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" <[email protected]> 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:* [email protected] [mailto: > [email protected]] *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 >
-- _____________________________________________________________________ -- 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
