> > no solution. I have a similar problem, when calling Freeswitch from my > > cell phone (via a SIP provider), sometimes DTMF is not recognized >> The important thing to note is that when using >> a SIP softphone (X-Lite) I have never had this problem, DTMF is
> So i guess that using latest version with a few changes in your config > should work unless there's any other issue related to your sip provider > ( PSTN / Media Gateway ), on this case you can get some captures of > sip/rtp traffic to check SDP and rtp Marks. I tried trunk and the values for the variables (all except rtp-timer-name=none are already default in trunk), but only two things are different: 1. When I press a key, the read app seem to always terminate, but not always the dtmf is captured in a variable. 2. The read app seems to ignore the variable name parameter: calling it with "1 1 104.wav choice_181152 10000 #" doesn't put the digit in variable_choice_181152, but to dmtf_digit. Why is that? > If it is coming from the sip provider as rfc 2833 dtmf, they are > probably doing inband detection and failing at it. If you look at an > rtp dump you can confirm this. If this is the case, there is nothing > you can do on the FreeSWITCH side and the provider will have to fix it. But the call goes through the same SIP provider even when using the soft phone and there it works fine. The difference might be that then it is SIP to SIP within the same provider.. How do I do the RTP dump? Also I should have mentioned that DTMF is not captured only DURING the file is being played. It is always captured correctly when I wait until the playback is finished. Does this sound familiar? I thought this would be somet obvious misconfiguration on my side. Jan _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org