[Asterisk-Users] Frustrated with echo...

Lorentz Hinrichsen lorentz.wulf at gmail.com
Wed Apr 5 05:56:51 MST 2006


http://www.voipsupply.com/index.php?cPath=99_359_360

A20002 no echo can = 359
A2002D with echo can = 659

Both above are with 4 FXO

I don't know if these are the best prices you can find, but for comparison -
from the same vendor:

http://www.voipsupply.com/index.php?cPath=99_103

DGM-TDM04B = 378.90

The above is without echo can.  There is no option for echo can, you are
stuck tuning the gain, which when I tried - left my call volume too low or
caused the echo to be worse!

I agree that rebooting sometimes brings the digium card back to life.

I've yet to have a problem with the Sangoma, also -- I have not tried their
board without the echo cancellation.  I'm thinking you'd be stuck adjusting
the gain though.


On 4/5/06, Steve Jones <sjones at ftdata.com> wrote:
>
> Can you tell me what model Sangoma cards you're talking about??  The ones
> I saw that had HW echo cancellation were substantially more expensive than
> the Digiums..  I'm hoping I was looking at the wrong model or something!
> Thanks
> -Steve
>
> ________________________________
>
> From: Lorentz Hinrichsen [mailto:lorentz.wulf at gmail.com]
> Sent: Tue 4/4/2006 10:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Frustrated with echo...
>
>
> I've had very poor results with the Digium cards, I am using a couple of
> the new Sangoma ones now (they are cheaper and have hardware echo
> cancellation).
>
> The digium boards proved almost impossible to completely eliminate echo,
> and I had random failures over time.
>
>
> On 4/4/06, Steve Jones <sjones at ftdata.com> wrote:
>
>         For phones, I've got a GS 101, a Sipura 841, and two analog phones
> hooked to an GS386 ATA (one phone per port).
>
>         My troubles seem to be regardless of which phone is used, so I
> dont think it's on the phone-end of asterisk, but rather where I interface
> w/ Vonage and Verizon via POTS FXO...  My SIP connections to the outside
> world have so far been good [frantically knocking on wood]
>
>         I did go ahead and order the digium card yesterday evening, so I'm
> hopeful this will help.  I had played with the gain, and was able to discern
> a difference, but it seemed to make some scenarios better, while making
> others worse, so I'm hoping the real digium card/drivers will just be
> smarter about handling it dynamically.  Of course, my wife, who's a
> stay-at-home-mom is the biggest user of the system, but she's not interested
> in being a techy, so getting her to interrogate all callers about which
> number they dialed, etc.. and logging her opinions of the quality of the
> call hasn't worked!  ;-)
>
>         I also have some Cisco phones, but I haven't configured SCCP on my
> system yet, and dont want to use SIP on these phone (mostly to force myself
> to learn to configure SCCP on *) so that's another aspect that may help me
> after this weekend!
>
>         Good point about the interrupts - I dont know the answer to that,
> but hopefully that'll also be a non-issue after I get the new card, and
> therefore have only one PCI slot handling everything.
>
>         Thanks for the ideas!!
>         -Steve
>
>
>         ________________________________
>
>         From: Mike Dent [mailto:mcdent at gmail.com]
>         Sent: Mon 4/3/2006 3:46 PM
>         To: Asterisk Users Mailing List - Non-Commercial Discussion
>         Subject: Re: [Asterisk-Users] Frustrated with echo...
>
>
>
>         On 4/3/06, Steve Jones <sjones at ftdata.com> wrote:
>         >
>         > I've been using my Asterisk (At my house - 2 modem-type fxos,
> and an
>         > assortment of SIP endpoints for phones) for about 5 weeks now,
> and I've been
>         > really happy with it, but I'm still having an echo problem that
> I've
>         > exhausted google with, and can't get straight...
>         >
>         > I think I've determined that because I'm using $7 voice modem
> clones for my
>         > FXOs that bad echo is going to just keep being a pain to
> me...  I think I
>         > should have only tried going through "proof of concept" state
> with them,
>         > switching to something a little better quality when it was time
> to actually
>         > commit to Asterisk.
>         >
>         > So, my question is "What's better and why:  1: a 'real' digium
> PCI card with
>         > two fxo plugins, or using a couple external SIP fxo units like a
>         > grandstream, zoom, or similar"  Personally, I think it would be
> desirable to
>         > keep the FXOs out of the asterisk box itself, just to give me
> future
>         > flexability to move to whatever the platform of the day I want
> to put
>         > asterisk on, without dealing with a PCI card to move, but if the
> consensus
>         > is that the voice quality and support for the digium board is
> the best, then
>         > that's what I'll do..
>         >
>         > So, any comments on relative quality of these devices, and/or
> ones I've
>         > missed?
>         > 1:  Grandstream HT-488
>         > 2:  Zoom 5801/5802
>         > 3:  DGM-TDM02B  (TDM 400P with two FXOs)
>         >
>         > Are there any IAX2 FXOs that I'm missing?  That seems to be an
> area that's
>         > oddly not taken care of...
>         >
>         > Any hints would be greatly appreciated!
>
>         Steve,
>         I have a similar setup at home, although I am in the UK. I've got
> the
>         echo fairly well under control, however it seems much less when
> using
>         my Cisco 7960 rather than
>         the Grandstrean BT102 phone.
>         Have you tried dropping the gain?
>         Have you made sure you have both cards on seperate IRQ's which are
> not
>         in use by network, video etc? I disabled USB and on board audio in
> the
>         BIOS to help free up IRQ's.
>         I think your best option is the TDM400 card, or perhaps consider
> the
>         Sangoma card with a dual FXO module, maybe slightly cheaper!
>         I'd be interested what SIP phones you are using and if echo
> differs
>         between them.
>
>         Mike
>
>
>
>
>         _______________________________________________
>         --Bandwidth and Colocation provided by Easynews.com --
>
>         Asterisk-Users mailing list
>         To UNSUBSCRIBE or update options visit:
>            http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/bf383d99/attachment.htm


More information about the asterisk-users mailing list