[Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call

[Michael J. Tubby B.Sc (Hons) G8TIC]

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
    mailbox=2001 at default
    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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