asterisk users - Jun 2005 - Cisco 7960 (SIP) with Asterisk: how to get # to work during a call

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@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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# and * are mapped later in the SIP(Default/MAC).cnf it has a section
in the manual if you want to see why.

On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC <mike.tubby@thorcom.co.uk>
wrote:
>  
> 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@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 
>   
>   
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 

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