FOP2 Community Forum

I failed to specify that the one with the Chrome issue is running on Druid and was installed via tar.

Having issues with 2.24. I have upgraded our PBX to Elastix 2.2 and had FOP2 2.2.4 loaded in via the add-ons, but just noticed I am getting a manager error no matter what I try. Verified it is set up properly.

I only noticed this as I upgraded another PBX from FOP2 2.23 to 2.24 yesterday to see if it fix issues with Chrome they were having, but in fact pretty much nothing worked after the upgrade, especially with the queues not showing members. I finally rolled back to 2.23 and everything immediately worked again.

Also curious that the add-on screen lists the version to download as 2.25, but 2.24 is what downloads. Just tried an uninstall and reinstall but no change in Elastix 2.2.

To fix I verified the external IP, which turned out to somehow be wrong, but it was adding a canreinvite=no that appears to have resolved the problem.

I set up a new FOP2 install this weekend on a Paetec SIP service and I am getting one way audio. The settings are the same as in my other installs. Dialing from the phone audio is fine, dialing from FOP2 the number called you cannot hear what they say. Paetec has a different setup than my other SIP providers so I assume that is the issue. Wondering where I can try to hunt for a solution. For now I just turned the phonebook off so no one tries to use it.

BTW: Bought two licenses this weekend, one for this client and one for another. Love FOP2!

When I pause someone in a queue, their state icon doesn't change to show as paused until the 60 seconds are up for what I have it set for poll_interval.

I didn't have it, but I have added it now. Still same behavior though.

[SIP/phone-aastra-1]
type=extension
extension=5001
context=druid-default
label="David"
mailbox=5001@druid-default
extenvoicemail=5001@vm
queuecontext=ael-druid-originate
queuechannel=Local/5001@ael-druid-originate/n
originatechannel=Local/5001@ael-druid-originate
email=dave@sollos.com
#privacy=all
group=sales

I just tried it with context =ael-druid-originate, but still not changing the button state.

Set the poll_interval to 60 and it changes correctly every minute.

Here is some of the fop2 debug:

192.168.4.120   <= <msg data="QUEUE/VOIP|queuepause|Local/5001@ael-druid-originate/n|20c0b629d5ec95ddb6177162cbcdaf30" />

-- PROCESS_FLASH_COMMAND origen QUEUE/VOIP accion queuepause destino Local/5001@ael-druid-originate/n

-- PROCESS_FLASH_COMMAND password 20c0b629d5ec95ddb6177162cbcdaf30

VALIDAR USUARIO 5001

Validation ok, have all permissions

127.0.0.1       -> Action: QueuePause
127.0.0.1       -> Interface: Local/5001@ael-druid-originate/n
127.0.0.1       -> Paused: 1

** MAIN End of block from localhost

127.0.0.1       <- Response: Success
127.0.0.1       <- Message: Interface paused successfully
127.0.0.1       <- Server: 0

** MAIN AMI event received...
** MAIN No 'Event' nor 'End'. Erasing block...

Response: Success
Message: Interface paused successfully
Server: 0

** MAIN End of block from localhost

127.0.0.1       <- Event: QueueMemberPaused
127.0.0.1       <- Privilege: agent,all
127.0.0.1       <- Queue: voip2
127.0.0.1       <- Location: Local/5001@ael-druid-originate/n
127.0.0.1       <- MemberName: Local/5001@ael-druid-originate/n
127.0.0.1       <- Paused: 1
127.0.0.1       <- Status: 1
127.0.0.1       <- Server: 0

** MAIN AMI event received...
** MAIN There's an 'Event' in the event block
** MAIN Event detected block_count = 0
** MAIN There are 0 blocks for processing
** MAIN Answer block cleared

** DIGEST_EVENT: start

** PROCESA_BLOQUE QUEUEMEMBERPAUSED 0

Not a reference at all

** MAIN End of block from localhost

127.0.0.1       <- Event: QueueMemberPaused
127.0.0.1       <- Privilege: agent,all
127.0.0.1       <- Queue: voip
127.0.0.1       <- Location: Local/5001@ael-druid-originate/n
127.0.0.1       <- MemberName: Local/5001@ael-druid-originate/n
127.0.0.1       <- Paused: 1
127.0.0.1       <- Status: 1
127.0.0.1       <- Server: 0

** MAIN AMI event received...
** MAIN There's an 'Event' in the event block
** MAIN Event detected block_count = 0
** MAIN There are 0 blocks for processing
** MAIN Answer block cleared

** DIGEST_EVENT: start

** PROCESA_BLOQUE QUEUEMEMBERPAUSED 0

Not a reference at all

** MAIN AMI event received...
** MAIN Processing command received from flash clients...

192.168.4.120   <= <msg data="QUEUE/VOIP|queueunpause|Local/5001@ael-druid-originate/n|20c0b629d5ec95ddb6177162cbcdaf30" />

-- PROCESS_FLASH_COMMAND origen QUEUE/VOIP accion queueunpause destino Local/5001@ael-druid-originate/n

-- PROCESS_FLASH_COMMAND password 20c0b629d5ec95ddb6177162cbcdaf30

VALIDAR USUARIO 5001

Validation ok, have all permissions

127.0.0.1       -> Action: QueuePause
127.0.0.1       -> Interface: Local/5001@ael-druid-originate/n
127.0.0.1       -> Paused: 0

** MAIN End of block from localhost

127.0.0.1       <- Response: Success
127.0.0.1       <- Message: Interface unpaused successfully
127.0.0.1       <- Server: 0

** MAIN AMI event received...
** MAIN No 'Event' nor 'End'. Erasing block...

Response: Success
Message: Interface unpaused successfully
Server: 0

** MAIN End of block from localhost

127.0.0.1       <- Event: QueueMemberPaused
127.0.0.1       <- Privilege: agent,all
127.0.0.1       <- Queue: voip2
127.0.0.1       <- Location: Local/5001@ael-druid-originate/n
127.0.0.1       <- MemberName: Local/5001@ael-druid-originate/n
127.0.0.1       <- Paused: 0
127.0.0.1       <- Status: 1
127.0.0.1       <- Server: 0

** MAIN AMI event received...
** MAIN There's an 'Event' in the event block
** MAIN Event detected block_count = 0
** MAIN There are 0 blocks for processing
** MAIN Answer block cleared

** DIGEST_EVENT: start

** PROCESA_BLOQUE QUEUEMEMBERPAUSED 0

Not a reference at all

** MAIN End of block from localhost

127.0.0.1       <- Event: QueueMemberPaused
127.0.0.1       <- Privilege: agent,all
127.0.0.1       <- Queue: voip
127.0.0.1       <- Location: Local/5001@ael-druid-originate/n
127.0.0.1       <- MemberName: Local/5001@ael-druid-originate/n
127.0.0.1       <- Paused: 0
127.0.0.1       <- Status: 1
127.0.0.1       <- Server: 0

** MAIN AMI event received...
** MAIN There's an 'Event' in the event block
** MAIN Event detected block_count = 0
** MAIN There are 0 blocks for processing
** MAIN Answer block cleared

** DIGEST_EVENT: start

** PROCESA_BLOQUE QUEUEMEMBERPAUSED 0

Not a reference at all

** MAIN AMI event received...
** MAIN Processing command received from flash clients...

192.168.4.120   <= <msg data="1|ping||" />

-- PROCESS_FLASH_COMMAND origen 1 accion ping destino 

-- PROCESS_FLASH_COMMAND password 

192.168.4.120   => { 'btn': '0', 'cmd': 'pong', 'data': '0', 'slot': '' }

Not using FreePBX, this is Druid UCS 5.3.9. I will try the debug thanks!

If I pause or unpause a member of a queue, they are properly paused, but the interface is not showing this at all.

So if each of my locations has Aastra phones which are automatically named phone-aastra-1, phone-aastra-2, etc I would have to rename the phones at all but one location in order to use one FOP2 interface to view and manage all locations, correct? If location A has phone-aastra-1 and so does location B this isn't going to work even if a line is added in the button configuration to specify what server to associate that button with.

Either is fine with me. It may be something nutty on my system. I am pasting the buttons.cfg below with email removed:

[SIP/phone-aastra-1]
type=extension
extension=5001
context=druid-default
label="David"
mailbox=5001@druid-default
extenvoicemail=5001@vm
email=xxxxxx@xxx.com
#privacy=all
group=sales

[SIP/phone-aastra-4]
type=extension
extension=5025
context=druid-users
label="Natalie"
mailbox=5025@druid-default
extenvoicemail=995025@druid-default
email=xxxxxx@xxx.com
#privacy=all

[SIP/phone-aastra-5]
type=extension
extension=5026
context=druid-users
label="Daniel"
mailbox=5026@druid-default
extenvoicemail=995026@druid-default
email=xxxxxx@xxx.com
#privacy=all
group=sales

[QUEUE/voip]
type=queue
label=Main Queue
extension=6002
context=druid-default

[CONFERENCE/7000]
type=conference
label=Conference
extension=7000
context=druid-default

[SIP/Vitelity]
type=trunk
label=Vitelity

[PARK/default]
type=park
extension=700
label=Parking
context=parkedcalls

I had the "No Way" message when I configured my phone book last week and it turned out to be my Safari had private browsing enabled by mistake. It worked fine in Firefox and was good in Safari once Private Browsing was disabled. Double check your browser settings or try another browser to be sure.

Actually I was about to reset the cache when I saw that private browsing had been enabled by mistake. I think that is what caused it not to work. Suddenly everything looks correct, even the park moves to the right place.

Here is a link to the result of that test. [url:3jq6foma]http://www.sollos.com/about-sollos-technology-solutions/downloads/doc_download/52-fop21-pick-up-from-park.html[/url:3jq6foma]

In the habit of using Safari 4.0.5 on Mac OS X and had many features not working for me with FOP 2.1. Finally launched Firefox 3.6.3 and the Visual Address Book works (in Safari it simply says "No Way"), picking up a call works (it was working with Safari in version 2.0) and the Call Park shows the parked call correctly. Might be more, but these are what I have seen. I have no problem at all using Firefox instead - it looks much better actually, just leaving this note in case anyone else spends most of a day working on problems that don't exist.

Send a call the the call park works fine and if you pick up the call by dialing that extension from the phone all is well, but if I use the pick up option from the call park in FOP2.1, FOP2.1 crashes and I have to start it again.

Restarted everything and the problem with ringing all extensions is now gone.

I just saw you had 2.1 out so I upgraded to that and used the new fop2cfg file. One of those fixed the problem for me as it works now. However its dialing on all the available lines on my Aastra so I need to work that out. Progress comes in steps.

I have the context issue seemingly resolved, but now my ITSP is rejecting the call as failing to authenticate.

Its not passing my caller ID as seen below. Calling between extensions this information is properly passed.

 Executing [s@macro-ael-druid-trunkdial:71] NoOp("SIP/phone-aastra-1-0828ae58", "Finish if-ael-druid-trunkdial-946") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:72] Set("SIP/phone-aastra-1-0828ae58", "exten=""") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:73] Set("SIP/phone-aastra-1-0828ae58", "name=""") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:74] Set("SIP/phone-aastra-1-0828ae58", "cid=""") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:75] GotoIf("SIP/phone-aastra-1-0828ae58", "0?76:83") in new stack
    -- Goto (macro-ael-druid-trunkdial,s,83)
    -- Executing [s@macro-ael-druid-trunkdial:83] GotoIf("SIP/phone-aastra-1-0828ae58", "0?84:86") in new stack
    -- Goto (macro-ael-druid-trunkdial,s,86)
    -- Executing [s@macro-ael-druid-trunkdial:86] Set("SIP/phone-aastra-1-0828ae58", "options="WTKo"") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:87] NoOp("SIP/phone-aastra-1-0828ae58", "Finish if-if-ael-druid-trunkdial-947-949") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:88] NoOp("SIP/phone-aastra-1-0828ae58", "Finish if-ael-druid-trunkdial-947") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:89] Verbose("SIP/phone-aastra-1-0828ae58", "3|- Checking if trunk is available") in new stack
    -- - Checking if trunk is available
    -- Executing [s@macro-ael-druid-trunkdial:90] ChanIsAvail("SIP/phone-aastra-1-0828ae58", "SIP/Vitelity") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:91] GotoIf("SIP/phone-aastra-1-0828ae58", "0?92:99") in new stack
    -- Goto (macro-ael-druid-trunkdial,s,99)
    -- Executing [s@macro-ael-druid-trunkdial:99] Verbose("SIP/phone-aastra-1-0828ae58", "3|- Normal trunk operation") in new stack
    -- - Normal trunk operation
    -- Executing [s@macro-ael-druid-trunkdial:100] Dial("SIP/phone-aastra-1-0828ae58", "SIP/Vitelity/4048867603||WTKo") in new stack
    -- Called Vitelity/4048867603
[Apr  1 19:12:20] NOTICE[2495]: chan_sip.c:12471 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@64.2.142.93>;tag=as42b9b750'
    -- SIP/Vitelity-082d9360 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)