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<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2>Jesse,</FONT></SPAN></DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff size=2>Thanks
for your feedback.</FONT></SPAN></DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff size=2>1. I
am running kernel 2.4.18.3 with linux 7.3, please let me know which version of
Redhat are you running on and which kernel are you running, I wonder if that
could make a difference too. I am surprised that you can run 25 channels with a
PIII 800, while I can only run less than 20 channels with a Xeon 2.4G. Please
see if you can run more channels with a better CPU and let me
know.</FONT></SPAN></DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2>2. I have also tried to use OH323 instead of H323, calls seem to go
through but I am just getting the ANSWERED indication even before the calls
start to ring, which is not right !! I have compiled the PWLIB 1.5.2 and OH323
1.12.2 and using OH323 version 0.5.7 which is the lastest version, are you
having the same experience? Is there anyway you can send me your OH323.conf
please? </FONT></SPAN></DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff size=2>3. I
would love to communicate with you privately to exchange experience. Please send
email to <A href="mailto:utitc@hotmail.com">utitc@hotmail.com</A>,
thanks</FONT></SPAN></DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=230002700-19012004><FONT face=Arial color=#0000ff
size=2>Tom</FONT></SPAN></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>Jesse
Peterson<BR><B>Sent:</B> Friday, January 16, 2004 12:32 PM<BR><B>To:</B>
Asterisk-Users (E-mail)<BR><B>Subject:</B> RE: [Asterisk-Users] capacity
testing<BR><BR></FONT></DIV>
<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
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