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 |