Thanks Eric, I breezed through the documentation and got the
impression that this was the case. Good luck on getting rid of that
echo Bilal!
N.
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New to Ast
On Tue, Aug 20, 2013 at 3:01 PM, Ghanshyam wrote:
> Shaun Ruffell digium.com> writes:
>
> >
> > On Thu, Jul 25, 2013 at 02:51:02AM -0700, bilal ghayyad wrote:
> > > Hello;
> > >
> > > If our Digium Telephony Card does not support echo cancellation
> > > like (1TDM410PLF or 1AEX410PLF), what is t
#!/bin/bash
IPTABLES='/sbin/iptables'
#Set interface values
INTIF1='eth0'
# Set Limits
LIMIT="2/sec"
LOGLIMIT="5/min"
LIMITBURST="5"
#flush rules and delete chains
$IPTABLES -F
$IPTABLES -X
#echo -e " - Dropping Forward Requests"
$IPTABLES -P FORWARD DROP
#echo -e " - Dropping Inpu
They are sending requests from his own public ip huh? Trade secrets
H, IPTaibles, Fail2Ban (as a preventative), there is something
I am missing What the f is it called again? Oh yeah Pike!!!
>> alwaysauthreject = yes
I don't know about that However, using the mac address of the dev
k-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
> Sent: Wednesday, August 14, 2013 11:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] G729 Passthrough How To
>
> Hey Eric, I do h
I wanted to mention that I do not mind posting the converted files on
this list for future individuals, given that I am not doing anything
illegal...
N.
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N
Hey Eric, I do have the codec installed, and I remember hearing about
the CLI command to convert. Is there a recent how-to of blog already
discussing this somewhere?
N.
On 8/14/13, Nick Khamis wrote:
> I wanted to mention that I do not mind posting the converted files on
> this list for
Hello Ashgar,
Thank you so much for your response. As removing A2B is not an option
we would first like to begin by converting all audio files (Asterisk,
VM, A2B prompts etc...) to G729 to minimize unneeded trascoding. Linux
commands and the list of recording would be a great help. Sorry, not
new
I forgot to mention that all our equipment (phones etc..) are using
G729, and this is for internal use over the net. The problem,
concurrent calls, and bad bandwidth at some locations...
N.
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Hey!!! Eric thank you so much for your response. Could you guys please
direct us in achieving as much as possible. For example:
* What linux command can we use to convert all recording to G729
* Which files do we need to convert and there locations
* For *testing* how do we make sure Asterisk NEVER
Anyone? :)
N.
On 8/13/13, Nick Khamis wrote:
> Hello Everyone,
>
> We are currently experiencing some higher load on our servers, and
> since signaling comes into our servers on G729, we would like to
> implement G729 pass-through. A few questions arise, do we need to
&g
Hello Everyone,
We are currently experiencing some higher load on our servers, and
since signaling comes into our servers on G729, we would like to
implement G729 pass-through. A few questions arise, do we need to
convert all the recording to the codec, and what about voicemail?
We are also using
Asterisk does fine in a virtual instance. The key is finding hardware that
would
support more than just virtualization (i.e., SR-IOV) Not sure if such a
card
exist.
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On 6/25/13, Jai Rangi wrote:
> Not a problem, I wanted to tell you the diff between PRI and sip trunking.
> I am sure there are lots of option we are just fine what ever works best
> for you.
>
> Back to subject we strongly believe that sip trunking is far better option
> than PRI and that's the w
Any other experts out there?
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http://www.asterisk.org/hello
asterisk-u
Thank you mitul.
N.
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asterisk-users mai
Hello Everyone,
We are currently having talks with various service providers, and
trying to determine what the best way is to interconnect in order to
have access to the PSTN network. As you know there are two ways of
doing this:
Traditional PRI: Have trunks grouped into a transport layer such as
Hello James,
Thank you so much for your response. I should have chose my words
carefully. PCI pass-through in terms of virtualization of devices and
it's draw back are well know. I was leaning more towards near host
performance virtualization using SR-IOV.
This moves emphasis back to the producti
Anyone?
N.
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asterisk-users mailing list
Anyone try this? I saw a post here:
http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html
But not sure if it's possible. What I am asking is if there are any T1
cards with virtual functions implemented in their drivers to allow
p
What about projects like YATE, DiaStar, and mobicents (even though I
have no idea how to approach that project in terms of downloading
etc..). Are there any mature SS7/SIGTRAN stacks?
Kind Regards,
Nick.
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Hello James, thank you so much for your response!
On 6/14/13, James Cloos wrote:
> If they will do atm over oc-n, perhaps that would work better.
Yes they will do atm over oc-n only not sure if they will ring or spur it...
> Ie, a perm virt circ for SS7 and as-needed vc's for ulaw.
I know you'
Hello Mitul,
Thank you so much for your response. During the testing phase
we would like to employ an open source solution, and wanted
to know what people have had success with, given the different
user part etc..
On a side note, anyone know of service providers offering SIGTRAN?
Kind Regards,
Hello Everyone,
I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model.
We are looking to interconnect with the PSTN world, and our supplier
has given us
a few options. We can either do this over traditional PRIs, A-Links or
the SS7IP new.
I am really interested in SIGTRAN,
Hello Eric,
Thank your for your reponse. We are discussing interconnects at a
different level. We are more interested in SS7 or ISUP-IP SS7IP type
interconnects. There are many people that offer DIDs channels etc.
over the internet. Including us.
N.
--
___
On 6/13/13, Eric Wieling wrote:
> Verizon (NE ILEC) has SIP handoff.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
> Sent: Thursday, June 13, 2013 8:11 AM
> To: Asteri
Correction:
"I think VT1.5s mappings are more flexible?"
Sorry!
N.
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On 6/12/13, Don Kelly wrote:
> Is there an OC-n to SIP solution that makes sense?
>
> --Don
Hello Don, what will be coming out of the network discussed above would be SIP.
Kind Regards,
Nick.
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Hello Brian,
Thank you so much
On 6/12/13, Brian LaVallee wrote:
> Hi Nick,
>
> Going from DS1 to OC-n is a multi-step process. Typically requiring a
> hardware device to handle each MUX step. But you can find hardware that
> handles multiple MUX steps together.
The connection is coming i
You mean the SDP payload? You kind of need that
c= is used for RTP transmission. o= always confuses
me so I will just say it's important at well.
You can put a proxy in the middle and do topology
hiding I guess however, that is beyond the scope of
this list?
Kind Regards,
Nick.
On 6/12/13,
Hello Everyone,
We are looking to interconnect with a local ILEC over an OC-n transport layer.
They basically gave us two options in terms of mapping the SONET to the DS3:
* VT1.5s mapping
* DS1s mapping
The second option is quite clear. We would MUX the connection, and plug
the lines into qaud
Thank you so much for your responses!!! With this route we would have
to manage so many * boxes with T1s, not to mention, the hit we would
take on the MUX. Any decent DS/T3 cards out there?
N.
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Anyone?
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asterisk-users mailing list
To
Hello Everyone,
Anyone know of a way of bypassing the 90K audiocodes mediant 3000
equipped for STM-1 interface using line cards and a linux box :).
Kind Regards,
Nick.
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We would like implement G729 passthrough for our calls and get rid of
the encoding overhead, and a little confused as to how to do this, and
some unanswered questions. Do we need the open source G729? If so, do
we still need the patent license. Not so much of an issue, just
checking. Finally, a rec
Anyone?
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asterisk-users mailing list
To
Hello Everyone,
I am looking to getting converged with the local ILEC here in Canada
(Bell or Telus), and was wondering if I can get some more information
about typical setups. DIDs and channel offerings from third party
clecs does not fit our business model and that's why we are looking to
purcha
Hello Doug,
A quick sift through
http://www.mail-archive.com/search?l=asterisk-users%40lists.digium.com&q=solaris+10,
yielded
many unanswered questions, questions with returning questions etc... There
was even an email that had the same
subject line. Surely, the creator of that email could take a
Bump
On 5/23/13, Nick Khamis wrote:
> Hello Everyone,
>
> I have bumped into the thralling penguin page on linux vs solaris for
> asterisk. Does the benchmark still hold with the newer versions of
> kernels? Curious to know of your thoughts. Also, they mentioned
> runn
Hello Everyone,
I have bumped into the thralling penguin page on linux vs solaris for
asterisk. Does the benchmark still hold with the newer versions of
kernels? Curious to know of your thoughts. Also, they mentioned
running it on Sun Fire x2100, but no benchmarks were given for that.
Can increas
Hello Roel,
Thank you so much for your response. We currently employ a number of
similar companies. Given our increasing traffic we are really looking
towards the incumbents for various reasons.
The purpose of my post is in the hopes that someone watching will let
us know how to setup interconnect
On 5/10/13, Nick Khamis wrote:
> Anyone here using Level 3 or AT&T wholesale sip terminations services? I
> would like to know on any minimums they would require? Also, an idea of how
> competitive the rates are. I am not asking to disclose your custom rate
> deck, just a
Sorry to chime in here, is it possible to change the "Server: Asterisk
", "s=Asterisk", and "o=" within sip.conf? What are the directives
exactly please?
Thanks in Advance,
Nick.
On 5/10/13, Asghar Mohammad wrote:
> hi,
> you can try to change sip user agent and sdp session s , owner in sip
> c
Anyone here using Level 3 or AT&T wholesale sip terminations services? I
would like to know on any minimums they would require? Also, an idea of how
competitive the rates are. I am not asking to disclose your custom rate
deck, just a "what to expect". Finally, if you guys can PM me contact info
to
Are these both caller id presentation related? If not, which on is
currently being used. Finally, is there a "latest" sip_peers table
structure to use with 1.8, without the obvious hacks, deprecations.
and redundancies?
Thanks in Advance,
Nick.
--
For anyone else that may be interested in the future, I found a
detailed depiction here:
http://wiki.sangoma.com/ntg-theory-of-operation
Thanks again,
N.
On 4/12/13, Nick Khamis wrote:
> Sorry for the missing info. Our current architecture is as such:
>
> NAT <-> SIP/RT
Sorry for the missing info. Our current architecture is as such:
NAT <-> SIP/RTP Proxy <-> *(n)
Our concurrent sessions usually peak at between 700-800 channels. On
average about 450. I will of course look at the documentation to
better understand how a transcoding appliance would fit in our
arch
Hello Gentlemen,
Thank you so much for your response, we have adopted transcoding cards
in our old system, and they do have some limitations, especially when
it comes to concurrent calls. We were looking more into the lines of a
scalable multi server router like a cisco 3745. And loading it with
m
Hello Everyone,
We are looking for solutions where the transcoding is abstracted away
from our * box (i.e., to the network layer) using some carrier grade
gateway, or router.
The reason for my post is to know about solutions people have used in
the past, and how it fits into their overall archite
increase exponentially till
something starts clunking and pinging?
What I am asking is what is the general rule of thumb when performing
such tests?
Thanks in Advance,
Nick.
On 4/9/13, Steve Edwards wrote:
> On Tue, 9 Apr 2013, Nick Khamis wrote:
>
>> We have a clustered asterisk setup, a
013, at 23:43, Nick Khamis wrote:
>
>> That's just it! Nothing! It just does not pass the 91 mark. There are
>> no failed calls during the test:
>>
>> Successful call|0 |20802
>> Failed call|0
it or call limit
thing set somewhere by accident?
N.
On 4/9/13, Paul Belanger wrote:
> On 13-04-09 02:49 PM, Nick Khamis wrote:
>> Hello Everyone,
>>
>> We are running some torcher tests on our * box using SIPP. The overall
>> idea
>> of the test is to contact ast
On Tue, Apr 9, 2013 at 3:22 PM, Joshua Colp wrote:
> Nick Khamis wrote:
>
>>
>> Hello Joshua,
>>
>> Thanks again for your response. I can understand how * does not rewrite
>> anything. When you mention the difference in call id, are you referring
>&
On Tue, Apr 9, 2013 at 3:04 PM, Joshua Colp wrote:
> Nick Khamis wrote:
>
>>
>> Hey Joshua,
>>
>> It was a poor choice of words on my part. What I meant to say was
>> whether the problem was due to our asterisk configuration re-writing
>> the RR w
On Tue, Apr 9, 2013 at 2:31 PM, Joshua Colp wrote:
> Nick Khamis wrote:
>
>> Is our asterisk server not relaying the RR along with the INVITE? If so,
>> can we configure the PBX to do so using one of it's variables? * Mailing
>> list CC'ed in this email...
>
Hello Everyone,
We are running some torcher tests on our * box using SIPP. The overall idea
of the test is to contact asterisk and play a g729 encoded recording. On
the asterisk side, we are initiating the echo app for the contacted
extension, simulating a two way conversation.
For some reason we
dr, so SIP routing is
> impossible.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 04/09/2013 08:05 PM, Nick Khamis wrote:
>
> Hello Everyone,
>
> I saw an earlier post about this issue:
> h
Hello Everyone,
Was wondering what some of you for stand alone LCR implementations. I
am aware of the LCR module within asterisk and a2billing however, we
are looking for a standalone self less coupled solution. Not sure if
such thing exist. Kind of like CDR Tool but for LCR...
Thanks in Advance,
Hello Everyone,
Just looking to secure our * box, and stumbled on the following
"This advice may run counter to the majority of documentation, sample
files and examples shown on the voip-info.org site and on Asterisk
forums, but you’ll have to take my word for it – using “type=friend”
is a big mi
sessions. Not sure if this was the best bang for our
buck?
N.
On 3/25/13, Nick Khamis wrote:
> Hello Guys,
>
> Thank you so much for your response. We reran the sipp test:
>
> ./sipp -sf uac_pcap.xml -s 1001 vancouver.example.com -l 250
> -trace_err -mp 3 -d 1
>
l capacity".
Thank you so much for your help,
Nick.
On 3/24/13, Steve Edwards wrote:
>> On Sat, Mar 23, 2013 at 09:33:38AM -0400, Nick Khamis wrote:
>
>>> We are getting some rather poor results (relative) with our Asterisk
>>> setup.
>
> On Sun, 24 Mar 2013,
Hello guys, no we do not do any recording of any kind. It was my
assumption that processing media in g729 requires some sort of
transcoding on the box?
N.
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gium solutions. Again, we would love to support the
cause.
Nick.
On 3/23/13, Andrew Latham wrote:
> On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp wrote:
>> Nick Khamis wrote:
>>>
>>> Oh no secret. Some things I do is increase the ulimit size. I was
>>> wondering i
Oh no secret. Some things I do is increase the ulimit size. I was
wondering if there was a way to increase allocated memory. I have been
reading about a -p option but when I start asterisk using "asterisk -p
-10" it does not accept it but "asterisk -p 10" works fine. Not sure
if that was the intend
Hello Everyone,
We are getting some rather poor results (relative) with our Asterisk
setup. Not sure if we are using the sipp correctly etc.. but
nevertheless, is there any documentation that describes how we can get
the most our of our Asterisk box. For example when we hit the "too
many file" err
good
software.
Kind Regards,
Nick Khamis
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Hello Asghar,
I fixed the issue after I realized that I was specifying allow before
disallow. Sorry for the noise!!!
Nick.
On 3/21/13, Asghar Mohammad wrote:
> please post sip.conf.
>
> On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis wrote:
>
>> Hello Everyone,
>>
&g
Hello Everyone,
I have disallow=all and allow=g729 set in sip.conf however, it seems
that asterisk still thinks it support other codecs:
Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How
can I disable gsm,ulaw,alaw.
Thanks in Advance,
Nick.
--
Hello Everyone,
I have gone through a few really good tutorials from the OpenSIPS
site, Asterisk resources etc.. The unanswered question (and final
piece of our puzzle) is if it's possible to have a register free
environment in an OpenSIPS/Asterisk integration. Most approaches have
OpenSIPS relay
Hello Osama, and Hisham,
At 1330GMT there was some malicious activity coming from your network
IP 37.75.210.90. Please act accordingly. Things that may be of use
"972599779558"
N.
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Hello Ishfaq, and Isrlgb,
The "canreinvite" value for UA "friend" entries are set to no, and for
the OpenSIPS "peer" entry it's set to yes. I do have esternip and
localnet cid set in sip.conf.
I did not want to start a new email, but part of my problem right now
is that OpenSIPS is in charge of pe
look registered correctly. This has now become a sip proxy issue :S.
Thank you so much for your time guys!!!
N.
On 1/3/13, Nick Khamis wrote:
> Oooops yes of course 10004-10007!! Simple math does not come easy
> anymore... Anyhow, I singled out Opensips and I have two way audio
> form UA
lf Of Jason Parker
> Sent: Thursday, January 03, 2013 2:26 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Moving User Agent To Remote Location
>
> On 01/03/2013 02:23 PM, Markus Weiler wrote:
>> Am 03.01.2013 21:21, schrieb Ni
y not able to create the SIP channel between the two UA? I
will try taking opensips out of the picture and work outwards...
N.
On 1/3/13, Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I would imagine. If anything, this email would make
google search
Hello Everyone,
Is there any way we can delete the following message sent to asterisk
ml, instead of the
actual user please? I appologize for the inconvenience however, my
personal info is in the
email.
http://markmail.org/message/gwhg4trnw4wei74k
Thanks in Advance!!
Nick.
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http://www.asterisk.or
How can we get thise license? Who do we have to pay.
Nick.
On Tue, Dec 20, 2011 at 9:52 AM, khalid touati wrote:
> Thank you Raj,
> I hope it will soon require no license as I heard there is a project to
> change this law, for now I believe I will recommend our office in India to
> go for li
SIP in India is illegal.
Nick.
On Mon, Dec 19, 2011 at 3:06 PM, khalid touati wrote:
> Hi All,
> Because I am pretty sure we have people in this DL from India, I was hoping
> to get the 100% accurate information, is it legal to make calls from any
> coutry to Indian mobile phones through an Aste
Hello Everyone,
For inbound, I am trying to specify a specific context. Everything
works fine using the IP address, however with domain name
it's not working at all. I tried changing the:
Via: SIP/2.0/UDP test.com, and the
Record-Route:
If I have a peer with the host, fromdomain, and outboundpr
Hello Everyone,
Can someone please let me know what the correct way to deal with
extensions for a particular user
using asterisk reatime. For a user 1001, we would like to support:
Local Calls: 123-456-7890
LD Calls: 1-123-456-7890
INT Calls: 011-64-03-123-456-7890
PBX EXT:1002
Do I
re,
> you're just as well off recording using the normal record function.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
> Sent: Tuesday, December 06, 2011 8:22 AM
> To: Aste
could
share what tools and tricks you use to get that "professional" look?
Thanks in Advance,
Nick.
On Tue, Dec 6, 2011 at 5:07 AM, Hans Witvliet wrote:
> On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote:
>> Hello Everyone,
>>
>> Are there any descent generic I
Hello Tzafrir,
Thank you so much for your response. I was aware of the sounds extra however.
I did forget to ask if there was an up-to-date configuration example
(extensions.conf, sip.conf etc.),
that makes use of the sounds to get an IVR up an running. This is for
a SIP platform.
Thanks Again,
Hello Everyone,
Are there any descent generic IVR recordings, that we can
use to quickly get our PBX up and running? It will obviously
not include the company name.
A sexy female's voice always do well yeah?
Cheers,
Nicholas.
--
__
sql or odbc, I would do
> - grep "Europe" /etc/asterisk/*
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
> Sent: Thursday, December 01, 2011 2:36 PM
> To
Hello Everyone,
The timezone is set correctly on the OS America/Toronto:
mv /etc/localtime /etc/localtime.bak
cp /usr/share/zoneinfo/America/Toronto /etc/localtime
I even tried adding the timezone setting to sip.conf:
timezone=America/Toronto
However. Asterisk wants to be in Bucharest? Thinkin
You want to talk SIP, you need to talk SIP proxy.
Hint: http://www.kamailio.org/w/ ;)
Nick from Toronto.
On Sun, Nov 27, 2011 at 5:19 PM, Alex Balashov
wrote:
> On 11/27/2011 04:53 PM, Faraj Khasib wrote:
>
>> I tried that with my SIP Cleint but the custom Header is not reaching
>> the cleint
; I am pretty sure I read that phantom rings are sip calls to a phone where
> they are probing for extensions or something; cant remember.
>
> --E
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Beh
Nick.
On Fri, Nov 18, 2011 at 9:38 AM, Danny Nicholas wrote:
> If your phones are being “hacked” you have a firewall problem. Your phones
> should only be registering to your local DHCP server and your Asterisk box.
>
> DHCP Server 192.X.X.X
>
> Asterisk Server 192.X.Y.Y
>
> Phone 192.X.Z.Z
>
>
o apologies for
> brevity, errors, and general sloppiness.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Nov 1
; On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote:
>> The ride is over before it even began A local ILEC here in Canada,
>> is already offering
>> Unlimited World service. And this on a Tier 1 network, not the crap
>> we're use to doing
>> business on. Choo
The ride is over before it even began A local ILEC here in Canada,
is already offering
Unlimited World service. And this on a Tier 1 network, not the crap
we're use to doing
business on. Choose a different angle before you get anymore grey
hairs on that head...
http://www.bell.ca/Home_phone/Pr
Smart card? I think we should be leaning more towards the network devices?
Cheers,
Nick.
On Wed, Nov 9, 2011 at 5:23 PM, Hans Witvliet wrote:
> On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote:
>> Is anybody using pci-passthrough?
>>
> Yes, though quite a while ago.
> About three years
Hahah... I was waiting on the sideline for this question.
Nick.
On Wed, Nov 9, 2011 at 8:10 AM, Anton Kvashenkin
wrote:
> Is anybody using pci-passthrough?
>
> 2011/11/9 Nick Khamis
>>
>> Hans,
>>
>> Thank you so much for your response. We will be moving every
Hans,
Thank you so much for your response. We will be moving everything to VM soon.
Cheers,
Nick.
On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet wrote:
> On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote:
>> That sucks! What about KVM or XEN?
>>
>> Nick.
>
>
That sucks! What about KVM or XEN?
Nick.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hell
Could you give a little more detail please? We have been running
asterisk on vmware for years as our test bed.
Nick.
On Mon, Nov 7, 2011 at 8:00 AM, Michelle Dupuis wrote:
> Although you say "SIMPLE"...not all virtualization hosts allow software
> installation. On VMware the host has become an
Do you gents feel that KVM and XEN hog too much resources which in
turn effects the functionality of Asterisk?
I really like the idea of Asterisk as an appllicance, for reasons
stated in this email. It just makes life all pretty and green.
Cheers,
Nick.
--
___
work reach to that region. The goal
> is to achieve the highest quality lowest cost routes to regions our
> customers are willing to pay for.
>
> ____
> From: "Nick Khamis"
> Sent: Friday, November 04, 2011 10:40 AM
> To: "Asteris
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