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

Tue Oct 12 01:14:25 CDT 2010

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:                jpeeler
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Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Target Version:             1.4.38
Asterisk Version:           SVN 
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:              2010-10-12 01:14 CDT
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Summary:                    [patch] 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
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0016764 Sip Channels Colapse
related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
related to          0017643 [patch] dialplan reload deadlocks in as...
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---------------------------------------------------------------------- 
 (0127850) lftsy (reporter) - 2010-10-12 01:14
 https://issues.asterisk.org/view.php?id=16382#c127850 
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Good morning, sorry not answering sooner but we've got jet-lag here -:) On
the server we have reinstalled together, current configuration for sip.conf
is:

[general]  
context=default ; Default context for incoming calls 
allowguest=no ; Allow or reject guest calls (default is yes) 
allowoverlap=no ; Disable overlap dialing support. (Default is yes) 
realm=voip ; Realm for digest authentication 
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) 
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) 
srvlookup=no ; Enable DNS SRV lookups on outbound calls 
pedantic=no ; Enable checking of tags in headers 
tos_sip=cs3 ; Sets TOS for SIP packets. 
tos_audio=ef ; Sets TOS for RTP audio packets. 
maxexpiry=180 ; Maximum allowed time of incoming registrations 
minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)

defaultexpiry=120 ; Default length of incoming/outgoing registration 
disallow=all ; First disallow all codecs 
allow=alaw ; Allow codecs in order of preference 
language=en ; Default language setting for all users/peers 
useragent=voipua ; Allows you to change the user agent string 
dtmfmode=rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833

rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity 
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP
activity 
rtpkeepalive=20 ; Number of seconds, when a RTP Keepalive packet will be
sent if no other RTP traffic on that connection. 
notifyringing=yes ; Control whether subscriptions already INUSE get sent 
nat=yes ; Global NAT settings (Affects all peers and users) 
rtcachefriends=yes ; Cache realtime friends by adding them to the internal
list 
rtupdate=yes ; Send registry updates to database using realtime? (yes|no)

rtautoclear=60 ; Auto-Expire friends created on the fly. If yes the
autoexpire will be in 120 seconds. Default yes.
qualify=yes ; Check if client is reachable. If yes, the checks occur every
60 seconds 
t38pt_udptl=yes ; T.38 faxing only works in SIP to SIP calls, with no
local or agent channel being used.
progressinband=never ; never|no|yes : If we should generate in-band
ringing always. Default never. 

And so rtautoclear is indeed set to 60s

Realtime SIP Settings:
----------------------
  Realtime Peers:         Yes
  Realtime Users:         Yes
  Cache Friends:          Yes
  Update:                 Yes
  Ignore Reg. Expire:     No
  Save sys. name:         No
  Auto Clear:             60

If you want me to change anything and report back, I'm available. Have a
nice morning. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-10-12 01:14 lftsy          Note Added: 0127850                          
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