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No recognition of incoming calls

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  • N
    N
    nicolas
    wrote on last edited by
    #12

    Most probably yes.. a difference in the ring group dialplan for FreePBX, or something to that effect.

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      CTJackal
      wrote on last edited by
      #13

      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.

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      • N
        N
        nicolas
        wrote on last edited by
        #14

        Do not check the modules, but the dialplans or agi scripts. there are parts of the dialplan that are static (not _additional or _custom files).

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          CTJackal
          wrote on last edited by
          #15

          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

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            CTJackal
            wrote on last edited by
            #16

            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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              CTJackal
              wrote on last edited by
              #17

              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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              • N
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                nicolas
                wrote on last edited by
                #18

                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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                  CTJackal
                  wrote on last edited by
                  #19

                  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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                  • N
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                    nicolas
                    wrote on last edited by
                    #20

                    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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                    • N
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                      nicolas
                      wrote on last edited by
                      #21

                      The thing is that as far as I know , FreePBX 2.6 or 2.7 introduced the from-queue context, so from-internal is used for older freepbx install.

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                        CTJackal
                        wrote on last edited by
                        #22

                        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: 7fb80dba1efb090159448854183948a7@192.168.200.49
                        127.0.0.1 <- SIPfullcontact: sip:500@192.168.200.37:5068;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: 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: 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

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                        • N
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                          nicolas
                          wrote on last edited by
                          #23

                          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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                            CTJackal
                            wrote on last edited by
                            #24

                            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.

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                              CTJackal
                              wrote on last edited by
                              #25

                              Nicolas,

                              This issue looks similar to one posted earlier which I've just noticed:

                              View Post 328

                              Did you manage to find an answer to that one?

                              Carlos.

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                              • N
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                                nicolas
                                wrote on last edited by
                                #26

                                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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                                • C
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                                  CTJackal
                                  wrote on last edited by
                                  #27

                                  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.

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