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Call Centre Hot Desking

July 30, 2019, 7:18 am
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I’m trying to understand the best way to set up my call centre in the move over to FreePBX/PBXAct and have come up against Device & User Mode vs Extension Mode. The former is not supported but does allow an extension to exist wherever a user logs in and that would allow the user to be automatically logged into their queues. To my mind, this is exactly what I want by the term Hot Desking.

Extension mode seems to be completely opposite to what a call centre needs so how have other people handled this? Are there commercial modules that simply resolve this issue or is there some configuration I’m not seeing?

I have a test box of FreePBX that I’m playing around with and going through the Sangoma PBXAct essentials videos but nothing I’ve seen there or in the wiki/forum makes a clear suggestion of what a call centre should do to handle hot desking and queues.

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From-internal-custom wildcard extensions

July 30, 2019, 7:18 am
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Not sure if it helps in what I am trying to achieve but, as an example, a normal extension 2003, when this dials, I want it to log in the stasis application using my ARI :frowning: Hope this helps…

I am not familiar with ARI’s and have a developer running the wscat commant in Node.Js prompt.

Thank you for your support here.

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Call Centre Hot Desking

July 30, 2019, 7:22 am
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Device and user mode is completely supported by the system. It is not supported by a lot of the modules that make your life easier as an administrator.

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Trunk stops working after some idle time

July 30, 2019, 7:23 am
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Integrating SIP with Simplex 5100 PA System

July 30, 2019, 7:25 am
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Only problem with that is I have no idea how to integrate one of those into the panel. It seems like all of the functions like swapping inputs and whatnot require the admin phone to dial a string to initiate it. There are no buttons on the panel itself.

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Ring Group will not send to IVR as Destination if no answer

July 30, 2019, 7:25 am
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Turns out that if I call the system from an outside line, it works as intended.

We have another cloud hosted PBX that services our internal office network.

I setup new production systems within the office on the same network.

Something in house was causing this. If I use our handsets for our office PBX it doesn’t work.

Thanks for the input guys, but forcing to answer is not the solution.

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From-internal-custom wildcard extensions

July 30, 2019, 7:30 am
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Would anyone else have any other suggestions here? I am really struggling to get the extension to ring successfully and show up in the stasis logs :tired_face:

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Connect to Door Striker Recommendations?

July 30, 2019, 7:30 am
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This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.

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WebRTC call button do nothing

July 30, 2019, 7:31 am
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Your webrtc extensions are probably still stuck as “SIP” so here is what i suggest (I have had this happen before).

go to https://yourdomain/admin/config.php?display=devices

Find the devices that start with 99 - these are the hidden webrtc devices. (They do not show up in your Extensions list)

If they are shown as sip (not pjsip) then delete them and apply changes.

After you do this you might have to go back and save your extension again so that the webrtc device gets created properly. If you go back to the ?display=devices URL again, this time you should see the 99… extension listed as a pjsip.

See whether that helps. If not, I am sorry but I am fully spent on ideas and help for today. :slight_smile:

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From-internal-custom wildcard extensions

July 30, 2019, 7:42 am
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I feel like I am getting closer using PJSIP

With the below in extension_custom.conf:
from-internal-test]
include => from-internal
exten => _XXXX,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
same => n,Answer()
same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
same => n,Hangup()

I dial now on any extension and it logs into the stasis application but now I cannot hear anyone on the phone… any ideas why this could be?

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WebRTC call button do nothing

July 30, 2019, 7:53 am
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Latest version DHCP bug, system drops after 1 hour

July 30, 2019, 7:53 am
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This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.

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From-internal-custom wildcard extensions

July 30, 2019, 7:57 am
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Since no one knows what “Tess_Asterisk” does then I would say no, there won’t be any ideas.

Since you are taking on a development effort then the best place to start would be to read up on it: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573

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Multiple “night modes”

July 30, 2019, 8:00 am
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Thanks for the reply!

To address the things you mentioned

  1. They are in as static agents. The queue is functioning as more of a ring group (the call volume is very low there is rarely more than one call at a time, even more so at night).
    2&3. This is the desired behavior as there is only one person, in the very rare event that there is more than one call at a time, the same person would be responsible for answering both calls using hold or call waiting.
  2. Okay, thats what I was afraid of.

Thank you!

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Integrating SIP with Simplex 5100 PA System

July 30, 2019, 8:02 am
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Get an ATA with 1 FXO port e.g. Obihai, Grandstream or Linksys. Set it up as a trunk. Create an Outbound Route that maps your desired page code to #10. No need for any custom dial plan; the ATA will generate the DTMF automatically.

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From-internal-custom wildcard extensions

July 30, 2019, 8:08 am
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Apologies. Tess is the application they are wanting to use… I have provided them ARI details in Asterisk Rest Interface Users.

In my scenario:
The exact scenario we need to achieve - In TESS Application I log in as 2003 extension as user and i will dial extension 2003 from 2002 and need to check whether i we will get call in 2003 SIP phone and event in the computer where TESS app is running

Efforts to try and get this working are:

  1. Executing the following command using Node.js prompt:
    wscat -c "ws://10.38.240.1:8088/ari/events?api_key=3rd_Party_Yucca:1d8c16f0eebafafceb7e0f6ab181e3f5&app=Tess_Asterisk"

  2. It shows connected but when I dial from extension 2003 to 2004 no logs show.

  3. When add the following context in extensions_custom.conf
    [from-internal-test]
    exten => 2003,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
    same => n,Answer()
    same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
    same => n,Hangup()
    [from-internal-custom]
    exten => 2004,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
    same => n,Answer()
    same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
    same => n,Hangup()

And then in the extension settings page set the context to “from-internal-test” I get some logs.

I want to make it so I do not have to define every extension in the custom conf.

  1. I changed the extensions_custom.conf to:
    [from-internal-test]
    exten => _XXXX,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
    same => n,Answer()
    same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
    same => n,Hangup()

When I point the extension to this context and dial, it logs but no sound can be heard from either extension at either end.

Hope this explains my problem and the efforts exhausted to try and get this working.

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Set 99* devices (WebRTC) Tech Type to pjsip

July 30, 2019, 8:09 am
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Set 99* devices (WebRTC) Tech Type to pjsip

July 30, 2019, 8:11 am
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When i’m trying to call with WebRTC that instant report a :

Session failed
Session terminated

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Calls via DAHDI FXO to IVR routing not working, SIP calls work fine

July 30, 2019, 8:31 am
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Ok, thanks. But it is worth mentioning that this problem is a change from years of working correctly. I always like to figure out what changed if I can.

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Multiple “night modes”

July 30, 2019, 8:33 am
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Right and put them on two devices. If I’m on my phone and the second call comes in, it will see my ip phone as IN_USE and then could ring the cell phone so the agent will have calls coming in on multiple devices in those cases.

I get the queue for the hold music but this should be a single extension that has FollowMe enabled and the Queue honors that so the IP phone is always called first and then based on the FollowMe goes out the cell phone if needed. You can still send multiple calls to the IP phone and thus allow them to use Call Waiting/Hold if needed.

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