[asterisk-dev] capture the time before Answer of call in dialplan

Ludovic Gasc gmludo at gmail.com
Tue May 19 11:22:55 CDT 2015


You can follow AMI events with linkedid. you have a newchannel event for
that.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
On 18 May 2015 08:09, "Muteesa Fred" <fred.muteesa at infocom.ug> wrote:

> Thanks Ludovic,
>
> I have tried the AMI,
>
> But I only see the variables I need after the call has ended. Look below.
> I want to capture AnswerTime, and Starttime as soon as the call has been
> answered.
>
>
>
>
>
> Event: Cdr
>
> Privilege: cdr,all
>
> AccountCode:
>
> Source: asterisk
>
> Destination: 123456
>
> DestinationContext: from_talklite
>
> CallerID: "asterisk" <asterisk>
>
> Channel: SIP/sip_virtual-00000001
>
> DestinationChannel:
>
> LastApplication: Playback
>
> LastData: custom/0414-holdmusic6
>
> StartTime: 2015-05-18 08:48:59
>
> AnswerTime: 2015-05-18 08:49:04
>
> EndTime: 2015-05-18 08:49:25
>
> Duration: 26
>
> BillableSeconds: 21
>
> Disposition: ANSWERED
>
> AMAFlags: DOCUMENTATION
>
> UniqueID: 1431928139.2
>
> UserField:
>
>
>
>
>
> Thanks and regards,
>
> Fred
>
>
>
> *From:* asterisk-dev-bounces at lists.digium.com [mailto:
> asterisk-dev-bounces at lists.digium.com] *On Behalf Of *Ludovic Gasc
> *Sent:* Friday, May 15, 2015 1:37 PM
> *To:* Asterisk Developers Mailing List
> *Subject:* Re: [asterisk-dev] capture the time before Answer of call in
> dialplan
>
>
>
> Hi,
>
> I've no idea to do that with pure dialplan, however you can do that via a
> daemon that talks AMI protocol, I already do that. Via AGI/FastAGI it may
> be possible, I'm not really sure. ARI should also usable for this use case,
> never tested yet.
>
> For the AMI daemon, you can use the programming language you want,
> Personally, I use Python for that.
>
> BTW, I recommend you to use an external daemon instead of to write a C
> module because if your business logic crashes, it's less grave outside of
> Asterisk instead of inside, it should crash all of your telephony.
>
> However, apparently it's possible via dialplan, it's better to use that
> with an external daemon.
>
> Regards.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
> Hello Everyone,
>
> I apologise this might look like a user mailing list question but it's more
> of a developers question. I think this is the only forum that can help me.
>
> I have spent a week trying to achieve this. I realize I might have to write
> a C or PERL extension to achieve it but I don't know where to start. Or
> might need to rebuild the dial funtion.
> Objective:
> I want to dial a number in asterisk via a sip-trunk, and when the other
> party answers the call, I want to test if the call was answered before 5
> seconds, or after 5 seconds.
> If it was answered in less than 5 seconds I want to cancel the call,
> otherwise the call should continue.
>
> Your guidance will highly be appreciated.
>
> Regards,
> Fred Muteesa
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20150519/c3fb36af/attachment-0001.html>


More information about the asterisk-dev mailing list