[asterisk-bugs] [JIRA] (ASTERISK-22851) Asterisk/SIP stops responding
Jeremy Kister (JIRA)
noreply at issues.asterisk.org
Tue Nov 12 23:26:03 CST 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22851?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Jeremy Kister updated ASTERISK-22851:
-------------------------------------
Attachment: lsof.txt
here's an LSOF 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}
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