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
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