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<div>I am running 1.4.0-beta2</div>
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<div>Date: Tue, 31 Oct 2006 10:57:06 -0600 (CST)<br>From: Anthony LaMantia <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:alamantia@digium.com">alamantia@digium.com</a>><br>Subject: Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>>
<br>Message-ID:<br> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:18975956.15221162313826243.JavaMail.root@jupiler.digium.com">18975956.15221162313826243.JavaMail.root@jupiler.digium.com
</a>><br>Content-Type: text/plain; charset=utf-8<br><br>Which asterisk release are you running chan_skinny under?<br><br>----- Original Message -----<br>From: Will Roy <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:willroyvi@gmail.com">
willroyvi@gmail.com</a>><br>To: <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central
<br>Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone<br><br><br>Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny.<br><br>The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call.
<br><br>When I debug Skinny on the console after the call has connected I see the following messag:<br><br>Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7]<br><br>What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :)
<br><br>regards<br>Wil<br><br> </div>