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

[Mahilal Silva]

Mike,
Were you able to get this working?
Even after with a entry in the dialplan.xml does not work for me.

Thanks,
Ken


On 6/20/05, Michael J. Tubby B.Sc (Hons) G8TIC <mike.tubby at thorcom.co.uk>
wrote:
>
> 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
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
> --
> <sig>
> Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
> WWW: http://lathama.com
> Email: lathama at lathama.com - lathama at yahoo.com - lathama at gmail.com
> If any of the above are down we have bigger problems than my email!
> </sig>
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