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

[Sammy Govind]

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_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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