@jhayes wrote:
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:701@166.172.185.222:24628;rinstanc 826c683d6a Avail 131.972 Contact: 701/sip:701@172.16.8.50:5060 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]#
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