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[asterisk-biz] Transfering calls with ARI


 
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damnjan.lukovic at abb...
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PostPosted: Fri Feb 27, 2015 1:46 pm    Post subject: [asterisk-biz] Transfering calls with ARI Reply with quote

Hi everyone,


I am looking for someone to hire to help us with step by step guidance on transferring calls with ARI.


We are running Asterisk 13.2.0 on Debian Wheezy. We want to achieve the following scenario:


Let’s say we have two extensions: 1000 and 1050. Someone wants to talk to 1050, so he calls 1000 (operator) from external SIP trunk (for example using Zoiper client). 1000 answers, and puts the user on hold. Then 1000 dials 1050 and asks if he wants to talk to the user. If 1050 agrees, 1000 hangs up and joins them two together.


We are using ARI for this. We are using Node.js but without a library, so we just use plain REST request to ARI. When the user asks to talk to 1050, we try to create a new channel with ARI, and than put that channel into the bridge between the user and operator 1000. We can not do that because we get error “Bridge not in Stasis application”. We understand that we can not mix channels that are not in Stasis, but how could we solve this problem? We also tried sending DTMF to a channel but we get the same error. We don’t understand Stasis at all, how to configure it, how it works, etc. We tried node-ari-client with this example: https://github.com/asterisk/node-ari-client/blob/master/examples/originate.js
But the “StasisStart” event never gets fired.


We tried messing around with /etc/asterisk/extensions_custom.conf, we put this:
same => n,Answer()
same => n,Stasis(myApp)
same => n,Hangup()


But we still don’t get anything.


Since we don’t understand what is Stasis and how it works, can someone explain to us step by step how can we achieve the scenario mentioned above?


Thanks


Damnjan
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