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Last active 7 years ago
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=ABC299:1234:all
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=299:1234:all
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 :-)