[asterisk-bugs] [JIRA] (ASTERISK-22851) Asterisk/SIP stops responding

Jeremy Kister (JIRA) noreply at issues.asterisk.org
Tue Nov 12 23:28:03 CST 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-22851?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211723#comment-211723 ] 

Jeremy Kister edited comment on ASTERISK-22851 at 11/12/13 11:27 PM:
---------------------------------------------------------------------

here's a [^lsof.txt] of asterisk during the brokenness.
                
      was (Author: jkister):
    here's an {{lsof.txt}} of asterisk during the brokenness.
                  
> Asterisk/SIP stops responding
> -----------------------------
>
>                 Key: ASTERISK-22851
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22851
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.6.0, 11.7.0
>         Environment: Debian6
>            Reporter: Jeremy Kister
>            Severity: Blocker
>         Attachments: lsof.txt
>
>
> I have regularly (once a week, once per few hundred calls?) been having 
> problems with Asterisk's SIP stack not responding to packets from any of 
> my registered devices.  In the past, I could not tolerate the outage, so 
> i would restart asterisk to make things happy.
> My Asterisk server is currently in this broken state and I can leave it 
> this way for a short while.  Devices are registered to it and I can 'sip 
> qualify peer xxx':
> {code}
> pbx1*CLI> sip qualify peer 111
> [Nov 13 00:16:33] NOTICE[27681]: chan_sip.c:29489 sip_poke_peer: Still have a QUALIFY dialog active, deleting
> Really destroying SIP dialog '0513566165179f7a3e2ccbd64424e614 at 10.1.0.3:5060' Method: OPTIONS
> Reliably Transmitting (NAT) to 10.1.0.111:5060:
> OPTIONS sip:111 at 10.1.0.111:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.3:5060;branch=z9hG4bK276d50c3;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 10.1.0.3>;tag=as7e156001
> To: <sip:111 at 10.1.0.111:5060;transport=udp>
> Contact: <sip:asterisk at 10.1.0.3:5060>
> Call-ID: 07d95a2a12b438373c441a227877202f at 10.1.0.3:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 11.7.0-rc1
> Date: Wed, 13 Nov 2013 05:16:33 GMT
> Session-Expires: 7200
> Min-SE: 90
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> {code}
> on the network, this shows:
> {code}
> IP 10.1.0.3.5060 > 10.1.0.111.5060: SIP, length: 592
> E..l.... at .p#
>         ..
>         .o.....X..OPTIONS sip:111 at 10.1.0.111:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.3:5060;branch=z9hG4bK276d50c3;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 10.1.0.3>;tag=as7e156001
> To: <sip:111 at 10.1.0.111:5060;transport=udp>
> Contact: <sip:asterisk at 10.1.0.3:5060>
> Call-ID: 07d95a2a12b438373c441a227877202f at 10.1.0.3:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 11.7.0-rc1
> Date: Wed, 13 Nov 2013 05:16:33 GMT
> Session-Expires: 7200
> Min-SE: 90
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 909
> E`...... at ..w
>         .o
>         ..........SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.3:5060;branch=z9hG4bK276d50c3;rport
> From: "asterisk" <sip:asterisk at 10.1.0.3>;tag=as7e156001
> To: <sip:111 at 10.1.0.111:5060;transport=udp>;tag=0013c401da4a2525291558f9-6a2c8580
> Call-ID: 07d95a2a12b438373c441a227877202f at 10.1.0.3:5060
> Date: Wed, 13 Nov 2013 05:22:18 GMT
> CSeq: 102 OPTIONS
> Server: Cisco-CP7940G/8.0
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
> Accept: application/sdp,multipart/mixed,multipart/alternative
> Accept-Encoding: identity
> Accept-Language: en
> Supported: replaces,join,norefersub
> Content-Length: 235
> Content-Type: application/sdp
> Content-Disposition: session;handling=optional
> v=0
> o=Cisco-SIPUA 27539 0 IN IP4 10.1.0.111
> s=SIP Call
> t=0 0
> m=audio 0 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> {code}
> 'sip show peer xxx' all show Status OK:
> {code}
>   * Name       : 111
>   Description  : 
>   Secret       : <Set>
>   MD5Secret    : <Not set>
>   Remote Secret: <Not set>
>   Context      : extensions
>   Record On feature : automon
>   Record Off feature : automon
>   Subscr.Cont. : <Not set>
>   Language     : 
>   Tonezone     : <Not set>
>   AMA flags    : Unknown
>   Transfer mode: open
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup    : 1
>   Pickupgroup  : 1
>   Named Callgr : 
>   Nam. Pickupgr: 
>   MOH Suggest  : 
>   Mailbox      : 
>   VM Extension : asterisk
>   LastMsgsSent : 0/0
>   Call limit   : 0
>   Max forwards : 0
>   Dynamic      : Yes
>   Callerid     : "Family Rm" <111>
>   MaxCallBR    : 384 kbps
>   Expire       : -15413
>   Insecure     : no
>   Force rport  : Yes
>   Symmetric RTP: Yes
>   ACL          : Yes
>   DirectMedACL : No
>   T.38 support : Yes
>   T.38 EC mode : FEC
>   T.38 MaxDtgrm: -1
>   DirectMedia  : No
>   PromiscRedir : No
>   User=Phone   : No
>   Video Support: No
>   Text Support : No
>   Ign SDP ver  : No
>   Trust RPID   : Yes
>   Send RPID    : Yes
>   Subscriptions: Yes
>   Overlap dial : Yes
>   DTMFmode     : rfc2833
>   Timer T1     : 500
>   Timer B      : 32000
>   ToHost       : 
>   Addr->IP     : 10.1.0.111:5060
>   Defaddr->IP  : (null)
>   Prim.Transp. : UDP
>   Allowed.Trsp : UDP
>   Def. Username: 111
>   SIP Options  : (none)
>   Codecs       : (ulaw)
>   Codec Order  : (ulaw:20)
>   Auto-Framing :  No 
>   Status       : OK (110 ms)
>   Useragent    : Cisco-CP7940G/8.0
>   Reg. Contact : sip:111 at 10.1.0.111:5060;transport=udp
>   Qualify Freq : 300000 ms
>   Keepalive    : 0 ms
>   Sess-Timers  : Originate
>   Sess-Refresh : uas
>   Sess-Expires : 7200 secs
>   Min-Sess     : 90 secs
>   RTP Engine   : asterisk
>   Parkinglot   : 
>   Use Reason   : No
>   Encryption   : No
> {code}
> but whenever one of the devices tries to make a new call, Asterisk just 
> doesnt respond.  'sip set debug on' shows no packets.
> from the asterisk server (10.1.0.3), i can see one of my phones 
> (10.1.0.111) trying to make a call:
> {code}
> # tcpdump -i eth0 -s 0 -t -n host 10.1.0.111
> ARP, Request who-has 10.1.0.3 tell 10.1.0.111, length 46
> ARP, Reply 10.1.0.3 is-at 00:0c:29:07:39:8e, length 28
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
> IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.123 > 10.1.0.3.123: NTPv3, Client, length 48
> IP 10.1.0.3.123 > 10.1.0.111.123: NTPv3, Server, length 48
> ARP, Request who-has 10.1.0.111 tell 10.1.0.3, length 28
> ARP, Reply 10.1.0.111 is-at 00:13:c4:01:da:4a, length 46
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> IP 10.1.0.111.5060 > 10.1.0.3.5060: SIP, length: 926
> {code}

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira



More information about the asterisk-bugs mailing list