[asterisk-users] Full Duplex

[Ken Williams]

We are currently on Asterisk 1.4.19 for anyone reading along.

Great idea on the phone to phone, I'll try that later and go from there.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gordon
Henderson
Sent: Monday, November 17, 2008 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Full Duplex

On Mon, 17 Nov 2008, Ken Williams wrote:

> We've had an issue since we went live nearly two years ago on Asterisk
> where people complain about not being able to talk while someone else
is
> talking.  I had assumed for a very long time this was because of the
> phones we went live with (Grandstream GXP-2000's) and for the longest
> time I believed this was a speakerphone problem only.
>
> Last week during budgets, a request to buy new phones was put in to
fix 
> this problem.  It was then that I finally researched and found that
our 
> phones do in fact support full-duplex on the speakerphone and handset.
>
> So, I'm looking for where I may have missed, perhaps an option on the
> Dial() command?  Something in the phone config?

I've used Grandstreams for a couple of years now. Generally gotten on OK

with them, although there are people here who've been badly "bitten" by 
them..

One thing to check is the hardware revision and software revision - the 
later software is much better - I'm using Program-- 1.1.6.16
Bootloader-- 
1.1.6.5, but new software isn't as reliable on old phones as it could be
- 
in particular my very first GXP2000 with that software experiences the 
occasional glitch/speech lock-up for a few seconds every now and then 
during a call.

The speaker-phones do work OK though. Actually one of the better ones
I've 
tried!

I've not put anything special in the phone config - dial flags I use are

WTon

> We use SIP for phone to phone conversations, IAX for site to site
> conversations and Zaptel for PSTN lines.  I've been told it happens
> regardless of protocol, so I assume it's one of the above options.

I use the same in various locations - everything working fine. Asterisk 
1.2.30 if that makes any different for you...

Have you tried to make a direct IP call from one Grandstream to another?

Try that, and bypass asterisk altogether and see what happens... (best
to 
be in different rooms though!)

Gordon


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