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