Hi Nikolay, On the advice of the FS team I was able to fix my issue by using spandsp tone detection instead of the original one. Basically, in the dialplan use spandsp_start_dtmf instead of start_dtmf. 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. On Tue, Feb 14, 2017 at 7:46 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
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).
I was wondering if enabling jitter buffer on FS might help with this. Anyone has experience with it?
On Sun, Feb 12, 2017 at 12:46 AM, Nikolay Zaytsev <nzaytsevc@gmail.com> wrote:
Hello,
I have a problem with an inbound provider: sometimes dtmfs do not reach the freeswitch. Provider told me that occurred because I am using G711 with rfc2833, and I should use only G711.
Inbound call from the carrier to FreeSwitch:
SDP in the INVITE carrier --> FreeSwitch v=0 o=Sansay-VSXi 188 1 IN IP4 x.x.x.x s=Session Controller p=+X XXXXXXXX c=IN IP4 x.x.x.x t=0 0 m=audio 17774 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
SDP in the 200 OK message from FreeSwitch-->carrier:
v=0 o=FreeSWITCH 1486794789 1486794790 IN IP4 x.x.x.x s=FreeSWITCH c=IN IP4 x.x.x.x t=0 0 m=audio 30088 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Carrier wants me to get rid of those session attributes in my 200 OK message: a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 and media to be like this m=audio 30088 RTP/AVP 0.
Could you please advise me how to do it?
I have already tried to set up dtmf-type=none and liberal-dtmf=false in SIP profiles, dtmf_type=none in Dialplan. In addition I tried start_dtmf before application answer. Also in the vars.xml I specified to use PCMU and PCMA only for both global_codec_prefs and outbound_codec_prefs.
My dialplan instruction for inbound call:
<extension name="inbound_call" continue="false"> <condition field="destination_number" expression="^79" > <action application="answer"/> <action application="sleep" data="1000"/> <action application="socket" data="127.0.0.1:8800 async full"/> </condition> </extension>
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.
Thank you for your time and best regards, Nikolay Zaytsev
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