Asterisk 13 Debian FOP2 2.29 See attached file, in a conference call using MeetMe, the status of the listener is not shown correctly. Even if a person is muted (*1) the icon shows they are talking. The buttons seem to work but is very confusing of who is doing the talking when several icons are showing "talker".
If you need more info, let me know and will provide.
Another picture of what is not working. The extension dialing into the conference should be showing "656" as the number that was called to access the conference, not "dstring" or "out". This worked prefectly with Asterisk 11 and FOP2 2.27. That is why I am seeing a problem. The difference between the two is very noticeable.
And you can see the numbers are out of order in the conference.
Gene
I will have to test it.. Asterisk 13 changed the AMI in a deep and profound way, so every little event and bit needs to be addressed to make it work and so it is also backwards compatible. I do not know why someone decide to change so many headers on AMI, perhaps just for the sake of breaking applications.
I do not have an Asterisk 13 machine handy, so it will take some time to test it and get it working.
Just tried, both with confbridge and meetme, and I cannot reproduce anything. Talker detection works fine and is shown correctly on conference buttons. To get you an idea of the events that modify the participant talking icon, you can look at those when you run fop2_server in debug level 1 mode at least:
127.0.0.1 <- Event: MeetmeTalking
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Meetme: 5000
127.0.0.1 <- Channel: PJSIP/309-0000000b
127.0.0.1 <- ChannelState: 6
127.0.0.1 <- ChannelStateDesc: Up
127.0.0.1 <- CallerIDNum: 609
127.0.0.1 <- CallerIDName: Maria
127.0.0.1 <- ConnectedLineNum: <unknown>
127.0.0.1 <- ConnectedLineName: <unknown>
127.0.0.1 <- AccountCode:
127.0.0.1 <- Context: from-internal
127.0.0.1 <- Exten: STARTMEETME
127.0.0.1 <- Priority: 4
127.0.0.1 <- Uniqueid: 1425640211.99
127.0.0.1 <- Status: on
127.0.0.1 <- User: 2
127.0.0.1 <- Event: MeetmeTalking
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Meetme: 5000
127.0.0.1 <- Channel: PJSIP/309-0000000b
127.0.0.1 <- ChannelState: 6
127.0.0.1 <- ChannelStateDesc: Up
127.0.0.1 <- CallerIDNum: 609
127.0.0.1 <- CallerIDName: Maria
127.0.0.1 <- ConnectedLineNum: <unknown>
127.0.0.1 <- ConnectedLineName: <unknown>
127.0.0.1 <- AccountCode:
127.0.0.1 <- Context: from-internal
127.0.0.1 <- Exten: STARTMEETME
127.0.0.1 <- Priority: 4
127.0.0.1 <- Uniqueid: 1425640211.99
127.0.0.1 <- Status: off
127.0.0.1 <- User: 2
127.0.0.1 <- Duration: 1
Regarding the dialed number, not sure where are you seeing 'dstring' or 'out', but those come from your own dialplan broadcasted via AMI events. As the manager changed, perhaps the actual numbers are not shown on the same headers as before.
However, dialing locally into a conference on my test system worked well, but I am dialing the conference locally, not from a DID or external numbers.
In any case, I cannot do anything without having access to the debug level 1 from fop2_server, to inspect the manager events you are receiving.
Best regards,
For the benefit of others who need to do a debug:
I created directory /var/log/fop2
Then:
changed the /etc/default/fop2 to :
OPTIONS="-d -p $PIDFILE -X 1 --logdir /var/log/fop2 --audit /var/log/fop2/audit.log"
I see the capture is working and will forward you info from a conference.
The debug file is 8000kb, I keep getting an error message after uploading that the file is empty.
How do I get it to you?
I sent you an email, there was one conference in this debug file that shows the problem.
custom-conf9520 it had four participants. While one person was talking, all the icons were "lit up" showing that all were talking at the same time, even the ones who were muted.
Hi Gene, I have sent you download links for you to try an updated version. let me know the outcome.
Best regards,
Hi admin!
I'am testing version 2.29 of FOP2 with Asterisk 13 and I've also get a "dstring" and "recordcheck" when dialing, please take look at the screenshot.
Hi macros,
Your screenshots do not show anything related to conference buttons? You should have posted on a different/new thread... What did you dial? An external number? Going through a trunk? What kind, SIP?
When I dial internal virtual number 5001 redirected to IVR it shows dstring, when I dial external phone number through a SIP trunk I get recordchek.
Same on Asterisk 12.
Are you using 2.29.00 ? /usr/local/fop2/fop2_server -v
Yep, sure:
[root@TestCC ~]# /usr/local/fop2/fop2_server -v
fop2_server version 2.29.00
[root@TestCC ~]# asterisk -V
Asterisk 12.8.2
Do you use any particular backend to configure asterisk? FreePBX? Thirdlane? PBXWare?
Yes, FreePBX 12.0.54.
Hi, Nicolas!
Is there any problem with FreePBX 12 backend? Which back could you advice please?
No, there is no problem at all
Than what could be the problem with my installation? Our company is licensed user of fop2, we are using fop2 2.27 and planning to buy a license update, so we could move to Asterisk 13, FreePBX 12 and FOP 2.29.
I have tested again in Asterisk 13, latest FreePBX. What you see happens when a call is NOT bridged, and because FreePBX changed the way that runs checks on recording and other stuff for doing Gosubs.. that generates events NewExten with priority 1 and an extension used for what before was a macro (not a numerical extension). That text is cleared as soon as the call is bridged/connected to the other leg.
Anyways, I made some checks to not set a dialed number on newexten/priority 1 if it is not a number. So it will skip no numeric extensions that you might traverse in your dialplan before the actual dial is being done.
Is a fairly minor issue, and related to the way the dialplan changed in the latest FreePBX version. FOP2 2.29.01 will prevent those from happen.
http://download.fop2.com/fop2-2.29.01-centos-i386.tgz
http://download.fop2.com/fop2-2.29.01-centos-x86_64.tgz
(2.29.01 is not yet final)
Hi Nicolas,
Thanks for the update, now called numbers shows correctly.
But I have another question. Is it normal to have no possibility to edit or even view user setting in FOP2 Manager? I'am clicking on user but nothing happens, I just can remove it and create again. If you need any additional info please let me know.
No idea what could be happening... maybe opening the javascript console in the browser will output some error? Perhaps you have some kind of extension/plugin that interferes with the page? (once I have issues when I was not able to click on some parts of an application, and it was an extension that added a non visible absolute positioned div above that part of the page) Did you try a different browser?
