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

Jeremy Kister (JIRA) noreply at issues.asterisk.org
Tue Nov 12 23:32:03 CST 2013


     [ https://issues.asterisk.org/jira/browse/ASTERISK-22851?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jeremy Kister updated ASTERISK-22851:
-------------------------------------

    Description: 
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' - see [^qualify.txt]

on the network, this shows [^qualify-tcpdump.txt]

'sip show peer xxx' all show Status OK [^sip-show-peer.txt]

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}

  was:
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' - see [^qualify.txt]

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}

    
> 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, qualify-tcpdump.txt, qualify.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' - see [^qualify.txt]
> on the network, this shows [^qualify-tcpdump.txt]
> 'sip show peer xxx' all show Status OK [^sip-show-peer.txt]
> 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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