Hey Mark<div><br></div><div>A few thoughts:</div><div><br></div><div>1. If you are using start_dtmf on the FS box then you are making it do media processing which is fine but this is a change from what the OpenSER / RTPproxy were doing for you. If you want it to be a drop-in replacement you might want to look at some of the following:</div>
<div><br></div><div>proxy_media</div><div>inbound-proxy-media</div><div><a href="http://wiki.freeswitch.org/wiki/Proxy_Media">http://wiki.freeswitch.org/wiki/Proxy_Media</a></div><div>disable-transcoding</div><div>codec_string / ep_codec_string</div>
<div>inherit_codec</div><div><a href="http://wiki.freeswitch.org/wiki/Codec_negotiation">http://wiki.freeswitch.org/wiki/Codec_negotiation</a></div><div><br></div><div>2. It&#39;s dtmf-type in the profile or dtmf_type in the dialplan. Not dtmf_mode as you mention in an earlier message.</div>
<div><br></div><div>3. Your post mentioning PT 18 and the annexb parameter suggests you are using G729, which does not support inband DTMF reliably, if at all. </div><div><br></div><div>4. You will need to put the sip_append_audio_sdp before the pre_answer (which you should get rid off as suggested already).</div>
<div><br></div><div>5. if you do want FS to convert from inband to rfc2833, there was a previous post suggesting spandsp_start_dtmf may perform better than start_dtmf (although I can&#39;t verify this).</div><div><br></div>
<div>Regards,</div><div>Richard</div><div><br></div><div><div class="gmail_quote">On 18 January 2013 16:12, Steven Ayre <span dir="ltr">&lt;<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>Did you try removing the rfc2833-pt param and setting &lt;param name=&quot;dtmf-type&quot; value=&quot;none&quot;/&gt; on the outgoing sofia profile to the PBX?</div>
<div><br></div><div>Inband tones will always be passed over, FS would actively need to detect the tone and remove it from the media to avoid doing so. So the PBX is not looking inband for it. Since there&#39;s no way in SIP to notify that inband tones are in use the PBX is probably checking for out-of-band DTMF and not checking inband at all if out-of-band has been negotiated - so what you probably need to is disable out of band media entirely.</div>


<div><br></div><div>As far as comparing FS behaviour to OpenSER goes I would avoid it. They&#39;re very different products. Understand the difference between a SIP proxy (OpenSER) and a B2BUA (FreeSWITCH). OpenSER only forwards SIP messages with media terminating on the endpoints. FS negotiates 2 separate calls including media and terminates both calls on FS and passes media between them. As such OpenSER doesn&#39;t do any media negotiation and FS does which will make their behaviour rather different.</div>


<div><br></div><div>-Steve</div><div><br></div><div><br></div><div><br></div><br><br><div class="gmail_quote"><div class="im">On 18 January 2013 15:20, <a href="mailto:support@ecn.net.au" target="_blank">support@ecn.net.au</a> <span dir="ltr">&lt;<a href="mailto:support@ecn.net.au" target="_blank">support@ecn.net.au</a>&gt;</span> wrote:<br>


</div><div><div class="h5"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Yes, we read that and tried to set that when attempting multiple configurations (attempting to pass inband dtmf).<br>

<br>
Currently we can&#39;t get Freeswitch in an SBC role to pass the inband in the media stream at all (see earlier email) - also having odd issues with start_dtmf when accessing Asterisk IVR&#39;s (again very odd).  the only difference we can see (between FS and our legacy openser) is the Invite pushed on the bleg on Openser included the fmtp:18 annexb=no directive (as per the telco&#39;s SDP to us); however on FS it doesn&#39;t forward these (and we can&#39;t seem to force FS to add these attributes to the SDP on the bleg).<br>



<br>
Again the start_dtmf issues exists only in 1.2 and 1.3 FS , in the 1.0.6 the issue is not present (is this a bug possibly?)<br>
<br>
Any ideas?<br>
<br>
Kind Regards,<br>
<br>
Mark<br>
<br>
________________________________________<br>
From: Paul Cupis [<a href="mailto:paul@cupis.co.uk" target="_blank">paul@cupis.co.uk</a>]<br>
Sent: Friday, 18 January 2013 8:26 PM<br>
<div>To: FreeSWITCH Users Help<br>
Subject: Re: [Freeswitch-users] SBC In-band DTMF<br>
<br>
</div><div><div>On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote:<br>
&gt;      &lt;param name=&quot;rfc2833-pt&quot; value=&quot;0&quot;/&gt;<br>
&gt;<br>
&gt;    What&#39;re you hoping to do here? This sets the payload type value. 0 is a<br>
&gt;    valid pt. This won&#39;t disable it, it&#39;ll instead use the value 0. That<br>
&gt;    happens to be the one for G711, so if that codec is enabled you&#39;re just<br>
&gt;    going to get a conflict.<br>
<br>
If the rfc2833-pt is set to a value &lt;96 it is outside the allowed range<br>
for dynamic payload negotiation and FS will not offer RFC2833 support<br>
in the SDP.<br>
<br>
ref: sofia_glue.c<br>
<br>
Regards,<br>
<br>
<br>
<br>
<br>
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