[asterisk-users] get start-time of all active calls

[Tony Mountifield]

In article <CAJUJwthT=mpYxQ+OmT5HreXTL1iQVd0kbs+jHtQLVsqScaYqOA at mail.gmail.com>,
Sammy Govind <govoiper at 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_kmr at 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:100 at xxx.xxx.xxx.xxx:5060
> >   Caller-ID:              100
> >   Need Destroy:           No
> >   Last Message:           Rx: ACK
> >   Promiscuous Redir:      No
> >   Route:                  sip:100 at 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: govoiper at gmail.com
> > To: asterisk-users at 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_kmr at 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
> >
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-- 
Tony Mountifield
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