[asterisk-bugs] [Asterisk 0016054]: Asterisk 1.6.1.6 not closing RTP ports after connection

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16054 
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Reported By:                alexrecarey
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16054
Category:                   Core/Channels
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.6 
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:             2009-10-11 18:53 CDT
Last Modified:              2009-10-13 18:17 CDT
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Summary:                    Asterisk 1.6.1.6 not closing RTP ports after
connection
Description: 
At the start of the day, asterisk is using a couple of ports and
filehandles. After every call, the number of used file handles (lsof)
increases, and the number of open ports (netstat -a) also increases. At the
end of the day, with no calls in the system, after one full day of traffic,
the system shows 13000 open files and 12500 open ports. When asterisk is
restarted (not reloaded, it must be closed and reopened) all of the ports
and file handles are closed. If this is not done every night the system
will run out of ports and stop accepting calls.
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 (0112247) alexrecarey (reporter) - 2009-10-13 18:17
 https://issues.asterisk.org/view.php?id=16054#c112247 
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Yes, I can confirm that it was a configuration issue regarding sip-timers,
sessions are now closing normally.

Perhaps session-expires should also have a default value?

I am pasting the error messages that asterisk threw up at the console just
in case somebody with the same problem does a search on the bugtracker. If
it's not appropriate please say so and I'll remove them.

Again, thank you for the prompt replies!

WARNING[15784]: channel.c:828 __ast_channel_alloc_ap: Channel allocation
failed: Can't create alert pipe!
WARNING[15784]: chan_sip.c:5865 sip_new: Unable to allocate AST channel
structure for SIP channel
WARNING[15784]: app_dial.c:1528 dial_exec_full: Unable to create channel
of type 'SIP' (cause 0 - Unknown)
WARNING[15796]: res_agi.c:636 launch_script: unable to create fromast
pipe: Too many open files
WARNING[3034]: rtp.c:2445 rtp_socket: Unable to allocate RTCP socket: Too
many open files
WARNING[3034]: channel.c:828 __ast_channel_alloc_ap: Channel allocation
failed: Can't create alert pipe!
WARNING[3034]: chan_sip.c:5865 sip_new: Unable to allocate AST channel
structure for SIP channel
NOTICE[3034]: chan_sip.c:18817 handle_request_invite: Unable to
create/find SIP channel for this INVITE
WARNING[3051]: pbx_spool.c:464 scan_thread: Unable to open directory
/var/spool/asterisk/outgoing: Too many open files
ERROR[3034]: acl.c:472 ast_ouraddrfor: Cannot create socket 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-13 18:17 alexrecarey    Note Added: 0112247                          
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