ASTERISK-27104: After 10 minutes after reboot get Peer 'XXX' is now UNREACHABLE!

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Summary: ASTERISK-27104: After 10 minutes after reboot get Peer 'XXX' is now UNREACHABLE!

Reporter: Pablo Parodi (pparodi) Labels:

Date Opened: 2017-07-02 03:10:46 Date Closed: 2017-07-02 05:34:21

Priority: Critical Regression? No

Status: Closed/Complete Components: Channels/chan_sip/Registration

Versions: 11.25.0 Frequency of
Occurrence Constant

Related
Issues:

Environment: PBX with public IP - Distribuited IP phones in several places Attachments:

Description: 10 minutes after restart the pbx, start to get the followin messages:
- Extension unreacheble
- Correct auth, but based on stale nonce received from

This issue started yesterday at the end of the working day, before that time the PBX was working fine from some year.
There was no changes before this issue.

If I reboot the pbx, it start to work great by 10 minutes. After that the extensions can't get incomming and outgoing calls.

I have tcpdump, sip set debug and configuration files.

Comments: By: Asterisk Team (asteriskteam) 2017-07-02 03:10:48.506-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Pablo Parodi (pparodi) 2017-07-02 03:20:15.272-0500

Tests and changes done:
- set nonce=true
- qualify=no and qualify=yes
- change extension's type from friend to peer and viceversa
- update Asterisk to11.25.0

Same issue than: http://forums.asterisk.org/viewtopic.php?p=144955
and ASTERISK-16062

No anwer or workaround found

By: Pablo Parodi (pparodi) 2017-07-02 03:20:47.162-0500

Sip set debug ip PHONE-EXTERNAL-IP-ADDRESS

<------------->
Really destroying SIP dialog 'ZDIxYjViZDQ4NWZmNjFhODUwOWQxZjhkM2QyMTk2OTE.' Method: BYE

<--- SIP read from UDP:PHONE-EXTERNAL-IP-ADDRESS:61446 --->
INVITE sip:299@PBX-IP-ADDRESS:5060 SIP/2.0
Via: SIP/2.0/UDP PHONE-LOCAL-IP-ADDRESS:61446;branch=z9hG4bK-d8754z-4c06ca335b60f675-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:104@PHONE-LOCAL-IP-ADDRESS:61446;rinstance=b04569da076865f7>
To: <sip:299@PBX-IP-ADDRESS:5060>
From: "104 - Softphone"<sip:PHONE-EXTENSION-A@PBX-IP-ADDRESS:5060>;tag=c80b6155
Call-ID: NDkzYzVhMDgwYTM0Y2Q0MmZmODlkZDEyYjFlZTNiZGM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 407

v=0
o=3cxVCE 390460005 231604545 IN IP4 PHONE-LOCAL-IP-ADDRESS
s=3cxVCE Audio Call
c=IN IP4 PHONE-LOCAL-IP-ADDRESS
t=0 0
m=audio 40030 RTP/AVP 3 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 PHONE-LOCAL-IP-ADDRESS
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv

<------------->
--- (13 headers 18 lines) ---
Sending to PHONE-EXTERNAL-IP-ADDRESS:61446 (NAT)
Sending to PHONE-EXTERNAL-IP-ADDRESS:61446 (NAT)
Using INVITE request as basis request - NDkzYzVhMDgwYTM0Y2Q0MmZmODlkZDEyYjFlZTNiZGM.
Found peer '104' for '104' from PHONE-EXTERNAL-IP-ADDRESS:61446

<--- Reliably Transmitting (NAT) to PHONE-EXTERNAL-IP-ADDRESS:61446 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP PHONE-LOCAL-IP-ADDRESS:61446;branch=z9hG4bK-d8754z-4c06ca335b60f675-1---d8754z-;received=PHONE-EXTERNAL-IP-ADDRESS;rport=61446
From: "104 - Softphone"<sip:104@PBX-IP-ADDRESS:5060>;tag=c80b6155
To: <sip:299@PBX-IP-ADDRESS:5060>;tag=as3617d26f
Call-ID: NDkzYzVhMDgwYTM0Y2Q0MmZmODlkZDEyYjFlZTNiZGM.
CSeq: 1 INVITE
Server: FPBX-2.8.1(11.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e715fa3"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'NDkzYzVhMDgwYTM0Y2Q0MmZmODlkZDEyYjFlZTNiZGM.' in 32000 ms (Method: INVITE)
Retransmitting #1 (NAT) to PHONE-EXTERNAL-IP-ADDRESS:61446:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP PHONE-LOCAL-IP-ADDRESS:61446;branch=z9hG4bK-d8754z-4c06ca335b60f675-1---d8754z-;received=PHONE-EXTERNAL-IP-ADDRESS;rport=61446
From: "104 - Softphone"<sip:104@PBX-IP-ADDRESS:5060>;tag=c80b6155
To: <sip:299@PBX-IP-ADDRESS:5060>;tag=as3617d26f
Call-ID: NDkzYzVhMDgwYTM0Y2Q0MmZmODlkZDEyYjFlZTNiZGM.
CSeq: 1 INVITE
Server: FPBX-2.8.1(11.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e715fa3"
Content-Length: 0

By: Pablo Parodi (pparodi) 2017-07-02 03:21:08.780-0500

CALL FLOW:
xINVITE sip:299@PBX-IP-ADDRESS:5060 SIP/2.0
PHONE-EXTERNAL-IP-ADDRESS:61446 PBX-IP-ADDRESS:5060 192.168.10.254:5060xVia: SIP/2.0/UDP PHONE-LOCAL-IP-ADDRESS:61446;branch=z9hG4bK-d8754z-9d4780076e6b886e-
qqqqqqqqqqwqqqqqqqqq qqqqqqqqqqwqqqqqqqqq qqqqqqqqqqwqqqqqqqqqx--d8754z-;rport
x INVITE (SDP) x x xMax-Forwards: 70
04:01:38.758185 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x xContact: <sip:104@PHONE-LOCAL-IP-ADDRESS:61446;rinstance=b04569da076865f7>
+0.000548 x 401 Unauthorized x xTo: <sip:299@PBX-IP-ADDRESS:5060>
04:01:38.758733 x <qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq x xFrom: "104 - Softphone"<sip:104@PBX-IP-ADDRESS:5060>;tag=8c2d262b
+0.498311 x INVITE (SDP) x x xCall-ID: NjQ2Yjc2NmVjYzkzYTk2Y2E1ZDY2MTNhMzRmYzE3OWY.
04:01:39.257044 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x xCSeq: 1 INVITE
+0.001416 x 401 Unauthorized x xAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER
04:01:39.258460 x <<<qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq x xINFO, MESSAGE
+0.999521 x 401 Unauthorized x xContent-Type: application/sdp
04:01:40.257981 x <<<qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq x xSupported: replaces
+0.000968 x INVITE (SDP) x x xUser-Agent: 3CXPhone 6.0.26523.0
04:01:40.258949 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x xContent-Length: 407
+1.999782 x INVITE (SDP) x x x
04:01:42.258731 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x xv=0
+0.000326 x 401 Unauthorized x xo=3cxVCE 348437625 213370980 IN IP4 PHONE-LOCAL-IP-ADDRESS
04:01:42.259057 x <<<qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq x xs=3cxVCE Audio Call
+3.999173 x 401 Unauthorized x xc=IN IP4 PHONE-LOCAL-IP-ADDRESS
04:01:46.258230 x <<<qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq x xt=0 0
+0.000944 x INVITE (SDP) x x xm=audio 40036 RTP/AVP 3 8 0 101
04:01:46.259174 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x xa=rtpmap:3 GSM/8000
+3.999158 x 401 Unauthorized x xa=rtpmap:8 PCMA/8000
04:01:50.258332 x <<<qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq x xa=rtpmap:0 PCMU/8000
x x x xa=rtpmap:101 telephone-event/8000
x x x xa=fmtp:101 0-15
x x x xa=ptime:20
x x x xa=sendrecv
x x x xm=video 40034 RTP/AVP 34
x x x xc=IN IP4 PHONE-LOCAL-IP-ADDRESS
x x x xa=rtpmap:34 H263/90000
x x x xa=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
x x x xa=sendrecv

By: Pablo Parodi (pparodi) 2017-07-02 03:22:31.589-0500

PBX*CLI> sip show settings

Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.8.1(11.25.0)
SDP Session Name: Asterisk PBX 11.25.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: Yes
Jitterbuffer forced: Yes
Jitterbuffer max size: 80
Jitterbuffer resync: 1000
Jitterbuffer impl: fixed
Jitterbuffer log: No

Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|g729)
Codec Order: g729:20,gsm:20,alaw:20,ulaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 360 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 360 secs
Outbound reg. timeout: 40 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 60
Use ClientCode: No
Progress inband: Never
Language: es
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97

----

PBX*CLI> sip show peer 104

* Name : 104
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language : es
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 104@device
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "device" <104>
MaxCallBR : 384 kbps
Expire : 91
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : info
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : PHONE-EXTERNAL-IP-ADDRESS:61446
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 104
SIP Options : (none)
Codecs : (gsm|ulaw|alaw|g729)
Codec Order : (g729:20,gsm:20,alaw:20,ulaw:20)
Auto-Framing : No
Status : OK (44 ms)
Useragent : 3CXPhone 6.0.26523.0
Reg. Contact : sip:104@PHONE-LOCAL-IP-ADDRESS:61446;rinstance=b04569da076865f7
Qualify Freq : 60000 ms
Keepalive : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

AFTER ISSUE:
PBX*CLI> sip show peer 104

* Name : 104
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language : es
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 104@device
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "device" <104>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : info
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 104
SIP Options : replaces replace
Codecs : (gsm|ulaw|alaw|g729)
Codec Order : (g729:20,gsm:20,alaw:20,ulaw:20)
Auto-Framing : No
Status : UNKNOWN
Useragent : 3CXPhone 6.0.26523.0
Reg. Contact : sip:104@192.168.1.115:61446;rinstance=b04569da076865f7
Qualify Freq : 60000 ms
Keepalive : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

PBX*CLI>

By: Joshua C. Colp (jcolp) 2017-07-02 05:34:05.163-0500

Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions