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  <title>SIP phone calls through C60</title>
  <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_thread?p_l_id=&amp;threadId=6310098" />
  <subtitle>SIP phone calls through C60</subtitle>
  <id>http://developer.cisco.com/c/message_boards/find_thread?p_l_id=&amp;threadId=6310098</id>
  <updated>2013-06-18T07:12:15Z</updated>
  <dc:date>2013-06-18T07:12:15Z</dc:date>
  <entry>
    <title>RE: SIP phone calls through C60</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6318976" />
    <author>
      <name>David Bruun-Lie</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6318976</id>
    <updated>2012-08-21T08:14:08Z</updated>
    <published>2012-08-21T08:14:08Z</published>
    <summary type="html">Hi Donald!
hm, no reason why this should not work. It is of course designed for accepting SIP audio calls too, not just video :)
What kind of audio system is on the far-end side? Have you tried using TC consoles audio routing to see that audio is being routed from the far-end site to the loudspeaker output? (there should be a line between "Remote input x" to "Loudspeaker")
If this shows that everything is OK then we need to have more details of what kind of remote system is sending the audio before we can dig further into debugging...

Cheers,
David</summary>
    <dc:creator>David Bruun-Lie</dc:creator>
    <dc:date>2012-08-21T08:14:08Z</dc:date>
  </entry>
  <entry>
    <title>SIP phone calls through C60</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6310097" />
    <author>
      <name>Donald Depperschmidt</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6310097</id>
    <updated>2012-08-20T16:52:09Z</updated>
    <published>2012-08-20T16:52:09Z</published>
    <summary type="html">A customer is trying to recieve SIP phone calls to the C60.  
 
[color=#1f497d]The problem is when a phone calls into the codec via SIP, we cannot hear audio from the room system on the phone.  But if the room system initiates the call out to a phone we can hear audio both ways.  When we open the audio monitor in the codec, we can see our audio levels in the codec, so we are confident that the audio is getting into the codec.  But, we can hear DTMF tones that are sent from the codec to the phone, but not voice.[/color]
[color=#1f497d][/color]
[color=#1f497d]Any suggestions on why this may not be working?  Is it designed to accept calls?[/color]</summary>
    <dc:creator>Donald Depperschmidt</dc:creator>
    <dc:date>2012-08-20T16:52:09Z</dc:date>
  </entry>
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