REMOTE AGENTS HANGUP CALL ISSUE

[rrb555]

ViciBox Server v.8.0.1
VERSION: 2.14-694a
BUILD: 181005-1738
© 2018 ViciDial Group
Asterisk 13.21.0-vici

Vicidial Bug Tracker 0001144


Remote Agents
User ID: 9900
Number of lines: 1
Server IP: 192.168.XX.XX
External Extension: 501
Extension Group: NONE
Status: Active
Campaign: TESTCAMP
On-Hook Agent: Y
On-Hook Ring Time: 15
Inbound Groups: TESTCAMPINB

Remote Agents are working fine, calls are pushing to User ID 9900, though when User 9900 hangup the call, the call is still in the system and Real-Time Main Report still show calls is still connected. Call will only be disconnected if the customer disconnects the call.

Asterisk logs upon UserID 9900 ends the call

Code: Select all

  == Spawn extension (default, 78600072, 1) exited non-zero on 'SIP/501-000002ed'
    -- Executing [h@default:1] AGI("SIP/501-000002ed", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
    -- <SIP/501-000002ed>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0


[williamconley]

Your signature shows Vicibox 6.0 (instead of "6.0.X" as it should) but your post shows Vicibox 8.0 (also instead of 8.0.X as it should). So please do two things: Fix the conflict (choose an installation CD) and put the full version of the installer in your update. Onward

I'm going out on a limb and assuming that the above is the result of you originally installing on an older version of Vicibox and upgrading a few times, then re-installing on a freshly configured server and restored your DB onto the new server.

If that's the case, then your settings in Admin->Servers no longer agree with the version of Asterisk that you actually have installed. Additionally it's possible you have other settings that aren't matched properly. Performing a new "perl install.pl --copy_sample_conf_files". You may want to make a copy of your /etc/asterisk folder first since all those will be hammered by the update. Any customizations you have on any of those files will need to be "remade".

If that's not the case, consider posting a single example call log (not 3000 lines of unrelated code, just one single test call with no other traffic).

[rrb555]

Ah yes forgot about the signature, i already removed it.

This is a year old fresh server, I only did SVN upgrades.

[rrb555]

Call log

1. Call to the DID
2. Extension 501 receiving the call
3. Answering the call
4. Extension 501 hanging up the call
5. Caller disconnects the call

Code: Select all

== Using SIP RTP CoS mark 5
       > 0x7f84c018a6a0 -- Strict RTP learning after remote address set to: 27.111.12.66:34724
    -- Executing [1909XXXXXXX@trunkinbound:1] AGI("SIP/VOIP-00000303", "agi-DID_route.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
    -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20181119223856_1909XXXXXXX_16092255799)
    -- <SIP/VOIP-00000303>AGI Script agi-DID_route.agi completed, returning 0
    -- Executing [99909*3***DID@default:1] Answer("SIP/VOIP-00000303", "") in new stack
    -- Executing [99909*3***DID@default:2] AGI("SIP/VOIP-00000303", "agi-VDAD_ALL_inbound.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
       > 0x7f84c018a6a0 -- Strict RTP switching to RTP target address 27.111.12.66:34724 as source
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 192*168*xxx*xxx*501@default
    -- Executing [192*168*xxx*xxx*501@default:1] Goto("Local/192*168*xxx*xxx*501@default-0000020d;2", "default,501,1") in new stack
    -- Goto (default,501,1)
    -- Executing [501@default:1] Dial("Local/192*168*xxx*xxx*501@default-0000020d;2", "SIP/501,60,") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/501
    -- SIP/501-00000304 is ringing
    -- Local/192*168*xxx*xxx*501@default-0000020d;1 is ringing
    -- Started music on hold, class 'default', on channel 'SIP/VOIP-00000303'
       > 0x7f85080191a0 -- Strict RTP learning after remote address set to: 192.168.4.23:51428
    -- SIP/501-00000304 answered Local/192*168*xxx*xxx*501@default-0000020d;2
    -- Local/192*168*xxx*xxx*501@default-0000020d;1 answered
    -- Channel SIP/501-00000304 joined 'simple_bridge' basic-bridge <260aa255-27d9-45bf-a5ac-5a2dcde0ca6c>
    -- Executing [138331*531*Y1192238570000023016*9900*@default:1] AGI("Local/192*168*xxx*xxx*501@default-0000020d;1", "agi-VDAD_local_optimize.agi,RINGAGENT00000000531") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi
    -- Channel Local/192*168*xxx*xxx*501@default-0000020d;2 joined 'simple_bridge' basic-bridge <260aa255-27d9-45bf-a5ac-5a2dcde0ca6c>
    -- <Local/192*168*xxx*xxx*501@default-0000020d;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
    -- Executing [138331*531*Y1192238570000023016*9900*@default:2] Wait("Local/192*168*xxx*xxx*501@default-0000020d;1", "2") in new stack
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Channel SIP/501-00000304 left 'simple_bridge' basic-bridge <260aa255-27d9-45bf-a5ac-5a2dcde0ca6c>
    -- Channel Local/192*168*xxx*xxx*501@default-0000020d;2 left 'simple_bridge' basic-bridge <260aa255-27d9-45bf-a5ac-5a2dcde0ca6c>
    -- Executing [8331*531*Y1192238570000023016*9900*@default:1] Playback("SIP/501-00000304", "sip-silence") in new stack
  == Spawn extension (default, 501, 1) exited non-zero on 'Local/192*168*xxx*xxx*501@default-0000020d;2'
    -- Executing [h@default:1] AGI("Local/192*168*xxx*xxx*501@default-0000020d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0-----SIP 200 OK)") in new stack
    -- <SIP/501-00000304> Playing 'sip-silence.gsm' (language 'en')
    -- <Local/192*168*xxx*xxx*501@default-0000020d;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0-----SIP 200 OK) completed, returning 0
  == Spawn extension (default, 138331*531*Y1192238570000023016*9900*, 2) exited non-zero on 'Local/192*168*xxx*xxx*501@default-0000020d;1'
[Nov 19 22:39:00] WARNING[27097][C-00000561]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/192*168*xxx*xxx*501@default-0000020d;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- Executing [8331*531*Y1192238570000023016*9900*@default:2] AGI("SIP/501-00000304", "agi-VDAD_RINGALL.agi,8331*531*Y1192238570000023016*9900*") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_RINGALL.agi
    -- <Local/192*168*xxx*xxx*501@default-0000020d;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
    -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/9900_6092255799_20181119-223900)
    -- <SIP/501-00000304> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/501-00000304> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/501-00000304> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/501-00000304> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/501-00000304>AGI Script agi-VDAD_RINGALL.agi completed, returning 0
    -- Executing [192*168*xxx*xxx*78600072@default:1] Goto("SIP/501-00000304", "default,78600072,1") in new stack
    -- Goto (default,78600072,1)
    -- Executing [78600072@default:1] MeetMe("SIP/501-00000304", "8600072,Fq") in new stack
    -- Created MeetMe conference 1022 for conference '8600072'
    -- Stopped music on hold on SIP/VOIP-00000303
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/VOIP-00000303> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
    -- <SIP/VOIP-00000303>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
    -- Executing [192*168*xxx*xxx*8600072@default:1] Goto("SIP/VOIP-00000303", "default,8600072,1") in new stack
    -- Goto (default,8600072,1)
    -- Executing [8600072@default:1] MeetMe("SIP/VOIP-00000303", "8600072,F") in new stack
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
       > 0x7f84c018a6a0 -- Strict RTP learning complete - Locking on source address 27.111.12.66:34724
  == Manager 'sendcron' logged off from 127.0.0.1
  == Spawn extension (default, 78600072, 1) exited non-zero on 'SIP/501-00000304'
    -- Executing [h@default:1] AGI("SIP/501-00000304", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
    -- <SIP/501-00000304>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Hungup 'DAHDI/pseudo-1967359221'
  == Spawn extension (default, 8600072, 1) exited non-zero on 'SIP/VOIP-00000303'
[Nov 19 22:39:08] WARNING[27090][C-0000055f]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("SIP/VOIP-00000303", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
    -- <SIP/VOIP-00000303>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0


[williamconley]

I only did SVN upgrades.

So ... the database was never imported? Good idea to verify the Asterisk version everywhere anyway. Remember that if the install.pl script says "11.X" and you enter "11", that's not a match.

Provide an asterisk CLI output example from a FULL single call. Beginning to end. Not a bad idea to check all the other logs for that one call as well. Turn on logging to FILE under admin->servers. Note that having NO other traffic (not even registered phones) during the test will make everything much easier to deal with. Delete all your vicidial logs. Rotate your asterisk logs. Then your logs will have ONLY data from this one test call. Copy them somewhere so you can process them without them continually growing.

Consider non-ring-all to see if that makes a difference.

[rrb555]

Update:

Test using Agent Interface with On-Hook Extension
1. Normal process when receiving a call through Agent interface
2. Softphone rings and agent answers
3. Agent interface changes as agent receives the call
4. Agent ending the call in Agent interface then hangup in Softphone
5. Call were successfully disconnected

It seems like the code ending the call is in the Agent interface, it is the one triggering the disconnection of the call. As when I am trying to disconnect the call in the softphone first, call were never disconnected, It needs to hangup the call in the Agent Interface. So I am guessing this Agent Interface On-Hook process is the same with Remote Agents, Softphone hangup is not sufficient enough to trigger the call disconnection.

Also Take Note: DO NOT add an entry in the Phone Extension Full Name field, as call will not push through. It will ring the Phone Extension but call will not push through upon answering it. Found an AGI log
2018-11-19 23:34:55|23:35:09|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found

[williamconley]

1) you still have not listed your full installer version. 8.0.X not 8.0 (probably best to go back and edit it ... for posterity)

2)

...So I am guessing this Agent Interface On-Hook process is the same with Remote Agents, Softphone hangup is not sufficient enough to trigger the call disconnection....

Not true. There is a different process, but that process will ultimately use the same method for final termination (thus the test). Once you get it configured properly, it'll work. Do you have any custom code in your asterisk configuration files? Are there any contexts which are missing the "h" hangup extension? Are the agent phones in admin->phones or are they external numbers for remote agents directly in the remote agent module? If external: What dialplan is used to reach the agents (standard carrier or something custom)?

Also Take Note: DO NOT add an entry in the Phone Extension Full Name field, as call will not push through. It will ring the Phone Extension but call will not push through upon answering it. Found an AGI log
2018-11-19 23:34:55|23:35:09|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found

3) Interesting. I'd love to see a comparison of two calls with and without this flaw to see the difference.

[rrb555]

1. Done. Vicibox 8.0.1

2. No custom asterisk configuration were done.
Agent phones are in admin>phones. I am using the local IP address of the server to connect to the phone extension using Zoiper5 and Bria


3. Another thing that I've noticed, Under Inbound > In-Groups you need to have a Generic On-Hook CID to make it work too or else you will be seeing the "No available balance agent found" issue.