<div dir="ltr"><div>Hello Metik,</div>
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<div class="gmail_quote">2009/12/6 Metik <span dir="ltr"><<a href="mailto:freeswitch-users-list@metik.com">freeswitch-users-list@metik.com</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">You previously stated that your Cisco gateway has some "bug" that<br>prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on<br>
the voip dial-peer that the call is using?<br><br></blockquote>
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<div>It is a PSTN dialpeer here, and it cannot be defined on it...</div>
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<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Unless you have configured the Cisco to support assymetric SDP or are<br>using a non-default "rtp payload-type nte" setting that does not agree<br>
to well with FS's (default) "rfc2833-pt" setting, you should not have to<br>use (SIP) INFO unless you want to.<br><br>I would recommend doing the following to ensure you are hitting the<br>correct dial-peer and it is configured for RFC 2833 ("rtp-nte"):<br>
<br>command: show dialplan number [number] | i (dtmf-relay|DTMF Relay)<br><br></blockquote>
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<div>Unfortunately this does not work on PSTN dial peers.</div>
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<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote"><br>Also, you can sift through "show sip-ua calls" for the call and ensure<br>that the value of "Negotiated Dtmf-relay" is "rtp-nte".<br>
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<div>This indeed shows that it has negotiated rtp-nte. Even when I do debug for CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them while it accepts them via INFO. As I said: I guess this is a bug.</div>
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<div>Since the gateway is on a remote site I hesitate on upgrading it until I hae the chance to go there.</div>
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<div> Thanks, __Yehavi:</div></div></div>