FOP2 Community Forum

Hi,

I would like to disable this fonction so the extension is always green, not faded when the extension is not connected. Is it possible ? I still need to keep track of the call status (ringing, on a call, etc...)

Hi,

I see theses events if I change my own status using the toolbar and it's working as expected. But if I change the status of someone else using the presence manager plugin the status is not updated and I see nothing in the debug.

David

Hi,

I'm trying to use the change presence plugin. When I click on another user, I can see the dropdown list but when I select a status, the status is not updated. I'm running ./fop2_server -X 3 and I don,t see anything when I change the status of another user.

If I change my own status, I do see this :

127.0.0.1 -> Action: Command
127.0.0.1 -> Command: database put fop2state SIP/200 "Out to lunch"

127.0.0.1 -> Action: UserEvent
127.0.0.1 -> UserEvent: FOP2ASTDB
127.0.0.1 -> Family: fop2state
127.0.0.1 -> Key: SIP/200
127.0.0.1 -> Channel: SIP/200
127.0.0.1 -> Value: Out to lunch

127.0.0.1 <- Response: Follows
127.0.0.1 <- Privilege: Command
127.0.0.1 <- Updated database successfully
--END COMMAND--

127.0.0.1 <- Response: Success
127.0.0.1 <- Message: Event Sent

127.0.0.1 <- Event: UserEvent
127.0.0.1 <- Privilege: user,all
127.0.0.1 <- UserEvent: FOP2ASTDB
127.0.0.1 <- Action: UserEvent
127.0.0.1 <- Family: fop2state
127.0.0.1 <- Key: SIP/200
127.0.0.1 <- Channel: SIP/200
127.0.0.1 <- Value: Out to lunch

Running fop2 on a vanilla Asterisk 11.X

Thx !

Hi,

It is working if I set the extension to 100 but CDR History is then broken. I found that you are using a query that look like this :

SELECT * FROM cdr WHERE (src='100' OR channel LIKE 'SIP/100-' OR dst='100' OR dstchannel LIKE 'SIP/100-' )

In my case, the channel is SIP/sdfhjk not SIP/100.

I would really like to have this fix so I could better integrate FOP2 in my solution. Could you please PM me ?

Thx !

Hi altsupport,

Are you using vanilla asterisk ? If yes, edit features.conf. There is a star code there to cancel the supervised transfer. Not sure if it's the default but on our system we use *0.

It would be a very good improvement if FOP2 would add a cancel button that would send the dtmf we define using the AMI on the live channel. That way we could push a button to have the transfer cancel.

Keep me posted if you can't find the starcode or your system. You could print the output of 'core show features'

HI,

I'm confuse by your answer. From the test I did, "extension=" need to be equal to "sdfghjk" in my latest example to have the action bar. I'm not sure why are are telling me to use extension = 233 and user=233 because my sip peer name is "SIP/sdfghjk".

To get back to my problem, do you plan on fixing the issue ? The problem is, if a peername have alphanumeric caracter, the presence dropdown is not working.

Hi,

I don't think you understand my problem. It has nothing to do with dialing or anything. It's only regarding the presence on the upper bar.

FYI but it's not relevant to my issue I'm using alphanumeric sip peer name but using digit only to ring them. For exemple, in my dialplan I'm using 233,1,Dial(SIP/sdfghjk). The phone can then be reach by dialing 233.

Hi,

I was able to track down the issue and it seem to be a bug in fop2.

If I have the following in buttong.cfg :

[SIP/ABC299]
type=extension
extension=ABC299
context=ABC
label=299 - Test
mailbox=299@ABC
extenvoicemail=7299@ABC
group=admin
originatechannel=Local/299@originate

and fop2.cfg :

user=ABC299all

The status is not updated because the event is not send the the Asterisk manager. (tested using : ./fop2_server -X 3 -i eth1)

But If I have :

[SIP/299]
type=extension
extension=299
context=ABC
label=299 - Test
mailbox=299@ABC
extenvoicemail=7299@ABC
group=admin
originatechannel=Local/299@originate

and fop2.cfg :

user=299all

It his working. I have also tested with abc lowercase and it's also NOT working. I have also test with "sdfghjk" only and it's also not working. that mean as soon as you have alpha character the presence is not working. That also mean FOP2 presence won't work with XiVO/WAZO because they use random alpha numeric name for the sip peers.

Also, if I manually put this in the asterisk manager using telnet :

Action: UserEvent
UserEvent: FOP2ASTDB
Family: fop2state
Key: SIP/sdfghjk
Channel: SIP/sdfghjk
Value: Out to lunch

It's working as expected.

Hi,

I have a very basic problem, I'm using a vanilla asterisk with the latest FOP2. If I login in FOP2 and I change the status of my extension to Do Not Disturb in FOP2, my little icon is not updated in FOP2. I'm not talking about any integration between Asterisk and FOP2.

Thx

I think you answered the wrong post

We tried this setting but it doesn't seem to work on the latest version. Is there a bug with this features or is there an equivalent we can set in fop2.cfg or any config files ?

Hi,

Is it possible to hide the extension number. We would like to display only the label.

Thx !

Hi,

What is the status of confbridge with FOP2 on Asterisk 13 ? Is there anything specific we need to do on the Conference button ?

Thx !

David

Hi,

Finally with the help of Nicolas, we found that the issue is not related to FOP2 and uppercase/lowercase. In my Asterisk I had the following :

[2016-09-23 19:40:35] WARNING[28278] channel.c: Unable to find a codec translation path: (h264) -> (g722)
[2016-09-23 19:40:35] WARNING[28278] channel.c: Unable to find a codec translation path: (g722) -> (h264)
[2016-09-23 19:40:35] VERBOSE[29910] dial.c: SIP/abc2001-00003726 answered
[2016-09-23 19:40:35] WARNING[29910] channel.c: Unable to find a codec translation path: (slin) -> (h264)
[2016-09-23 19:40:35] ERROR[29910] app_chanspy.c: Could Not Set Write Format.

abc2001 is a Bria softphone with video enabled and on asterisk in sip.conf I had videosupport=yes. Chanspy seem to terminate because video is involved. For the record, the destination abc2000 doesn't support video, it was a regular audio call.

Using Chanspy + Video enable in Asterisk/Softphone is the problem.

Hi,

Using Asterisk 13 and latest version of FOP2. I'm unable to get the Whisper to work. My peer name is SIP/abc202 for example. Everything is working except whisper. When I try to Whisper, my phone ring but when I answer, the line disconnect. After running fop2 in debug mode, I saw the Chanspy command launch with SIP/ABC203 as the destination to whisper instead of SIP/abc203. Asterisk being case sensitive, it doesn't work because the peer doesn't exist.

1. Why is fop2 using upper case why in the button.cfg the name of my channel is sip/abc203 ?
2. Is to possible to have fop2 to send the call in the dialplan instead of calling the chanspy application directly ?

Thx !

Hi,

I have done what you said but I still see 150 buttons (which see to be imported from somewhere.

1. Why do I need to have #exec autoconfig-users.sh ? I'm running a vanilla Asterisk and this script doesn't really seem to understand the way I create my peer. Can't I just disable this script completely and create the button in the fop2manager ?

2. If I understand well, the button need to be generated by script and the fop2manager is using the file button-custom.cfg to load the button ?

Thx !

Hi,

I'm confuse on FOP2 and Vanilla. I have commented the last line with the exec autoconfig-users.sh and I have created two users with a button files :

user=213:213:all
user=221:221:all
buttonfile=buttons.cfg

Then In the button files I have created 6 extensions + 3 queues + 2 trunk. This is all working fine.

Now I needed the have the Queue Alarm plugin, from what I understand I needed to use the FOP2 Manager to install the plugin. At first I could install the plugin but when I tried to check the "Globally Enable" check box. It didn't work for some reason. I have created a MySQL database for FOP2 manager and edited the configuration files to use that database. Now it seem to be working, the plugin is installed and enabled. But...

Now if I go to the button and users sections, FOP2 Manager imported some of my peers but they are all mess up. For example, like 30% of my user are defined with the type trunk instead of extensions. And I can't seem to edit thoses ???

Also, how do I created a new button manually ???
How are the button populated ??? (Which script is run ?)

Anyway, I decided to try to login in FOP2 as a user to see the result, but it's not working. Which context is fop2 manager using to create the buttons/users ? How to access the "tenant" created by FOP2 Manager 9 (http://192.168.1.X/fop2/?context=XXXX) ?

Finally, how to I enable a plugin on a user if I'm using command line (no fop2 manager) to a user ?

Thx for reading my wall of text but I'm sure if some take the time to answer my questions, this will help a lots of people because the documentation is not complete if you want to use fop2 manager + Vanilla Asterisk.

Thx