you can see in the trace that the exchange is using alaw.
So the complaint about trans coding is because it's going from PCMU on leg a
to PCMA when talking to exchange.


please revert any changes, upgrade to current SVN trunk and try configuring
exchange to use PCMA
or confiigure FS to only use PCMA by editing vars.xml and making the
codec_string be PMCA instead of PCMU



On Sat, Sep 20, 2008 at 10:08 PM, Matt Darnell <[EMAIL PROTECTED]>wrote:

> > In the meantime, Matt, try applying this non-standard change (it worked
> for
> > me).
> > Change src/switch_rtp.c (lines 1197-1198) from:
> > rtp_session->dtmf_data.out_digit_sofar = samples;
> > rtp_session->dtmf_data.out_digit_sub_sofar = samples;
> > to:
> > rtp_session->dtmf_data.out_digit_sofar = 0;
> > rtp_session->dtmf_data.out_digit_sub_sofar = 0;
> >
> > Let me know if that fixed your DTMF problem with your UM.
>
> I made the change, did configure, make clean, make, & make install.  I
> see this in the log now:
> 2008-09-20 16:54:32 [DEBUG] switch_rtp.c:1201 do_2833() Send start
> packet for [3] ts=12560 dur=0/0/2000 seq=15883
> Before the duration would start at 160/160/2000.  Is that what I should
> see?
>
> It still doesn't solve the issue.  It doesn't act like I would expect.
>  The server either 'hears' digits on a call or it doesn't; I would
> expect it to miss digits every now and then, not either completely
> pass or completely fail.
>
> What type of DTMF failure were you experiencing?
>
> I wonder if it is not a Freeswitch issue but an Exchange issue, I will
> get a SIP/TCP softphone and try to connect directly to the Exchange
> server.  I think SJPhone does SIP/TCP.
>
> -Matt
>
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-- 
Anthony Minessale II

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