vicidial.org

Hello;

vicibox 7.0.3, vicidial 2.12-15, Build 160508-0139, asterisk 11.22.0-vici, Single Machine

1) Using Agent On Hook (so when login, there will not be a ring at the agent phone), after receiving the call and finishing it and pressing on Hangup at the agent desktop screen, still I have to go for the phone and hangup from the extension!! Any solution for this?

2) And because I am using outbound and inbound at the same time (the dial method is Inbound_man), so when there are leads to be dialed, I have to press at ring link which is existed in the left upper side corner of the agent desktop (it is existed beside the SIP/Phone_Exten). If I did not press at ring link, then I will not hear any thing for the outbound call ... So until now, I have to press at ring link and answering the call before dial the outbound !! Is there any solution for this?

3) Now, because of pressing at ring link to do outbound call, I will receive call on the phone extension and I have to answer it (it will be the conference call), the big problem when receiving inbound call and answering it while the conference call that resulted from pressing the ring link is still existed, I will hear music. What is the solution for this?

Regards
Bilal

For the first two, sounds like On-Hook Agent Phones are working just as they were designed to. They were built for very low volume call centers, and are not designed for agents that receive or place many phone calls.

For the third one, you would have to post the Asterisk Output when this happens, as I have never heard of music being sent to an agent while receiving an inbound call with an On-Hook agent phone.

Hello Matt;

For the third one, you would have to post the Asterisk Output when this happens, as I have never heard of music being sent to an agent while receiving an inbound call with an On-Hook agent phone.

* It is happening if we were doing outbound, because as you know that we have to click on ring link first and then dialing outbound number. The only way to not hear this music is to hangup the conference call that resulted from ring the phone and then answering the incoming call.

Below is the asterisk CLI:

Code: Select all

[Apr 28 02:20:16]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:20:16]   == Using SIP RTP CoS mark 5
[Apr 28 02:20:18]        > Channel SIP/2299-0000001f was answered
[Apr 28 02:20:18]     -- Executing [8600051@default:1] MeetMe("SIP/2299-0000001f", "8600051,F") in new stack
[Apr 28 02:20:18]   == Parsing '/etc/asterisk/meetme.conf': Found
[Apr 28 02:20:18]   == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Apr 28 02:20:18]     -- Created MeetMe conference 1023 for conference '8600051'
[Apr 28 02:20:18]     -- <SIP/2299-0000001f> Playing 'conf-onlyperson.gsm' (language 'en')
[Apr 28 02:20:18]        > 0x7f8c80012880 -- Probation passed - setting RTP source address to 192.168.43.32:5062
[Apr 28 02:20:19]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:20:20]   == Using SIP RTP CoS mark 5
[Apr 28 02:20:20]     -- Executing [888888@defaultlog:1] AGI("SIP/223344-00000020", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Apr 28 02:20:20]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Apr 28 02:20:20]     -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20180428022020_223344_888888)
[Apr 28 02:20:20]     -- <SIP/223344-00000020>AGI Script agi-NVA_recording.agi completed, returning 0
[Apr 28 02:20:20]     -- Executing [888888@defaultlog:2] Goto("SIP/223344-00000020", "default,888888,1") in new stack
[Apr 28 02:20:20]     -- Goto (default,888888,1)
[Apr 28 02:20:20]     -- Executing [888888@default:1] AGI("SIP/223344-00000020", "agi-DID_route.agi") in new stack
[Apr 28 02:20:20]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Apr 28 02:20:20]     -- <SIP/223344-00000020>AGI Script agi-DID_route.agi completed, returning 0
[Apr 28 02:20:20]     -- Executing [s@Main:1] Answer("SIP/223344-00000020", "") in new stack
[Apr 28 02:20:20]        > 0x7f8c90016110 -- Probation passed - setting RTP source address to 192.168.43.1:43488
[Apr 28 02:20:20]        > 0x7f8c90016110 -- Probation passed - setting RTP source address to 192.168.43.1:43488
[Apr 28 02:20:20]     -- Executing [s@Main:2] AGI("SIP/223344-00000020", "agi-VDAD_inbound_calltime_check.agi,CALLMENU-----YES-----Main-------------------------NO") in new stack
[Apr 28 02:20:20]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
[Apr 28 02:20:20]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:20]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:20]     -- <SIP/223344-00000020>AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
[Apr 28 02:20:20]     -- Executing [s@Main:3] Set("SIP/223344-00000020", "INVCOUNT=0") in new stack
[Apr 28 02:20:20]     -- Executing [s@Main:4] BackGround("SIP/223344-00000020", "AlBarakaSalamAleikom") in new stack
[Apr 28 02:20:20]     -- <SIP/223344-00000020> Playing 'AlBarakaSalamAleikom.slin' (language 'en')
[Apr 28 02:20:22] DTMF[29251][C-00000052]: channel.c:4215 __ast_read: DTMF begin '0' received on SIP/223344-00000020
[Apr 28 02:20:22] DTMF[29251][C-00000052]: channel.c:4219 __ast_read: DTMF begin ignored '0' on SIP/223344-00000020
[Apr 28 02:20:22] DTMF[29251][C-00000052]: channel.c:4129 __ast_read: DTMF end '0' received on SIP/223344-00000020, duration 60 ms
[Apr 28 02:20:22] DTMF[29251][C-00000052]: channel.c:4199 __ast_read: DTMF end passthrough '0' on SIP/223344-00000020
[Apr 28 02:20:22]     -- Executing [0@Main:1] AGI("SIP/223344-00000020", "agi-VDAD_ALL_inbound.agi,CIDLOOKUP-----LB-----GenericInGroup-----Main--------------------100-----961-----Generic------------------------------") in new stack
[Apr 28 02:20:22]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Apr 28 02:20:22]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:23]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:23]     -- <SIP/223344-00000020> Playing 'AlBarakaSalamAleikom.slin' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:29]     -- <SIP/223344-00000020> Playing 'AlBarakaRecTran.slin' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:38]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:20:38]     -- Executing [192*168*043*091*2299@default:1] Goto("Local/192*168*043*091*2299@default-00000035;2", "default,2299,1") in new stack
[Apr 28 02:20:38]     -- Goto (default,2299,1)
[Apr 28 02:20:38]     -- Executing [2299@default:1] Dial("Local/192*168*043*091*2299@default-00000035;2", "SIP/2299,60,") in new stack
[Apr 28 02:20:38]   == Using SIP RTP CoS mark 5
[Apr 28 02:20:38]     -- Called SIP/2299
[Apr 28 02:20:38]     -- SIP/2299-00000021 is ringing
[Apr 28 02:20:40]     -- Started music on hold, class 'Main', on SIP/223344-00000020
[Apr 28 02:20:40]     -- SIP/2299-00000021 answered Local/192*168*043*091*2299@default-00000035;2
[Apr 28 02:20:40]        > Channel Local/192*168*043*091*2299@default-00000035;1 was answered
[Apr 28 02:20:40]     -- Executing [8331*21*Y4280220220001000037*agent1*2299@default:1] Playback("Local/192*168*043*091*2299@default-00000035;1", "sip-silence") in new stack
[Apr 28 02:20:40]     -- <Local/192*168*043*091*2299@default-00000035;1> Playing 'sip-silence.gsm' (language 'en')
[Apr 28 02:20:40]        > 0x7f8c98016e10 -- Probation passed - setting RTP source address to 192.168.43.32:5066
[Apr 28 02:20:40]     -- Executing [h@default:1] AGI("Local/192*168*043*091*2299@default-00000035;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0") in new stack
[Apr 28 02:20:40]        > 0x7f8c80012880 -- Probation passed - setting RTP source address to 192.168.43.32:5062
[Apr 28 02:20:40]     -- <Local/192*168*043*091*2299@default-00000035;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0 completed, returning 0
[Apr 28 02:20:40]   == Spawn extension (default, 2299, 1) exited non-zero on 'Local/192*168*043*091*2299@default-00000035;2'
[Apr 28 02:20:40]     -- Executing [8331*21*Y4280220220001000037*agent1*2299@default:2] AGI("SIP/2299-00000021", "agi-VDAD_RINGALL.agi,8331*21*Y4280220220001000037*agent1*2299") in new stack
[Apr 28 02:20:40]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[Apr 28 02:20:40]        > 0x7f8c80012880 -- Probation passed - setting RTP source address to 192.168.43.32:5062
[Apr 28 02:20:41]     -- <SIP/2299-00000021> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:41]     -- <SIP/2299-00000021> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:41]     -- <SIP/2299-00000021> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:41]     -- <SIP/2299-00000021> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:41]     -- <SIP/2299-00000021>AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[Apr 28 02:20:41]     -- Executing [192*168*043*091*78600051@default:1] Goto("SIP/2299-00000021", "default,78600051,1") in new stack
[Apr 28 02:20:41]     -- Goto (default,78600051,1)
[Apr 28 02:20:41]     -- Executing [78600051@default:1] MeetMe("SIP/2299-00000021", "8600051,Fq") in new stack
[Apr 28 02:20:41]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:20:41]     -- Executing [192*168*043*091*78600051@default:1] Goto("Local/192*168*043*091*78600051@default-00000036;2", "default,78600051,1") in new stack
[Apr 28 02:20:41]     -- Goto (default,78600051,1)
[Apr 28 02:20:41]     -- Executing [78600051@default:1] MeetMe("Local/192*168*043*091*78600051@default-00000036;2", "8600051,Fq") in new stack
[Apr 28 02:20:41]        > Channel Local/192*168*043*091*78600051@default-00000036;1 was answered
[Apr 28 02:20:41]     -- Executing [83047777777777@vicidial-auto:1] Answer("Local/192*168*043*091*78600051@default-00000036;1", "") in new stack
[Apr 28 02:20:41]     -- Executing [83047777777777@vicidial-auto:2] Playback("Local/192*168*043*091*78600051@default-00000036;1", "ding") in new stack
[Apr 28 02:20:41]     -- <Local/192*168*043*091*78600051@default-00000036;1> Playing 'ding.gsm' (language 'en')
[Apr 28 02:20:41]     -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/192*168*043*091*78600051@default-00000036;1", "") in new stack
[Apr 28 02:20:41]   == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/192*168*043*091*78600051@default-00000036;1'
[Apr 28 02:20:41]     -- Executing [h@vicidial-auto:1] AGI("Local/192*168*043*091*78600051@default-00000036;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 28 02:20:41]     -- <Local/192*168*043*091*78600051@default-00000036;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 28 02:20:41]   == Spawn extension (default, 78600051, 1) exited non-zero on 'Local/192*168*043*091*78600051@default-00000036;2'
[Apr 28 02:20:41]     -- Executing [h@default:1] AGI("Local/192*168*043*091*78600051@default-00000036;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 28 02:20:41]     -- <Local/192*168*043*091*78600051@default-00000036;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 28 02:20:41]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:20:42]     -- Stopped music on hold on SIP/223344-00000020
[Apr 28 02:20:42]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:42]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:42]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:20:42]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:42]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:42]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:42]     -- <SIP/223344-00000020> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Apr 28 02:20:42]     -- <SIP/223344-00000020>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Apr 28 02:20:42]     -- Executing [192*168*043*091*8600051@default:1] Goto("SIP/223344-00000020", "default,8600051,1") in new stack
[Apr 28 02:20:42]     -- Goto (default,8600051,1)
[Apr 28 02:20:42]     -- Executing [8600051@default:1] MeetMe("SIP/223344-00000020", "8600051,F") in new stack
[Apr 28 02:20:42]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:20:42]     -- Executing [58600051@default:1] MeetMe("Local/58600051@default-00000037;2", "8600051,Fmq") in new stack
[Apr 28 02:20:42]        > Channel Local/58600051@default-00000037;1 was answered
[Apr 28 02:20:42]     -- Executing [8309@default:1] Answer("Local/58600051@default-00000037;1", "") in new stack
[Apr 28 02:20:42]     -- Executing [8309@default:2] Monitor("Local/58600051@default-00000037;1", "wav,GenericInGroup_20180428-022041_agent1_223344") in new stack
[Apr 28 02:20:42]     -- Executing [8309@default:3] Wait("Local/58600051@default-00000037;1", "3600") in new stack
[Apr 28 02:20:43]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:21:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:21:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:21:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:21:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:21:17]        > 0x7f8c80012880 -- Probation passed - setting RTP source address to 192.168.43.32:5062
[Apr 28 02:21:17]        > 0x7f8c98016e10 -- Probation passed - setting RTP source address to 192.168.43.32:5066
[Apr 28 02:21:18]        > 0x7f8c98016e10 -- Probation passed - setting RTP source address to 192.168.43.32:5066
[Apr 28 02:21:20]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/2299-0000001f'
[Apr 28 02:21:20]     -- Executing [h@default:1] AGI("SIP/2299-0000001f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 28 02:21:20]     -- <SIP/2299-0000001f>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 28 02:21:20]        > 0x7f8c98016e10 -- Probation passed - setting RTP source address to 192.168.43.32:5066
[Apr 28 02:21:25]   == Spawn extension (default, 78600051, 1) exited non-zero on 'SIP/2299-00000021'
[Apr 28 02:21:25]     -- Executing [h@default:1] AGI("SIP/2299-00000021", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 28 02:21:25]     -- <SIP/2299-00000021>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 28 02:21:30]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:21:30]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/223344-00000020
[Apr 28 02:21:30]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/223344-00000020'
[Apr 28 02:21:30]     -- Executing [h@default:1] AGI("SIP/223344-00000020", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 28 02:21:30]     -- <SIP/223344-00000020>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 28 02:21:31]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 28 02:21:31]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600051@default-00000037;2
[Apr 28 02:21:31]     -- Hungup 'DAHDI/pseudo-682634294'
[Apr 28 02:21:31]   == Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-00000037;2'
[Apr 28 02:21:31]     -- Executing [h@default:1] AGI("Local/58600051@default-00000037;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 28 02:21:31]     -- <Local/58600051@default-00000037;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 28 02:21:31]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-00000037;1'
[Apr 28 02:21:31]     -- Executing [h@default:1] AGI("Local/58600051@default-00000037;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 28 02:21:31]     -- <Local/58600051@default-00000037;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 28 02:21:31]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 28 02:21:32]   == Manager 'sendcron' logged off from 127.0.0.1



Regards
Bilal

I don't see any indication of Asterisk MoH being on when the call goes to the meetme room. It says that MoH stops before then.

Hello Matt;
First of all, when inbound call is reaching to the agent softphone and the agent is answering the call, I see two calls at the softphone: the meetme conf call and the inbound call. So they are two calls and not one call. In other words, the inbound call is not coming through the meetme conf call.
From another side: if I hanged up the meetme conf call, this problem is not happening. So before answering the incoming call, I have to hang up the meet me conf call that resulted from ring the phone to be able to do outbound calls. Or, if I answered the inbound call and I heard the music, then I have to hangup the meet me conf call.
I believe the reason for the problem: there is a conflict with using the meet me conf call that resulted from ring the phone while doing the outbound, I do not know why the inbound call is using this meet me conf, here is the reason for hearing the music, and here is the reason why this music disappears and I can talk if I hanged up the meet me conf call that resulted from ring the phone.

Last point: this music is heard by the caller and the agent and as I mentioned, the agent has to hangup the meet me conf call to let the music disappears and becoming able to talk with the caller.

Regards
Bilal

If you are using On-Hook agent phones, you should never have a call in the meetme room unless you are about to place a manual outbound call. You are supposed to hang up your agent phone after handling every call.

Ok that means the on hook and outbound calls from list is not a practical method.

So I have to use off hook, but the problem of the off hook that agents should keep looking for screen to see if call came or not (specially if there is no outbound calls need to be done). If I used agent alert, and the headset was connected, so the alert will come from the headset or from the speaker? If it will come from the headset then it will be without any benefit as the agent will not hear this alert.
I am sorry that maybe I have to check this before asking, but I am in a place that no phone and I have only softphone and I dont have the USB for headset to check if alert will come from speaker or through headset.

Regards
Bilal

How frequently do your agents place outbound manual dial calls?

How many inbound calls do your agents handle per day?

What is the average idle time between phone calls?

The outbound calls are about 10 - 15 call / day for each agent.
The inbound calls about 2 - 4 call / day for each agent.
And he might do the outbound calls between 9 and 11 am and he receive the little inbound calls between 12 and 4 pm.
That is the situation.
It is very important to hear alert for the agent in case off hook, how can we resolve this? This alert is not possible to be heard if the headset was not at agent head?

Regards
Bilal

There really isn't an optimal solution for something like that at this time.

Ok matt thank u.
I will do testing on alert for agent and come back for you but I have to reach my office where I have enough tools and I am in another city.
Actually my aim is to check how can we let the alert to come from the speaker in case of using softphone or in case of using phone device.

Regards
Bilal

Hello Matt.
First of all, regarding to the below problem which I placed it in my this post in the beginning:

3) Now, because of pressing at ring link to do outbound call, I will receive call on the phone extension and I have to answer it (it will be the conference call), the big problem when receiving inbound call and answering it while the conference call that resulted from pressing the ring link is still existed, I will hear music. What is the solution for this?

This problem did not appear when I tested using hard phone and not the softphone. It might be a problem related to the softphone it self (I was using phonerlite). So it might differs from one softphone to another, but it is not existed when using hard phone.

Now regarding to the agent alert and using Off Hook, it will be useful on two cases: if the speaker button at the phone is always pressed (and the headset is connected), so the alert will be heard form the speaker, but agent should always keep speaker button on after each call unless he is getting calls one after other directly and no ready for alert. Another case, when the agent is working at some screen and the headset on his head, then he will hear the alert before the conversation starts with the caller.

I believe if there were CTI then it is possible to have some rings (one or two ) before auto answer or it is possible to ring the phone until the agent will answer. But vicidial is depending on meetme conference instead of CTI.

Thanks a lot for your kindly help and answering.

Regards
Bilal