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<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
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