I had some issues with some previous versions of FS , in trunk looks that is fixed. ( Notice current svn revision is 10609 ) in sip profiles i have : ... <param name="rfc2833-pt" value="101"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="none"/> <param name="inbound-codec-negotiation" value="greedy"/> ... As codecs g711 ULAW (PCMU): in vars.xml.conf : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM"/> 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. El vie, 05-12-2008 a las 12:08 +0100, Jan Kubr escribió:
Hi, recently someone was mentioning an issue with DTMF here, but there was no solution. I have a similar problem, when calling Freeswitch from my cell phone (via a SIP provider), sometimes DTMF is not recognized (read app doesn't terminate). I could not find any regularity in this, sometimes it is recognized just fine, sometimes I had to wait for the file to be played etc. The important thing to note is that when using a SIP softphone (X-Lite) I have never had this problem, DTMF is recognized perfectly. So it's probably related to GSM or something.
I was wondering whether anyone experienced the same and whether there is something I can do about it. There are a few DTMF-related variables in the config files (dtmf-duration, pass-rfc2833, rfc2833-pt, dtmf-type, default_dtmf_duration, max_dtmf_duration) and I played with them a bit, but I don't really know what I'm doing.. Couldn't find any docs, either. Any ideas would be appreciated.
Jan Kubr
Cheers, -- Angel Carpintero ack ( at ) telefonica ( dot ) net Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 6EF1 B90D