[Asterisk-Users] Frustrated with echo...

Steve Jones sjones at ftdata.com
Tue Apr 4 06:58:16 MST 2006


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



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