finally I got it with 'core show channel' <channel-id> thanks for your support.
Date: Wed, 14 Dec 2011 15:11:49 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] get start-time of all active calls oops, you got it. On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield <t...@softins.co.uk> wrote: In article <CAJUJwthT=mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com>, Sammy Govind <govoi...@gmail.com> wrote: > Hi, > Not sure why you didnt get it, when I did thta command for originator > channel it showed me the CDR variables list which included That's from "show channel", not "sip show channel". Cheers Tony > CDR Variables: > level 1: dnid=XXXX > level 1: clid="XXX" <XXXX> > level 1: src=XXXX > level 1: dst=XXXX > level 1: dcontext=SIP-incoming > level 1: channel=XXXX > level 1: dstchannel=XXXX > level 1: lastapp=Dial > level 1: lastdata=SIP/XXXX > *level 1: start=2011-12-14 09:15:54* > level 1: answer=2011-12-14 09:16:01 > level 1: duration=11 > level 1: billsec=4 > level 1: disposition=ANSWERED > level 1: amaflags=DOCUMENTATION > level 1: uniqueid=1323854154.856 > level 1: linkedid=1323854154.856 > level 1: sequence=1096 > > Thats valid for an ongoing bridged call-initiator side only. > > Regards, > Sammy > On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar <kamlesh_...@hotmail.com>wrote: > > > Hello, > > > > 'sip show channel' also does not give this info. > > > > sip show channel f600ed29f561d57 > > localhost*CLI> > > * SIP CallI> > > Curr. trans. direction: Incoming > > Call-ID: f600ed29f561d57f > > Owner channel ID: SIP/100-00000000 > > Our Codec Capability: 14 > > Non-Codec Capability (DTMF): 1 > > Their Codec Capability: 302 > > Joint Codec Capability: 14 > > Format: 0x2 (gsm) > > T.38 support No > > Video support No > > MaxCallBR: 384 kbps > > Theoretical Address: xxx.xxx.xxx.xxx:5060 > > Received Address: xxx.xxx.xxx.xxx:5060 > > SIP Transfer mode: open > > NAT Support: Always > > Audio IP: xxx.xxx.xxx.xxx (local) > > Our Tag: as2a60820a > > Their Tag: 1b7d6a7d > > SIP User agent: eyeBeam release 3007n stamp 17816 > > Username: 10036 > > Peername: 10036 > > Original uri: sip:1...@xxx.xxx.xxx.xxx:5060 > > Caller-ID: 100 > > Need Destroy: No > > Last Message: Rx: ACK > > Promiscuous Redir: No > > Route: sip:1...@xxx.xxx.xxx.xxx:5060 > > DTMF Mode: rfc2833 > > SIP Options: (none) > > Session-Timer: Inactive > > > > regards, > > Kamlesh > > > > ------------------------------ > > Date: Wed, 14 Dec 2011 12:43:14 +0500 > > From: govoi...@gmail.com > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] get start-time of all active calls > > > > > > Hi, > > I think you need to use the command "sip show channel <channel-id>" > > Regards, > > Sammy > > > > On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar > > <kamlesh_...@hotmail.com>wrote: > > > > Hello, > > > > asterisk version 1.6.2.7 > > > > 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. > > > > thanks, > > Kamlesh > > > > -- > > _____________________________________________________________________ > > -- 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 > > > > > > > > -- _____________________________________________________________________ > > -- 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 > > > > -- > > _____________________________________________________________________ > > -- 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 > > > > -=-=-=-=-=- > [Alternative: text/html] > -=-=-=-=-=- > -=-=-=-=-=- > > -- > _____________________________________________________________________ > -- 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 > -=-=-=-=-=- -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
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