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
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 >
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