Realtime report_Status "RING" although call hang up

[luckypig]

When I have a call inbound. But no agent anserver, so timeout and call hang up. But, On Realtime Report screen, status of agent "RING" athoung this call hangup and no call is coming.
I sent link to everyon can see image

https://drive.google.com/file/d/1ssYSD_ ... sp=sharing

Please help me. thanks so much

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) To be clear: You walked up to a car dealer and said "my car doesn't work" without providing make/model/year/color/age or even allowing them to look at the car.

4) It's not very cool to put images on sites that log entry. Use a "public access without logging" system. That is not facebook, google, or any one of several others. I will give you credit, though, at least it's not locked (often google drive and such require an active account and the image can not be seen anonymously ... so you did better than most! lol).

5) Does this happen for ALL inbound calls? Some? Or do you only have one type of inbound call from a single provider?

6) Describe the pathway of the call. EG: Ordinarily it would arrive at your server from a Carrier, then pass to "trunkinbound" based on the sip account settings in "admin->carriers", then use the stock "trunkinbound" in extensions.conf (since you didn't mention having edited that file), and then to the DID module and then what route does it take? Call Menu? Ingroup? Does it DROP to a different ingroup or call menu after the first one because it's after hours? give us a hint, here.

7) In case that wasn't clear enough: If you have edited extensions.conf (or any "included" file related to extensions.conf) this would be a really good time to UNedit that file and test again.

Did you follow the Vicidial Manager's Manual to configure the inbound route/DID?

[luckypig]

At first, I'm sorry for mistake.
I install virtual server on ESXi include :2 servers: server (Web and DB), server Telephony
I install from Vicibox_v.6.0.x86_64-6.0_standard.iso. Asterisk version 1.4.21.2. VERSION: 2.14-717a
BUILD: 190724-1603
© 2019 ViciDial Group. Carrier is trunked with ITSP via sip trunk. I tested inbound and outboud call. IT's ok

I have this error for INBOUND calls have status TIMEOUT/WAITO.
EX: one inbound call, customer press 1 into CSKH group. And agent in this group status CLOSER. So that, this call was ringing to this agent. A long time no anserver from agent, customer press 1 for recall (WAITO) or TIMEOUT, call hangup. But on realtime report screen, status of agent still appear RING stead of (CLOSER/READY)

We have many log from server, which log do you want send to you? please help me. Thanks

[williamconley]

Why did you install with a version of Vicibox that is so old? Why not install with Vicibox 8.1.2?

Note that sometimes the Asterisk Version in /etc/astguiclient.conf and/or "admin->servers" is not properly matched to the actual Vicidial Version installed in the server. So what Vicidial version is listed for each of those two places?

Next up: Virtual Vicidial is a Very Bad Idea. Use a physical server unless you are only using this for testing with one or two agents to get the system up and running. Before you launch for real, put it in physical hardware or it will very likely fail as soon as you try to fully load it with very unusual problems.

[luckypig]

I update realtime report

NO LIVE CALLS WAITING
1 agents logged in 0 agents in calls 0 agents waiting 0 paused agents 0 agents in dead calls 0 agents in dispo

Agents Time On Calls Campaign: |ALL-ACTIVE| 2019-08-21 14:32:00
+----------------+------------------------+-----------+-----------------+---------+------------+-------+------+------------------
| STATION | USER SHOW ID INFO | SESSIONID | STATUS PAUSE | MM:SS | CAMPAIGN | CALLS | HOLD | IN-GROUP
+----------------+------------------------+-----------+-----------------+---------+------------+-------+------+------------------
| SIP/9998 *| phuongtest + | 8600051 | RING | 32:10 | INBOUND | 1 |
+----------------+------------------------+-----------+-----------------+---------+------------+-------

1 agents logged in on all servers

[luckypig]

I update realtime report

NO LIVE CALLS WAITING
1 agents logged in 0 agents in calls 0 agents waiting 0 paused agents 0 agents in dead calls 0 agents in dispo

Agents Time On Calls Campaign: |ALL-ACTIVE| 2019-08-21 14:32:00
+----------------+------------------------+-----------+-----------------+---------+------------+-------+------+------------------
| STATION | USER SHOW ID INFO | SESSIONID | STATUS PAUSE | MM:SS | CAMPAIGN | CALLS | HOLD | IN-GROUP
+----------------+------------------------+-----------+-----------------+---------+------------+-------+------+------------------
| SIP/9998 *| phuongtest + | 8600051 | RING | 32:10 | INBOUND | 1 |
+----------------+------------------------+-----------+-----------------+---------+------------+-------

1 agents logged in on all servers

[luckypig]

At the currently, i install on virtual machine to test cluster server. I used to install this version, but all in one server, and calls're good

[luckypig]

I'm sorry about my information.
Version asterisk in /etc/astguiclient.conf VARasterisk_version => 1.8
And on admin/servers/TEle: Asterisk Version: 1.8.29.0-vici

[williamconley]

But your original post shows "Asterisk version 1.4.21.2"?

If this is true, then you will need to re-run the "install.pl" script after fixing the asterisk version. Be sure to choose "sample conf files" so the proper version of sip.conf and extensions.conf are utilized, and be sure to re-edit externip in sip.conf (along with any other manual edits to .conf files you've made).

And reboot.

[luckypig]

I found calls had status :WAITO/TIMEOT, ring_callerid field on vicidial_live_agents table is not empty, although calls hangup. And I think this a reason why on realtime report, status of agent always RING although calls end. How to make this field is empty. Because, I noticed, for calls had status: DROP or answer, this field's empty and after call, status of agent on realtime report is CLOSER.

[ambiorixg12]

I'm sorry about my information.
Version asterisk in /etc/astguiclient.conf VARasterisk_version => 1.8
And on admin/servers/TEle: Asterisk Version: 1.8.29.0-vici

You will need to upgrade and test using the latest stable version of VICIbox, because you re using an old version of VICI and Asterisk, and in case there is a bug with these versions you wont get support unless you upgrade your system

[luckypig]

I installed the lasted version vicibox
/etc/astuguiclient.conf VARasterisk_version => 13
admin/server/tele: Asterisk Version 13.21.1-vici

I still get this error. Although, inbound calls hang up (TIMEOUT), on realtime report screen, status of agent is "RING"

[williamconley]

Have you customized any of the .conf files in your system?

Do you have any custom dialplan entries anywhere in the system outside the Carrier settings?

Provide your Carrier "dialplan entry" field (X anything private, like carrier name if you don't want to share that, or just replace with CARRIER).

Also provide a single example of the entire call from start to end (path? DID -> routes to where? and so on). Consider posting an asterisk CLI example from start to finish of one call (not 3000 lines of unrelated code, just one call preferably with no other traffic on the system If Possible )