[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16382 
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Reported By:                lftsy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/DatabaseSupport
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.27.1 
JIRA:                       SWP-478 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-12-03 10:04 CST
Last Modified:              2009-12-04 10:30 CST
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Summary:                    SIP OPTIONS qualify message forever
Description: 
Hello, I have a trouble with different Asterisk versions (1.4.26, 1.4.27,
1.4.27.1). When I use the steps below, Asterisk starts to send SIP OPTIONS
to the previous IP/port used by a SIP realtime peer (that has been pruned)
and will keep trying to send SIP OPTIONS pings forever, event if the peer
is connected with a new IP/port address.

I have just checked with my old Asterisk 1.2.27 with the same sip.conf and
I do not have the problem, the SIP OPTIONS stops once register timer has
expired.

During my experience to reproduce the bug, I have been able to have 10
IP/port currently pinged by the Asterisk server for one single peer.
And the only way to stop it is to restart Asterisk...

Thank you for your attention!
Marc Leurent
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Relationships       ID      Summary
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related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
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 (0114758) lftsy (reporter) - 2009-12-04 10:30
 https://issues.asterisk.org/view.php?id=16382#c114758 
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Ok, I have set these 2 commands and launched a pcap capture.
 bas-flu-vp-ast-03*CLI> core set debug 20
 Core debug was 0 and is now 20
 bas-flu-vp-ast-03*CLI> sip history
 SIP History Recording Enabled (use 'sip show history')

I'm going to add the pcap file with the filter sip contains "0245667911"
* I have simply registered the 0245667911
* I use sip prune realtime 0245667911 to remove it (or reload command)
from memory and I unplug the phone
   bas-flu-vp-ast-03*CLI> sip show peers like 0245667911
   Name/username              Host            Dyn Nat ACL Port     Status 
   Realtime
   0245667911/0245667911      212.147.47.91    D   N      5060     OK (17
ms) Cached RT
   1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0
offline]


bas-flu-vp-ast-03*CLI> sip show peers like 0245667911
Name/username              Host            Dyn Nat ACL Port     Status    
Realtime
0245667911/0245667911      212.147.47.91    D   N      5060    
UNREACHABLE Cached RT

* I call 0245667911 from 0245667945, it recreates the peer with MySQL DB
contents and Expire -1 (IP 213.162.3.159:5060 is pinged)
* Even if I connect the phone from a different port/IP
(194.38.160.113:5060), the first port/IP (IP 213.162.3.159:5060) is flooded
by SIP OPTIONS forever... As you can see, the register time was 600s, but
after 15 min after, the qualify OPTIONS message are still there, and it
last until next restart. You will also notice that the new IPs is qualified
too. And I can do it again and again with other IP/ports and there will all
be SIP OPTIONS qualified

Thank you for your time 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-04 10:30 lftsy          Note Added: 0114758                          
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