<div>Mike,</div>
<div>Were you able to get this working?</div>
<div>Even after with a entry in the dialplan.xml does not work for me.</div>
<div> </div>
<div>Thanks,</div>
<div>Ken<br><br> </div>
<div><span class="gmail_quote">On 6/20/05, <b class="gmail_sendername">Michael J. Tubby B.Sc (Hons) G8TIC</b> <<a href="mailto:mike.tubby@thorcom.co.uk">mike.tubby@thorcom.co.uk</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Andrew,<br><br>I presume you mean in the Cisco 7940/7960 SIP Phone Administrator's Guide?<br><br>When you say "mapped", dou mean that it needs an explicit entry in the
<br>dialplan.xml like:<br><br> <TEMPLATE MATCH="#" Timeout="0" User="Phone"/> <!--<br>Explicit # for Asterisk --><br><br>Mike<br><br>----- Original Message -----
<br>From: "Andrew Latham" <<a href="mailto:lathama@gmail.com">lathama@gmail.com</a>><br>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br><<a href="mailto:asterisk-users@lists.digium.com">
asterisk-users@lists.digium.com</a>><br>Sent: Thursday, June 16, 2005 2:53 PM<br>Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get #<br>towork during a call<br><br><br># and * are mapped later in the SIP(Default/MAC).cnf it has a section
<br>in the manual if you want to see why.<br><br>On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC <<a href="mailto:mike.tubby@thorcom.co.uk">mike.tubby@thorcom.co.uk</a>><br>wrote:<br>><br>> Gents,<br>><br>> I've built an Asterisk system to replace our PBX at work and have Cisco
<br>> 7960 phones (SIP 7.4) running with Asterisk 1.0.7.<br>><br>> How to I get Asterisk to recognise the '#' being pressed during a call?<br>><br>> In sip.conf I have entries likle this:<br>><br>> [2001]
<br>> type=friend<br>> context=local-phone<br>> auth=md5<br>> username=2001<br>> secret=xyzzy<br>> callerid=Jack Tubby <2001><br>> host=dynamic<br>> nat=no<br>
> canreinvite=no<br>> dtmfmode=rfc2833<br>> incominglimit=2<br>> mailbox=2001@default<br>> disallow=all<br>> allow=alaw<br>> allow=ulaw<br>> callgroup=2<br>> pickupgroup=2
<br>><br>> and in the SIPDefault.cnf for the phones I have:<br>><br>> # Inband DTMF Settings (0-disable, 1-enable (default))<br>> dtmf_inband: 1<br>><br>> # Out of band DTMF Settings (none-disable, avt-avt enable (default),
<br>> avt_always - always avt )<br>> dtmf_outofband: avt<br>><br>> # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default),<br>> 4-3db up, 5-6dB up)<br>> dtmf_db_level: 3<br>>
<br>> DTMF works for voicemail and for remote services over both analogue Zap<br>> channels and digital (ISDN) channels.<br>><br>> Asterisk doesn't appear to be 'monitoring' the audio so I can't get to<br>> Asterisk
<br>> features like Asterisk's transfer, parked calls and one-tuch-record...<br>><br>> Am I missing something?<br>><br>><br>> Mike<br>><br>><br>> _______________________________________________<br>
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--<br><sig><br>Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)<br>WWW: <a href="http://lathama.com">http://lathama.com</a><br>Email: <a href="mailto:lathama@lathama.com">lathama@lathama.com</a> - <a href="mailto:lathama@yahoo.com">
lathama@yahoo.com</a> - <a href="mailto:lathama@gmail.com">lathama@gmail.com</a><br>If any of the above are down we have bigger problems than my email!<br></sig><br>_______________________________________________<br>
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