asterisk users - Jul 2007 - Asterisk with iax2 over satellite

Hi guys,
I'm in the process of setting up an Asterisk server over a satellite
connection to allow people on a remote island to place and receive calls
over the pstn.
What are the ideal settings I should use in iax.conf for the optimal
operation over satellite besides the normal options for the type=friend
peer?

Does anyone have this working? I an place calls as things are now, but there
is a lot of drop out in the audio.
I get a lot of receiving mini frame before full voice frame errors
especially the first 5 or 10 seconds of the call.

Thanks,
Tom

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007
5:44 PM

I have a client that is using SIP over satellite with G729, VAD and Jitter 
buffer. The calls are coming in great.

----- Original Message ----- 
From: "Tom Moore" <tmoore at issvsat.com>
To: <asterisk-users at lists.digium.com>
Cc: <jchomen at issvsat.com>
Sent: Sunday, July 15, 2007 5:50 PM
Subject: [asterisk-users] Asterisk with iax2 over satellite

> Hi guys,
> I'm in the process of setting up an Asterisk server over a satellite
> connection to allow people on a remote island to place and receive calls
> over the pstn.
> What are the ideal settings I should use in iax.conf for the optimal
> operation over satellite besides the normal options for the type=friend
> peer?
>
> Does anyone have this working? I an place calls as things are now, but 
> there
> is a lot of drop out in the audio.
> I get a lot of receiving mini frame before full voice frame errors
> especially the first 5 or 10 seconds of the call.
>
> Thanks,
> Tom
>
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007
> 5:44 PM
>
>
>
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On Sun, 2007-07-15 at 10:50 -0400, Tom Moore wrote:
> What are the ideal settings I should use in iax.conf for the optimal
> operation over satellite besides the normal options for the type=friend
> peer?
If you have a suitable timing source (ztdummy or a Digium card) and more
than 2 simultaneous calls between the two Asterisk boxes, I'd set
"trunk=yes" to reduce the IP bandwidth overhead.

This won't help with audio quality, but it will allow you to fit more
calls down a pipe with limited bandwidth.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.