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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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 <[EMAIL PROTECTED]>
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
[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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