<div dir="ltr">Actually I don't think I need to worry about this, the DTMF in question always arrives from an Asterisk box which supports SIP INFO via dtmfmode=info<div><br></div><div>Thanks for reading!</div><div>Mark.</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, 18 Mar 2019 at 16:18, Mark Farmer <<a href="mailto:farmorg@gmail.com">farmorg@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div dir="ltr">Thanks Bogdan. I was beginning to fear as much but I do need a solution for this. I will try to explain the problem in more detail.<div><br></div><div>Our 3rd party is a secure payment gateway and only receives audio following a SIP RE-INVITE. This is to re-route the audio so that DTMF tones can be removed. At this point the DTMF is required to be transmitted to the 3rd party as SIP INFO messages. Once the payment process is complete, there is another SIP RE-INVITE to switch the audio back to it's original path.<br></div><div><br></div><div>What I have is OpenSIPS & RTPProxy combination but this is a PoC platform so I can change things. Is there a method to have an RTP proxy instruct OpenSIPS to generate SIP INFO messages and send them to the 3rd party? It doesn't matter if we end up with both RFC2833 and SIP INFO - the 3rd party will just act on the INFO messages.</div><div><br></div><div>TIA</div><div>Mark.</div><div><br></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, 18 Mar 2019 at 15:37, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<div bgcolor="#FFFFFF">
<tt>Hi Mark,<br>
<br>
You cannot do this with OpenSIPS, mainly as OpenSIPS has no access
to RTP in order to "see" the RFC2833 DTMFs.<br>
<br>
Regards,<br>
</tt>
<pre class="gmail-m_-4494738480563567478gmail-m_-6881269663754811909moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="gmail-m_-4494738480563567478gmail-m_-6881269663754811909moz-txt-link-freetext" href="https://www.opensips-solutions.com" target="_blank">https://www.opensips-solutions.com</a>
OpenSIPS Summit 2019
<a class="gmail-m_-4494738480563567478gmail-m_-6881269663754811909moz-txt-link-freetext" href="https://www.opensips.org/events/Summit-2019Amsterdam/" target="_blank">https://www.opensips.org/events/Summit-2019Amsterdam/</a>
</pre>
<div class="gmail-m_-4494738480563567478gmail-m_-6881269663754811909moz-cite-prefix">On 03/18/2019 04:24 PM, Mark Farmer
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hi everyone
<div><br>
</div>
<div>I have an OpenSIPS server with trunks to a provider and to
a 3rd party and routing calls between the two. The provider
sends DTMF as RFC2833 but the 3rd party requires SIP INFO.
Somehow I need to translate from RFC2833 to SIP INFO - is this
possible?</div>
<div><br>
</div>
<div>Many thanks</div>
<div>Mark.</div>
<div>
<div><br>
</div>
-- <br>
<div dir="ltr" class="gmail-m_-4494738480563567478gmail-m_-6881269663754811909gmail_signature">Mark Farmer<br>
<a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a></div>
</div>
</div>
<br>
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<br>
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</pre>
</blockquote>
<br>
</div>
</blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail-m_-4494738480563567478gmail_signature">Mark Farmer<br><a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a></div>
</blockquote></div><br clear="all"><div><br></div>-- <br><div dir="ltr" class="gmail_signature">Mark Farmer<br><a href="mailto:farmorg@gmail.com" target="_blank">farmorg@gmail.com</a></div>