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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100426/ba654126/attachment.htm
