<div dir="ltr">Hi Nikolay,<div><br></div><div>On the advice of the FS team I was able to fix my issue by using spandsp tone detection instead of the original one. </div><div><br></div><div>Basically, in the dialplan use spandsp_start_dtmf instead of start_dtmf. </div><div><br></div><div>If you miss some inband DTMF but some are still coming through from that provider, you might want to give it a try to improve detection.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 14, 2017 at 7:46 AM, Oleg Stolyar <span dir="ltr"><<a href="mailto:olegstolyar@gmail.com" target="_blank">olegstolyar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">I am having a similar issue with a provider that only supports inband DTMF. About half the time DTMF does not reach FS (or does not register).<div><br></div><div>I was wondering if enabling jitter buffer on FS might help with this. Anyone has experience with it?</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Sun, Feb 12, 2017 at 12:46 AM, Nikolay Zaytsev <span dir="ltr"><<a href="mailto:nzaytsevc@gmail.com" target="_blank">nzaytsevc@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div dir="ltr"><span style="font-size:12.8px">Hello,</span><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">I have a problem with an inbound provider: sometimes dtmfs do not reach the freeswitch.</div><div style="font-size:12.8px">Provider told me that occurred because I am using G711 with rfc2833, and I should use only G711.</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Inbound call from the carrier to FreeSwitch:</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">SDP in the INVITE carrier --> FreeSwitch</div><div style="font-size:12.8px"><div>v=0</div><div>o=Sansay-VSXi 188 1 IN IP4 x.x.x.x</div><div>s=Session Controller</div><div>p=+X XXXXXXXX</div><div>c=IN IP4 x.x.x.x</div><div>t=0 0</div><div>m=audio 17774 RTP/AVP 18 0 8 101</div><div>a=rtpmap:18 G729/8000</div><div>a=fmtp:18 annexb=no</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>a=ptime:20</div></div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">SDP in the 200 OK message from FreeSwitch-->carrier:</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px"><div>v=0</div><div>o=FreeSWITCH 1486794789 1486794790 IN IP4 x.x.x.x</div><div>s=FreeSWITCH</div><div>c=IN IP4 x.x.x.x</div><div>t=0 0</div><div>m=audio 30088 RTP/AVP 0 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div></div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Carrier wants me to get rid of those session attributes in my 200 OK message:</div><div style="font-size:12.8px"><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div></div><div style="font-size:12.8px">and media to be like this</div><div style="font-size:12.8px">m=audio 30088 RTP/AVP 0.</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Could you please advise me how to do it?</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">I have already tried to set up</div><div style="font-size:12.8px">dtmf-type=none and liberal-dtmf=false in SIP profiles, dtmf_type=none in Dialplan. In addition I tried start_dtmf before application answer.</div><div style="font-size:12.8px">Also in the vars.xml I specified to use PCMU and PCMA only for both global_codec_prefs and outbound_codec_prefs.</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">My dialplan instruction for inbound call:</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px"><div> <extension name="inbound_call" continue="false"></div><div> <condition field="destination_number" expression="^79" ></div><div> <action application="answer"/></div><div> <action application="sleep" data="1000"/></div><div> <action application="socket" data="<a href="http://127.0.0.1:8800/" target="_blank">127.0.0.1:8800</a> async full"/></div><div> </condition></div><div> </extension></div></div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Im using several FreeSwitch boxes, one of them is 1.6.2 version another one is 1.6.14, but this problem occurres on both.</div><div style="font-size:12.8px"><br></div><div style="font-size:12.8px">Thank you for your time and best regards,</div><div style="font-size:12.8px">Nikolay Zaytsev</div></div>
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