Endpoint Unavailable over time

Margon · February 6, 2019, 8:11am

Good afternoon, I’m just learning asterisk.
I am using FreePBX Distro 14.0.5.25. All extention work on pjsip.
I was able to configure everything so that my 200 phones (snom D715) would work fine.
I also have 2 phones nortel 1200 with which there are problems.
These phones are registered on asterisk and work normally for some time.
after about an hour, asterisk stops seeing them.
It looks like this:

pjsip show endpoint 2237

Endpoint: <Endpoint/CID…> <State…> <Channels.>
I/OAuth: <AuthId/UserName…>
Aor: <Aor…>
Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>
Transport: <TransportId…> <BindAddress…>
Identify: <Identify/Endpoint…>
Match: <criteria…>
Channel: <ChannelId…> <State…> <Time…>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>

Endpoint: 2237/2237 Unavailable 0 of inf
InAuth: 2237-auth/2237
Aor: 2237 1

ParameterName : ParameterValue

100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|alaw|g726|g723|g722|g729)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 2237
asymmetric_rtp_codec : false
auth : 2237-auth
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : “Чистякова И.” <2237>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : from-internal
cos_audio : 5
cos_video : 4
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
ice_support : false
identify_by : username,ip
inband_progress : false
incoming_mwi_mailbox :
language : ru
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 : true
outbound_auth :
outbound_proxy :
pickup_group :
preferred_codec_only : false
record_off_feature : apprecord
record_on_feature : apprecord
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : 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 : true
send_rpid : false
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 : 184
tos_video : 136
transport :
trust_id_inbound : true
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no

In this case, the call to this extetion passes, but it is impossible to call from it.

I will be immensely grateful for the tips in which direction I need to move.

arielgrin · February 6, 2019, 11:48am

Are the phones in the same network as FreePBX ?

Margon · February 6, 2019, 12:32pm

Phones and FeePbx are on the same network but on different subnets. NAT is not used.
In the subnet where the problem device is located (nortel) there are also devices (snom) which do not have such problems.

Here is how it looks in the logs:
Why the device(nortel) does not want to re-register.
[2019-02-06 13:18:59] VERBOSE[7706] res_pjsip/pjsip_configuration.c: Endpoint 9999 is now Reachable
[2019-02-06 13:18:59] VERBOSE[7706] res_pjsip/pjsip_options.c: Contact 9999/sip:[email protected]:5060 is now Reachable. RTT: 92.178 msec
[2019-02-06 15:18:59] VERBOSE[22498] res_pjsip/pjsip_options.c: Contact 9999/sip:[email protected]:5060 has been deleted
[2019-02-06 15:18:59] VERBOSE[22498] res_pjsip/pjsip_configuration.c: Endpoint 9999 is now Unreachable

arielgrin · February 6, 2019, 8:02pm

Just to be on the safe side, have you tried connecting the nortel on the same subnet as the freepbx server to check if it behaves the same or differently?

Margon · February 7, 2019, 8:35am

I tried it, the problem was repeated in the same subnet as asterisk.

Margon · February 7, 2019, 10:52am

Here’s a config for the Nortel phone:

more DeviceConfig.cfg
#DNS_DOMAIN
SIP_DOMAIN1 192.168.x.y
SERVER_IP1_1 192.168.x.y
SERVER_PORT1_1 5060
SERVER_TCP_PORT1_1 0
#SERVER_TLS_PORT1_1 0
IPV6_ENABLE NO
#PCPORT_ENABLE YES
LLDP_ENABLE YES

SIP_PING NO
NAT_SIGNALLING NONE
USE_RPORT NO
ENABLE_PRACK YES

TRANSFER_TYPE STANDARD
REDIRECT_TYPE RFC3261
MADN_DIALOG YES
MADN_TIMER 1800

Session Timer RFC4028

SESSION_TIMER_MIN_SE 90
REG_REFRESH_INTERVAL 86400
REGISTER_RETRY_TIME 30
REGISTER_RETRY_MAXTIME 1800

Settings to disable extended license

MAX_LOGINS 1
USB_HEADSET LOCK
EXP_MODULE_ENABLE NO
ENABLE_SERVICE_PACKAGE NO
IM_MODE DISABLED
AVAYA_AUTOMATIC_QoS NO
VQMON_PUBLISH NO
SIP_TLS_PORT 0
ENABLE_BT NO

Enable SSH

SSH YES
SSHID admin
SSHPWD password

#SERVER_RETRIES1 30
#DEF_USER1 user1

BANNER COMPANY NAME
AUTOLOGIN_ENABLE USE_AUTOLOGIN_ID

#DEF_AUDIO_QUALITY Low Medium High
DEF_LANG Russian
DST_ENABLED NO
TIMEZONE_OFFSET 10800
SNTP_ENABLE YES
SNTP_SERVER 192.168.z.w

SIP LOGGING

LOGSIP_ENABLE NO

with this config in the logs wrote:

[2019-02-05 11:59:39] VERBOSE[22498] res_pjsip_registrar.c: Added contact ‘sip:[email protected]:5060’ to AOR ‘9999’ with expiration of 7200 seconds
[2019-02-05 11:59:39] VERBOSE[4983] res_pjsip/pjsip_configuration.c: Endpoint 9999 is now Reachable
[2019-02-05 11:59:39] VERBOSE[4983] res_pjsip/pjsip_options.c: Contact 9999/sip:[email protected]:5060 is now Reachable. RTT: 110.658 msec
[2019-02-05 13:59:50] VERBOSE[7706] res_pjsip/pjsip_options.c: Contact 9999/sip:[email protected]:5060 has been deleted
[2019-02-05 13:59:50] VERBOSE[7706] res_pjsip/pjsip_configuration.c: Endpoint 9999 is now Unreachable

I tried to play with the parameter REG_REFRESH_INTERVAL set it to 1500.
with this parameter, the logs look like this:

[2019-02-07 11:24:18] VERBOSE[31690] res_pjsip_registrar.c: Added contact ‘sip:[email protected]:5060’ to AOR ‘9999’ with expiration of 1500 seconds
[2019-02-07 11:24:18] VERBOSE[31690] res_pjsip/pjsip_configuration.c: Endpoint 9999 is now Reachable
[2019-02-07 11:24:18] VERBOSE[31690] res_pjsip/pjsip_options.c: Contact 9999/sip:[email protected]:5060 is now Reachable. RTT: 145.096 msec
[2019-02-07 11:49:26] VERBOSE[12571] res_pjsip/pjsip_options.c: Contact 9999/sip:[email protected]:5060 has been deleted
[2019-02-07 11:49:26] VERBOSE[12571] res_pjsip/pjsip_configuration.c: Endpoint 9999 is now Unreachable

system · February 14, 2019, 10:52am

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.