Re: Fwd: Asterisk Call Control

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You could call a local channel  that executes dialplan that dials the phone(s) with a pre-dial handler that sends a SIP header to answer the call.

It's more steps but I think you could accomplish what you want. 

On Thu, Sep 7, 2017 at 7:40 AM, Arne Herbots <arne.herbots@xxxxxxxxxxxx> wrote:
Hey, 

I tried using ARI already, but that did not help either.
We are able to make the dial but we still had to manually click on Answer. 
Since the project is about automating the complete test-scenarios, we have to eliminate every human action.  


On 7 September 2017 at 11:36, Marin Odrljin <marin@xxxxxxxxx> wrote:

Hi Arne,

 

Try Asterisk ARI interface. We are using it lately and we are quite happy with it. A long time ago we have tried AGI and AMI but we have concluded that it was not worth to spend too much time on it because we couldn’t achieve all what we wanted.

 

I hope ARI will continue to develop because it is correct way how APIs should look like. Unfortunately there are still a lot of bugs related with Asterisk and ARI itself which are not being fixed for a long time, but you can always do some kind of workaround to prevent it or just try some other version of Asterisk. I recommend Asterisk 13 for ARI because I have discovered a memory leak problem in Asterisk 14 and ARI: https://issues.asterisk.org/jira/browse/ASTERISK-27067 . I really don’t understand why such an obvious bug is not resolved till now. There are also other ARI bugs issued by me on Asterisk tracker site.

 

Good luck.

Marin

 

 

From: asterisk-app-dev-bounces@lists.digium.com [mailto:asterisk-app-dev-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Arne Herbots
Sent: Wednesday, September 06, 2017 4:25 PM
To: asterisk-app-dev@lists.digium.com
Subject: Fwd: Asterisk Call Control

 

 

 

Hello everyone,

 

We are creating a setup/framework to test pbx's from several vendor (one of them being Asterisk).

 

Let me just sketch what we have now and what issues we are having at the moment.

We started with programming everything for Cisco (CUCM) using jtapi which works.

We can tell the pbx to set up calls, put calls on hond, extend calls,... everything works perfectly so far..

 

After that we started working on the same for Asterisk. Since we read that Astrisk works with jtapi we were hoping for a smooth development. But apparently the version of Asterisk we need to test doesn't work with jtapi.

 

After some searching we started using Asterisk-Java. But this gives some problems also(or we are doing it wrong...).

We used the Dail-example and after some corrections we were able to do some stuff. 

We have a set-up with 2 Zoiper phones and try to make the pbx setup a call between these two using the following line of code.

DialActivity dial = pbx.dial(from, fromCallerID, to, toCallerID);

At first we thought we finally did it, we saw on one of the phones that we were being called. But problem was that this phone is the phone that should make the call, not get called. The call it receives is from the 'to' number but on the second phone nothing happens there is no call to be seen there...

We should be able to control who hangs up the phone, in some cases it is the called party and others the calling party

 

If anyone has any idea where we went wrong or how we are supposed to do it, could you please help us out? 

 

 

 

 


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