<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">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 <br />
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Thanks<br />
Bryant<br />
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<b>From</b>: "Zeeshan Zakaria" <zishanov@gmail.com><br />
<b>Sent</b>: Wednesday, October 13, 2010 10:41 AM<br />
<b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-users] DMTF Mode</span><br />
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<p>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?</p>
<p>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. </p>
<p>Zeeshan A Zakaria</p>
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<a href="http://www.ilovetovoip.com">www.ilovetovoip.com</a></p>
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<blockquote type="cite">On 2010-10-13 10:19 AM, "Dan Journo" <<a href="mailto:dan@keshercommunications.com">dan@keshercommunications.com</a>> wrote:<br />
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<p><span style="color: #500050;">> It depends upon whether you are receiving DTMF or sending, and whether you are using a VoIP protoc...</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">Sorry about the lack of info.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">It's a simple SIP only setup. A handful of sip phones, an asterisk server, and a sip provider.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">The DTMF signals from the sip phones are received by Asterisk because they can access features like *1.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">The DTMF signal from the called party are received by Asterisk because they can also access features like *1.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">But, the DTMF tones are not passed through from the Sip Phone to the Called Party.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">The same happens regardless of whether its an incoming or outgoing call.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">That means, if any of my users try to call a company with a menu system, they can't select any options.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">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.</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">Any ideas?</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">Thanks</span></p>
<p><span style="color: #1f497d; font-size: 11pt;">Dan</span></p>
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