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finally I got it with 'core show channel' <channel-id><BR>
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thanks for your support.<BR> <BR>
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Date: Wed, 14 Dec 2011 15:11:49 +0500<BR>From: govoiper@gmail.com<BR>To: asterisk-users@lists.digium.com<BR>Subject: Re: [asterisk-users] get start-time of all active calls<BR><BR>oops, you got it.<BR><BR>
<DIV class=ecxgmail_quote>On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield <SPAN dir=ltr><<A href="mailto:tony@softins.co.uk">tony@softins.co.uk</A>></SPAN> wrote:<BR>
<BLOCKQUOTE style="BORDER-LEFT: #ccc 1px solid; PADDING-LEFT: 1ex" class=ecxgmail_quote>In article <CAJUJwthT=<A href="mailto:mpYxQ%2BOmT5HreXTL1iQVd0kbs%2BjHtQLVsqScaYqOA@mail.gmail.com">mpYxQ+OmT5HreXTL1iQVd0kbs+jHtQLVsqScaYqOA@mail.gmail.com</A>>,<BR>
<DIV class=ecxim>Sammy Govind <<A href="mailto:govoiper@gmail.com">govoiper@gmail.com</A>> wrote:<BR>> Hi,<BR>> Not sure why you didnt get it, when I did thta command for originator<BR>> channel it showed me the CDR variables list which included<BR><BR></DIV>That's from "show channel", not "sip show channel".<BR><BR>Cheers<BR>Tony<BR>
<DIV class=ecxim><BR>> CDR Variables:<BR>> level 1: dnid=XXXX<BR>> level 1: clid="XXX" <XXXX><BR>> level 1: src=XXXX<BR>> level 1: dst=XXXX<BR>> level 1: dcontext=SIP-incoming<BR>> level 1: channel=XXXX<BR>> level 1: dstchannel=XXXX<BR>> level 1: lastapp=Dial<BR>> level 1: lastdata=SIP/XXXX<BR></DIV>> *level 1: start=2011-12-14 09:15:54*<BR>
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<DIV class=h5>> level 1: answer=2011-12-14 09:16:01<BR>> level 1: duration=11<BR>> level 1: billsec=4<BR>> level 1: disposition=ANSWERED<BR>> level 1: amaflags=DOCUMENTATION<BR>> level 1: uniqueid=1323854154.856<BR>> level 1: linkedid=1323854154.856<BR>> level 1: sequence=1096<BR>><BR>> Thats valid for an ongoing bridged call-initiator side only.<BR>><BR>> Regards,<BR>> Sammy<BR>> On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar <<A href="mailto:kamlesh_kmr@hotmail.com">kamlesh_kmr@hotmail.com</A>>wrote:<BR>><BR>> > 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></DIV>> > ------------------------------<BR>
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<DIV class=h5>> > Date: Wed, 14 Dec 2011 12:43:14 +0500<BR>> > From: <A href="mailto:govoiper@gmail.com">govoiper@gmail.com</A><BR>> > To: <A href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A><BR>> > Subject: Re: [asterisk-users] get start-time of all active calls<BR>> ><BR>> ><BR>> > Hi,<BR>> > I think you need to use the command "sip show channel <channel-id>"<BR>> > Regards,<BR>> > Sammy<BR>> ><BR>> > On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar <<A href="mailto:kamlesh_kmr@hotmail.com">kamlesh_kmr@hotmail.com</A>>wrote:<BR>> ><BR>> > 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<BR>> > please let me know the best way to get it.<BR>> ><BR>> > thanks,<BR>> > Kamlesh<BR>> ><BR>> > --<BR>> > _____________________________________________________________________<BR>> > -- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<BR>> > <A href="http://www.asterisk.org/hello" target=_blank>http://www.asterisk.org/hello</A><BR>> ><BR>> > asterisk-users mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>> ><BR>> ><BR>> ><BR>> > -- _____________________________________________________________________<BR>> > -- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> -- New<BR>> > to Asterisk? Join us for a live introductory webinar every Thurs:<BR>> > <A href="http://www.asterisk.org/hello" target=_blank>http://www.asterisk.org/hello</A> asterisk-users mailing list To UNSUBSCRIBE<BR>> > or update options visit:<BR>> > <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>> ><BR>> > --<BR>> > _____________________________________________________________________<BR>> > -- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR>> > New to Asterisk? Join us for a live introductory webinar every Thurs:<BR>> > <A href="http://www.asterisk.org/hello" target=_blank>http://www.asterisk.org/hello</A><BR>> ><BR>> > asterisk-users mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR>> ><BR>><BR></DIV></DIV>> -=-=-=-=-=-<BR>> [Alternative: text/html]<BR>> -=-=-=-=-=-<BR>> -=-=-=-=-=-<BR>
<DIV class=ecxim>><BR>> --<BR>> _____________________________________________________________________<BR>> -- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR>> New to Asterisk? Join us for a live introductory webinar every Thurs:<BR>> <A href="http://www.asterisk.org/hello" target=_blank>http://www.asterisk.org/hello</A><BR>><BR>> asterisk-users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></DIV>> -=-=-=-=-=-<BR><SPAN class=ecxHOEnZb><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>
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<DIV class=h5><BR>--<BR>_____________________________________________________________________<BR>-- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR>New to Asterisk? Join us for a live introductory webinar every Thurs:<BR> <A href="http://www.asterisk.org/hello" target=_blank>http://www.asterisk.org/hello</A><BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></DIV></DIV></BLOCKQUOTE></DIV><BR><BR>-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users</DIV> </div></body>
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