[asterisk-bugs] [JIRA] (ASTERISK-25107) Phone losing registration 10 seconds after reboot.
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Thu May 21 09:52:33 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25107?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-25107:
------------------------------------
Description:
I am using Asterisk 13.3.2 realtime with the new pjsip driver.
For some reason, if I remove the network cable from a registered phone and I reconnect it, the phone registers and
after about 10 seconds, the phone loose its registration and reboot. The phone stays registered after it rebooted. Am I missing something?
I am not experiencing the issue with the chan_sip driver.
Here is my phone config:
[Edit by Rusty - Excessive debug in the description field. Please attach your debug and configuration to the issue via More > Attach files. Attach them as .txt for easy access by others.]
was:
I am using Asterisk 13.3.2 realtime with the new pjsip driver.
For some reason, if I remove the network cable from a registered phone and I reconnect it, the phone registers and
after about 10 seconds, the phone loose its registration and reboot. The phone stays registered after it rebooted. Am I missing something?
I am not experiencing the issue with the chan_sip driver.
Here is my phone config:
ParameterName : ParameterValue
==================================================
authenticate_qualify : false
contact : sip:U_8374 at 10.188.4.26:5061
default_expiration : 3600
mailboxes :
max_contacts : 1
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy :
qualify_frequency : 0
remove_existing : false
support_path : false
ParameterName : ParameterValue
======================================================
100rel : yes
accountcode :
aggregate_mwi : true
allow : (ulaw)
allow_subscribe : true
allow_transfer : true
aors : U_8374
auth : U_8374
call_group :
callerid : "Asterisk test" <2708>
callerid_privacy : allowed
callerid_tag :
connected_line_method : invite
context : garneau-interne
cos_audio : 6
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 : rfc4733
fax_detect : false
force_avp : false
force_rport : false
from_domain :
from_user : U_8399
ice_support : false
identify_by : username
inband_progress : false
language : fr
mailboxes : 2707 at default
media_address :
media_encryption : none
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 : U_8374
outbound_proxy :
pickup_group :
record_off_feature : automixmon
record_on_feature : automixmon
rewrite_contact : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_symmetric : false
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : true
send_rpid : true
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 : 184
tos_video : 0
transport : transport-udp
trust_id_inbound : true
trust_id_outbound : true
use_avpf : false
use_ptime : false
user_eq_phone : false
Asterisk PJsip debug log:
PJSIP Logging enabled
[May 20 08:57:38] <--- Received SIP request (464 bytes) from UDP:10.188.4.26:5061 --->
[May 20 08:57:38] REGISTER sip:206.167.100.39 SIP/2.0
[May 20 08:57:38] Via: SIP/2.0/UDP 10.188.4.26:5061;branch=z9hG4bK-1da3da5f
[May 20 08:57:38] From: <sip:U_8374 at 206.167.100.39>;tag=8e8f0ea47a7ee063o0
[May 20 08:57:38] To: <sip:U_8374 at 206.167.100.39>
[May 20 08:57:38] Call-ID: 9423e4bb-576aa77b at 10.188.4.26
[May 20 08:57:38] CSeq: 50108 REGISTER
[May 20 08:57:38] Max-Forwards: 70
[May 20 08:57:38] Contact: <sip:U_8374 at 10.188.4.26:5061>;expires=30
[May 20 08:57:38] User-Agent: Cisco/SPA514G-7.5.7
[May 20 08:57:38] Content-Length: 0
[May 20 08:57:38] Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
[May 20 08:57:38] Supported: replaces
[May 20 08:57:38]
[May 20 08:57:38]
[May 20 08:57:38] <--- Transmitting SIP response (478 bytes) to UDP:10.188.4.26:5061 --->
[May 20 08:57:38] SIP/2.0 401 Unauthorized
[May 20 08:57:38] Via: SIP/2.0/UDP 10.188.4.26:5061;received=10.188.4.26;branch=z9hG4bK-1da3da5f
[May 20 08:57:38] Call-ID: 9423e4bb-576aa77b at 10.188.4.26
[May 20 08:57:38] From: <sip:U_8374 at 206.167.100.39>;tag=8e8f0ea47a7ee063o0
[May 20 08:57:38] To: <sip:U_8374 at 206.167.100.39>;tag=z9hG4bK-1da3da5f
[May 20 08:57:38] CSeq: 50108 REGISTER
[May 20 08:57:38] WWW-Authenticate: Digest realm="asterisk",nonce="1432126658/9e6371431e236030a6d01bb82ecddc53",opaque="0ac8fae7159f136b",algorithm=md5,qop="auth"
[May 20 08:57:38] Server: Asterisk PBX 13.3.2
[May 20 08:57:38] Content-Length: 0
[May 20 08:57:38]
[May 20 08:57:38]
[May 20 08:57:39] <--- Received SIP request (722 bytes) from UDP:10.188.4.26:5061 --->
[May 20 08:57:39] REGISTER sip:206.167.100.39 SIP/2.0
[May 20 08:57:39] Via: SIP/2.0/UDP 10.188.4.26:5061;branch=z9hG4bK-f131a7b7
[May 20 08:57:39] From: <sip:U_8374 at 206.167.100.39>;tag=8e8f0ea47a7ee063o0
[May 20 08:57:39] To: <sip:U_8374 at 206.167.100.39>
[May 20 08:57:39] Call-ID: 9423e4bb-576aa77b at 10.188.4.26
[May 20 08:57:39] CSeq: 50109 REGISTER
[May 20 08:57:39] Max-Forwards: 70
[May 20 08:57:39] Authorization: Digest username="U_8374",realm="asterisk",nonce="1432126658/9e6371431e236030a6d01bb82ecddc53",uri="sip:206.167.100.39",algorithm=MD5,response="5f45d42aa4fcd7d48e64efca4a049842",opaque="0ac8fae7159f136b",qop=auth,nc=00000001,cnonce="56508fed"
[May 20 08:57:39] Contact: <sip:U_8374 at 10.188.4.26:5061>;expires=30
[May 20 08:57:39] User-Agent: Cisco/SPA514G-7.5.7
[May 20 08:57:39] Content-Length: 0
[May 20 08:57:39] Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
[May 20 08:57:39] Supported: replaces
[May 20 08:57:39]
[May 20 08:57:39]
[May 20 08:57:39] -- Added contact 'sip:U_8374 at 10.188.4.26:5061' to AOR 'U_8374' with expiration of 60 seconds
[May 20 08:57:39] <--- Transmitting SIP response (409 bytes) to UDP:10.188.4.26:5061 --->
[May 20 08:57:39] SIP/2.0 200 OK
[May 20 08:57:39] Via: SIP/2.0/UDP 10.188.4.26:5061;received=10.188.4.26;branch=z9hG4bK-f131a7b7
[May 20 08:57:39] Call-ID: 9423e4bb-576aa77b at 10.188.4.26
[May 20 08:57:39] From: <sip:U_8374 at 206.167.100.39>;tag=8e8f0ea47a7ee063o0
[May 20 08:57:39] To: <sip:U_8374 at 206.167.100.39>;tag=z9hG4bK-f131a7b7
[May 20 08:57:39] CSeq: 50109 REGISTER
[May 20 08:57:39] Date: Wed, 20 May 2015 12:57:39 GMT
[May 20 08:57:39] Contact: <sip:U_8374 at 10.188.4.26:5061>;expires=59
[May 20 08:57:39] Server: Asterisk PBX 13.3.2
[May 20 08:57:39] Content-Length: 0
[May 20 08:57:39]
[May 20 08:57:39]
[May 20 08:57:39] <--- Transmitting SIP request (491 bytes) to UDP:10.188.4.30:5061 --->
[May 20 08:57:39] OPTIONS sip:U_9172 at 10.188.4.30:5061 SIP/2.0
[May 20 08:57:39] Via: SIP/2.0/UDP 206.167.100.39:5060;rport;branch=z9hG4bKPj0780b124-ef75-49a2-b085-5098efc190dc
[May 20 08:57:39] From: <sip:445ea362-ee78-475d-a2fa-b58ec04aad36 at 206.167.100.39>;tag=120412c8-bb40-4080-8ec6-e4f1b8fe47c0
[May 20 08:57:39] To: <sip:U_9172 at 10.188.4.30>
[May 20 08:57:39] Contact: <sip:445ea362-ee78-475d-a2fa-b58ec04aad36 at 206.167.100.39:5060>
[May 20 08:57:39] Call-ID: 01f7db06-cd4f-4d48-8dee-3641441ba799
[May 20 08:57:39] CSeq: 6248 OPTIONS
[May 20 08:57:39] Max-Forwards: 70
[May 20 08:57:39] User-Agent: Asterisk PBX 13.3.2
[May 20 08:57:39] Content-Length: 0
[May 20 08:57:39]
[May 20 08:57:39]
[May 20 08:57:39] <--- Received SIP response (475 bytes) from UDP:10.188.4.30:5061 --->
[May 20 08:57:39] SIP/2.0 200 OK
[May 20 08:57:39] To: <sip:U_9172 at 10.188.4.30>;tag=70a3a0f156c02e8fi0
[May 20 08:57:39] From: <sip:445ea362-ee78-475d-a2fa-b58ec04aad36 at 206.167.100.39>;tag=120412c8-bb40-4080-8ec6-e4f1b8fe47c0
[May 20 08:57:39] Call-ID: 01f7db06-cd4f-4d48-8dee-3641441ba799
[May 20 08:57:39] CSeq: 6248 OPTIONS
[May 20 08:57:39] Via: SIP/2.0/UDP 206.167.100.39:5060;branch=z9hG4bKPj0780b124-ef75-49a2-b085-5098efc190dc
[May 20 08:57:39] Server: Cisco/SPA514G-7.5.7
[May 20 08:57:39] Content-Length: 0
[May 20 08:57:39] Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
[May 20 08:57:39] Supported: replaces
*************** 10 seconds later the phone reboot *****************************
[May 20 08:57:45] <--- Transmitting SIP request (487 bytes) to UDP:10.188.6.23:5061 --->
[May 20 08:57:45] OPTIONS sip:2727 at 10.188.6.23:5061 SIP/2.0
[May 20 08:57:45] Via: SIP/2.0/UDP 206.167.100.39:5060;rport;branch=z9hG4bKPj702b5494-cd16-4af6-887a-edd8bb2cffbc
[May 20 08:57:45] From: <sip:034068e2-b9c7-4305-b165-488bf37f6d7f at 206.167.100.39>;tag=2c779ce5-890a-46d5-9d9d-9849888dc187
[May 20 08:57:45] To: <sip:2727 at 10.188.6.23>
[May 20 08:57:45] Contact: <sip:034068e2-b9c7-4305-b165-488bf37f6d7f at 206.167.100.39:5060>
[May 20 08:57:45] Call-ID: ff69ca34-c32a-4339-8baa-1bdfc860f99d
[May 20 08:57:45] CSeq: 3283 OPTIONS
[May 20 08:57:45] Max-Forwards: 70
[May 20 08:57:45] User-Agent: Asterisk PBX 13.3.2
[May 20 08:57:45] Content-Length: 0
[May 20 08:57:45]
[May 20 08:57:45]
[May 20 08:57:46] <--- Received SIP response (473 bytes) from UDP:10.188.6.23:5061 --->
[May 20 08:57:46] SIP/2.0 200 OK
[May 20 08:57:46] To: <sip:2727 at 10.188.6.23>;tag=425a3bd2b3bae70bi0
[May 20 08:57:46] From: <sip:034068e2-b9c7-4305-b165-488bf37f6d7f at 206.167.100.39>;tag=2c779ce5-890a-46d5-9d9d-9849888dc187
[May 20 08:57:46] Call-ID: ff69ca34-c32a-4339-8baa-1bdfc860f99d
[May 20 08:57:46] CSeq: 3283 OPTIONS
[May 20 08:57:46] Via: SIP/2.0/UDP 206.167.100.39:5060;branch=z9hG4bKPj702b5494-cd16-4af6-887a-edd8bb2cffbc
[May 20 08:57:46] Server: Cisco/SPA514G-7.5.7
[May 20 08:57:46] Content-Length: 0
[May 20 08:57:46] Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
[May 20 08:57:46] Supported: replaces
[May 20 08:57:46]
[May 20 08:57:46]
[May 20 08:57:54] <--- Received SIP request (721 bytes) from UDP:10.188.4.26:5061 --->
[May 20 08:57:54] REGISTER sip:206.167.100.39 SIP/2.0
[May 20 08:57:54] Via: SIP/2.0/UDP 10.188.4.26:5061;branch=z9hG4bK-fbd168b4
[May 20 08:57:54] From: <sip:U_8374 at 206.167.100.39>;tag=8e8f0ea47a7ee063o0
[May 20 08:57:54] To: <sip:U_8374 at 206.167.100.39>
[May 20 08:57:54] Call-ID: 9423e4bb-576aa77b at 10.188.4.26
[May 20 08:57:54] CSeq: 50110 REGISTER
[May 20 08:57:54] Max-Forwards: 70
[May 20 08:57:54] Authorization: Digest username="U_8374",realm="asterisk",nonce="1432126658/9e6371431e236030a6d01bb82ecddc53",uri="sip:206.167.100.39",algorithm=MD5,response="4ce611ce455c5bce840d5e1eaab69e6f",opaque="0ac8fae7159f136b",qop=auth,nc=00000002,cnonce="56508fed"
[May 20 08:57:54] Contact: <sip:U_8374 at 10.188.4.26:5061>;expires=0
[May 20 08:57:54] User-Agent: Cisco/SPA514G-7.5.7
[May 20 08:57:54] Content-Length: 0
[May 20 08:57:54] Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
[May 20 08:57:54] Supported: replaces
[May 20 08:57:54]
[May 20 08:57:54]
[May 20 08:57:54] -- Removed contact 'sip:U_8374 at 10.188.4.26:5061' from AOR 'U_8374' due to request
[May 20 08:57:54] <--- Transmitting SIP response (358 bytes) to UDP:10.188.4.26:5061 --->
[May 20 08:57:54] SIP/2.0 200 OK
[May 20 08:57:54] Via: SIP/2.0/UDP 10.188.4.26:5061;received=10.188.4.26;branch=z9hG4bK-fbd168b4
[May 20 08:57:54] Call-ID: 9423e4bb-576aa77b at 10.188.4.26
[May 20 08:57:54] From: <sip:U_8374 at 206.167.100.39>;tag=8e8f0ea47a7ee063o0
[May 20 08:57:54] To: <sip:U_8374 at 206.167.100.39>;tag=z9hG4bK-fbd168b4
[May 20 08:57:54] CSeq: 50110 REGISTER
[May 20 08:57:54] Date: Wed, 20 May 2015 12:57:54 GMT
[May 20 08:57:54] Server: Asterisk PBX 13.3.2
[May 20 08:57:54] Content-Length: 0
[May 20 08:57:54]
[May 20 08:57:54]
dti-asterisk*CLI>
> Phone losing registration 10 seconds after reboot.
> --------------------------------------------------
>
> Key: ASTERISK-25107
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25107
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 13.3.2
> Environment: CentOS release 6.6 (Final)
> Asterisk 13.3.2 realtime PJsip driver
> Phones: Cisco SPA514G firmware 7.5.7
> Reporter: Carl Fortin
> Severity: Minor
> Attachments: 10_sec_reboot.pcapng, Capture_Cisco_registration_lost_call_flow.PNG, Capture_Cisco_registration_lost.PNG
>
>
> I am using Asterisk 13.3.2 realtime with the new pjsip driver.
> For some reason, if I remove the network cable from a registered phone and I reconnect it, the phone registers and
> after about 10 seconds, the phone loose its registration and reboot. The phone stays registered after it rebooted. Am I missing something?
> I am not experiencing the issue with the chan_sip driver.
> Here is my phone config:
> [Edit by Rusty - Excessive debug in the description field. Please attach your debug and configuration to the issue via More > Attach files. Attach them as .txt for easy access by others.]
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