<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<TITLE>RE: [Asterisk-Users] capacity testing</TITLE>
<META content="MSHTML 6.00.2800.1276" name=GENERATOR></HEAD>
<BODY dir=ltr>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=935085716-16012004>1)
Yes, I did get that. I've never seen a segmentation fault message, but that
should be b/c I've been running the process in the background since it is
obviously seg-faulting. I believe you are also correct that most people are not
trying to put the load on it that we are.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=935085716-16012004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=935085716-16012004>2) I
always see 'safe_asterisk' and 'asterisk -vvvg' running. my monitoring was
always done with top, but I've checked w/ ps a couple times and I believe only
ever see 1 of each of those processes. I may have to do some tests again to
double check that. My CPU problems did not come until the last 10 - 30 seconds
before asterisk crashed. This is still odd that our memory & processor
observations are opposite... the next thing I'm going to try is a dual xeon pIII
800 or 1ghz machine to see what happens.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=935085716-16012004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=935085716-16012004>3) I'm
running oh323. It was the one I could get to register w/ my gatekeeper as a
gateway - that made it much easier for me to do call routing on both sides. I
have also noticed some inconsistencies in the call flows like you mention, but
haven't taken the time yet to pinpoint exactly what and when they are
happening.</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]<B>On Behalf Of </B>T.
Chan<BR><B>Sent:</B> Thursday, January 15, 2004 22:54<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Cc:</B> Alan Chan<BR><B>Subject:</B> RE:
[Asterisk-Users] capacity testing<BR><BR></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=562063803-16012004>Hi
all, and Jesse</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=562063803-16012004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=562063803-16012004>1.
So, you did get the experience of crashing all of a sudden with the
"Disconnected from Asterisk server" error message. I got both this and the
segmentation error when crashing. I am running the version of asterisk, libpri
and zaptel updated to about 5 days ago, but I have had tested Asterisk for
more than a month already and needless to say I have had this experience since
Day 1, meaning it has always been a problem even in the previous revisions.
Henceforth, I feel that it is an intrinsic Asterisk problem, rather than just
the problem with specific versions / revisions. I have posted this problem a
few times before, I feel that this is a major problem but surprisingly, I was
not getting any feedback at all. I have this feeling that more than 90% of the
Asterisk community is using the system for PBX application rather than VOIP,
may be, just may be, Asterisk has not been tested with a good number of
simultaneous calls.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=562063803-16012004>2. I
am using Xeon 2.6G chip, much more powerful than yours, I have not got any
problem with CPU usage, at least not during the time that I was watching. The
thing is when I start 'safe_asterisk' , I could see when doing a PID, 1
"safe_asterisk" PID session and at least 10 (or more especially when there are
more calls) "asterisk -vvvg -c" PID session. Each session takes up about 18M
to 20M RAM, when that is why I am seeing all very high memory usage. How many
sessions of Asterick do you see running after you loaded it?
</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=562063803-16012004>3.
Are you running H323 (Jeremy) and OH323 (Michael)? I am running Jeremy's and
have had this inbound H323 problem. I tried OH323 (Michael) as well, but for
some reasons, I am getting this false connect signal, that is, I made an
outbound H323 call to a CiscoAS5300 for example, I heard the ring and
immediately on my "Asterisk", it showed call answered when it was still
ringing. Do you have that experience?? What setting you have if you do not
have that experience?</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=562063803-16012004>4.
Lets talk off list at <A
href="mailto:utitc@hotmail.com">utitc@hotmail.com</A>.</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=562063803-16012004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=562063803-16012004>Thanks</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=562063803-16012004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=562063803-16012004>Tom</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Jesse Peterson
[mailto:asterisk-users-admin@lists.digium.com]<B>On Behalf Of </B>Jesse
Peterson<BR><B>Sent:</B> Thursday, January 15, 2004 8:21 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> RE: [Asterisk-Users]
capacity testing<BR><BR></FONT></DIV>
<DIV><FONT size=2>Sorry for the malformed mail. My responses are marked with
'***' below.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>jesse</FONT></DIV>
<DIV><FONT size=2>======</FONT></DIV>
<DIV><FONT size=2>Hi,<BR><BR>I am a newbie in Asterisk as well, intending to
use it in a similar way as<BR>you are, communicating with AS5300 as well as
other gateways including<BR>MAXTNT.<BR><BR>I have had similar, but yet
different experiences than yours.<BR><BR>1. Asterisk does crash with the
number of calls, but in my case, about or<BR>less than 20 calls, then I
would get either a Segmentation Error and then<BR>crashed OR it would just
crash saying "Disconnected from Asterisk server"<BR>all of a
sudden.</FONT><FONT size=2></DIV>
<DIV>*** The crashes I experienced were fairly transparent. When I had
the console (asterisk -r) running, I saw the 'Disconnected' message you
mention.<BR><BR>2. I am using Pentium Xeon chip and hence more powerful than
yours with 512M<BR>RAM, my CPU usage has always been low, however, I have
not had a chance to<BR>look at the CPU usage just before crashing, but all
the time that I was<BR>looking, it has been low. Rather the MEMORY has
always remained high at 450M<BR>usage even with no calls. This is a
different experience as compared to<BR>yours.<BR>*** A Xeon of the same
speed (800mhz in my case) should certainly perform better - lower, I don't
know. I find it a little odd that you experienced basically the opposite of
my testing. What version are you running?</DIV>
<DIV><BR>3. I have also noticed that with more calls, and after a certain
random<BR>period of time, any H323 calls going into the Asterisk would fail,
my AS5300<BR>and MAXT TNT would get their calls all rejected from Asterisk.
However,<BR>Asterisk was still running at the time and I could actually call
in and out<BR>the zap interface and outbound H323 from Asterisk was not a
problem. It<BR>seems that something got hung with H323, causing inbound H323
calls into<BR>Asterisk to all fail. In this situation, I would have to stop
the Asterisk<BR>and rerun it to fix the problem.<BR>*** Interesting - I have
not experienced that (yet...).</DIV>
<DIV><BR>4. I have not tried the 0.7.0 version, but with existing version, I
am not<BR>getting reliable and stable system, nothing close to Cisco and
Lucent which<BR>are rock solid. However, I really love the power and the
features of<BR>Asterisk, and I remain in good faith to see
improvements.<BR><BR>Any associate out there who can shed some lights into
this? I am rather<BR>curious as to why I seem to be using up all memory
although I am not running<BR>any unnecessary processes, or should I actually
disable all modules, other<BR>than really necessary ones to support
VOIP?<BR></DIV>
<DIV>*** Since you and I are working in what sounds to be a familiar
environment, maybe we should communicate about our test scenarios, etc off
list to both help each other and see if we can find some similarities to
share with others.</DIV>
<DIV><BR>Thanks !<BR><BR>Tom<BR><BR>-----Original Message-----<BR>From:
asterisk-users-admin@lists.digium.com<BR>[<A
href="mailto:asterisk-users-admin@lists.digium.com">mailto:asterisk-users-admin@lists.digium.com</A>]On
Behalf Of Jesse<BR>Peterson<BR>Sent: Thursday, January 15, 2004 2:40
PM<BR>To: Asterisk-Users (E-mail)<BR>Subject: [Asterisk-Users] capacity
testing<BR><BR><BR>Hello all. I'm new to asterisk and have been using and
testing it for about<BR>a week now. My initial hope has been to use it as a
sip<->h323 gateway to<BR>tie SIP & H323 based ip phones together
with my Cisco AS5300 and Lucent<BR>MaxTNT/MVAM networks.<BR><BR>I am
currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800<BR>with
256megs RAM. I have tried a couple CVS version from the past week<BR>(maybe
01/09/04 and 01/14/04) and have not been able to get them to
work<BR>semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has
supported<BR>those ok. Primarily test cases have involved sending ip phone
calls via SIP<BR>to Asterisk and having Asterisk route the calls using h323
via a gatekeeper<BR>to my TNT network which then sends it out the PSTN...
and the opposite path,<BR>PSTN->TNT->Asterisk->SIP Phone. Another
test has been sending a call from a<BR>AS5300 using SIP to Asterisk, out
H323 to a TNT. Both of those have worked<BR>very well with the voice quality
being excellent (actually better than a<BR>SIP->ISDN T1 hardware solution
we've been working with - audiocodes mediant<BR>2k for those interested).
This is the test case I describe below as it was<BR>the one the allowed me
to load Asterisk up with the most calls.<BR><BR>Anyway, I know that what I'm
doing is not exactly the intended primary use<BR>of Asterisk. That said,
here's what I found.<BR><BR>Voice quality was very good until I had approx.
25 calls up. At that point<BR>there were intermittent issues with garbled
voice, a little echo, etc. When<BR>it reached a little over 30 calls,
Asterisk just died (oops).<BR>During the test, I was trying to keep an eye
on proc. & memory util. Memory<BR>never seemed to be an issue - even
right before the crash the Asterisk<BR>process was not using more than 20 -
25MB.<BR>Processor utilization was interesting to watch though. I couldn't
make any<BR>direct/firm correlation, but it seemed like my spikes were
coming when<BR>Asterisk was doing call setup. Even up to about 25 calls,
utilization didn't<BR>spike to more the 25% for long, and with ~25 calls
seemed to 'idle' around<BR>15%. Above the 25 (when also started noticing
voice quality issues), the<BR>proc. util. seemed to start going wacky -
spikes up to 40, 50, even 60%.<BR>Then it went to 99% for a moment, voice
quality was horrible if you could<BR>hear anything, and Asterisk
crashed.<BR><BR>I did not find anything in the logs to inidicate any
problems, though I've<BR>found that to be the case pretty much everytime
Asterisk crashes.<BR><BR>I saw a list thread in which a developer asked for
some gdb output... in it,<BR>he said this:<BR>> Run asterisk with
"-vvvcg".<BR>> Do your test (core file generated).<BR>> Run "gdb
/usr/sbin/asterisk <core_filename>"<BR>> From within gdb run
"bt" and send me the output<BR>> of it.<BR><BR>if it is of use, here it
is (from asterisk v.0.5.0)<BR>-----------------------------<BR>(gdb)
bt<BR>#0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at
frame.c:72<BR>#1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8)
at<BR>chan_oh323.c:1504<BR>#2 0x0805884f in ast_write (chan=0x8214488,
fr=0x5de5c4a8) at<BR>channel.c:1385<BR>#3 0x0805afa1 in
ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0,<BR>fo=0x6ef20e50,
rc=0x6ef20e54) at channel.c:2262<BR>#4 0x418bdd7a in ast_bridge_call
(chan=0x5de5ed98, peer=0x8214488,<BR>allowredirect_in=0,
allowredirect_out=0, allowdisconnect=0) at<BR>res_parking.c:224<BR>#5
0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b)
at<BR>app_dial.c:668<BR>#6 0x08061a5a in pbx_exec (c=0x5de5ed98,
app=0x80f0f98, data=0x6ef216e8,<BR>newstack=1) at pbx.c:396<BR>#7
0x08068c61 in pbx_extension_helper (c=0x5de5ed98,
context=0x5de5eeec<BR>"longdistance", exten=0x8214488 "H323:8257",
priority=2,<BR> callerid=0x5de10048 "\"Jesse Peterson\"
<2474766>", action=1104606132)<BR>at pbx.c:1150<BR>#8 0x0806392c
in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634<BR>#9 0x08069321 in
pbx_thread (data=0x84a5038) at pbx.c:1855<BR>#10 0x40026484 in start_thread
() from /lib/tls/libpthread.so.0<BR>-----------------------------<BR><BR>If
anyone has tried something like this or has any comments, I'd
be<BR>interested in hearing from
them.<BR><BR><BR><BR>jesse<BR><BR><BR>_______________________________________________<BR>Asterisk-Users
mailing list<BR>Asterisk-Users@lists.digium.com<BR><A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>To
UNSUBSCRIBE or update options visit:<BR> <A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR><BR>---<BR>Incoming
mail is certified Virus Free.<BR>Checked by AVG anti-virus system (<A
href="http://www.grisoft.com">http://www.grisoft.com</A>).<BR>Version:
6.0.558 / Virus Database: 350 - Release Date:
1/2/2004<BR><BR>---<BR>Outgoing mail is certified Virus Free.<BR>Checked by
AVG anti-virus system (<A
href="http://www.grisoft.com">http://www.grisoft.com</A>).<BR>Version:
6.0.558 / Virus Database: 350 - Release Date:
1/2/2004<BR><BR>_______________________________________________<BR>Asterisk-Users
mailing list<BR>Asterisk-Users@lists.digium.com<BR><A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>To
UNSUBSCRIBE or update options visit:<BR> <A
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></DIV></FONT><BR>
<P><FONT size=2>---<BR>Incoming mail is certified Virus Free.<BR>Checked by
AVG anti-virus system (http://www.grisoft.com).<BR>Version: 6.0.558 / Virus
Database: 350 - Release Date:
1/2/2004<BR></FONT></P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>