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Gents,
I've built an Asterisk system to replace our PBX at
work and have Cisco
7960 phones (SIP 7.4) running with Asterisk
1.0.7.
How to I get Asterisk to recognise the '#' being
pressed during a call?
In sip.conf I have entries likle this:
[2001]
type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001> host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I
have:
# Inband DTMF Settings
(0-disable, 1-enable (default))
dtmf_inband: 1 # Out of band DTMF Settings
(none-disable, avt-avt enable (default), avt_always - always avt
)
dtmf_outofband: avt # DTMF dB Level Settings (1-6dB
down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3 DTMF works for voicemail and for remote services
over both analogue Zap
channels and digital (ISDN) channels.
Asterisk doesn't appear to be 'monitoring' the
audio so I can't get to Asterisk
features like Asterisk's transfer, parked calls and
one-tuch-record...
Am I missing something?
Mike
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