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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Nov 19 09:58:51 CST 2009


The following issue requires your FEEDBACK. 
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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:                     feedback
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:              2009-11-19 09:58 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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---------------------------------------------------------------------- 
 (0114002) lmadsen (administrator) - 2009-11-19 09:58
 https://issues.asterisk.org/view.php?id=16270#c114002 
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Can you provide some additional information such as console output, sip
debug and history (per the bug guidelines), and output of 'sip show
channels'?

That might get us started. I'm not sure what else to ask for at this time. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-19 09:58 lmadsen        Note Added: 0114002                          
2009-11-19 09:58 lmadsen        Status                   new => feedback     
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