My asterisk equipped with a BN8S0 is configured as an gateway between my
PBX and the PSTN. Now when I establish a call from a PBX extension to
the PSTN, DTMF tones from the extensions are not transmitted to the
PSTN. So I can not dial passwords closed by # into conference bridges.
The # is interpreted as hold. If I have established call between SIP
Phones or SIP phones and an PBX extension the DTMF tones are transmitted.
The connection to the PBX is done by four NT ports and the connection to
the PSTN is done by four TE ports.
They are configured in the misdn.conf in the following way:
[general]
debug=0
bridging=yes
append_digits2exten=yes
[default]
context=default
language=de
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
dialplan=0
echocancelwhenbridged=no
[TEports]
context=telecom
ports=1ptp,2ptp,7ptp,8ptp
msns=*
[NTports]
context=integral-nooverlap
ports=3ptp,4ptp,5ptp,6ptp
dialplan=4
How can I disable the DTMF detection in the asterisk.
Jörg Simon
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