to start blocking them at the edge. Let their customers complain to them
instead.
-Original Message-
From: Erik L
Sent: April 11, 2010 10:38
To: na...@nanog.org
Subject: Seeking Amazon EC2 abuse contact
Could someone from Amazon EC2 please contact me off-list regarding an abuse
issue from one
,
Erik de Wild
Tripple-o: your asterisk migration partner
the Netherlands
On 6 feb 2010, at 03:54, Thomas Perron wrote:
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses
of communicating ;-)
erik
On 6 feb 2010, at 22:04, Ira wrote:
At 11:17 AM 2/6/2010, you wrote:
Actually bottom-posting without trimming anything (SCNR) is about as
annoying as top-posting.
Interleaved posting is fine, quoting just enough text so everyone
can understand the context.
Seems to me if you
(10.0.50.107)
;; WHEN: Mon Nov 16 14:14:05 2009
;; MSG SIZE rcvd: 111
enum.conf:
[general]
;
; The search list for domains may be customized. Domains are searched
; in the order they are listed here.
;
search = ns3.e164.xxx.com
search = e164.arpa
Regards,
Erik
it should look something like
exten = 4000,1,Dial(SIP/4000,30,t)
exten = 4000,2,Goto(4001,1)
exten = 4001,1,Dial(SIP/4001,30,t)
If 4000,1 is answered it will never reach 4000,2
if 4000 is busy or not available for another reason it wil goto 4001,1
hope this is useful
Erik de Wild
Tripple-o
just a hint. you might have # assigned the moh in feature.conf and #3
to starting the recording. check your feature.conf and makesure that #
isn't assigned to anything.
erik de wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Verstuurd vanaf mijn iPhone
Op 7 sep 2009 om 20:40
you should check dialstatus and gotoif. if you use both in the proper
way ( see the wiki) then you have the dialplan behaviour you are
looking for.
erik de wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Verstuurd vanaf mijn iPhone
Op 7 sep 2009 om 21:26 heeft Miguel
idea. after mixing it should be stored in a
retrievable way.
Erik de Wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Verstuurd vanaf mijn iPhone
Op 8 sep 2009 om 00:25 heeft Miguel Molina mmol...@millenium.com.co
het volgende geschreven:\
I imagine this setup will need those
Just a hint based on experience. Run top from de linux prompt to
check if any proces causes an enormous cpu load. I once ran into the
same behaviour because some asterisk related php script looped and
took almost all the cpu power available.
erik de wild
Tripple-o
Your Asterisk migration
You can use the M parameter to run a macro after the channel picks up
or the g parameter to jump to a given context. there is also a
parameter to run an AGI script. Check the dial() cmd on the wiki for
further details.
Erik de Wild
Tripple-o
Your Asterisk migration partner
the Netherlands
the telco role and the other the enduser
role.
See www.asteriskguru.com/tutorials/e1t1.HTML
Erik de Wild
Tripple-o
Your Asterisk migration partner
The Netherlands
Op 4 sep 2009 om 23:23 heeft Juan Cardoza jcard...@tpmex.com het
volgende geschreven:\
Hello All
I am looking for the cable I need
if
building a working system from source is so simple?
Hope this is useful for someone
\erik
Date: Sat, 13 Jun 2009 09:51:24 +1000
From: Alex Samad a...@samad.com.au
Subject: Re: [asterisk-users] Help building dahdi for debian
To: asterisk-users@lists.digium.com
Message-ID: 20090612235124.gb17
it is useful
/Erik
Step 1 Installing needed packages/libs on your system
install this packages (I'm not sure if all the packages are needed but
with this packages it works)
apt-get install libwww-perl
apt-get install openssl
apt-get install libcrypt-ssleay
apt-get install libnet-ssleay
city and province?
--
Erik
Caneris
Tel: 647-723-6365
Fax: 647-723-5365
Toll-free: 1-866-827-0021
www.caneris.com
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. If its core purpose does not consist of interfacing with ASR
and TTS engines, then some would argue that it's best to keep such features to
a separate platform.
Regards,
--
Erik
Caneris
Tel: 647-723-6365
Fax: 647-723-5365
Toll-free: 1-866-827-0021
www.caneris.com
://www.trafficondemand.ca/
I believe that it's still considered beta for non-Toronto.
Regards,
--
Erik
Caneris
Tel: 647-723-6365
Fax: 647-723-5365
Toll-free: 1-866-827-0021
www.caneris.com
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith (lists
that the client/stakeholders are aware
of limitations. There's a certain expectation of it will speak perfectly
these days, followed by disappointment and blame when reality hits them.
Regards,
--
Erik
Caneris
Tel: 647-723-6365
Fax: 647-723-5365
Toll-free: 1-866-827-0021
www.caneris.com
Erik -
Have you found RealSpeak to be worth the cost?
Actually my last note was probably a bit misleading because in the particular
cases I mentioned RealSpeak, the platform wasn't Asterisk and Cepstral wasn't
even on the radar.
Can Cepstral, with
the hourly $ spent on tuning, be made
to asterisk
through an ATA box?
Best Regards,
Erik
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Hi Giorgio,
Thanks for your answer.
Your setup is exactly what we're thinking of. We have 1100 DID's, so
that shouldn't be a problem at all. Which ATA box are you using?
Erik
On Sep 23, 2008, at 2:06 PM, Giorgio Incantalupo wrote:
Hi Olivier,
We DO NOT use faxdetect because it does
by ~ 600 employees.
Erik
On Sep 23, 2008, at 3:25 PM, Matthew Marion wrote:
Hey Erik,
You can also check out pika technologies which supply chan_pika.
This comes with a fax application that will let you do your faxes in
asterisk (even using non-pika boards). Works pretty good
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http
?
Regards
Bilal
--- On Tue, 9/2/08, Erik Anderson [EMAIL PROTECTED] wrote:
From: Erik Anderson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk Trunk and normal
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday
giving it a try. I've wanted to get
it set up for several months now, but haven't been able to due to lack
of play time in my work schedule.
-Erik
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.
Ideas?
Thanks!
-Erik
--
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http://andersonfam.org
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openfire and
asterisk on the same box. If needed, you could even just nice the
openfire process down to a lower priority than asterisk - it's not as
latency-sensitive as asterisk is. I'd doubt you'll need to do that,
though.
-Erik
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of someone's call history: http://1ezphone.com/callhistory.html#
I, for one, won't be giving this a try any time soon.
-Erik
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I have been doing some reading about gtalk and asterisk. Most of it is
pointed to enable using gtalk for making phonecalls. Would it be
possible to use gtalk instant messaging input (just some text send to
the gtalk account configured on an asterisk box) into the dialplan.
This way you
As far as I know there is no Dutch Asterisk mailing list but there is
a Dutch Asterisk forum. See http://forum.asteriskportal.nl/ It is not
an answer to your question but you are more then welcome to join the
forum.
Erik de Wild
Tripple-o
Your Asterisk migration partner
was almost tempted to try it, but time was short at the time, and
holding an N770 to my head seemed a bit silly.. (built in mic and
speakers, but no socket for an external mic)
Gordon
I run Asterisk on a Nokia 770 and as a mini pbx it run pretty smooth.
I used it, just for fun and
starts I will join ;-)
Erik de Wild
Tripple-o
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.
exten = s,1(answer),Answer()
exten = s,n(gotoif),Gotoif($[ ${DIALSTATUS} : NOANSWER]?wait:go_on)
exten = s,n(wait),Wait(1)
exten = s,n,Goto(gotoif)
exten = s,n(go_on),Playback(special_message)
Erik de Wild
Tripple-o
Your Asterisk migration partner
Another solution that works for me is to add
and it seems to work ok but
I would like to add ztdummy for conferences. Any suggestion to solve
this problem is very welcome.
Friendly regards,
Erik de Wild
output uname -a
Linux debian 2.6.18-6-486 #1 Sun Feb 10 22:06:33 UTC 2008 i686 GNU/Linux
#
gcc -g -O2 -I. -g -fPIC -Wall -DBUILDING_TONEZONE
I pass a value from a macro by storing the value needed to the $
{MACRO_RESULT} variable. This is returned and because of this
available after finishing the macro. I'm not sure that it works in the
way you are looking for but it works for me.
Erik de Wild
Tripple-o
list. I hope someone
will find this usefull.
Below are the actual Asterisk lines. It is pure old fashioned Asterisk
without any additional AGI scripts or whatever.
With friendly regards,
Erik de Wild
Tripple-o
Your Asterisk migration partner
; this is where te inbound call
didn't mention whether you are doing traffic shaping on your
upstream connection, so I'll assume you're not. That would be
something good to look into - with traffic shaping, you can prioritize
your VoIP traffic over all other types of network traffic.
-erik
the
phone using this wav file before. Does anyone know what it is used for?
It's played at the completion of the boot process. It's always been
very quiet on the models I've worked with.
-erik
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On Wed, Apr 2, 2008 at 9:37 AM, Ruben Zamora [EMAIL PROTECTED] wrote:
I just want to know if anyone have problems with server DELL 1600,
Like: Hangup Call.
Give us some more details of your setup and you'll probably have
better chances of getting an answer.
-Erik
On Wed, Apr 2, 2008 at 10:24 PM, Al Baker [EMAIL PROTECTED] wrote:
Clearly all of this not feasible in a IVR environment, so, in the
absence of all this, just how good , and how sophisticated of a voice
recognition can one achieve ?
Have you ever called Google 411?
1-800-GOOG-411
It'll
and simple. Please stop abusing the list for your
own business opportunities.
-Erik
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On Wed, Mar 19, 2008 at 12:48 PM, Carlos Carvalhar
[EMAIL PROTECTED] wrote:
How do I install phpagi?
http://phpagi.sourceforge.net/
Since phpagi is really just a set of php libraries, all you need to do
to install is dump it somewhere and add that location to your php
include_path.
-Erik
can I get the php files of the class phpagi?
How did you download it?
$ wget http://superb-east.dl.sourceforge.net/sourceforge/phpagi/phpagi-2.14.tgz
$ tar zxvf phpagi-2.14.tgz
$ cd phpagi-2.14
$ ls
-Erik
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want the hard copy :)
Agreed - I'm sure you'll be much more happy with the stability of your
vanilla asterisk implementation (assuming you're running on a stable
OS and server-class hardware) as well as being much more comfortable
with what's going on behind the scenes.
-Erik
On Mon, Mar 17, 2008 at 12:09 PM, Brett Crapser [EMAIL PROTECTED] wrote:
Then I noticed how all the asterisk files/directorys had been 777'ed.
Ouch - I think I'll pass as well.
--
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http://andersonfam.org
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through that server. If this isn't an option, you need to make sure
that your asterisk server has a valid publicly-available DNS record
(and reverse DNS). That's most likely the reason the remote server is
rejecting these emails.
-erik
server.
-Erik
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- these usually fall in the 10k-20k udp range. 5060/udp is used
for call signalling only, the actual voice data can use a variety of
ports, depending on how you're set up. You can specify what RTP ports
you want to use in your rtp.conf.
-erik
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On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK.
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://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom
-erik
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into the new Sun servers? I've been researching them
lately, and they have some compelling offerrings. They also offer
full support for linux as well...
-erik
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, which in turn would dump it into a certain meetme room.
Alternatively, you could have the central server call out to the
branch servers and join them to the meetme room.
In practice, though, I have no idea how the audio quality would be.
-erik
.
-erik
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On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
It also consumes more CPU.
True, a fraction more. If you have that little overhead on your
server, though, that this would cause a problem, you probably should
upgrade your hardware, IMHO.
-eriik
All - I've been trying to pick out a bluetooth conference phone that I
could use with a softphone along with my asterisk server. I've been
looking at the TrendNet TVP-SP4BK. Have any of you used this device
or any other bluetooth conference phone? How have your experiences
been?
Thanks!
-Erik
discussed ad nauseam on the list and is documented
quite well on the wiki - search there and you'll most likely find the
answers you're looking for.
-erik
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On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote:
How about http://www.mgamble.ca/oss/iphone_asterisk/ ?
Hah! Cool, but quite ridiculous. :-)
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on the list: HP Proliant
DL360IBM x206IBM x346
Does anyone has a most recent list and I will be adding the digium cards for
T1 the 220 series with echo cancellation?
--
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http://andersonfam.org
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...not sure
exactly what you're looking for.
-erik
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sip.conf?
That would be quite helpful in helping you troubleshoot this problem.
Also, please post any messages that appear on the asterisk console
when you try and register your x-lite phone.
-Erik
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On Jan 20, 2008 7:47 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
Here are my log information.
[Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from
'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device does
not match ACL
[Jan 20 12:35:33] NOTICE[2637] chan_sip.c:
On Jan 20, 2008 8:06 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
Windows XP.
Andrew - you're going to need to get us your sip.conf before we can
really assist you any further.
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server motherboards. I've become *completely* addicted to the
DRAC cards on the high-end PowerEdges, to the point that I now refuse
to order a server without a DRAC card.
That said, I'm sure this server would run a small/medium asterisk
install just fine.
-Erik
. That wasn't an option when I ordered the SC440.
-erik
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On Jan 10, 2008 8:24 AM, Drew Gibson [EMAIL PROTECTED] wrote:
It has 5 ports! Although the ports are labeled as 1 Internet port and 4 LAN
ports, each can be assigned to a VLAN of your choosing and you can use them
as you please (at least you can under openWRT).
Yup - you can do the same with
on it. That will traffic
shape any type of traffic you want. I have installed several of these
around the country and they work great for prioritizing VoIP traffic.
-Erik
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page on its QoS implementation:
http://www.dd-wrt.com/wiki/index.php/Quality_of_Service
Looks like they don't recommend HFSC currently due to some lag issues.
That might have been fixed, though, in the more recent firmware
builds.
-Erik
= 1002,10,Hangup
Kind Regards,
Erik
Am Mittwoch, 2. Januar 2008 18:59 schrieb Tilghman Lesher:
On Wednesday 02 January 2008 09:34:24 Erik Wartusch wrote:
No it's even simpler. ( I dont need an IF case)
I just want to add e.g. 15 minutes to the current date / time:
So simply said:
${STRFTIME
the math function in 1.4 but how can I manage easily the
time operations?
Kind Regards,
Erik
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manipulation)
Cheers,
Erik
Am Mittwoch, 2. Januar 2008 13:23 schrieb Doug Lytle:
Erik Wartusch wrote:
Hi all,
Im using Asterisk 1.4.11 and I want to proceed some time and date
operations in my dial plan. (for a time shifted callback).
If you'll be using call files to do this, you can 'touch
increase +1 and the hours start with 0:x the
minutes with 12 ( and not 72 as the normal addition would result).
Kind Regards,
Erik
Am Mittwoch, 2. Januar 2008 16:02 schrieb dave cantera:
erik,
you can start here:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
http://www.asteriskguru.com
that I use:
http://www.voipreview.org/voipspeedtester.aspx
This sort of test is ideal for VoIP because unlike most other speed
tests, it measures latency, jitter, etc.
-erik
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asterisk
tools that will
take care of the patching/compiling/installing/configuring for you.
-erik
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You should be able to issue a stop gracefully command to asterisk.
That'll cause it to stop accepting new calls, but will let existing
calls continue until complete.
-erik
On Nov 27, 2007 12:06 PM, Alex Balashov [EMAIL PROTECTED] wrote:
In other words, what I need is a way for the upstream
what additional
functionality it has over the standard version, though.
-erik
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that will get high
priority if that would help in your setup.
HTH-
Erik
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you're convincing any people to check out this
product by doing this?
Please go away until you can figure out a way to contribute in a meaningful way.
-Erik
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verizon told me is that its b8zs/esf, that's it.
One end of your T1 link will need to be pri_net and one will need to be pri_cpe.
-erik
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if you prefer.
-erik
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to one of the report pages.
It doesn't do the query manually.
-erik
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-- randomly, dropped
calls). So you have to install the BETAS whether you want or not...
That you have to use unique ports is a rumour and not SIP standard. As John
said -- IP:Port must be unique . I definitely not understand why I should
use random ports.
Kind Regards,
Erik
I`m
correctly for me. That said,
this sort of thing can happen frequently if, instead of composing a
new email to the list, you hit Reply to an existing message and just
change the subject line.
-erik
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On Nov 13, 2007 11:44 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
HI Erik,
thanks for your post, Actually im sending new posts not replying but if you
see them correct, how come its wrongly viewed for me. Are you using a
speciall software to view mailing lists? Im just using firefox
]
type=friend
context=outgoing
username=test1
secret=987454
qualify=yes
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
callerid=Test 0
insecure=very
Kind Regards,
Erik
Yep - as Doug mentioned, give esf framing and national switchtype a try.
I have a PRI from ATT in one of my offices, and use this setup.
-erik
On 11/1/07, Lutgring, Sam [EMAIL PROTECTED] wrote:
I am in the process of implementing a new ISDN pri and have a couple of
questions
in this configuration.
-erik
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quad-port card.
Re-seating the PCI expansion board seemed to solve the problem.
-erik
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On 10/22/07, Vincent [EMAIL PROTECTED] wrote:
2008 might be a good year to update * - The future of telephony :-)
Version 2 of TFOT was just released a few weeks ago...
http://downloads.oreilly.com/books/9780596510480.pdf
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On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Why is that? I'm using the xml-style config and they're working just fine.
--
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http://andersonfam.org
.
-erik
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:
http://www.newegg.com/Product/Product.aspx?Item=N82E16826275009
Haven't heard of a single problem thus far.
-erik
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recognize that vanilla asterisk text configuration isn't for everyone.
-Erik
P.S. By the way - don't misquote me. I said nothing about laying down
a config in 3 minutes.
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asterisk-users
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
I don't do text editing so please indulge me. Why would someone want to do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of
for asterisk.
-erik
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DID charge.
If you are just wanting to receive incoming calls, check out IPKall -
they'll give you a DID and a SIP trunk to your PBX for incoming calls.
-erik
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asterisk-users
://en.wikipedia.org/wiki/Load_%28computing%29#Unix-style_load_calculation
For systems with multiple CPUs, the number needs to be divided by the
number of processors in order to get a percentage.
- Erik
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On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I wouldn't be too happy about a system with a
loadavg of 3.
The system he mentioned had 8 cores, though. So a load average of 3
is less than 50% usage.
-erik
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A400D PCI card).
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 ;-)
-erik
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in conversation with our Sales rep
today, and he's quite adamant that they currently only support Cisco
Call Manager and CCM Express. I believe they're using CCM to provice
the SIP trunks - if this is indeed the case, I don't see
interoperability with asterisk as a problem.
Thanks
-Erik
On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote:
Hi all,
Probably this is the wrong place to ask,
but is there an estimated time of arrival of the future?
i.e. TFOT--next generation dealing with * -1.4
I attended a workshop some time ago, and the book was part of the
package
The
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