[asterisk-users] get start-time of all active calls
Kamlesh Kumar
kamlesh_kmr at hotmail.com
Wed Dec 14 06:02:20 CST 2011
finally I got it with 'core show channel' <channel-id>
thanks for your support.
Date: Wed, 14 Dec 2011 15:11:49 +0500
From: govoiper at gmail.com
To: asterisk-users at 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 <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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