[asterisk-users] DTMF from SIP phone to FXS/FXO

[Andres Marquez]

Hello,
 
I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1
 
In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as long as I am pressing the key and I can hear the tone correctly). If I call from SIP phone to SIP phone everything works fine. When I set dtfmode to any other option, I can hear voice in both sides (SIP phone and analog line) but DTMF is not transfered.
 
I remember in the past (about 5 months ago) it worked for me with dtmfmode=inband (even dialing through an FXO line. I have been working on other things and just now came to realize that there is a problem there).
 
I have tried other options in sip.conf and rtp.conf (relaxdtmf, directrtpsetup, dtmftimeout) but none seem to make a difference.
 
Any help is greatly appreciated.
 
 
ANDRES
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