[asterisk-bugs] [Asterisk 0018855]: Asterisk hangs when generating SIP calls with sipp
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Feb 21 07:37:00 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18855
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Reported By: nbhatti
Assigned To:
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Project: Asterisk
Issue ID: 18855
Category: Core/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.2.16.1
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: 2011-02-21 03:59 CST
Last Modified: 2011-02-21 07:37 CST
======================================================================
Summary: Asterisk hangs when generating SIP calls with sipp
Description:
Hello, I have the following specs:
=============
Linux voip 2.6.32-4-pve https://issues.asterisk.org/view.php?id=1 SMP Wed Dec 15
14:04:31 CET 2010 x86_64
GNU/Linux
# cat /etc/debian_version
6.0
# ulimit -a
core file size (blocks, -c) 0
data seg size (kbytes, -d) unlimited
scheduling priority (-e) 0
file size (blocks, -f) unlimited
pending signals (-i) 16382
max locked memory (kbytes, -l) 64
max memory size (kbytes, -m) unlimited
open files (-n) 19096
pipe size (512 bytes, -p) 8
POSIX message queues (bytes, -q) 819200
real-time priority (-r) 0
stack size (kbytes, -s) 8192
cpu time (seconds, -t) unlimited
max user processes (-u) unlimited
virtual memory (kbytes, -v) unlimited
file locks (-x) unlimited
=============
Asterisk 1.6.2.16.1
and also have tried with 1.8.2.3, both reported the same results.
Asterisk hangs when I try to generate calls with sipp. I am using simple
UAC testing with a max call limit of 20, sipp -sn uac -d 2000 -s 2005
192.168.1.100 -l 20. After a few seconds Asterisk simply hangs. (or if
there is any better word to describe the situation). In this state, it
won't accept any SIP calls, now core or sip command will work through CLI.
Even, core restart now won't result in any action. The only solution would
be a kill -9. I reproduced this many times same environment. My asterisk
configuration is very simple and minimal.
======================
cat sip_additional.conf
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=ulaw
======================
cat extensions.conf
[sipp]
exten => 2005,1,Answer
exten => 2005,2,SetMusicOnHold(default)
exten => 2005,3,WaitMusicOnHold(10)
exten => 2005,4,Hangup
======================
cat sip.conf
[general]
context=from-sip-external
callerid=Unknown
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=MyAgent(1.0)
disallow=all
allow=g723
allow=g729
callevents=no
jbenable=no
maxexpiry=3600
defaultexpiry=120
minexpiry=60
allowguest=yes
srvlookup=no
registerattempts=0
registertimeout=6
g726nonstandard=no
t38pt_udptl=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=7
notifyhold=yes
rtpkeepalive=0
notifyringing=yes
checkmwi=10
rtpholdtimeout=50
nat=yes
tcpenable=no
preferred_codec_only=yes
language=en
===============
If someone would like to login at the time of crash if this is not
reproducible on your machine, I am help with that as well. Let me know if
any more information is required.
======================================================================
----------------------------------------------------------------------
(0132235) loloski (reporter) - 2011-02-21 07:37
https://issues.asterisk.org/view.php?id=18855#c132235
----------------------------------------------------------------------
Slightly modified again the dialplan, with the same effect, this fixed the
issue
[sipp]
exten => 9999,1,Progress
same => n,Ringing
same => n,Answer
same => n,MusicOnHold(10) # SetMusicOnHold(default) is deprecated!!!!
same => n,Hangup
root at stealth:/usr/src/1.6.2# sipp -sn uac -d 20000 -s 9999 192.168.0.101
-l 10 -r 2
Resolving remote host '192.168.0.101'... Done.
------------------------------ Scenario Screen -------- [1-9]: Change
Screen --
Call-rate(length) Port Total-time Total-calls Remote-host
2.0(20000 ms)/1.000s 5061 140.67 s 61
192.168.0.101:5060(UDP)
0 new calls during 0.000 s period 0 ms scheduler resolution
0 calls (limit 10) Peak was 10 calls, after 5 s
0 Running, 11 Paused, 0 Woken up
0 dead call msg (discarded) 0 out-of-call msg (discarded)
1 open sockets
Messages Retrans Timeout
Unexpected-Msg
INVITE ----------> 61 0 0
100 <---------- 61 0 0 0
180 <---------- 61 0 0 0
183 <---------- 61 0 0 0
200 <---------- E-RTD1 61 0 0 0
ACK ----------> 61 0
Pause [ 20.0s] 61 0
BYE ----------> 61 0 0
200 <---------- 61 0 0 0
------------------------------ Test Terminated
--------------------------------
----------------------------- Statistics Screen ------- [1-9]: Change
Screen --
Start Time | 2011-02-21 21:31:24:875 1298295084.875448
Last Reset Time | 2011-02-21 21:33:45:576 1298295225.576463
Current Time | 2011-02-21 21:33:45:576 1298295225.576763
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:000 | 00:02:20:701
Call Rate | 0.000 cps | 0.434 cps
-------------------------+---------------------------+--------------------------
Incoming call created | 0 | 0
OutGoing call created | 0 | 61
Total Call created | | 61
Current Call | 0 |
-------------------------+---------------------------+--------------------------
Successful call | 0 | 61
Failed call | 0 | 0
-------------------------+---------------------------+--------------------------
Response Time 1 | 00:00:00:000 | 00:00:00:019
Call Length | 00:00:00:000 | 00:00:20:024
------------------------------ Test Terminated -------------------------
Issue History
Date Modified Username Field Change
======================================================================
2011-02-21 07:37 loloski Note Added: 0132235
======================================================================
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