[asterisk-bugs] [JIRA] (ASTERISK-27104) After 10 minutes after reboot get Peer 'XXX' is now UNREACHABLE!

Pablo Parodi (JIRA) noreply at issues.asterisk.org
Sun Jul 2 03:23:57 CDT 2017


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27104?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237617#comment-237617 ] 

Pablo Parodi commented on ASTERISK-27104:
-----------------------------------------

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 at 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 at 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 at 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 at 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>




> After 10 minutes after reboot get Peer 'XXX' is now UNREACHABLE! 
> -----------------------------------------------------------------
>
>                 Key: ASTERISK-27104
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27104
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Registration
>    Affects Versions: 11.25.0
>         Environment: PBX with public IP - Distribuited IP phones in several places
>            Reporter: Pablo Parodi
>            Severity: Critical
>
> 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.



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