No recognition of incoming calls
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Carlos, popups will not show if the callerid matches the extension, as you do not want self popups (as when doing originations). I have other reports of FreePBX using a RINGGROUP number as the called number, and not the callerid of the person calling. That change is in the ringgroup macro for some freepbx version. And it is freepbx that uses that callerid and fop2 just displays it. I do not have reports of this behavior on anything but ringgroups.
If you can provide me access to your trixbox 2.8 I can take a deeper look, but it is a dialplan issue, not a fop2 issue...
Best regards,
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Hi Nicolas,
Here's a snip from the screen:
http://www.ripon.org/carlos/screen.jpg
showing the incoming call to the queue, but each extension showing its own calling number.
Carlos
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Nicolas,
Seem to be getting somewhere at last. If an agent logs into a queue via FOP2, then the CALLING number is shown as ringing the agent's extension and hence the pop-up works.
If an agent logs into a queue via (queue number)* e.g. 803* or the agent is placed permanently into the queue via trixbox static agents, then it's the AGENT number which is shown as ringing the agent's extension, and hence no pop-up.
Any thoughts?
Carlos
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Not sure how is the agent-add macro doing its thing, you should look at the "queue show" output from the asterisk cli to see how are the agents added, and if they use Local/XXX@YYY, verify that the button configuration use extension=XXX and queuecontext=YYY. You can see that by running autoconfig-buttons-freepbx.sh. There must be a coincidence between extension/queuecontext in both the queue member and the button config so fop2 can match the inbound call as a queue member call.
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Hi Nicolas,
Just had a further look and Trixbox adds to the queue:
member=Local/500@from-internal/n,0
when adding a static member. If I change this to:
member=SIP/500
then it works as expected. At least until Trixbox is relaoded and overwrites the change.
Tried "queue show" and it gives Local/500@from-internal/n in both v2.6 and 2.8
Also autoconfig-buttons-freepbx.sh gives:[QUEUE/803]
type=queue
extension=803
label=Sales
context=ext-queues
queuecontext=from-internal
extenvoicemail=*again in both v2.6 and 2.8
To add yet more, setting the destination of an inbound route directly to an extension results in the CALLED extension being displayed against the extension in FOP2.
Carlos.
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Look at the extension configuration in the buttons.cfg file, not the queue configuration, and see if they have the queuecontext set and to what value it is set.. it should be "from-internal", but maybe it is set to "from-queue". If that is the case, you must change the autoconfiguration button script to use the correct queuecontext for your setup.
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Yet more info Nicolas,
Using 2 setups, one with static queue members (not working) and one with dynamic members (working), I've had a look at the output of fop2_server -X 15 and here are the major differences:
Static:
127.0.0.1 <- Event: Newchannel
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: Local/500@from-internal-0f29;1
127.0.0.1 <- ChannelState: 0
127.0.0.1 <- ChannelStateDesc: Down
127.0.0.1 <- CallerIDNum:
127.0.0.1 <- CallerIDName:
127.0.0.1 <- AccountCode:
127.0.0.1 <- Uniqueid: 1290532841.6
127.0.0.1 <- Server: 0Dynamic:
127.0.0.1 <- Event: ChannelUpdate
127.0.0.1 <- Privilege: system,all
127.0.0.1 <- Channel: SIP/500-00000009
127.0.0.1 <- Uniqueid: 1290532944.13
127.0.0.1 <- Channeltype: SIP
127.0.0.1 <- SIPcallid: 7fb80dba1efb090159448854183948a7@192.168.200.49
127.0.0.1 <- SIPfullcontact: sip:500@192.168.200.37:5068;transport=udp
127.0.0.1 <- Server: 0127.0.0.1 <- Event: ChannelUpdate
127.0.0.1 <- Privilege: system,all
127.0.0.1 <- Channel: SIP/500-00000009
127.0.0.1 <- Channeltype: SIP
127.0.0.1 <- SIPcallid: 7fb80dba1efb090159448854183948a7@192.168.200.49
127.0.0.1 <- SIPfullcontact: sip:500@192.168.200.37:5068;transport=udp
127.0.0.1 <- Peername: 500
127.0.0.1 <- Server: 0127.0.0.1 <- Event: ExtensionStatus
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Exten: 500
127.0.0.1 <- Context: ext-local
127.0.0.1 <- Hint: SIP/500&Custom:DND500
127.0.0.1 <- Status: 8
127.0.0.1 <- Server: 0Further on we see the reason for the CID, i.e.
Static:
127.0.0.1 <- Event: Newstate
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/500-00000006
127.0.0.1 <- ChannelState: 5
127.0.0.1 <- ChannelStateDesc: Ringing
127.0.0.1 <- CallerIDNum: 500
127.0.0.1 <- CallerIDName:
127.0.0.1 <- Uniqueid: 1290532841.10
127.0.0.1 <- Server: 0127.0.0.1 <- Event: Newstate
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/501-00000007
127.0.0.1 <- ChannelState: 5
127.0.0.1 <- ChannelStateDesc: Ringing
127.0.0.1 <- CallerIDNum: 501
127.0.0.1 <- CallerIDName:
127.0.0.1 <- Uniqueid: 1290532841.11
127.0.0.1 <- Server: 0Dynamic:
127.0.0.1 <- Event: Newstate
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/500-00000009
127.0.0.1 <- ChannelState: 5
127.0.0.1 <- ChannelStateDesc: Ringing
127.0.0.1 <- CallerIDNum: 07866676869
127.0.0.1 <- CallerIDName: Trial: Stem Carl Marshall
127.0.0.1 <- Uniqueid: 1290532944.13
127.0.0.1 <- Server: 0127.0.0.1 <- Event: Newstate
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/501-0000000a
127.0.0.1 <- ChannelState: 5
127.0.0.1 <- ChannelStateDesc: Ringing
127.0.0.1 <- CallerIDNum: 07866676869
127.0.0.1 <- CallerIDName: Trial: Stem Carl Marshall
127.0.0.1 <- Uniqueid: 1290532944.14
127.0.0.1 <- Server: 0It looks as though Asterisk is passing the CallerIDNum as the called extension! Any idea as to where to look in FreePBX to find where this may be generated?
Carlos
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Hi Carlos,
I am not sure where to look, but you might want to traverse the dialplan priority by priority in both cases from the asterisk full log.. quite difficult to do, but I do not see any other way to spot differences? When you refer to static vs dynamic, it is always with Local/XXX@YYY/n queue members?
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Hi Nicolas,
No, the static queue agents show as Local/xxx@yyy/n either when added into the queue via Trixbox queues (same as FreePBX queues) or when added as dynamic using 803* to join queue 803. Both show this when looking at -X 15 output, i.e.:
Static:
127.0.0.1 <- Event: QueueMember
127.0.0.1 <- Queue: 803
127.0.0.1 <- Name: Local/500@from-internal/n
127.0.0.1 <- Location: Local/500@from-internal/n
127.0.0.1 <- Membership: static
127.0.0.1 <- Penalty: 0
127.0.0.1 <- CallsTaken: 0
127.0.0.1 <- LastCall: 0
127.0.0.1 <- CallAnswered: 0
127.0.0.1 <- TalkTime: 0
127.0.0.1 <- Status: 1
127.0.0.1 <- Paused: 0
127.0.0.1 <- Server: 0Or when added via 803* :
127.0.0.1 <- Event: QueueMember
127.0.0.1 <- Queue: 803
127.0.0.1 <- Name: Local/500@from-internal/n
127.0.0.1 <- Location: Local/500@from-internal/n
127.0.0.1 <- Membership: dynamic
127.0.0.1 <- Penalty: 0
127.0.0.1 <- CallsTaken: 0
127.0.0.1 <- LastCall: 0
127.0.0.1 <- CallAnswered: 0
127.0.0.1 <- TalkTime: 0
127.0.0.1 <- Status: 1
127.0.0.1 <- Paused: 0
127.0.0.1 <- Server: 0However when added as a queue agent using FOP2, we get:
127.0.0.1 <- Event: QueueMember
127.0.0.1 <- Queue: 803
127.0.0.1 <- Name: SIP/500
127.0.0.1 <- Location: SIP/500
127.0.0.1 <- Membership: dynamic
127.0.0.1 <- Penalty: 0
127.0.0.1 <- CallsTaken: 0
127.0.0.1 <- LastCall: 0
127.0.0.1 <- CallAnswered: 0
127.0.0.1 <- TalkTime: 0
127.0.0.1 <- Status: 1
127.0.0.1 <- Paused: 0
127.0.0.1 <- Server: 0Carlos.
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Nicolas,
This issue looks similar to one posted earlier which I've just noticed:
Did you manage to find an answer to that one?
Carlos.
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Hi Carlos,
I already said it before, if the AMI events says the calleris is "INCORRECTNUMBER", fop2 will display that as the callerid number. Regarding the queue members, you can configure fop2 to use the same type as FreePBX by setting the queuechannel for the button.
So from what you say it seems that the problem is with Local channels, it is not static vs dynamic, but Local vs SIP.
Please capture the output from fop2_server -X 1 while doing a call which shows the incorrect clid and send me the capture privately so I can review it here and see if I can find the issue.
Best regards,
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Nicolas,
Thanks for all you help on this one. I'll add this section even though you sorted it out privately since there may be others with the same issues with Trixbox 2.8.x
/etc/asterisk/manager.conf has the recommended lines of:
read=system,call,log,verbose,command,agent,user,originate
write=system,call,log,verbose,command,agent,user,originateChanging this to:
read=all
write=allsolved the problem.
Many thanks for all your hard work.
Carlos.
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