[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Feb 11 04:54:48 CST 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:
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Project: Asterisk
Issue ID: 16382
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
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: 2010-02-11 04:54 CST
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Summary: 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
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Relationships ID Summary
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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
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(0117982) lftsy (reporter) - 2010-02-11 04:54
https://issues.asterisk.org/view.php?id=16382#c117982
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Here is all the details you asked. I'm going to keep the server flooding
for a few yours if you need further information, but I will have to restart
it before tonight....
* SIP CHOW CHANNELS is chowing many many SIP OPTIONS transactions...
(16670 active SIP channels)
212.147.45.91 (None) 308bd32143e 00102/00000 0x0 (nothing) No
Init: OPTIONS
212.147.45.91 (None) 21d699095af 00102/00000 0x0 (nothing) No
Init: OPTIONS
212.147.45.91 (None) 73df27e6236 00102/00000 0x0 (nothing) No
Init: OPTIONS
212.147.45.91 (None) 7e27d4444ef 00102/00000 0x0 (nothing) No
Init: OPTIONS
212.147.45.91 (None) 41e8bf2d66d 00102/00000 0x0 (nothing) No
Init: OPTIONS
212.147.45.91 (None) 146b3535531 00102/00000 0x0 (nothing) No
(d) Init: OPTIONS
212.147.45.91 (None) 41008d2b599 00102/00000 0x0 (nothing) No
(d) Init: OPTIONS
212.147.20.222 0009004141 25bfed474d3 00102/00000 0x8 (alaw) No
Tx: ACK
212.147.45.91 0323440749 3c2fabb6a42 00101/00002 0x100 (g729) No
Rx: ACK
16670 active SIP channels
* CORE SET DEBUG 20 is showing many many things.... I cannot read the
screen, I got all the transaction destroy below
pul-lav-vp-ast-03*CLI> core set debug 20
Core debug was 0 and is now 20
Really destroying SIP dialog
'6ee0fd1c263972af67a4e37702b2cd6b at 212.147.47.83' Method: OPTIONS
Really destroying SIP dialog
'75f3ab7f25a26e3d539b67510b7e9be8 at 212.147.47.83' Method: OPTIONS
Really destroying SIP dialog
'075157740620306c1ccc1e065c1bc32b at 212.147.47.83' Method: OPTIONS
Really destroying SIP dialog
'6adbda4500c070b36328e47425ff0cfd at 212.147.47.83' Method: OPTIONS
Really destroying SIP dialog
'32f7f8694761279e0ea1db58685ca106 at 212.147.47.83' Method: OPTIONS
Really destroying SIP dialog
'4006ff767b3101b93fc938b12239f557 at 212.147.47.83' Method: OPTIONS
Really destroying SIP dialog
'7538ed357da4d5933acf29054e2555a5 at 212.147.47.83' Method: OPTIONS
* CORE SET VERBOSE 20 is not showing anything, because no new calls are
possible since the server is flooding the proxy and has been blacklisted
pul-lav-vp-ast-03*CLI> core set verbose 20
Verbosity was 3 and is now 20
pul-lav-vp-ast-03*CLI>
pul-lav-vp-ast-03*CLI>
pul-lav-vp-ast-03*CLI>
* The topology I use is a simple OpenSIPs 1.4.5 server in front of 2
Asterisk servers, the proxy does nothing except Keepalive and loadbalancing
on source IP. All AAA is done on Asterisk. BUT, I can reproduce the bug on
a direct IP connection to the Asterisk server!
* And here is my sip.conf
[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 the includes:
#include sip.local
regcontext=pul-lav-vp-ast-03
and another include (there are other trunks of course like the one below)
#include sip-trunk.conf
[sip_trunk_vm]
host=212.147.45.17
type=peer
context=default
dtmfmode=info
insecure=port,invite
nat=never
sendrpid=yes
disallow=all
allow=alaw
Issue History
Date Modified Username Field Change
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2010-02-11 04:54 lftsy Note Added: 0117982
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