Apologies. The freeswitch software is receiving incoming calls from a voip gateway. I'm using voiptalk in the UK. The DTMF method was efault which I believe is "info" but I've now set it explicitly to rfc2833 inband to see if that helps. Is there a way I can tell from the logs that this is the case and that my config changes have worked. Most of the phones are mobiles but some landlines as well. I've done a detailed analysis of the logs and the calls that don't work are missing what appear to be critical actions in the debug. Namely: 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 And then a little later in the call.... 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2734 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2120 sofia_glue_activate_rtp() Audio params changed for sofia/external/07540526194@194.145.190.143 from 194.145.190.143:11780 to 87.238.72.155:16968 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2127 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/07540526194@194.145.190.143] 77.86.49.249 port 21054 -> 87.238.72.155 port 16968 codec: 8 ms: 20 2009-05-15 09:47:45 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp() AUDIO RTP CHANGING DEST TO: [87.238.72.155:16968] 2009-05-15 09:47:45 [DEBUG] sofia.c:3241 sofia_handle_sip_i_state() Processing Reinvite 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526194@194.145.190.143 entering state [completed][200] 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526194@194.145.190.143 entering state [ready][200] These lines appear for calls that work and not when they don't. Hope that helps. Cheers Andy -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jason White Sent: 15 May 2009 08:47 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF not comming through on some calls Andy <andy@fabulous4.co.uk> wrote:
It's not that digits get dropped some calls semm to handle dtmf perfectly and others don't seem to get dtmf at all.
Can anyone shed any light opn this or suggest any solutions?
I can't help, but you could make it a lot easier for others to help you by including the necessary information with your question. For example, what DTMF method is configured in the SIP profiles - RFC2833 or Info, or are you using inband DTMF detection? What are the phones, and how are they connected to your FreeSWITCH system? What relevant information appears in your FreeSWITCH logs? For example, when debug-level logging is enabled, you should see log entries related to the DTMF detection. Check whether there are differences between the calls that work and those which don't. If it appears to be a bug, test whether you can reproduce it with the latest code taken from svn trunk. _______________________________________________ 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