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<div class="moz-cite-prefix">Hi Salvatore,<br>
<br>
Be careful with SIP softphones. I've seen them do weird stuff
with AMI commands (i.e.: Commands received are messed up/cut-out
in parts), especially when it comes to DTMFs. I've found SFLphone
(not Ring) to work the best, but even it had issues. In the end I
opted for Digium cards (analog and PRI work best) and hardware SIP
devices.<br>
<br>
Sorry, I don't really have any experience using queues and even
less with ARI (like I said, I went pure AMI from a C-application
and I've never tried controlling/handling a call via a web
interface). I learnt a few queue dialplan concepts years ago, but
don't recall the details. In all of this you didn't mention how
you were trying to answer the calls. Have you used the
AddQueueMember application (with no members defined in
queues.conf, as specified within the "Return codes" section of
<a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue">http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue</a>) to add
someone with whom to receive/answer calls? If not, that might be
something to read up on..<br>
<br>
To answer your question, I imagine either you need to use Asterisk
queue's or write your own queue. I mean, there has to be some way
of tracking all the calls and allowing you to select one to deal
with at a time.<br>
<br>
Here's a tutorial on queues which I came across. Maybe you could
try following along with it and see if it helps you progress with
queues:<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.orderlyq.com/asterisk-queues-tutorial/">http://www.orderlyq.com/asterisk-queues-tutorial/</a><br>
<br>
Hope this helps,<br>
<br>
Daniel<br>
<br>
<br>
<br>
<br>
On 11/23/2016 03:24 AM, Salvatore Franco wrote:<br>
</div>
<blockquote
cite="mid:OH3603$3C244CC1AFD09C0720776704D6DF2B92@fast-auto.it"
type="cite">
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<div>Hi Daniel,<br>
the Queue works well if I use a SIP Client (Zoiper) but if I try
to answer using AMI i receive that error :(<br>
<br>
ARI command ANSWER only works if I use it on a single channel,
the queue is a strange application that cannot be in Stasis
directly.<br>
Is writing my own queue the only way to handle calls via http in
queue?<br>
<br>
Thanks<br>
<br>
Regards<br>
<p class="MsoNormal"><b><span>Salvatore
Franco</span></b><span></span></p>
<p class="MsoNormal"><br>
</p>
<br>
<div>
<hr></div>
<br>
<div><span>Da</span><span>:
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a></span></div>
<div><span>A</span><span>: "Asterisk Developers Mailing List"
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a></span></div>
<div><span>Cc</span><span>: </span></div>
<div><span>Data</span><span>: Tue, 22 Nov 2016 12:33:38 -0500</span></div>
<div><span>Oggetto</span><span>: Re: [asterisk-dev] AGI:async</span></div>
<br>
<div class="moz-cite-prefix">Hi,<br>
<br>
Unfortunately, I wish I had a simple queue to share but my
implementation is quite complex due to how I control the call
on a per-operation basis. If you choose this route then
perhaps you can use store your channel/queue name(s) within
simple storage elements such as a stack, array, or even a
hashtable (although the last may not be ideal for retrieving
calls in order) and then use AMI to control the calls from
there. Sorry I can't be more helpful in providing you an
example.<br>
<br>
I imagine Asterisk queue's work though, so perhaps it might be
worth your while to pursue that approach a bit more
(unfortunately I've never tried them, so I can't tell you how
it's done!). That being said, perhaps getting Asterisk
Queue's going will require a more step-by-step approach to
figure out why it's not working for you. Here are my
suggestions:<br>
<br>
Have you tried the exact dialplans in the examples within <a
class="moz-txt-link-freetext"
href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue"
_djrealurl="http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue"
moz-do-not-send="true">http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue</a>
? i.e.: Have you been able to get Asterisk queue's working
without AMI? If not, that might be where to start and then
add AMI in afterward..<br>
<br>
Another suggestion I have is to use the QueueAdd AMI command,
rather then calling Queue within your dialplan. If you have
success on doing something like an Answer/Playback via AMI,
then you will probably have no issues with the QueueAdd AMI
command. Please try that and maybe you'll have more luck.
Here's the API reference: <a moz-do-not-send="true"
class="moz-txt-link-freetext"
href="http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueAdd"
_djrealurl="http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueAdd">http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueAdd</a><br>
<br>
Hope this helps & good luck,<br>
<br>
Daniel<br>
<br>
<br>
On 11/22/2016 10:35 AM, Salvatore Franco wrote:<br>
</div>
<blockquote
cite="mid:OH1VAX$94F64C4506FD1FAE9471144FCF266EF9@fast-auto.it"
type="cite">
<div><br>
Hi,<br>
Thanks for your suggestions. <br>
The comma was an error in my email, not in my dialplan.<br>
Even if i use Answer command before Queue i received the
same error back, so i think the best way is to have my own
queue.<br>
Have you got any example of how to implement a simple queue
that use the "fewestcalls" strategy?<br>
<br>
Best Regards<br>
<br>
Salvatore <br>
<br>
<br>
<div>
<hr></div>
<br>
<div><span>Da</span><span>: <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:asterisk-dev-bounces@lists.digium.com"
_djrealurl="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a></span></div>
<div><span>A</span><span>: "Asterisk Developers Mailing
List" <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:asterisk-dev@lists.digium.com"
_djrealurl="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a></span></div>
<div><span>Cc</span><span>: </span></div>
<div><span>Data</span><span>: Tue, 22 Nov 2016 10:04:58
-0500</span></div>
<div><span>Oggetto</span><span>: Re: [asterisk-dev]
AGI:async</span></div>
<br>
<div class="moz-cite-prefix">Hi Salvatore,<br>
<br>
I don't have any experience with the "Queue" command, as I
basically pass the call directly to agi:async and control
the call every step of the way from my C program (where
I've implemented my own queue/call tracking).<br>
<br>
However, here are my suggestions:<br>
<br>
1) Upon taking a quick look at <a
class="moz-txt-link-freetext"
href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue"
_djrealurl="http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue"
moz-do-not-send="true">http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue</a>,
I think your "failed to add agi command to channel.."
error may be since the call is not answered yet. You'll
notice on that web page that the dial plan calls "Answer"
before Queue in example 1, and Playback (which itself
internally answers the call to be able to then play the
announcement) before Queue in example 3 (I'm skipping
example 2, as it's a link to a German site and not
directly on the page). Please try one of those commands,
before calling your Queue command, and I think you'll get
better results.<br>
<br>
2) You are correct that AGI(agi:async) stops execution of
the dialplan. In your attempt to execute AGI(agi:async)
before Queue, that's exactly what happened. The call
entered agi:async mode and from that point on you are
supposed to issue commands to control the call from your
AGI program.<br>
<br>
That's probably the reason you don't hear ringing on the
phone, as (in your AGI call on priority 1 example) your
inbound call never reached the "Queue" dialplan priority.<br>
<br>
3) This is a minor detail, but I'm not sure your syntax is
correct. Try removing the comma before the "n", so that
you have the following in your dial plan: "same =>
n,AGI(agi:async)"<br>
<br>
Hope this helps,<br>
<br>
Daniel<br>
<br>
<br>
<br>
On 11/22/2016 05:03 AM, Salvatore Franco wrote:<br>
</div>
<blockquote
cite="mid:OH1FYH$E0E0D9B067EFB835A9435710DECDA23D@fast-auto.it"
type="cite">
<div>Hi, <br>
I'm trying to answer a call that is in Queue through
AMI, launching AGI command ANSWER.<br>
I read about putting AGI(agi:async) in extensions.conf
dialplan but if i set in this order:<br>
<br>
exten=> 1000,1,Queue(queuename)<br>
same=> ,n,AGI(agi:async)<br>
<br>
and try to answer using AMI the response is : failed to
add agi command to channel XXX queue.<br>
<br>
Else if i set the dial plan in this order:<br>
<br>
<br>
exten=>1000,1,AGI(agi:async)<br>
same=> ,n,Queue(queuename)<br>
<br>
the phone doesn't ring and the command execute
successful, but nothing happens. Isn't agi:async
"really" async but stops execution of the dialplan?<br>
<br>
Any suggestions?<br>
<br>
<br>
Thanks<br>
<br>
Best Regards <br>
<br>
<p class="MsoNormal"><b><span>Salvatore Franco</span></b><span></span></p>
<p class="MsoNormal"><br>
</p>
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<pre class="moz-signature" cols="72">--
Daniel McFarlane
Szeto Technologies Inc.
885 Ellingham Ave
Pointe Claire, Québec
Canada, H9R 5E8
tel: (514) 630-7878
fax: (514) 630-6394
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.szeto.ca" _djrealurl="http://www.szeto.ca">http://www.szeto.ca</a>
</pre>
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<pre class="moz-signature" cols="72">--
Daniel McFarlane
Szeto Technologies Inc.
885 Ellingham Ave
Pointe Claire, Québec
Canada, H9R 5E8
tel: (514) 630-7878
fax: (514) 630-6394
<a class="moz-txt-link-freetext" href="http://www.szeto.ca">http://www.szeto.ca</a>
</pre>
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