[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:           1.6.0.17 
JIRA:                        
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:             2009-11-18 07:48 CST
Last Modified:              2009-11-20 09:06 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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 (0114041) corruptor (reporter) - 2009-11-20 09:06
 https://issues.asterisk.org/view.php?id=16270#c114041 
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I've managed to find out what causes this problem.
Some of our call center employees use softphone SJphone/1.65.377a. I've
uploaded the sip dialog  (file callflow) after which asterisk doesn't free
udp two ports. You can also see the full version of it. Asterisk is at
10.150.32.194.
SjPhone is behind NAT 10.150.32.98. 
I've checked all sip call flows related to used UDP port and they are all
the same. This happens if call isn't answered and caller (SjPhone) cancels
the call.
I guess that SjPhone shouldn't send second invite. Should asterisk send OK
to that INVITE?
It seems that earlier asterisk have had protection against this because we
use this softphone for a quite long and the problem appeared when we moved
from 1.4.21.2 to 1.4.26 but I am not sure about this...
Is this asterisk bug or should we change our softphones? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-20 09:06 corruptor      Note Added: 0114041                          
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