oops, you got it.<br><br><div class="gmail_quote">On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield <span dir="ltr">&lt;<a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
In article &lt;CAJUJwthT=<a href="mailto:mpYxQ%2BOmT5HreXTL1iQVd0kbs%2BjHtQLVsqScaYqOA@mail.gmail.com">mpYxQ+OmT5HreXTL1iQVd0kbs+jHtQLVsqScaYqOA@mail.gmail.com</a>&gt;,<br>
<div class="im">Sammy Govind &lt;<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>&gt; wrote:<br>
&gt; Hi,<br>
&gt; Not sure why you didnt get it, when I did thta command for originator<br>
&gt; channel it showed me the CDR variables list which included<br>
<br>
</div>That&#39;s from &quot;show channel&quot;, not &quot;sip show channel&quot;.<br>
<br>
Cheers<br>
Tony<br>
<div class="im"><br>
&gt;   CDR Variables:<br>
&gt; level 1: dnid=XXXX<br>
&gt; level 1: clid=&quot;XXX&quot; &lt;XXXX&gt;<br>
&gt; level 1: src=XXXX<br>
&gt; level 1: dst=XXXX<br>
&gt; level 1: dcontext=SIP-incoming<br>
&gt; level 1: channel=XXXX<br>
&gt; level 1: dstchannel=XXXX<br>
&gt; level 1: lastapp=Dial<br>
&gt; level 1: lastdata=SIP/XXXX<br>
</div>&gt; *level 1: start=2011-12-14 09:15:54*<br>
<div><div class="h5">&gt; level 1: answer=2011-12-14 09:16:01<br>
&gt; level 1: duration=11<br>
&gt; level 1: billsec=4<br>
&gt; level 1: disposition=ANSWERED<br>
&gt; level 1: amaflags=DOCUMENTATION<br>
&gt; level 1: uniqueid=1323854154.856<br>
&gt; level 1: linkedid=1323854154.856<br>
&gt; level 1: sequence=1096<br>
&gt;<br>
&gt; Thats valid for an ongoing bridged call-initiator side only.<br>
&gt;<br>
&gt; Regards,<br>
&gt; Sammy<br>
&gt; On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar &lt;<a href="mailto:kamlesh_kmr@hotmail.com">kamlesh_kmr@hotmail.com</a>&gt;wrote:<br>
&gt;<br>
&gt; &gt;  Hello,<br>
&gt; &gt;<br>
&gt; &gt; &#39;sip show channel&#39; also does not give this info.<br>
&gt; &gt;<br>
&gt; &gt; sip show channel f600ed29f561d57<br>
&gt; &gt; localhost*CLI&gt;<br>
&gt; &gt;   * SIP CallI&gt;<br>
&gt; &gt;   Curr. trans. direction:  Incoming<br>
&gt; &gt;   Call-ID:                f600ed29f561d57f<br>
&gt; &gt;   Owner channel ID:       SIP/100-00000000<br>
&gt; &gt;   Our Codec Capability:   14<br>
&gt; &gt;   Non-Codec Capability (DTMF):   1<br>
&gt; &gt;   Their Codec Capability:   302<br>
&gt; &gt;   Joint Codec Capability:   14<br>
&gt; &gt;   Format:                 0x2 (gsm)<br>
&gt; &gt;   T.38 support            No<br>
&gt; &gt;   Video support           No<br>
&gt; &gt;   MaxCallBR:              384 kbps<br>
&gt; &gt;   Theoretical Address:    xxx.xxx.xxx.xxx:5060<br>
&gt; &gt;   Received Address:       xxx.xxx.xxx.xxx:5060<br>
&gt; &gt;   SIP Transfer mode:      open<br>
&gt; &gt;   NAT Support:            Always<br>
&gt; &gt;   Audio IP:               xxx.xxx.xxx.xxx (local)<br>
&gt; &gt;   Our Tag:                as2a60820a<br>
&gt; &gt;   Their Tag:              1b7d6a7d<br>
&gt; &gt;   SIP User agent:         eyeBeam release 3007n stamp 17816<br>
&gt; &gt;   Username:               10036<br>
&gt; &gt;   Peername:               10036<br>
&gt; &gt;   Original uri:           sip:100@xxx.xxx.xxx.xxx:5060<br>
&gt; &gt;   Caller-ID:              100<br>
&gt; &gt;   Need Destroy:           No<br>
&gt; &gt;   Last Message:           Rx: ACK<br>
&gt; &gt;   Promiscuous Redir:      No<br>
&gt; &gt;   Route:                  sip:100@xxx.xxx.xxx.xxx:5060<br>
&gt; &gt;   DTMF Mode:              rfc2833<br>
&gt; &gt;   SIP Options:            (none)<br>
&gt; &gt;   Session-Timer:          Inactive<br>
&gt; &gt;<br>
&gt; &gt; regards,<br>
&gt; &gt; Kamlesh<br>
&gt; &gt;<br>
</div></div>&gt; &gt;  ------------------------------<br>
<div><div class="h5">&gt; &gt; Date: Wed, 14 Dec 2011 12:43:14 +0500<br>
&gt; &gt; From: <a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a><br>
&gt; &gt; To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
&gt; &gt; Subject: Re: [asterisk-users] get start-time of all active calls<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt; Hi,<br>
&gt; &gt; I think you need to use the command &quot;sip show channel &lt;channel-id&gt;&quot;<br>
&gt; &gt; Regards,<br>
&gt; &gt; Sammy<br>
&gt; &gt;<br>
&gt; &gt; On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar &lt;<a href="mailto:kamlesh_kmr@hotmail.com">kamlesh_kmr@hotmail.com</a>&gt;wrote:<br>
&gt; &gt;<br>
&gt; &gt;  Hello,<br>
&gt; &gt;<br>
&gt; &gt; asterisk version 1.6.2.7<br>
&gt; &gt;<br>
&gt; &gt; I want to get the start time of all active calls from console, could you<br>
&gt; &gt; please let me know the best way to get it.<br>
&gt; &gt;<br>
&gt; &gt; thanks,<br>
&gt; &gt; Kamlesh<br>
&gt; &gt;<br>
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</div></div>&gt; -=-=-=-=-=-<br>
&gt; [Alternative: text/html]<br>
&gt; -=-=-=-=-=-<br>
&gt; -=-=-=-=-=-<br>
<div class="im">&gt;<br>
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<span class="HOEnZb"><font color="#888888"><br>
<br>
--<br>
Tony Mountifield<br>
Work: <a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a> - <a href="http://www.softins.co.uk" target="_blank">http://www.softins.co.uk</a><br>
Play: <a href="mailto:tony@mountifield.org">tony@mountifield.org</a> - <a href="http://tony.mountifield.org" target="_blank">http://tony.mountifield.org</a><br>
</font></span><div class="HOEnZb"><div class="h5"><br>
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</div></div></blockquote></div><br>