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: 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
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.
Hi Nicolas,
After some more experimentation, here are my finding:
Trixbox 2.6 + FOP 2.11
Inbound route straight to extension:
FOP2 shows calling number at extension, Calling number and name shown on pop-up.
Inbound route to queue:
FOP2 shows calling number on all extensions which are queue agents. Shows calling number and name in queue. Shows calling name and number in pop-up.
Trixbox 2.8 + FOP 2.11
Inbound route straight to extension:
FOP2 shows [u:3bfxb89z]called[/u:3bfxb89z] extension number at extension e.g ext 502 shows 502 calling. No pop-up.
Inbound route to queue:
FOP2 shows [u:3bfxb89z]called[/u:3bfxb89z] extension number at extension. Shows [u:3bfxb89z]calling[/u:3bfxb89z] number [u:3bfxb89z]and[/u:3bfxb89z] name in queue. No pop-up.
(Each extension shows own number as calling number e.g. ext 511 shows 511 as calling number, 512 shows 512)
I can only assume that FOP2 must be being given the calling number as it appears in the queue, but why isn't the extension showing the calling number (only shows own extension number calling) and there's no pop-up?
Thanks
Carlos.
Just to add a little more!
I realised that I needed to stop the fop2 server and restart with -X 15 . Silly me!
I've also noticed that internal calls show:
500 Carlos
Line1 device
Line2 inactive
501 Mobile
Line1 501
Line 2 inactive
when calling extension 501 from extension 500
What's the "device" mean?
Carlos
Sorry, forgot to say that netstat -nta gives:
0.0.0.0:4444 0.0.0.0:* LISTEN
0.0.0.0:4445 0.0.0.0:* LISTEN
as I changed FOP1 to port 4444
Carlos
Output of autoconfig-buttons-freepbx.sh is normal.
However /usr/local/fop2/fop2_server -X 15 generates: Can't listen to port 4445
Tried to search for this on forum, but the search seems to be not working at present. Regards this as too common words.
Thanks,
Carlos
Hi,
Yes. Received "Connection to manager OK!"
Carlos
Installed FOP2 2.11 on Trixbox v 2.8.0.5 and am getting no recognition of incoming calls in FOP2. I've used the visual phonebook database as a lookup source for Trixbox and that's picking the correct record to display on the phone. No SQL query originated by FOP2 though. Extension simply shows inactive on both lines.
Tried to compare this to a working setup using Trixbox 2.6 and can't see any difference in my configuration. Any ideas on what's going wrong or how I can investigate further?
Thanks
Carlos.
Thanks Nicolas,
You're correct, there's no appropriate output. Checking the MySQL databases, there's no such table as 'fop2users' within the 'asterisk' database, hence autoconfig-users.sh produces no output other than the button file.
How was that supposed to be added? During the installation of FOP2?
Cheers,
Carlos.
Hi,
Just upgraded to v2.11 and find that I can't get a login as an extension. FOP2 is running on port 4445 and appears to be OK, however I enter the extension number and password on starting the flash client, then it attempts to connect to the server but fails. fop2_server -X 15 gives:
192.168.200.33 => <?xml version="1.0"?>
<!DOCTYPE cross-domain-policy SYSTEM "http://www.macromedia.com/xml/dtds/cross-domain-policy.dtd">
<cross-domain-policy>
<allow-access-from domain="*" secure="false" to-ports="4445" />
</cross-domain-policy>
** CLEAN SOCKET connection lost removing socket IO::Socket::INET=GLOB(0xab01ce8)
** CLEAN SOCKET flash client connection lost
No flash clients connected
192.168.200.33 <= <msg data="GENERAL|contexto|1|" />
192.168.200.33 => { 'btn': '0', 'cmd': 'key', 'data': 'ZiQmv0sh6uNJNcm7WbgeuLpCg', 'slot': '' }
192.168.200.33 => { 'btn': '0@GENERAL', 'cmd': 'version', 'data': '2.11', 'slot': '' }
192.168.200.33 <= <msg data="1|auth|501|e1a60eeb9f4026059a177f3f3e22bb4e" />
192.168.200.33 => { 'btn': '0', 'cmd': 'incorrect', 'data': '0', 'slot': '' }
** CLEAN SOCKET connection lost removing socket IO::Socket::INET=GLOB(0xaa38c44)
** CLEAN SOCKET flash client connection lost
No flash clients connected
Any ideas?
Cheers,
Carlos
Hi Nicolas,
Just installed Asterisk 1.4 in order to test this. The equivalent output is:
127.0.0.1 <- Event: Newchannel
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/87.127.240.98-08595bf8
127.0.0.1 <- State: Down
127.0.0.1 <- CallerIDNum: 0769
127.0.0.1 <- CallerIDName: 0769
127.0.0.1 <- Uniqueid: 1266495006.279
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/87.127.240.98-08595bf8
127.0.0.1 <- State: Ring
127.0.0.1 <- CallerID: 0769
127.0.0.1 <- CallerIDName: 0769
127.0.0.1 <- Uniqueid: 1266495006.279
127.0.0.1 <- Server: 0
The main difference between this and Asterisk 1.6 appears to be the "CallerIDName:" being presented as the caller ID in v1.4, but presented as the CID Name Prefix + caller ID in v1.6.
Could this be why it's working in 1.4 and not in 1.6?
Carlos.
Hi Nicolas,
Just a thought, but could it be that the problem lies here:
127.0.0.1 <- Event: NewCallerid
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/87.xxx.xxx.242-09ea55c0
127.0.0.1 <- CallerIDNum: 01700
127.0.0.1 <- CallerIDName: Main Number - 01700
127.0.0.1 <- Uniqueid: 1266074985.21
127.0.0.1 <- CID-CallingPres: 0 (Presentation Allowed, Not Screened)
127.0.0.1 <- Server: 0
This is the output from Asterisk 1.6, and I think that the 1.4 version would output:
127.0.0.1 <- CallerID: 017******00
Rather than CallerIDNum:
Could this be the issue?
Carlos.
