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
--
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