[asterisk-app-dev] How do I bind a outbound call?

[Oscar]

I'm sorry for my bed english ...but in wiki there is:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-redirect

so I thought that I could do a originate call by ARI and deviate it from PSTN to sip without bridge, but only when originate call answer.is it more clear?
ps. I wouldn't use AMI, but only ARI so I manage anything by WebSocket.
thanks _______________________________________________________________________
Oscar OXY

      Da: Ben Merrills <b.merrills at mersontech.co.uk>
 A: Asterisk Application Development discussion <asterisk-app-dev at lists.digium.com> 
 Inviato: Mercoledì 25 Febbraio 2015 10:23
 Oggetto: Re: [asterisk-app-dev] How do I bind a outbound call?
   
> that you have written is like me, or do I mistake?

> thanks

You DO need a bridge, if that's what you're asking. The bridge is used to connect the two channels. ARI Does not support Originate to anything else but a Stasis Application. Unlike AMI's Originate which takes a source and destination (channel/extension/app etc), Stasis will only allow you to originate a channel into a Stasis app. I think the reasoning here was mostly to do with security etc, but either way, you need to bridge both the calls.

You might just be better off using AMI's originate! It would be easier for you, you could set the PSTN as the first leg and your sip extension as the second.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate




> > I'm doing a predictive dialer by ARI, that is:
> >
> > asterisk call 2 number phone (to PSTN) in outbound, when one phone PSTN answer (we suppose answer first phone) then asterisk call my sip phone, when I answer then asterisk bind, by a bridge, the first channel (PSTN) and my sip phone and hang up the second calling (PSTN).
> >
> > is it right? exist a ARI function redirect from PSTN to my sip phone without bridge?
> > thanks

> The general pattern here would be, in my opinion.

> Originate 1st channel (PSTN)
> On Answer Originate 2nd channel (local sip extension)
> On Answer Create a bridge and add 1st and second channels to the bridge
> On Hang-up destroy bridge


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