Unless it's Monster cable ground wire at $100 a foot that has depleted
oxygen and is bombarded by Xrays to free up the quantum particles which
makes everything work better.
[/sarcasm off]
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 09, 2007
Try some of these suggestions.
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 15, 2007 12:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TDM400P FXO
Please explain to me how FXO tune would fix popping and clicking sounds???
-Original Message-
From: Matthew Fredrickson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 15, 2007 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM400P
I've gotten burned by software raid so I'll probably be sticking with
hardware in the future. If your drive dies for some reason it could affect
the SATA bus and cause the system to crash. That's what happened to me. It
wouldn't come back up on the second Raid 1 drive until I removed the bad
Still true on CentOS 5. You can only RAID partitions unless you do the LVM
thing. What are the disadvantages compared to being able to RAID the whole
disk? Maybe for monitoring it's just more to deal with but does it make a
RAID 1 any less reliable?
-Original Message-
From: Zane C.B.
It compiles fine for me but I can't change the soft EC. It always compiles
with MG1 no matter what I select in zconfig.h. Downgraded back to 1.4.4 and
it works fine again.
-Original Message-
From: Jan du Toit [mailto:[EMAIL PROTECTED]
Sent: Friday, August 24, 2007 1:30 AM
To:
Well there are a couple fine examples of FUD if I do say so myself. Just do
a search and see what cards the 'serious' companies out there are using.
Nuff said.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 26, 2007 8:49 AM
To: Asterisk Users Mailing
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma
On 8/26/07, shadowym [EMAIL PROTECTED] wrote:
Well there are a couple fine examples of FUD if I do say so myself. Just
do
a search and see what cards the 'serious' companies out
, Digium vs. Sangoma
shadowym wrote:
Well there are a couple fine examples of FUD if I do say so myself. Just
do
a search and see what cards the 'serious' companies out there are using.
Nuff said.
When did saying nothing at all become enough?
Eric ManxPower Wieling wrote:
Sangoma cards
We are at about 250 days of 24/7 uptime now. It would be more but we had a
long power outage and the UPS's ran out. We are using Sangoma cards though.
You can easily substitute the 2U for a 3U but I don't think you need it.
Qty 1 Supermicro SC823T-R500LP, 2U, redundant 500W ps w/ PFC, 6x1 SATA
I have found the response to bug reports extremely impressive! If something
happens and I spend a bit of time to get good information to post to
bugs.digium.com or put it in a bug thread that matches the problem I am
having the response often can be very quick and sometimes resolutions can
come
Tried the AsteriskNOW beta6 VMWare image yesterday. It's come a long way
since last time I looked at it a few months ago. Some things are nicely
polished and worked very smoothly but somethings were surprisingly flaky.
Maybe because I never had all the incoming/outgoing/user stuff configured as
when
you do what should be a dot upgrade, and find out a feature that had
worked just one dot below has now stopped working, or worse yet
asterisk segfaults.And when it's on a production system you can't
just keep trying and get traces.
On 8/29/07, shadowym [EMAIL PROTECTED] wrote:
I have found
Then you should probably use a commercial application like the Business
Edition. I've found that once I decide to go down the open source road it's
a different ball game. Test with the latest and greatest release that has
the features you need. If it's a fairly new release chances are it's not
- Non-Commercial Discussion
Subject: Re: [asterisk-users] where is 1.4.12?
shadowym wrote:
Then you should probably use a commercial application like the Business
Edition. I've found that once I decide to go down the open source road
it's
a different ball game. Test with the latest
Maybe his comments were taken out of context as they don't have the whole
interview posted. Why is he talking about queue games, Biologicall and
other extremely niche crap when there are huge holes in the basic offering
(SLA and SCA)?
From: Al lists [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
Message-
From: Matthew Fredrickson [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 13, 2007 1:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
shadowym wrote:
Maybe his comments were taken out
FreePBX is a beautiful thing but nothing will prevent the inevitable train
wreck if your hardware is garbage. Not saying yours is but..just sayin.
-Original Message-
From: Jay R. Ashworth [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 13, 2007 1:32 PM
To: Asterisk Users Mailing
[mailto:[EMAIL PROTECTED]
Sent: Friday, September 14, 2007 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
shadowym wrote:
Yes thank you for reminding me it is open source. Thank you for reminding
me
You never really specified what you want this for. If it is for enterprise
type installations then Aastra has a very robust SIP DECT solution
specifically designed for multiple roaming extensions. When you go through
their webinar training they provide all the calculations in terms of square
You cannot set up your dialplan with the CLI or am I missing something?
Creating relatively simple dialplans manually can be quite time consuming.
A GUI takes care of all that grunt work.
-Original Message-
From: SIP [mailto:[EMAIL PROTECTED]
Sent: Monday, September 17, 2007 2:12 PM
To:
The SME+ has a PCI bridge chip btw the 133MHz and 100Mhz slots so you might
want to try moving it to a slot on the other side of the bridge. I believe
that card will work in any of the slots.
-Original Message-
From: kido [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 19, 2007
Just forget it about the 1.2 mantra, it's not going to happen. Focus your
energy elsewhere.
Lot's of bug fixes are good. Even Cisco comes out with regular bug fixes
for IOS. Open source just makes things more visible.
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
I disagree with any argument for or against Digium in support of Asterisk as
much as I do for or against Sangoma or Rhino or one of the Chinese knock
offs in support of Asterisk. Digium uses the open source community to
create better commercial software products and their licensing policies
Like fixing the poor design of the TDM400P and TE110 with the newer cards
that advertise VoiceBus. For a company that supposedly embraces the open
source philosophy I don't think Digium has been very forthcoming with what
they are doing so they should not be surprised by any apparent lack of
Because there is still old hardware in the pipeline at distributors and
resellers. They may even still be manufacturing some of that old hardware
so there is probably a lot of unrealized money involved. So in that sense I
can't blame them for being a bit hush hush about it's short comings.
Astlinux does seem to be growing cob webs a bit. Askozia doesn't support
Zaptel cards in the GUI and not sure if it is possible to configure them
manually. There is no Voicemail storage mechanism yet. It's still very
basic but a nice start.
-Original Message-
From: Michael Graves
That's kinda high then. I wouldn't be happy about that either. You
shouldn't be over 30% ever for anything real time. Instantaneous spikes can
really start to make your life miserable at that point.
-Original Message-
From: Erik Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, October
I hope I am not opening a can of worms here but IMHO there is ABSOLUTELY NO
REASON TO USE SCSI anymore! For sure not for this application but most other
things too. SATA is mature now, does command queuing, and works well on 2.6
kernels. Oh, there is the issue of cost as well.
-Original
: [asterisk-users] Is there real benefits on a SMP machine for
Asterisk?
shadowym wrote:
I hope I am not opening a can of worms here but IMHO there is
ABSOLUTELY NO REASON TO USE SCSI anymore! For sure not for this
application but most other things too. SATA is mature now, does
command queuing
So nano just makes things too easy for you?
-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Monday, October 15, 2007 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
I use vi. Not
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 fancy things like call
queues, parking, forwarding,
- Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
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
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
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
Subject: Re: [asterisk-users] What web GUI are people happy with?
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
So how long would it take you to vi a 20 extension office with custom
dialplan involving a medium level of complexity? Including time to debug
etc.
Well - there's a large amount
. Some of them I posted to
astrecipes.net or voip-info as well so I have them handy all of the time.
OK, I'm on this list since 2004 so maybe I'm a bit biased, but it really
works fine for me.
l.
On Wed, 17 Oct 2007 05:23:14 +0200, shadowym [EMAIL PROTECTED] wrote:
That seems quite fast
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What web GUI are people happy with?
On 10/17/07, shadowym [EMAIL PROTECTED] wrote:
Ok so you use templates. I understand that. The problem is some people
on
here seem to be claiming they type it all in from
I have read all the wiki's and blogs and how to links about Bluetooth but so
far no luck. I can confirm that CentOS5 sees my Bluetooth adapter and my
cell phone. No Joy on Asterisk 1.4. The information out there is kind of
confusing as there is a lot of outdated info sometimes referring to
Size/Speed/write cycles have gone way up, price has gone way down. More
common than CompactFlash and no need for an adapter. So is it feasible to
run an Asterisk server on something like this? With a MTBF of 1million
write cycles coupled with dynamic wear management on a 4Gig USB drive,
Or your could use a touch screen with Flash Operator Panel. Just a
suggestion out of left field.
-Original Message-
From: Russell Brown [mailto:[EMAIL PROTECTED]
Sent: Friday, October 19, 2007 1:12 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Receptionists Phone
PROTECTED]
Sent: Friday, October 19, 2007 8:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
shadowym wrote:
Or your could use a touch screen with Flash Operator Panel. Just a
suggestion out of left
There is a bug in Zaptel 1.4.5.1 that prevents other Echo Can's from being
selected.
http://bugs.digium.com/view.php?id=10555
Use 1.4.6 or 1.4.4 or edit the source yourself.
-Original Message-
From: marcotasto [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 24, 2007 9:08 AM
To:
-users] Listening to Allison voicemail prompt on SIP
phone causes [pop] sounds.
shadowym wrote:
A bit hard to describe. Using a SIP hardphone I log into my voicemail at
which point Allison says you have x messages.. There are various
other prompts that exhibit the same problem but that is one
U..not really. Aastra is right up there. IMHO they are better for
several reasons.
From: Michael Graves [mailto:[EMAIL PROTECTED]
Sent: Saturday, March 01, 2008 6:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] which phones to use ??
When
This is what I have working
topsoftkey5 type: speeddial
topsoftkey5 label: Park
topsoftkey5 value: ##70
topsoftkey5 states: connected
-Original Message-
From: Russell Brown [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 04, 2008 1:22 AM
To: asterisk-users@lists.digium.com
Subject: Re:
I would rather stick needles in my eyes but that's just me.
-Original Message-
From: Matt Riddell [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 09, 2008 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Microsoft Office Communications Server
All arguments aside, I'll guarantee you MS OCS is much less stable than
Asterisk/Linux, much more buggy, and will be for the forseeable future. It
took M$ 5-10 years to get Exchange right.
So in a few more years I'll have another look at it but I won't be a guinea
pig for now.
Sounds like
That probably includes 5 years of support but still expensive.
John Faubion wrote:
Although this is a users list, I think it is more of a list for
Asterisk resellers. I'd be interested in how many of you are simply
using Asterisk as your phone system and NOT selling your services or an
You don't have to build Supermicro stuff yourself if you don't want to.
Most Supermicro dealers do it for you if you buy all the parts from them.
It's true that what your doing with Dell/HP is paying for emotional support.
When it comes to PBX's you not getting any value paying for Dell/HP
Supermicro with hotswap bays and KVM card does the same thing.
From: Darren Wright [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 26, 2008 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question
One of the
CallWeaver is the new name for OpenPBX
-Original Message-
From: Carlos Jerónimo [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 17, 2007 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX
i use xlite
Looks interesting.
I could not find out what the difference is between the Free and Pro version
of the Dialer. Can you explain or provide a link?
-Original Message-
From: S. A. Kamran [mailto:[EMAIL PROTECTED]
Sent: Monday, April 23, 2007 2:05 AM
To: asterisk-users@lists.digium.com
I have some general questions about marketing. Lot's of technical info but
I was wondering how people are getting the business to begin with. I'm from
the IT end of things but Telco is quite a bit different. Is cold calling
still the way to go or networking? General stuff like that.
Are
Yes, we found (at least with Aastra phones) that we had to disable the SIP
fixup protocols on a pix 501.
Here is the whole setup.
NOTE: I could be wrong but I believe the requirement to open ports
1-2 for remote extensions has become an urban myth. I don't think
you need to open any
Again, is the 1-2 not an urban myth? Someone correct me if I'm
wrong.
I run about 10 external extensions and limit the ports to 1-10025. I
just can't see why you would need to open 1 ports to the outside world
unless your going to have 1 simultaneous conversations.
shadowym,
best thing to do is talk to a lot of consultants, coaches, and marketing
people... take the approach you do with learning open source only reverse
it... instead of reading source (internal) ask people (external)... it is a
big undertaking and the most important task you
have... marketing
If you have an interest in learning a bit of Linux I would suggest looking
at Trixbox. I would not have said that 1 year ago but it has come a long
ways since then. Eventually as you learn more you can install your own
Linux/Asterisk/FreePBX from scratch just for the sake of being able to learn
That is probably because you did not disable SIP fixup protocol. When you
set up a PiX correctly it works. Guaranteed!
-Original Message-
From: Ed Nuñez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 25, 2007 6:31 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
You can use something like this which supposedly works well and is easy to
configure. Costs about $120 per port full retail and works with all sort of
phones including Nortel.
http://www.citel.com/Products/Portico.asp
-Original Message-
From: Dean Collins [mailto:[EMAIL PROTECTED]
SOFTWARE
FreePBX 2.1.3
CentOS 4.4
Asterisk 1.2.13
Zaptel 1.2.10
Sangoma Wanpipe 2.3.4.5
I had an Asterisk server lock up on me today after 95 days of up time. Had
to manually kill the Asterisk process and then restart. Nothing out of the
ordinary in terms of memory use as far as I could
Just wondering if anyone has tried using Asterisk 1.2 on CentOS 5. Is it
worth considering for a Production install yet? Did they fix that
spinlock.h Kernel problem?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
So how well does SLA work?
The FreePBX developers (well, at least one) seem to think it's a non-starter
and dumbs down the other features too much. I haven't experimented with it
yet so I'm eager to hear some real world feedback.
-Original Message-
From: David W. Rice [mailto:[EMAIL
You can use the digital phones with Asterisk. Just that you need a Citel
Portico external box to convert the proprietary (ie. Digital) non IP
protocol that cannot work on TCP/IP networks to standard IP SIP protocol
that can. At about $120 per port the advantages and potential issues may
or not
Octasic SoftEcho works very well for me.
_
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 16, 2007 6:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HPEC audio clipping
2007/5/15, George Pajari [EMAIL PROTECTED]:
If
Anyone have Windows Media streaming for MOH working? I followed the various
procedures on the Asterisk Wiki for using mplayer which seems to be the only
Linux player capable of playing windows media streaming audio (asf, wmv
etc.). Anyone get this working?
I can get shoutcast streams working
This is probably cold comfort but I have NEVER had any issues with MWI
working on Aastra phones. It always just works by default. No extra
configuration necessary on the phone for sure. Just reset it to factory
defaults. Explicit MWI is NOT checked by default and I have never had to
check it.
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it production ready
for fax? By production ready I mean
Here is what I am trying to do. I have a SIP soft phone running on a PC
that is streaming a local radio station. I assigned mono out in XP
Equalizer as the mic so now I have the softphone streaming audio. I then
create a conference room and dial that room from the softphone. Now anyone
who
Anybody??
-Original Message-
From: shadowym [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 24, 2007 9:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Bottom line on fax reception
So what is the bottom line? Does it work or not. I've heard stories it
works
Hi Sebastien,
I'm just a lowly user but I will tell you what I think I understand about
it.
There is nothing in the Octasic documentation that suggests you can have
continuosly updated statistics but I agree that would be a nice to have
feature. Have you tried contacting Octasic about that?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Monday, May 28, 2007 11:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bottom line on fax reception
Anybody??
-Original
lists for much bigger examples
Cheers Duncan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Tuesday, 29 May 2007 7:34 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Bottom line on fax reception
Can you cut and paste the last few relevant lines of your log file? That
should help determine what is causing the core dumps. After that is
determined you can file a bug report with the log file cut and paste if
necessary. Is there some reason you cannot test patches on a separate test
system.
If anything this should motivate the FreePBX developers a bit more.
-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 30, 2007 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X
I don't see any mention of you adjusting gains on the card/phones. Also,
what are you doing for echo cancellation? Can you post your zapata.conf
file?
-Original Message-
From: Stephen Bosch [mailto:[EMAIL PROTECTED]
Sent: Monday, June 04, 2007 10:01 PM
To: Asterisk Users Mailing List -
(Sangoma)
shadowym wrote:
I don't see any mention of you adjusting gains on the card/phones.
Also, what are you doing for echo cancellation? Can you post your
zapata.conf file?
I had actually tried to adjust the gains, but it actually seemed to make the
problem worse.
I turned echo cancellation
and
is not needed on Sangoma cards.
-Original Message-
From: Stephen Bosch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 05, 2007 11:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Noise on FXS ports (Sangoma)
shadowym wrote:
Every Sangoma
Yikes!
{sarcasm on}
Yea, why use good stuff when you can get stuff at less than half the price.
Who cares if it ACTUALLY works properly. As long as it's cheap.
{/sarcasm off}
Yikes!
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 09, 2007 10:55 PM
To:
Hi All,
Eagerly waiting for v1.4.x to mature a bit before getting serious about it.
Is it ready for production yet? If that's too general, where is it in terms
of stability compared to where 1.2.x is now. Anyone running it successfully
in production environment and if so what sort of config
I have not had any problems with hangup detection in N America. At least not
with Sangoma cards in Eastern and Western Canada. As long as the central
office your analog lines are connected to support kewlstart which means they
momentarily disconnect the loop current when the other end hangs up.
I can only speak for Aastra phones.
Central provisioning is very easy. All you need is one simple text file on
a TFTP, FTP, or HTTP server which all the phones point to. To customize
individual phones you add a second text file for each phone you want
customized. The custom text file is given
Have you tried them lately???
Firmware and documentation is light years ahead of where it was a year ago!
NONE of your issues are issues any more IMHO.
_
From: Michelle Dupuis [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 08, 2007 10:00 AM
To: 'Asterisk Users Mailing List -
I whole heartedly agree.
Trixbox/FreePBX are much more mature and feature rich. AsteriskNOW has
greater future potential because of it's tight integration and no need for
MySQL/Apache. However, it's not there yet so if I was to implement
something today I would go with FreePBX. Not Trixbox but
IMHO,
If you don't know enough about Linux/Asterisk/FreePBX to be able to set it
up yourself you should not be doing it for a Production install in a
business environment.
NOTE: Production install in a business environment does NOT include setting
it up in your house with extensions for the kids
Interesting,
Is this just a more advanced software echo canceller or software with
hardware hooks or software with hardware assisted processing?
How would it compare to a true hardware echo canceller like the one Sangoma
uses. Besides the extra CPU cycles required.
-Original Message-
Canceller (HPEC)
shadowym wrote:
Interesting,
Is this just a more advanced software echo canceller or software with
hardware hooks or software with hardware assisted processing?
A more advanced software canceller (there's no magical thing that makes
hardware echo cancellers better, it's still
Hi there,
I'm looking for a compact fanless solution preferrably wall mountable and
not too exotic. It needs to be commercial grade. I don't really consider
most of the Via ITX solutions I have seen commercial grade but perhaps
someone can convince me otherwise.
This solution is about the
: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Feb 14, 2007, at 10:17 AM, shadowym wrote:
I gotta take issue with your comments that a HWEC is just software
running on a DSP. In the case of Octasic, it's an ASIC. How it does
EC is VERY different because.it's done
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Wednesday 14 February 2007 11:17 am, shadowym wrote:
I gotta take issue with your comments that a HWEC is just software
running on a DSP. In the case of Octasic, it's an ASIC. How it does
EC is VERY different
] The High Performance Echo Canceller (HPEC)
shadowym wrote:
I gotta take issue with your comments that a HWEC is just software
running
on a DSP. In the case of Octasic, it's an ASIC. How it does EC is
VERY
different because.it's done completely in hardware
(HPEC)
On Wednesday 14 February 2007 4:12 pm, shadowym wrote:
The algorithms may be similar but EC is an infinitely variable
non-linear(analog) process. A CPU cannot do that. You can fake it by
performing cpu intensive rapid calculations one after another but it
is fundamentally
, shadowym wrote:
The algorithms may be similar but EC is an infinitely variable
non-linear(analog) process. A CPU cannot do that. You can fake it by
performing cpu intensive rapid calculations one after another but it
is fundamentally not an analog processor. HWEC is designed to deal
I looked into it once. As far as I can tell they took out the TAPI
interface a couple years ago. Probably too many support issues.
Without a TAPI interface I would say it would not be very easy if at all
possible/practical.
-Original Message-
From: Dean Collins [mailto:[EMAIL
Hi there,
Here is my dilema. I have a new small business customer that wants me to
put in a VoIP phone system for them. Based on their requirements, I have
determined that it needs to be a set it and forget it type of thing like a
lot of small business proprietary systems.
At the same
, shadowym [EMAIL PROTECTED] wrote:
Astlinux would work except it does not currently meet some key
requirements (GUI, Sangoma Analog card support). Otherwise it would
be a GREAT distribution for set it and forget it running without a Hard
Drive IMHO.
shadowym,
With a little work one could use
Thanks Tom and everyone else,
Based largely on your comments I decided to just stick with what works. I
have a site using entry level ATX server hardware that has been solid as a
rock. I'll just go with that instead of more specialized fanless hardware,
specialized power supply and 2.5 hard
I'll second that,
CentOS 4.4 + FreePBX 2.1.3 + Asterisk 1.2.13 + Sangoma A200D + Aastra
9133i's running 4 months without a reboot and no memory leaks fielding about
150 calls a day. Everyone loves the system. These are normal users used to
tradtitional phone systems.
I would not go as far as
I couldn't agree more. The Telco card is the LAST thing you should be
trying to cut corners on.
IMHO you should consider a Sangoma A200D which is even more money due to the
HWEC. It's worth every penny!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Ummm.
How about upgrading to production released drivers?
-Original Message-
From: Ron McCarthy [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 08, 2007 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Zap Channel Deadlocks
Hey List,
With hw echo cancellation you are pretty much guaranteed to not have any
problems. At least with Sangoma cards. I cannot speak for the other
manufacturers. I believe most of not all HWEC also does other things to
help clean up the sound and maybe even add background noise etc. so the over
all
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