[Asterisk-Users] Livevoip 800 Choppy Audio

[Wiley Siler]

Scott,

Hello again!
I currently only recommend two companies.

For dial time, VoipJet has been a rock for me.
For DIDs, Nufone has been a consistent performer as well.

After that, you may just need to experiment.  
I have heard good things about Teliax and when I tested them out, it
worked well.
YMMV as the saying goes!

Cheers!
Wiley

 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott
Wolfe
Sent: Friday, June 03, 2005 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Livevoip 800 Choppy Audio

Wiley,
  Long time no chat. I just got asterisk going with my Legacy Mitel PBX.
YEA!!! I have been following your emails on Livevoip and am wondering if
not them then who? I am still looking for something that will allow me
several concurrent connections for a small business setting.

Take care and good weekend.

-Scott


----- Original Message -----
From: "Wiley Siler" <wsiler at education2020.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Friday, June 03, 2005 3:36 PM
Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio


Go to dslreports.com and look in the forum for LiveVoip.

Or alternately you can search this list with google via the
site:lists.digium.com parameter.

I spent two months working through problems with LiveVOip.

I highly recommend against them.

Cheers,
Wiley


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John
Kington
Sent: Friday, June 03, 2005 3:08 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Livevoip 800 Choppy Audio

I just signed up with livevoip for 800 DID and have very choppy audio.
>From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am
using IAX and was assigned to server iax01.nyc.*. I do not believe it is
a bandwidth problem on my end and I have no problems using iax with
gafachi. I opened a ticket with livevoip but no response yet. Would I be
better off using sip with them? Is there a server with better
response/bandwidth?
I admit that I am running a cvs head may 2004 prior to 1.x.x release.
Could that be the problem?
Regards,
John


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