[asterisk-app-dev] ARI Bridge and Dial

Mon Jun 2 15:28:35 CDT 2014

Thanks again for the quick reply. I am using Java Websockets Java EE 7 JSR
356 - Tyrus part of the Oracle Glassfish package. I have retested on
various servers and am seeing the same behavior. The call itself is fine
but when one party hangs up I get the exception (onClose) from Asterisk. I
am not sending a corresponding ARI event - the Hangup or StasisEnd have not
yet fired.

The object I capture in the onClose has the following info:

 CloseReason object:
closeCode:
code:1007
Name:NOT_CONSISTENT
reasonPhrase:Illegal UTF-8 Sequence

On Sat, May 31, 2014 at 1:00 PM, <asterisk-app-dev-request at lists.digium.com>
wrote:

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>    1. Re: ARI Bridge and Dial (Jim Black)
>    2. Re: ARI Bridge and Dial (Matthew Jordan)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 30 May 2014 17:45:06 -0400
> From: Jim Black <jblack at mediu.com>
> To: asterisk-app-dev at lists.digium.com
> Subject: Re: [asterisk-app-dev] ARI Bridge and Dial
> Message-ID:
>         <CAMXHdJSjrKmhRdf41sWiMOXfY6kmDNvLiUMmACdz72Oat7=
> crQ at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> The example you gave really helped and I am able to complete a call.
> However, when the call-ee hangs-up my websocket listener gets an OnClose
> message with a 1007 - Illegal UTF-8 Sequence. I don't understand why it
> would be throwing this since I am not sending anything back from the
> websocket listener. Sometimes I get the ChannelHangupRequest and StasisEnd
> events before it closes... other times it just closes immediately after the
> hangup.
>
> I just moved servers and it wasn't exhibiting this type of behavior prior
> to the move. Any ideas are appreciated!!
>
> -Jim
>
>
>
> On Wed, May 14, 2014 at 4:46 PM, Jim Black <jblack at mediu.com> wrote:
>
> > Thanks for the quick response. I mistakenly assumed the create bridge
> > command took a list of types. After taking a look at the swagger UI I
> > figured out it was merely a list of acceptable values. I had an issue
> with
> > de-serializing the json response which confused the matter. I wrote a
> > custom deserializer and it works fine now.
> >
> > Thanks for the example from your python code. I followed that example and
> > it worked fine. Considering a bridge needs to be created for a simple
> dial
> > application - do you see any pitfalls of creating a pool of bridges for
> the
> > application to share - assuming I take care of the bridge-state
> internally?
> >
> > Also... with ARI, I see no hooks into provisioning devices, I assume I
> > need to use AMI *updateconfig*? Thanks!!!
> >
> >
> >> ------------------------------
> >>
> >> Message: 2
> >> Date: Thu, 8 May 2014 14:43:17 -0500
> >> From: Samuel Galarneau <sgalarneau at digium.com>
> >> To: Asterisk Application Development discussion
> >>         <asterisk-app-dev at lists.digium.com>
> >> Subject: Re: [asterisk-app-dev] ARI Bridge and Dial
> >> Message-ID:
> >>         <CAGZGSQ7Z8AeP7VBvT1o_aRRJJDS+8zY2J=
> >> ubAqn-1hBDMLmOXA at mail.gmail.com>
> >> Content-Type: text/plain; charset="utf-8"
> >>
> >> Jim, please see my responses in line.
> >>
> >>
> >> > Hi,
> >> >
> >> > I have a few questions regarding ARI after experimenting with it for a
> >> > while.
> >> >
> >> > Bridging. When I create a bridge, I provide a single type ('mixing') I
> >> get
> >> > a '200' OK back but when I retrieve details on the bridge, the type
> >> 'list'
> >> > is NULL. The bridge seems to work - but I wanted to make sure there
> >> wasn't
> >> > an issue.
> >> >
> >>
> >> What do you mean by type 'list'? What ARI operation are you using to get
> >> details for the bridge?
> >>
> >>
> >> >
> >> > Let's say I want to create a simple Dial application. By trial and
> >> error,
> >> > what seems to work is a call comes into my dial plan and off to my
> app.
> >> I
> >> > answer, create a bridge and add this channel to the bridge. I then
> >> create a
> >> > channel for the destination SIP when it picks-up and add this to the
> >> > bridge. I should now have 2 connected phones. Thanks!!
> >> >
> >>
> >> This sounds about right. After the first channel enters your
> application,
> >> you need to originate a call to the second channel and then put them
> both
> >> in the bridge. Once that is done, getting the details of that bridge
> will
> >> show both channels under the channels property, which will be an array
> of
> >> channel ids. Please see
> >>
> >>
> https://github.com/asterisk/ari-py/blob/master/examples/originate_example.pyfor
> >> an example of how to do this using ari-py. The same functionality
> >> could
> >> be accomplished by making direct calls to ARI of course.
> >>
> >>
> >> Samuel Fortier-Galarneau
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> ------------------------------
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> Message: 2
> Date: Fri, 30 May 2014 17:57:03 -0500
> From: Matthew Jordan <mjordan at digium.com>
> To: Asterisk Application Development discussion
>         <asterisk-app-dev at lists.digium.com>
> Subject: Re: [asterisk-app-dev] ARI Bridge and Dial
> Message-ID:
>         <
> CAN2PU+4jpRLkwq4Ys6D6uLaNVeu61nqviT+6bNn3Mepmw_t_Sw at mail.gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> On Fri, May 30, 2014 at 4:45 PM, Jim Black <jblack at mediu.com> wrote:
> > The example you gave really helped and I am able to complete a call.
> > However, when the call-ee hangs-up my websocket listener gets an OnClose
> > message with a 1007 - Illegal UTF-8 Sequence. I don't understand why it
> > would be throwing this since I am not sending anything back from the
> > websocket listener. Sometimes I get the ChannelHangupRequest and
> StasisEnd
> > events before it closes... other times it just closes immediately after
> the
> > hangup.
> >
> > I just moved servers and it wasn't exhibiting this type of behavior
> prior to
> > the move. Any ideas are appreciated!!
> >
> > -Jim
> >
>
> What WebSocket library are you using?
>
> Can you provide a dump of what the WebSocket receives?
>
>
>
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