PJSIP, in the extensions we’re having an issue with, it seems to ring whichever device it wants to ring. I have verified that the extensions are setup the exact same and still not seeing any possible configuration issues.
Here is the show endpoint of on of the extensions I am having issues with.
[root@sip asterisk]# asterisk -rx "pjsip show endpoint 701"
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: <ip/cidr.........................>
Channel: <ChannelId......................................> <State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
=========================================================================================
Endpoint: 701/701 Not in use 0 of inf
InAuth: 701-auth/701
Aor: 701 5
Contact: 701/sip:[email protected]:24628;rinstanc 826c683d6a Avail 131.972
Contact: 701/sip:[email protected]:5060 c60b5bc3e5 Avail 6.613
Identify: 701-identify/701
ParameterName : ParameterValue
====================================================
100rel : yes
accountcode :
aggregate_mwi : true
allow : (ulaw|g722)
allow_subscribe : true
allow_transfer : true
aors : 701
auth : 701-auth
call_group :
callerid : "device" <701>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
context : from-internal
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 : auto
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 : en
mailboxes :
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_suggest : default
mwi_from_user :
named_call_group :
named_pickup_group :
one_touch_recording : false
outbound_auth :
outbound_proxy :
pickup_group :
record_off_feature : automixmon
record_on_feature : automixmon
rewrite_contact : true
rpid_immediate : 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
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 :
trust_id_inbound : true
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
[root@sip asterisk]#