FOP2 Community Forum

Hello,

We run the console successfully on commercial deployments. However, from time to time, the console freezes.

If we have the console running in a browser on 2 PC's simultaneously, here is what happens :

• On PC1, the console is not refreshing anymore : the time counters are still running, but line monitoring is broken.
• On PC2, the console is perfectly running.

If you close the browser and restart it on PC1, the problem remains.

What could be the problem ?
How can we debug it ?

Thanks,

Damien

Hello,

It seems that when originatechannel is set, it is not taken into account when doing a pickup.

The value inside the []'s is used instead.

Is that possible to change that behavior ?

Damien

It works, thanks a lot for the very efficient help !

Damien

I ignored that setting even existed.

Thanks for the tip !

I have another (silly) question: the Pickup action does not appear when I click on a waiting call. How to add pickup right for a queue ?

Damien

Hi,

membername for statically configured members ?

Where should I specify that ?

Thanks,

Damien

Hi again,

That works, but the problem is that the same agent is in different queues with a queuecontext depending on the queue itself.

For example, 901@through-queue-71 and 901@through-queue-72

Is there something to do ?

Thanks,

Damien

Hi,

Thanks for the originateCOntext tip.

But is there a risk to use both channels between []'s ?

It's the only way to get the correct status for us.

Hi,

I know I become annoying, but if I do this :
[SIP/75&SIP/75-2]
type=extension
extension=75
context=ext-everything
label=DS
mailbox=75@default
extenvoicemail=855575@ext-everything-local
channel=SIP/75&SIP/75-2

then it works as expected except for one thing : click to call or dialing from the UI does not work anymore :

[2011-04-12 18:22:29] WARNING[30004]: chan_sip.c:3178 create_addr: No such host: 75&SIP
[2011-04-12 18:22:29] NOTICE[30004]: channel.c:3490 __ast_request_and_dial: Unable to request channel SIP/75&SIP/75-2

some escaping required ?

Damien

Hi,

I understand what you mean.

So, my only concern is when there are multiple channels : how can I consider the user as online if at least one of the channels is online ?

I guess it is not possible and will never be but I need to find a solution.

Damien

Hi,

Actually I checked our code and it is feasible.

If you define this :
hint => 75,hint,SIP/75&SIP/75-2
in context ext-local-users

then core show hints shows this :
75@ext-local-users : SIP/75-2&SIP/75 State:Idle Watchers 1

And when there is an incoming call, the AMI shows this event :
Event: ExtensionStatus
Privilege: call,all
Exten: 75
Context: ext-local-users
Status: 8

The only thing you have to change in your code is to add the ability to listen to that type of event.

Would it be possible ?

It is kind of mandatory if we want to go further with FOP2 in our solution

thank you,

Damien

Hi,

Perhaps the ideal would be to add a hint= setting in buttons.cfg where the user could specify the hint to watch.

Is that something you can do in a further release ?

Damien

Hi,

The hint as determined by exten => 74,hint,SIP/74&SIP/74-2 has the correct value.

Is it possible for you, technically, to monitor extension status through this hint ?

Our purpose is to integrate FOP2 (paid version) into our own product. So that is important for us.

Thank you,

Damien

Hello,

It seems to work to monitor channels, but the problem is that the extension appears to be offline as soon as at least one of the extensions is offline (or as soon as [SIP/75] is offline if [SIP/75] is the leading extension).

We use multi-devices for softphone support. The softphone is often disconnected.

Damien

Hello,

Thanks to your help, I have nearly finalized integration into our own product line. I'll need to contact you offline for licensing.

However, I have a few more questions :

1. We are implementing multi-devices into our system by having different SIP peers for the same extension. For example, user 74 can be called through Local/74 which will in turn trigger a Dial on SIP/74&SIP/74-2&SIP/74-3 . Is that possible to have one button for all of these and have sort of an integrated status view ?

We have hints defined like this :
exten => 74,hint,SIP/74&SIP/74-2

I have tried different combinations of user definitions without success so far:
[SIP/74]
...
channel=SIP/74&SIP/74-2&/SIP/74-3

2. concerning user language, I would like to dynamically set user language without having the user change his personal preferences. Is there a way to achieve this ?
   (ie either looking at the user asterisk language definition or any other tip)

Thank you,

Damien

Hi,

The css code works.

However, the transfer to cellphone button is not displayed at all when you do not have 'all' permissions, but if you specify this instead: user=211dial,transfer,hangupself (for example).

Is this a small bug ?

Damien

Hello everyone,

In our configuration, we have queue members defined as Local/xxxx@some-asterisk-context/n.

FOP2 is displaying the correctly, but with the @some-asterisk-context details.

Is it possible to either display the member name or only the xxxx part ?

Damien

Hello everyone,

We are integrating FOP2 into our own product base. However, I have a few questions regarding permissions :
a) when enabling the transfer button by using the 'transfer' keyword, only the transfer and transfer to voicemail buttons appear, but not the transfer to cell button. How can I enable it ?
b) is there a way to hide the logout button ?

Thank you,

Damien