[asterisk-users] innomedia ATA's

[Bryant Zimmerman]

We are testing the innomedia ATA's to possibly replace our current line up 
of ATA's that we are using. Has anyone used their product? What is their 
track record on stability, voice quality, DTMF talkoff, T.38 

Thanks
Bryant

----------------------------------------
 From: "Zeeshan Zakaria" <zishanov at gmail.com>
Sent: Wednesday, October 13, 2010 10:41 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] DMTF Mode

I would suggest first to make sure that asterisk is receiving DTMF fine 
from your IP devices/phones. Do you have a test IVR where you can dial and 
press digits and verify that asterisk is responding? 

Once you are sure that asterisk is receiving DTMF fine, then you should ask 
your provider what DTMF setting you should have on your system. Usually all 
of them support RFC2833, so if in your sip.conf where you have defined the 
trunk, dtmfmode is set to rfc2833, your provider should receive it and pass 
on to the next carrier or trunk.  

Zeeshan A Zakaria 

--
www.ilovetovoip.com 

  On 2010-10-13 10:19 AM, "Dan Journo" <dan at keshercommunications.com> 
wrote:

> It depends upon whether you are receiving DTMF or sending, and whether 
you are using a VoIP protoc... 

Sorry about the lack of info. 

It's a simple SIP only setup. A handful of sip phones, an asterisk server, 
and a sip provider. 

The DTMF signals from the sip phones are received by Asterisk because they 
can access features like *1. 

The DTMF signal from the called party are received by Asterisk because they 
can also access features like *1. 

But, the DTMF tones are not passed through from the Sip Phone to the Called 
Party. 

The same happens regardless of whether its an incoming or outgoing call. 

That means, if any of my users try to call a company with a menu system, 
they can't select any options. 

How can I tell if Asterisk is sending the tones through to the provider? I 
need to find out whether its something I'm doing, or something the provider 
is doing. 

Any ideas? 

Thanks 

Dan   
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

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/20101013/03e27402/attachment.htm