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

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

Andrew,

I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?

When you say "mapped", dou mean that it needs an explicit entry in the 
dialplan.xml like:

            <TEMPLATE MATCH="#"         Timeout="0" User="Phone"/> <!--  
Explicit # for Asterisk -->

Mike

----- Original Message ----- 
From: "Andrew Latham" <lathama at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, June 16, 2005 2:53 PM
Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # 
towork during a call


# 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 at 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 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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