[asterisk-users] get start-time of all active calls
Sammy Govind
govoiper at gmail.com
Wed Dec 14 04:11:49 CST 2011
oops, you got it.
On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield <tony at softins.co.uk>wrote:
> 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
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
>
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