[asterisk-app-dev] ARI transfer calls

Lukovic, Damnjan damnjan.lukovic at abbeoo.com
Fri Feb 27 14:26:29 CST 2015


Hi Paul,

thanks for replying - i am so sorry but i am too new to it understanding what u say. first i couldn't get swagger to run :-(

second i really don't know what u mean with dial plan and dump into it?!

could u please concrete your-self or send me some link with good working examples for me to read?

many thanx,

Damnjan.

On Thu, Feb 26, 2015 at 5:03 PM, Lukovic, Damnjan
<damnjan.lukovic at abbeoo.com> wrote:
> Hi all,
>
> I am digging now already for weeks getting the asterisk ARI to transfer
> calls with no success. I am on my end of knowledge and would appreciate any
> help indeed.
>
> scenario:
>
> 1) we got 2 extensions, 1000 and 1050.
> 2) a external person calls us via sip trunk to extension 1000, which takes
> the line.
> 3) extension 1000 wants now to transfer the call to 1050
> 4) extension 1050 accepts the external call and want to take the line
> 5) extension 1000 get disconnected.
>
> by the way:
>
> 1) i am using node.js
> 2) i don't use any 3rd libary
>
> I try to create a new channel from ARI and add it to the first bridge.
> the error we get is as followed:
>
> 1) "Bridge not in Stasis application"
>
> folks :-) how can that be done using ARI?
>
> many thanks for all answers,
>
> Damnjan.
>
Fire up http://ari.asterisk.org/ and log into your asterisk instance.
Then, call into your dialplan and dump the channel into Stasis().
Using the swagger UI, you can then test answering the channel,
creating a bridge then moving the channel into the bridge.

Obligatory video of me showing that at astricon:
https://www.youtube.com/watch?v=7nujWPFcftc

--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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