Vicidial Group queue to call the next number and not hang-up

[rahat651]

Hi,
we are new to Vicidial. We have created a Group in Vicidial and added few remote agents to the group. Remote agents have mobile phone numbers. Here are some of the options from the group configurations:

Code: Select all

Next Agent Call: longest_wait_time
Queue Priority: 0 - Even
On-Hook Ring Time: 15

Now the queue is working as expected. Agent numbers are dialed based on the wait time. But the problem is that for an incoming call, the system dialed only one number from the queue and hangs up if the call is not answered.
It does not dialed the next number from the queue if the previous number does not receive the call. It only dialed the next number if a new call comes in.

I was wondering how to have the next number from the queue dialed (or redirect the user to an IVR) if the previous number does not picks up?

Vicidial:

Code: Select all

Version: 2.14b0.5
SVN Version: 3254
DB Schema Version: 1596

[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) Look up "OnHook". This applies to both logged in agent and remote agents. In remote agents, it's a dropdown with Y/N. Change it to Y.

[rahat651]

1) 3) Look up "OnHook". This applies to both logged in agent and remote agents. In remote agents, it's a dropdown with Y/N. Change it to Y.

Thanks for the quick reply.

Pardon my lack ok knowledge, I thought on-hook was for the agents to have the calls displayed in web panel.
May I know how does on-hook allow the next number to be dialed if the previous one is busy?

[williamconley]

OnHook refers to "will not call you until you get a call". This makes it perfect for desk phones that don't actually have an agent or for agents who only get calls sporadically.

Vicidial was originally designed for agents to log in and stay on the phone without ever hanging up. The agent is placed into a conference call, and the prospects/leads are put into (and removed from) the conference all day. But the agent ... never leaves, never hangs up the phone. This allows extreme speed in call routing. No "wait for an agent to pick up the phone" since the agent is already ON the phone. Instant answer. Very cool ... a decade ago.

Since then, however, it became useful to have Vicidial fill a larger gap in call centers. The ability for a secretary or sales agent who is NOT always on the phone (or in some cases even RARELY on the phone) who can just answer the phone when it rings and get the call. But with this comes the inevitability that someone will forget to "log out" and those calls will ring forever. So Vicidial can no longer "assume" there is an agent in that desk waiting.

Thus we now have "OnHook" phones. These phones (those with OnHook assigned) ring and if nobody answers, the call will automatically ring over to the next agent. There is even an option (if properly configured) for "RingAll" so the first account rep who grabs that phone ... wins.

[rahat651]

1)
3) Look up "OnHook". This applies to both logged in agent and remote agents. In remote agents, it's a dropdown with Y/N. Change it to Y.

William, setting the Onhook to 'Y' for all remote agents does make the system call the next number if the previous one does not receive/rejects the call.

However there are few things I have noticed. Not sure if those can be bypassed/changed from the system. Issues are as followings:
1. If the caller or the receiver disconnects the call then the other end is still connected to the system. In that case can we play an audio file (e.g. "Disconnected") to the connected end?

2. If the caller disconnects the call before any remote agent receives the call then the phone still rings on the remote agent's side and disconnects as soon as the remote agent receives the call.

3. It takes random time to switch to the next numbers. I had tested with three mobile phones from the same operator and kept those on the same desk.
From the CLI log the call switching times (not the ring times) calculated based on the "Called SIP/DelTa9/123456789" lines are as followings:
16 Seconds
30 seconds
17 seconds
16 seconds
30 seconds

[dheeraj2709]

Hi,
If the caller disconnects the call before any remote agent receives the call then the phone still rings on the remote agent's side and disconnects as soon as the remote agent receives the call.

Did you find solution for it?

[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) "Before the agent receives the call" is a bit vague. For clarity: Is the call already ringing at the remote agent? Is this a ringall queue? What are the settings for the ingroup? Have you Upgraded?

[dheeraj2709]

Hi, Thanks for your reply
i installed ViciBox v.8.1.2 as virtual machine (Physical machine is UBUNTU desktop) no other third party software and no up gradation.
Asterisk Version Asterisk 13.21.1-vici

i am trying to built a inbound call center where all agents are remote agents(no login required and set ON HOOK option to 'Y').
i added following lines in extensions.conf to send the call selected agents. ingroup setting is ON HOOK CID=RA_AGENT_PHONE and Next Agent call=old_call_finish

[default]
exten => _XXXXXXXXXX,1,Set(CALLERID(num)=1725306871)
exten => _XXXXXXXXXX,n,Verbose(${EXTEN})
extten => _XXXXXXXXXX,n,Dial(SIP/SIPtrunk/0${EXTEN})
exten => _XXXXXXXXXX,n,Hangup()



i am facing the problem when inbound caller hangup call (in queue) agent number keep ringing,and when agent answer the call it dropped.
some time caller discoonnect the call immediately (no ringing started a agent) then agent get the call after 15-20 seconds.

This Hangup call issue occur only on if caller is not connected with remote agent otherwise no issue occur.


[May 1 01:06:39] -- <SIP/10.157.1.4-00000195>AGI Script agi-DID_route.agi completed, returning 0
[May 1 01:06:39] -- Executing [99909*2***DID@default:1] Answer("SIP/10.157.1.4-00000195", "") in new stack
[May 1 01:06:39] > 0x1ea5c10 -- Strict RTP switching to RTP target address 10.157.1.68:38424 as source
[May 1 01:06:39] -- Executing [99909*2***DID@default:2] AGI("SIP/10.157.1.4-00000195", "agi-VDAD_ALL_inbound.agi") in new stack
[May 1 01:06:39] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_request: agi-VDAD_ALL_inbound.agi
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_channel: SIP/10.157.1.4-00000195
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_language: en
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_type: SIP
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_uniqueid: 1651347399.1177
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_version: 13.21.1-vici
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callerid: 917508973789
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_calleridname: unknown
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callingpres: 0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callingani2: 0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callington: 0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callingtns: 0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_dnid: 01725306872
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_rdnis: unknown
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_context: default
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_extension: 99909*2***DID
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_priority: 2
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_enhanced: 0.0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_accountcode:
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> agi_threadid: 140058794321664
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >>
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << GET VARIABLE state
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << GET VARIABLE postal_code
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << GET VARIABLE source_id
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=0
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << STREAM FILE sip-silence ""
[May 1 01:06:39] -- <SIP/10.157.1.4-00000195> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=0 endpos=160
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << STREAM FILE sip-silence ""
[May 1 01:06:39] -- <SIP/10.157.1.4-00000195> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=0 endpos=160
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << SET CALLERID "Y5010106390000000197 <917508973789>"
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=1
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:39] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:40] == Manager 'sendcron' logged on from 127.0.0.1
[May 1 01:06:40] -- Called 172*017*058*095*9653018144@default
[May 1 01:06:40] -- Executing [172*017*058*095*9653018144@default:1] Goto("Local/172*017*058*095*9653018144@default-0000013d;2", "default,9653018144,1") in new stack
[May 1 01:06:40] -- Goto (default,9653018144,1)
[May 1 01:06:40] -- Executing [9653018144@default:1] Set("Local/172*017*058*095*9653018144@default-0000013d;2", "CALLERID(num)=1725306871") in new stack
[May 1 01:06:40] -- Executing [9653018144@default:2] Verbose("Local/172*017*058*095*9653018144@default-0000013d;2", "9653018144") in new stack
[May 1 01:06:40] 9653018144
[May 1 01:06:40] -- Executing [9653018144@default:3] Verbose("Local/172*017*058*095*9653018144@default-0000013d;2", "1651347400.1179") in new stack
[May 1 01:06:40] 1651347400.1179
[May 1 01:06:40] -- Executing [9653018144@default:4] Dial("Local/172*017*058*095*9653018144@default-0000013d;2", "SIP/SIPtrunk/09653018144,,F") in new stack
[May 1 01:06:40] == Using SIP RTP CoS mark 5
[May 1 01:06:40] -- Called SIP/SIPtrunk/09653018144
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:40] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:41] > 0x7f6254006a30 -- Strict RTP learning after remote address set to: 10.157.1.68:33792
[May 1 01:06:41] -- SIP/SIPtrunk-00000196 is making progress passing it to Local/172*017*058*095*9653018144@default-0000013d;2
[May 1 01:06:41] -- Local/172*017*058*095*9653018144@default-0000013d;1 is making progress
[May 1 01:06:41] > 0x7f6254006a30 -- Strict RTP switching to RTP target address 10.157.1.68:33792 as source
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Rx << CHANNEL STATUS SIP/10.157.1.4-00000195
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=6
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Rx << SET MUSIC ON default
[May 1 01:06:41] -- Started music on hold, class 'default', on channel 'SIP/10.157.1.4-00000195'
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> 200 result=0
[May 1 01:06:41] -- Stopped music on hold on SIP/10.157.1.4-00000195
[May 1 01:06:41] -- <SIP/10.157.1.4-00000195>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 4
[May 1 01:06:41] == Spawn extension (default, 99909*2***DID, 2) exited non-zero on 'SIP/10.157.1.4-00000195'
[May 1 01:06:41] WARNING[6890][C-0000026b]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[May 1 01:06:41] -- Executing [h@default:1] AGI("SIP/10.157.1.4-00000195", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[May 1 01:06:41] AGI Tx >> agi_network: yes
[May 1 01:06:41] AGI Tx >> agi_network_script: call_log--HVcauses--PRI-----NODEBUG-----0--------------------)
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_request: agi://127.0.0.1:4577/call_log--HVcauses ... ----------)
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_channel: SIP/10.157.1.4-00000195
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_language: en
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_type: SIP
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_uniqueid: 1651347399.1177
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_version: 13.21.1-vici
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callerid: 917508973789
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_calleridname: Y5010106390000000197
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callingpres: 0
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callingani2: 0
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callington: 0
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_callingtns: 0
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_dnid: 01725306872
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_rdnis: unknown
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_context: default
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_extension: h
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_priority: 1
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_enhanced: 0.0
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_accountcode:
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> agi_threadid: 140058794321664
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >>
[May 1 01:06:41] -- <SIP/10.157.1.4-00000195>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[May 1 01:06:41] <SIP/10.157.1.4-00000195>AGI Tx >> HANGUP
[May 1 01:06:46] > 0x7f6254006a30 -- Strict RTP learning complete - Locking on source address 10.157.1.68:33792
[May 1 01:06:46] -- SIP/SIPtrunk-00000196 is ringing
[May 1 01:06:46] -- SIP/SIPtrunk-00000196 is making progress passing it to Local/172*017*058*095*9653018144@default-0000013d;2
[May 1 01:06:46] -- Local/172*017*058*095*9653018144@default-0000013d;1 is ringing
[May 1 01:06:46] -- Local/172*017*058*095*9653018144@default-0000013d;1 is making progress
[May 1 01:06:54] == Spawn extension (default, 9653018144, 4) exited non-zero on 'Local/172*017*058*095*9653018144@default-0000013d;2'
[May 1 01:06:54] -- Executing [h@default:1] AGI("Local/172*017*058*095*9653018144@default-0000013d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------------SIP 180 Ringing)") in new stack
[May 1 01:06:54] AGI Tx >> agi_network: yes
[May 1 01:06:54] AGI Tx >> agi_network_script: call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------------SIP 180 Ringing)
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_request: agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 180 Ringing)
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_channel: Local/172*017*058*095*9653018144@default-0000013d;2
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_language: en
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_type: Local
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_uniqueid: 1651347400.1179
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_version: 13.21.1-vici
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_callerid: 1725306871
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_calleridname: RA_9653018144_917508
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_callingpres: 0
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_callingani2: 0
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_callington: 0
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_callingtns: 0
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_dnid: unknown
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_rdnis: unknown
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_context: default
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_extension: h
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_priority: 1
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_enhanced: 0.0
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_accountcode:
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> agi_threadid: 140059367556864
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >>
[May 1 01:06:54] -- <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 180 Ringing) completed, returning 0
[May 1 01:06:54] <Local/172*017*058*095*9653018144@default-0000013d;2>AGI Tx >> HANGUP
[May 1 01:06:55] == Manager 'sendcron' logged off from 127.0.0.1

[williamconley]

The single most important piece of informaton to post is your vicidial version with build. I note you posted your installer version and asterisk version, very good, but: The installer (any version) can install any version of Vicidial, so knowing the installer version only gives us the "environment", not the Vicidia version.

You appear to be trying to use a dialplen entry to call the agent phones at their normal phone numbers without the agi() function as the first line.

ALL carrier entries for calls auto-generated by Vicidial must have that AGI function before the dial command. The AGI function "attaches" Vicidial to the call. Then Vicidial will (for instance) set the caller ID.

Code: Select all

exten => _71NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _71NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:1},,tTor)
exten => _71NXXNXXXXXX,n,Hangup


Line 1: Attach Vicidial with perl script listening on port 4577
Line 2: DIAL
Line 3: Hang up (ONLY if there is no answer: this line is never reached if the call is answered because the dialplan is intercepted and goes elsewhere, but if the call is not answered the Hangup function must be called to trigger Vicidial functions to terminate the call and clean up any residual functions awaiting termination).

[dheeraj2709]

Hi,
I dont know installer version ..i downloaded the ViciBox_v8_1.x86_64-8.1.2.iso and installed ...when I login on machine i get this message "ViciBox v.8.1.2 181002"

as you told even i tried also _XXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) Before Dial() to call remote agent number....
But problem is agent number get ring and caller disconnected the call in that case agent's mobile keep ringing up to set ring time... i want hangup agent call.

in case of ringall all agent number keep ringing even caller disconnected the call...

[williamconley]

i downloaded the ViciBox_v8_1.x86_64-8.1.2.iso and installed

That's the installer

The Vicidial Version (with build) is at the bottom left corner of most administration screens.

as you told even i tried also _XXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) Before Dial() to call remote agent number....
But problem is agent number get ring and caller disconnected the call in that case agent's mobile keep ringing up to set ring time... i want hangup agent call.

in case of ringall all agent number keep ringing even caller disconnected the call...

_XXXXXXXXXX is not the same as _XXNXXXXXX, thus those will not be part of the same extension.

Test with this at the command line to see which lines are in the dialplan after asterisk reloads, this will tell you if your changes had the expected result:

Code: Select all

asterisk -rx "dialplan show 5555555555@default"

If your previous dialplan extension was:

Code: Select all

exten => _XXXXXXXXXX,1,Set(CALLERID(num)=1725306871)
exten => _XXXXXXXXXX,n,Verbose(${EXTEN})
extten => _XXXXXXXXXX,n,Dial(SIP/SIPtrunk/0${EXTEN})
exten => _XXXXXXXXXX,n,Hangup()

Then the new one should be:

Code: Select all

exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,n,Set(CALLERID(num)=1725306871)
exten => _XXXXXXXXXX,n,Verbose(${EXTEN})
exten => _XXXXXXXXXX,n,Dial(SIP/SIPtrunk/0${EXTEN})
exten => _XXXXXXXXXX,n,Hangup()

Note how the "exten => _XXXXXXXXXX" all line up perfectly, because they are all the same extension, but with different "priorities".

also worthy of note: If you modified the extensions.conf file (or the sip.conf, or any other .conf files) directly: You should not be doing that until you know what you are doing. Entries in sip-vicidial.conf and extensions-vicidial.conf are accomplished by putting entries in the Admin->Carriers in their relevent sections and auto-included by the vicidial system.

This is all covered, one section at at time, in the Vicidial Manager's Manual. For a reason. If you use the manager's manual (one page at a time, don't skip anything) you'll end up with a working system without having to pay someone or spending hours trying to re-invent Vicidial. The Free version of the Manager's Manual will take you from "fresh install" to "fully functional" without spending a penny but with a full understanding of what is needed. IMHO.

[dheeraj2709]

in my previous reply there was Typo mistake .... i am using _XXXXXXXXXX.

i have changed the dialplan as you todld and output of asterisk -rx "dialplan show 5555555555@default":
'_XXXXXXXXXX' => 1. AGI(agi://127.0.0.1:4577/call_log) [pbx_config]
2. Set(CALLERID(num)=1725306871) [pbx_config]
3. Verbose(${EXTEN}) [pbx_config]
4. Dial(SIP/SIPtrunk/0${EXTEN}) [pbx_config]
5. Hangup() [pbx_config]

-= 1 extension (5 priorities) in 1 context. =-

---------------------------------------------------------------------------------------------
and CLI out out is :

[May 5 10:49:18] == Using SIP RTP CoS mark 5
[May 5 10:49:18] > 0x10a18a0 -- Strict RTP learning after remote address set to: 10.157.1.69:2012
[May 5 10:49:18] -- Executing [01725306872@default:1] Verbose("SIP/10.157.1.4-00000000", "1651727958.0") in new stack
[May 5 10:49:18] 1651727958.0
[May 5 10:49:18] -- Executing [01725306872@default:2] AGI("SIP/10.157.1.4-00000000", "agi-DID_route.agi") in new stack
[May 5 10:49:18] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[May 5 10:49:18] -- <SIP/10.157.1.4-00000000>AGI Script agi-DID_route.agi completed, returning 0
[May 5 10:49:18] -- Executing [99909*2***DID@default:1] Answer("SIP/10.157.1.4-00000000", "") in new stack
[May 5 10:49:18] > 0x10a18a0 -- Strict RTP switching to RTP target address 10.157.1.69:2012 as source
[May 5 10:49:18] -- Executing [99909*2***DID@default:2] AGI("SIP/10.157.1.4-00000000", "agi-VDAD_ALL_inbound.agi") in new stack
[May 5 10:49:18] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 5 10:49:18] -- <SIP/10.157.1.4-00000000> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 5 10:49:19] -- <SIP/10.157.1.4-00000000> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 5 10:49:19] > Saved useragent "Z 5.3.4 rv2.9.22" for peer 8001
[May 5 10:49:19] == Manager 'sendcron' logged on from 127.0.0.1
[May 5 10:49:19] -- Called 172*017*058*095*9772527591@default
[May 5 10:49:19] -- Executing [172*017*058*095*9772527591@default:1] Goto("Local/172*017*058*095*9772527591@default-00000000;2", "default,9772527591,1") in new stack
[May 5 10:49:19] -- Goto (default,9772527591,1)
[May 5 10:49:19] -- Executing [9772527591@default:1] AGI("Local/172*017*058*095*9772527591@default-00000000;2", "agi://127.0.0.1:4577/call_log") in new stack
[May 5 10:49:19] -- <Local/172*017*058*095*9772527591@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 5 10:49:19] -- Executing [9772527591@default:2] Set("Local/172*017*058*095*9772527591@default-00000000;2", "CALLERID(num)=1725306871") in new stack
[May 5 10:49:19] -- Executing [9772527591@default:3] Verbose("Local/172*017*058*095*9772527591@default-00000000;2", "9772527591") in new stack
[May 5 10:49:19] 9772527591
[May 5 10:49:19] -- Executing [9772527591@default:4] Dial("Local/172*017*058*095*9772527591@default-00000000;2", "SIP/SIPtrunk/09772527591") in new stack
[May 5 10:49:19] == Using SIP RTP CoS mark 5
[May 5 10:49:19] -- Called SIP/SIPtrunk/09772527591
[May 5 10:49:20] -- Started music on hold, class 'default', on channel 'SIP/10.157.1.4-00000000'
[May 5 10:49:22] > 0x7f41380084a0 -- Strict RTP learning after remote address set to: 10.157.1.69:20508
[May 5 10:49:22] -- SIP/SIPtrunk-00000001 is making progress passing it to Local/172*017*058*095*9772527591@default-00000000;2
[May 5 10:49:22] -- Local/172*017*058*095*9772527591@default-00000000;1 is making progress
[May 5 10:49:22] > 0x7f41380084a0 -- Strict RTP switching to RTP target address 10.157.1.69:20508 as source
[May 5 10:49:22] -- SIP/SIPtrunk-00000001 is ringing
[May 5 10:49:22] -- SIP/SIPtrunk-00000001 is making progress passing it to Local/172*017*058*095*9772527591@default-00000000;2
[May 5 10:49:22] -- Local/172*017*058*095*9772527591@default-00000000;1 is ringing
[May 5 10:49:22] -- Local/172*017*058*095*9772527591@default-00000000;1 is making progress
[May 5 10:49:23] > 0x10a18a0 -- Strict RTP learning complete - Locking on source address 10.157.1.69:2012
[May 5 10:49:23] -- Stopped music on hold on SIP/10.157.1.4-00000000
[May 5 10:49:23] -- <SIP/10.157.1.4-00000000> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[May 5 10:49:23] -- <SIP/10.157.1.4-00000000>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 4
[May 5 10:49:23] == Spawn extension (default, 99909*2***DID, 2) exited non-zero on 'SIP/10.157.1.4-00000000'
[May 5 10:49:23] WARNING[3237][C-00000000]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[May 5 10:49:23] -- Executing [h@default:1] AGI("SIP/10.157.1.4-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[May 5 10:49:23] -- <SIP/10.157.1.4-00000000>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[May 5 10:49:27] > 0x7f41380084a0 -- Strict RTP learning complete - Locking on source address 10.157.1.69:20508
[May 5 10:49:48] == Manager 'sendcron' logged off from 127.0.0.1
[May 5 10:49:48] == Spawn extension (default, 9772527591, 4) exited non-zero on 'Local/172*017*058*095*9772527591@default-00000000;2'
[May 5 10:49:48] -- Executing [h@default:1] AGI("Local/172*017*058*095*9772527591@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------------SIP 180 Ringing)") in new stack
[May 5 10:49:48] -- <Local/172*017*058*095*9772527591@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 180 Ringing) completed, returning 0
[May 5 10:50:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 5 10:50:01] == Manager 'sendcron' logged off from 127.0.0.1

as you can check incoming caller disconnected the call at [May 5 10:49:23] and agent mobile is still ringing ,it ring up to set 30 seconds ... i want to diconnect the agent call immediately on caller hangup.

[williamconley]

Cool. Now we've connected the Vicidial process to the call from Vicidial to the Remote Agent, but we have not yet confirmed that the Vicidial processes are properly connected to the call from the Caller to Vicidial.

This would ordinarily be where we recommend (again) that you set up the system according to the manager's manual from beginning to end until all features you need are set up and running as expected, without skipping around.

At this stage I would recommend using the system with a Logged In Agent to test the ability of the Vicidial system to detect the "hang up". Ordinarily, with a logged in agent who is using the Vicidial Agent Interface, the agent screen during the call has an image in the top right corner indicating the connection status of the call. When a lead hangs up, that should immediately change to "Hung Up", thus demonstrating that Vicidial has detected the hung up condition and asserted that hangup throughout the system.

This would ordinarily be fairly early in the validation of a functioning system.

[dheeraj2709]

Hi,

Today we have created a phone with external protocol and we have put agent's phone number in dialplan number and on-hook is enabled.

after that we have logged in the agent via web interface and clicked on resume button, but still we are facing same issue as agent's phone is ringing until on-hook time get's over even customer has hangup the call immediately.

[williamconley]

The goal of that exercise was to check and see if the image in the top right corner of the agent screen changes to show that the customer has hung up when the customer hangs up.

When the customer is talking to the agent, and the image shows live call, at that point the customer hangs up ... does the image change immediately? Or does it change "after a while"? Or does it not change at all?

The answer to this question will tell us if Vicidial is informed of the hangup. This information determines where we go next.

[dheeraj2709]

Hi william,
Thanks for your every reply.. sorry i did not test vicidial as you told me but now i have solved the mentioned issue.
I added the extra colum for incoming call unigue id in vicidial_manager table and insert values from agi-inbound_vdad_all.agi And then send hanup on agent channels using script that is called on incoming call hangup handler.

Once again thank you

[williamconley]

As long as you got where you needed to be!

PS: Good job posting some details of your solution, rather than "fixed" without any info.