Member
Last active 14 years ago
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,originate
Changing this to:
read=all
write=all
solved the problem.
Many thanks for all your hard work.
Carlos.
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.
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: 0
Or 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: 0
However 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: 0
Carlos.
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: 0
Dynamic:
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: [email protected]
127.0.0.1 <- SIPfullcontact: sip:[email protected]:5068;transport=udp
127.0.0.1 <- Server: 0
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 <- Channeltype: SIP
127.0.0.1 <- SIPcallid: [email protected]
127.0.0.1 <- SIPfullcontact: sip:[email protected]:5068;transport=udp
127.0.0.1 <- Peername: 500
127.0.0.1 <- Server: 0
127.0.0.1 <- 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: 0
Further 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: 0
127.0.0.1 <- 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: 0
Dynamic:
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: 0
127.0.0.1 <- 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: 0
It 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
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.
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
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
Hi Nicolas,
I'll try and get a report from fop2_server -X 15 for you if this will help guide you to the issue.
Carlos
Hi Nicolas,
Done a "diff" on the freepbx core modules, extensions.conf, sip.conf, iax.conf, and features.conf, between the Trixbox 2.8 (not working) and the Trixbox 2.6 (working) versions. Both exactly the same!
Any other thoughts?
Carlos.
Hi Nicolas,
Thanks for the offer, but I won't put you to trouble just yet ;-)
Do you mean a difference in the extensions.conf file?
Carlos.