Jeremy Kister wrote:
using asterisk 11.6.0-rc1 i just converted my nat=yes to
nat=auto_force_rport,auto_comedia
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the
Asmaa,
You're getting ahead of yourself. How do you expect audio to work if
your firewall/NAT settings aren't even configured correctly to
establish SIP sessions?
Go back and read the message that I sent yesterday. Fix the SIP
three-way handshake problem. That is step 1 and you'll know you
Asmaa Ahmed wrote:
Indeed I missed your previous message!
After changing the externip, it worked successfully... The sip
session is established with the complete three-way handshake, and
the voice packet is exchanged with no problem!
Many thanks.
Asmaa,
That's great news!! I guess
Asmaa Ahmed wrote:
I am trying to make my first call on Asterisk to succeed. I have
Asterisk 1.8.10.1 running on Ubuntu machine.
The configuration is quite simple just for my first test, Trying to
have a call between two X-lite sipphone. The subscribers succeeded
to register and the
Chris Gentle wrote:
Is there any way I can improve the audio quality in a confbridge in
Asterisk 11? I've changed the internal_sample_rate setting to 44100
but that doesn't seem to make any difference. I would also think this
would make my confbridge recordings 44100 but they all end up as
Eloi,
My responses are inline.
Thanks a lot for this detailed answer :
You're welcome. Thank you for responding. A lot of people forget to do so and
future list readers are left wondering whether or not the proposed solution
worked.
- I managed to have it working disabling auth message
Eloi Bail wrote:
I am trying to enable SIP SIMPLE communication in my test environment.
I have the following env :
- one server (192.168.50.126) with Asterisk 11
- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143
I successfully had a phone call between clients.
I
Andreas Sikkema wrote:
On 6/13/13 16:20 , Matthew J. Roth wrote:
It's hard to be certain without seeing a full SIP trace, but I think the
INVITE
with the internal IP is actually a re-INVITE that Asterisk is sending to
establish a native bridge between the SIP friend and the SIP gateway
Mickael MONSIEUR wrote:
My version is Asterisk 1.6.2.9.
Or have you seen NAT ? I have no NAT on my network . Have you seen my little
diagram above ?
Here it is:
SIP friends (phones) - Asterisk - SIP gateway to PSTN converter
80.236.215.61109.69.217.6 internal
Carlos Chavez wrote:
I have been struggling with an audio issue for a week now and have
not been able to solve it.
We have an Asterisk server (now running 11.4 but started with 1.8)
with several sip phones on an internal network and a SIP trunk for
external calls. We recently put several
Mickael MONSIEUR wrote:
I have a standard Asterisk configuration:
SIP friends (phones) - Asterisk - SIP gateway to PSTN converter
80.236.215.61109.69.217.6 internal IP (
10.4.0.10/255.255.255.0 )
When analyzing traffic on a SIP friend/phone I see this:
INVITE
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received the DTMF-input ?
Jonas,
Please provide the
Jonas Kellens wrote:
Even if there *can* be more than 1 digit, in case there is only 1 digit it
should go faster.
Jonas,
Use the TIMEOUT function to set the maximum amount of time permitted between
digits when the user is typing in DTMF. As you've discovered, the default is 5
seconds.
Kamlesh Kumar wrote:
allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP
supports g729 codec as we are able to send the traffic from other soft switch.
There must be some difference between your Asterisk servers. Please set them
up for calling the ITSP with G.729
Kamlesh Kumar wrote:
SIP.conf
[100]
username=100
secret=password
type=friend
host=dynamic
nat=yes
canreinvite=no
insecure=port
disallow=all
allow=ulaw
allow=alaw
allow=g729
context=asterisk
qualify=no
Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer?
SIP
Mark Henry wrote:
1. Your softphone is not sending g729
This was a SIP trace of a successful u-law call. In an earlier post Kamlesh
provided a trace of a failed G.729 call which did not include the dialog between
the Asterisk server and the ITSP. I asked for this trace so that I could see
Kamlesh Kumar wrote:
Yes that's correct, when I use u-law call works fine.
In case of g729, I enabled sip debug with 'sip set debug on' and captured all
the sip traces and got whatever I posted in last email. There was no other
call on the system when I captured sip trace. Please suggest
Kamlesh Kumar wrote:
Call even doesn't go to the ITSP. I tried without AGI script and the results
were same.
Kamlesh,
Your first message stated that the call is successful if the codec is u-law, so
there must be communication between the Asterisk server and the ITSP. The key
to
Kamlesh,
Please provide SIP traces of both call legs for a failed call.
Your last message only included a SIP trace of the call leg from the SIP
softphone to the Asterisk server. There was no SIP trace for the call leg from
the Asterisk server to the ITSP and, as shown below, that is probably
asterisk users wrote:
Registration trace
(note that extension 88 is the voicemail extension, which the phone registers
to also for MWI)
-- http://pastebin.com/c3H700wa
There are no REGISTER requests in that trace. All I see are SUBSCRIBE, NOTIFY,
OPTIONS, and INVITE dialogs.
Call trace:
asterisk users wrote:
I'm having a strange problem recently with a Yealink SIP-T28P phone connected
to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I
dial anything from the phone, it shows Forbidden, and the Asterisk console
shows:
[May 21 10:47:49]
Brian,
KDDI does provide a list of supported equipment and vendors. Specific
hardware or license based software products that quickly become cost
prohibitive.
I doubt that Asterisk will find it's way on the list any time soon. Because
KDDI follows the traditional big telco method of
Brian LaVallee wrote:
My SIP provider is not happy that credentials (in the Authorization header
field) are provided in the initial REGISTER request.
The SIP provider ONLY wants the credentials AFTER rejecting the message with
a 401.
I know it's dumb, because the RFC says that the the
Joshua Colp wrote:
Most of your response is correct except it doesn't take into account the
rport RFC. Lack of implementation of an RFC doesn't make it
non-compliant, so their stuff really is fine for this scenario. It all
comes down to us forcing rport to be on by default.
This is now
Markus,
I'll take another shot at answering your questions. As before, if someone more
knowledgeable, like Joshua Colp, also responds please give more credibility to
their remarks.
Although I have to say I don't understand what is going on exactly. :)
As can be seen below, and as Joshua
Markus,
I think I know what's wrong here but I did a fair amount of research while
digging into your problem. I may have misinterpreted something along the way so
you should also consider other responses, especially if they come from someone
who claims greater expertise. I did this to help you
Joshua Colp wrote:
If you set nat=no for that specific peer it should work as you need.
'rport' is forced on these days which works for most situations, except
with some platforms and Cisco phones. _
Joshua,
That sounds much easier than what I came up with, so I'd recommend to Markus
that
Mitch Claborn wrote:
I get to go home on Saturday! The Digium phone deployment is simple
enough to manage remotely.
Glad to hear it. If the problem comes back on the hardphones, just post the
debug information to this thread and I'll take a look at it.
Regards,
Matthew Roth
InterMedia
Mitch Claborn wrote:
Interestingly, using Bria we sometimes see similar, though not exactly
the same, symptoms. That would make me wonder about the TCP stack on
the client machine, or similar.
With a softphone, you're dependent on the entire software stack up to the
softphone and at the
Florian Wolters wrote:
Does it make sense to have a more detailed tcpdump of the SIP session? If
so, how should such a thing been shared without posting too much ASCII
text to the list?
SIP sessions are generally small enough to post right to the list. Otherwise,
you can put them up on a
Florian Wolters wrote:
So I turned on SIP debug for this host and analyszed it with wireshark.
The last packets show an INVITE from my provider, that is answered by my
Asterisk with 200 OK, with session description. What follows is an ACK
by the provider and immediately a BYE sent by the
Mitch Claborn wrote:
Thank you for that most excellent post. I had guessed at most of the
SDP fields and meaning.
No problem. I actually like looking at SIP traces for some reason.
I have wireshark traces from the client and the RTP packets are not in
the trace, which I think means
Mitch Claborn wrote:
Where is a good place to find documentation on the various fields in the
INVITE SIP message and the response? I'd like to dig into them and try
to understand them more completely.
For the SIP headers:
http://en.wikipedia.org/wiki/Session_Initiation_Protocol
Jai Rangi wrote:
I am sure we all get lots if spam emails every day.
Yes, I do. Now ask yourself why I was able to immediately identify
where my address was harvested from for this particular piece of spam.
The answer has to do with DIDForSale's business practices as observed
on the Asterisk
Carlos Alvarez wrote:
Sounds like the same huge effort it takes to work with Qwest/
Centurylink, and in the long run we found it simply isn't worth it.
The few benefits of working with an RBOC are countered by the many
drawbacks of working with an RBOC.
Also we recently acquired a half
Michael L. Young wrote:
I should have probably stated that this is going to be going through
an MPLS network being setup with Verizon. They may not be requiring
that since it is within their network, not going over the internet.
They have not said anything about the the need to secure the
Justin Sherrill wrote:
We occasionally get a sort of feedback/echo noise on our phones
here. (Polycom IP550 / Asterisk 1.8). It lasts for about a second,
and it's described by users as 'jingle bells'.
That's very appropriate for the season. I just hope they're not
Todash Chimes. ; )
It
Michael L. Young wrote:
If I am reading this right, it looks like a BYE is coming in from
the far end, Bandwidth.com.
Prior to that, Asterisk retransmits the OK to Bandwidth.com's INVITE
twice. It doesn't look like Bandwidth.com receives any of them,
because they never respond with an ACK.
Rafael Visser wrote:
I replaced for the following sip.conf
[general]
context=default ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0
Administrator TOOTAI wrote:
10.0.70.12 is the IP of Asterisk server (kvm virtual machine) which is
replaced by externaddr parameter from sip.conf.
If you have other ideas, welcome ;-)
Considering that you made progress on your initial problem by setting
nat=force_rport (resulting in
Administrator TOOTAI wrote:
We tested this setting this WE, effectively this problem disappear but
another appears: call get connected but no audio. We installed Asterisk
10.3.1 - connection and no audio too, so same behaviour.
We did read those files, don't see which parameter we could
Administrator TOOTAI wrote:
we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn't accept our calls anymore, we receive a
timeout error Packet timed out after 32000ms with no response.
List users,
I have an AudioCodes nCite 1000 SBC that is end-of-life and I'm
looking to replace it with open source software. I believe one of the
SIP proxy projects will fit my needs, but I'm a bit overwhelmed by
the number of choices and I'd like the advice of experienced users
before I venture
Kaushal,
Your version of SoX does not have MP3 support. Since you have LAME
installed, use it as a first step to produce an intermediate file
that SoX supports. Then use SoX to convert the intermediate file
to the desired format.
Step 1
--
# lame --decode obd-demo.mp3 obd-demo.wav
input:
CDR wrote:
The point is that a minor change in the code would have a dramatic
effect on security, and carry a lower impact on CPU that using
Iptables. The simplicity of the change cannot understated.
You're in luck. Since Asterisk is open source, you can make the
unbelievably simple change
Michael,
It looks like your problem is caused by a phone with a non-standard
SDP session version implementation. The phone is sending an INVITE
with SDP that contains an a=sendonly line. Asterisk should respond
with an OK that contains an a=recvonly line, but it responds with
a=sendrecv
Kevin P. Fleming: The versions all go to ten. Look, right across the
board, ten, ten, ten and...
Asterisk Users: Oh, I see. And most open source projects upgrade to
two?
Kevin P. Fleming: Exactly.
Asterisk Users: Does that mean it's better? Is it any better?
Kevin P. Fleming: Well, it's eight
Michael,
Here are the differences between the systems that I determined from the two SIP
traces:
* Working system: no NAT, phone codec: G.729, Asterisk codec: G.729
* Non-working system: NAT, phone codec: G.729, Asterisk codec: A-law
Does the conversation have two-way audio prior to the
Michael wrote:
True. In the working system, LAN calls are also using G.729, while
in the non-working system, LAN calls are in G.711 (supported but
not prioritized by the phones) and only the SIP trunk to the ITSP
is set to G.729.
Can you set the phone to G.711 and try making a LAN call on
Steve Edwards wrote:
Also they tend to be used more by 'non-programmers' who get away with
'stupid' stuff like calling out to system() and piping a bunch of
commands together because they don't know how to use the language
properly :)
I'm not disparaging Perl programmers or the language.
Tzafrir Cohen wrote:
Well, there are a number of separate optimizations in systemd:
1. Delayed loading of services (or even not loading them at all, if not
needed. E.g.: don't load CUPS if nobody needs it.
2. Paralelized loading of services (though there have been other
Steve,
Apology accepted. As I said in the original post, I hold you in high
regard so your criticism was hard to take. I still think that the trade-
off between readability and optimization is up for debate, but it's
certainly nothing to hold a grudge over.
I can tell you one thing for
HB wrote:
Please excuse me for addressing this Linux issue on this list, however I
hope that some of you have found a solution thats matches the * use and
also easy to install without very deep knowledge of Linux.
My wish are a program that maintain a mirror copy of the HD.
Tilghman Lesher wrote:
Eric Smith wrote:
Using Asterisk 1.4.26.2
I can get a nice colored CLI if I run asterisk -c
But I cannot achieve this when I reattach to an existing instance
(as i want to do) with asterisk -r.
Is there a way to reattach and have color?
Yes, but you'll need to
Albert Bonomo wrote:
Well, I did tried using the source but couldn't make it work.
some problem with dependencies and kernel version.
It is really difficult to put all the stuff in order to make a source
work.
Albert,
It's really not that difficult if your system is in decent shape.
Leif Madsen wrote:
I have a client using QueueMetrics and they seem to be fairly pleased
with it. Their response times on issues has been pretty good from
what I can tell (I had the client communicate with them directly
where necessary).
Unless you build it yourself, I'm not sure there
Bruce,
These two links may be helpful to you:
PHP: SQL Injection - Manual
http://www.php.net/manual/en/security.database.sql-injection.php
PHP: mysql_real_escape_string - Manual
http://www.php.net/manual/en/function.mysql-real-escape-string.php
Regards,
Matthew Roth
InterMedia
Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to
be able to enhance existing IP-PBX solutions such as Asterisk, so maybe it
can be used as a simple means to provide secure/encrypted calls.
GNU SIP Witch - Summary http://savannah.gnu.org/projects/sipwitch
GNU SIP
Matthew J. Roth wrote:
Is anyone out there using SIP Witch in conjunction with Asterisk? It
claims to be able to enhance existing IP-PBX solutions such as
Asterisk, so maybe it can be used as a simple means to provide
secure/encrypted calls.
GNU SIP Witch - Summary http
Maria Cristina Bayno wrote:
I'm here again and need your help. Regarding Queue logs in asterisk,
do anyone knows how can I get all the logs for the specific/particular
date? Example is I want to get all the logs from September 9-12?
Maria,
Here's a Bash script that will do what you're
Miguel Molina wrote:
I recently upgraded a production machine to asterisk 1.4.25. It seems
quite stable but after ~5 days of normal operation it core dumped with
this result:
(gdb) bt
#0 0x00516402 in __kernel_vsyscall ()
#1 0x005b3d20 in raise () from /lib/libc.so.6
#2 0x005b5631 in
Alessandro Russo wrote:
Unfortunately echo is not due to speakerphone. Each participant calls
a geographical number that is redirected from the PBX to a call
manager which pass the flow to the asterisk machine which creates a
meetme voice conference, so user calls via traditional either
Alessandro Russo wrote:
we are using Asterisk 1.4.18.1 http://1.4.18.1/ on debian 4.0 etch,
pwlib 1.10 and openh323 1.18.
We are using MeetMe for conference calls and with two participants
there is no echo problems, but with more than two participants there
is a lot of echo that
Chris Rowson wrote:
I wanted to setup Oreka to monitor calls on a trixbox box I have
setup. Oreka doesn't seem to be catching all of the calls
though I have port mirroring setup on the port that trixbox is
connected to, mirrored to the port Oreka is connected to.
I
Carsten Maass wrote:
we are in the need to reach an external Conference-System, whos
numbering system is *2*. Unfortunately *2 is the featurecode for
attended transfer in our local asterisk, so the call doesn't come
through. Is there a way to somehow escape the featurecode, so we can
Mike wrote:
I tried using this iptables sample, and did not see duplicate packets
on '--to-ports' port
Has some verified this is working for them?
I listened on both ports with tcpdump command.
Mike,
I can confirm that it's working. Admittedly, I never looked at the
packets with tcpdump
Rizwan Hisham wrote:
Is it possible to make single asterisk server listen on two different
ports?
Rizwan,
There is no way to make a single instance of Asterisk listen on multiple
ports. However, you can use an iptables REDIRECT to achieve the same
functionality.
To redirect a single port
c james wrote:
I am trying to work in the console, figuring why it exits, but about 75%
is always taken up with
Really destroying SIP dialog '' Method: OPTIONS
Can anyone point me where I can stop this without turning down the
debugging/verbose on the entire console.
c james,
Tilghman and Jay,
Thanks for the licensing advice. If anyone is interested in replicate,
I'm now ready to distribute it under the GPL.
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
___
-- Bandwidth
Douglas Garstang wrote:
Not at all, just offering a workaround. If your master.csv is
complete and correct then it makes sense to use that data unless
someone can identify your problem and offer a fix.
Unfortunately, not really feesible. I didn't design the system but we
are using CDR's
Alex Balashov wrote:
A program like netcat?
Alex,
You're not the first person to suggest nc for this purpose. As I
understand it, it's a TCP/UDP swiss army knife so I'm sure it's up to
the task. However, in reading the man page, I don't see any trivial way
to buffer failed writes and
Dovid B wrote:
Thanks. I like to know my errors and what cause them. Anyone available to
help me pick at their brain to see where its coming from or am I really
barking up the wrong tree ?
Dovid,
The number of concurrent calls on the server is tightly related to the
number of file
Johansson Olle E wrote:
In the long run we're trying to move to using the manager for all
parsing by adding a lot of new manager events and actions.
If there's something missing that you only can do or information you
only can get in the CLI, please tell us.
Olle,
How does what you are
Soumya Kat wrote:
Thank you to all those who replied to my last query. For them and for
the suggestion, I can monitor asterisk using the asterisk -r -x
command option. What I would like to know is that using asterisk
-r -x way I can only use the *CLI commands. Is there any other way in
love U.all wrote:
i wanna build a production Asterisk box ,will RedHat Linux Enterprise
Server be more stable than Fedora core Linux or it makes no
significant difference
I started out running Fedora, but I have migrated away from it for a few
reasons. Fedora has a very short life cycle
Administrator TOOTAI wrote:
This is not true if you're using B410P cards. We always face timing
problem as we can't -Asterisk stability issues- add X100P or TDM400P
with those cards
Daniel,
I thought that using an empty TDM400P as a timing source may no longer
be the best solution due to
List users,
A recent post on MeetMe timing mentioned the internal_timing option,
which can be configured to have Asterisk asynchronously generate
outgoing RTP when a timing device (ie. ztdummy) is available. This
allows Asterisk to produce outgoing audio in situations where no
incoming audio
Tomasz Zieleniewski wrote:
ztttest results show value below 99,98:
[EMAIL PROTECTED]:~/src/zaptel-1.4$ ./zttest -v -c 5
snip
--- Results after 11 passes ---
Best: 50.003 -- Worst: 49.612 -- Average: 49.931827, Difference:
49.931827
This is the first thing I would address. Get that
Tomasz Zieleniewski wrote:
I am using Debian OS kernel 2.6.22-3-amd64
and zaptel driver 1.4 with ztdummy module for meetme application.
I use meetme with SIP channels.
I have such problem that when one connects to the conference voice is
cut.
Each voice sequence is disturbed.
Does any
Per Jessen wrote:
I don't know why it's stopping, but I'm pretty certain it's a segfault.
Next time it happens, I should be getting the core dump.
I'm running 1.4.13, no AGI scripts.
Per,
You should be able to determine if it was a segfault by looking at your
system log. For example, on one
Raúl Gómez C. wrote:
Thinking about my original post, I was reluctant of installing my PBX
on a shared system, is a Dell PowerEdge 2950 with 2 Intel Xeon Dual
Core CPUs @2GHz (4 totals cores) and 4GB RAM which serves as Domain
Controller and File Server (Samba), central backup server
Erik Anderson wrote:
For this load level (even with high-load transcoding), a multi-core
machine certainly would not be needed. That said, it certainly
wouldn't hurt anything to add on extra cores, especially if they're
free ;-)
Raul,
The points concerning overall load are valid, but I agree
Alan Lord wrote:
I'm building a test asterisk server and building the latest kernel I got
to wonder if there are any specific recommendations about schedulers and
so forth for optimum performance.
There are a few areas that raise questions in my mind and I wonder if
anyone has any
Atis Lezdins wrote:
This is available starting from 1.4, see UPGRADE.txt:
* ... The new behavior, enabled by setting autofill=yes in queues.conf
either at the [general] level to default for all queues or
to set on a per-queue level, makes sure that when the waiting
callers are
Ryan Parlee wrote:
I am experiencing extreme jitter/slowdown on Playback() or Background().
I've looked thoroughly on voip-info.org and elsewhere for help regarding
this issue but cannot figure this out. I can make outgoing calls with no
problems.
When I run zttest I get the following:
Call files and app_amd (Answering Machine Detection) come to mind.
app_amd can take a little time to tune, but you can get it to be pretty
reliable in most cases.
See: http://www.voipinfo.org/wiki/index.php?page=Asterisk+cmd+AMD
http://www.voipinfo.org/wiki/view/Asterisk+auto-dial+out
Paco Brufal wrote:
I have an asterisk 1.2.18 working fine, the only problem is that all
applications that play audio, sound like tremolo or vibrato, but
musiconhold plays fine.
The same audio file (wav, mp3, ...) works fine with Musiconhold()
but not with Playback() or Background()...
Do you
Remco Post wrote:
I guess that if I read these stats correctly, the bottleneck for * is
not so much cpu power, it's the cpu cache. As I see it, the cpu cache
becomes far less efficient for larger call volumes, eg. the cache is
unable to keep the most frequently used code and data in cache, due
List users,
This post contains the benchmarks for Asterisk at high call volumes on a
4 CPU, dual-core (8 cores total) server. It's a continuation of the
posts titled Scaling Asterisk: Dual-Core CPUs not yielding gains at
high call volumes. They contain a fair amount of information,
John Hughes wrote:
For me all these numbers look too small to be useful for benchmarking.
John,
They are small, and they are probably more useful as baseline numbers.
I'm working on writing up some data I've collected off of our production
switch. The call range is 0-450 at 10 call
John Hughes wrote:
OpenSSI can't (at the moment) migrate threads between compute nodes. It
can migrate separate processes, but doesn't Asterisk use threads?
John,
Asterisk uses 1 thread per call, plus about 10 to 15 background threads
that persist throughout the life of the process.
I'm
John Hughes wrote:
Matthew J. Roth wrote:
As far as Asterisk is concerned, at low call volumes the dual-core
server outperforms the single-core server at a similar rate.
Outperforms in what sense?
At low call volumes the cumulative CPU utilization, expressed as a
percentage
Luki wrote:
Perhaps a naive question, but how does 0.137% CPU utilization per call
equal 1735 MHz per call?
If 1735 MHz / 0.137% = 1735 MHz / 0.00137 = 1266423 MHz at 100%
utilization ??! Even with 4 CPUs, those would be 316 GHz CPUs.
I think you meant:
Average CPU utilization per call: 0.137%
JR Richardson wrote:
Do you get any errors at max call capacity about too many open files? You
may try increasing your file descriptors.
JR,
Thanks for the response, but I have the maximum number of open files
available to Asterisk set to 65536.
Thank you,
Matthew Roth
InterMedia
William Moore wrote:
Are you recording memory figures as well and have you checked the
total used memory? Or did I miss it somewhere? Thanks for doing
this, scalability testing is always good.
William,
This round of benchmarking is heavily focused on CPU utilization,
because it is causing
JR Richardson wrote:
The Dual-Core system you are working with must have cost a bundle, several
thousand. My approach has been to stick with single cpu, single core
servers and add more servers to the cluster, versus building bigger, faster
Proc servers. With sub $1000 servers, I can achieve
Mark Coccimiglio wrote:
Sounds like you are running into the hardware limitations of your
systems PCI or Front Side Bus (FSB) and not necessarily an issue of
asterisk. In short there is a limited amount of bandwidth on the
computer's PCI Bus (33 MHz) and the FSB (100-800MHz). One thing to
List users,
Using Asterisk in an inbound call center environment has led us to
pushing the limits of vertical scaling. In order to treat each caller
fairly and to utilize our agents as efficiently as possible, it is
desirable to configure each client as a single queue. As far as I know,
List users,
This post contains the benchmarks for Asterisk at low call volumes on
similar single and dual-core servers. I'd appreciate it greatly if you
took the time to read and comment on it.
Thank you,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
Sean M. Pappalardo wrote:
Just curious if you've checked out Linux clustering software such as
OpenSSI ( http://www.openssi.org/ ) and run Asterisk on it? It
features a multi-threaded cluster-aware shell (and custom kernel) that
will automatically cluster-ize any regular Linux executable (such
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