[asterisk-bugs] [Asterisk 0016270]: Asterisk doesn't free udp ports

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16270 
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Reported By:                corruptor
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16270
Category:                   Core/Netsock
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-444 
Regression:                 Yes 
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:             2009-11-18 07:48 CST
Last Modified:              2009-11-24 02:28 CST
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Summary:                    Asterisk doesn't free udp ports
Description: 
I've had this problem on 1.4.26 also.
I've recently upgraded to 1.6.0.17 and problem also exists here.
Concurrent number of calls is usually not more than 25. All calls are SIP
to SIP. 

Number of udp ports used by asterisk is growing. For example at the moment
of writing this note:
34 active channels
17 active calls
1871 calls processed

Total number of udp ports open  - 1023

At the other moment it could be a little bit lower but anyway it is
growing.

There are many queues and different ivrs on the server. We also use AMI
for originating calls for our CC. 
I've noticed that asterisk writes warnings to log, for example:
WARNING[26372] chan_sip.c: Trying to destroy
"6009EE42DD5A4BDA81A18571C82B76310x0ad20c43", not found in dialog list?!?!
I don't know if it's related or not.

Please help me to debug this problem.

Sorry if I've chosen the wrong category.
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Relationships       ID      Summary
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related to          0015403 [patch] Session timer is not activated
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 (0114218) luca942 (reporter) - 2009-11-24 02:28
 https://issues.asterisk.org/view.php?id=16270#c114218 
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After a lot of debugging I have revealed that this behavior is provoked by
auto-congesting channel.
NOTICE[25556] chan_sip.c: Auto-congesting SIP/test-00000007
I have attached 2 files sip_trace.log and asterisk_cosole.txt for more
details. A call from 52040449 was made to SIP line "test". SIP line was
successfully registered, but I have simulated network failure to provoke
auto-congestion on Asterisk. This is very common situation in real life,
when registered SIP line becomes unreachable due to end-user connection
failure. After unsuccessfull call all 4 UDP sockets 14536, 14537, 13526,
13527 remained opened. They are closed only if you reboot Asterisk. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-24 02:28 luca942        Note Added: 0114218                          
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