Call goes to queue but never connects to agent

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Call goes to queue but never connects to agent

Postby jheengoo » Tue Sep 11, 2018 1:29 pm

Hi

I have a call menu setup which lets user dial desired extension to connect to an agent. The issue I face is that when customer dials an extension, the system takes the customer on hold and stays there until call ends, and call never connects to agent. I have tried the following lines call menu's custom dialplan

agi-AGENT_route.agi,default---AGENTDIRECT---LOGGED_IN---if-u-know-ext-dial---X---invalid---please-try-again---3---TEST_IN3---pbxtransfer---outside-transfer"
exten => _XXXX,1,AGI(agi-AGENT_route.agi,default---AGENTDIRECT---ACTIVE)

as mentioned in agi-AGENT_route.agi

Also when I see in realtime report, I do see a cal waiting in agentdirect group for the dialed agent's extension. Kindly help.

Thanks
Waqar

Asterisk 13.21.1-vici
VERSION: 2.14-675a
BUILD: 180520-1749
© 2018 ViciDial Group
Vicibox 8
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Re: Call goes to queue but never connects to agent

Postby Rogger » Tue Sep 11, 2018 7:02 pm

Hello,

Check the following steps:
1) Is ingroup enabled in the campaign? Allowed Inbound Groups
2) Can your agent receive calls from this in-group? Check inside the in-group or through realtime

Good luck,

Rogger Faioli
VICIdialBrasil
GOSAT.org
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Re: Call goes to queue but never connects to agent

Postby jheengoo » Wed Sep 12, 2018 12:02 am

1) yes
2) yes
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Re: Call goes to queue but never connects to agent

Postby williamconley » Wed Sep 12, 2018 10:08 am

Post the agent information: Extension, dialplan number ... and the extension the prospect dialed ... and asterisk CLI output from an example call (start to finish of one call, but when there's no other traffic so we don't have 3000 lines of unrelated code).

Also post the locations you put those two lines, as it's easy to misplace things in Vicidial (plus I hate assumptions)
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Re: Call goes to queue but never connects to agent

Postby jheengoo » Thu Sep 13, 2018 9:34 am

This happens for any extension, not just one particular extension... The extension I'm using is 9999 with dialplan number 9999

The cli output is as below during call

[Sep 13 09:30:47] == Using SIP RTP CoS mark 5
[Sep 13 09:30:47] > 0x7fbeb0141ca0 -- Strict RTP learning after remote ad dress set to: 23.253.253.225:22160
[Sep 13 09:30:47] -- Executing [12149722769@trunkinbound:1] AGI("SIP/Skyetel OR-00000002", "agi-DID_route.agi") in new stack
[Sep 13 09:30:47] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID _route.agi
[Sep 13 09:30:47] -- <SIP/SkyetelOR-00000002>AGI Script agi-DID_route.agi co mpleted, returning 0
[Sep 13 09:30:47] -- Executing [s@IVRMAIN:1] Answer("SIP/SkyetelOR-00000002" , "") in new stack
[Sep 13 09:30:47] -- Executing [s@IVRMAIN:2] AGI("SIP/SkyetelOR-00000002", " agi-VDAD_inbound_calltime_check.agi,CALLMENU-----YES-----IVRMAIN---------------- ---------NO") in new stack
[Sep 13 09:30:47] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDA D_inbound_calltime_check.agi
[Sep 13 09:30:47] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (esc ape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:30:47] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (esc ape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:30:47] -- <SIP/SkyetelOR-00000002>AGI Script agi-VDAD_inbound_cal ltime_check.agi completed, returning 0
[Sep 13 09:30:47] -- Executing [s@IVRMAIN:3] Set("SIP/SkyetelOR-00000002", " INVCOUNT=0") in new stack
[Sep 13 09:30:47] -- Executing [s@IVRMAIN:4] BackGround("SIP/SkyetelOR-00000 002", "IVR-MAIN") in new stack
[Sep 13 09:30:47] -- <SIP/SkyetelOR-00000002> Playing 'IVR-MAIN.slin' (langu age 'en')
[Sep 13 09:30:48] > 0x7fbeb0141ca0 -- Strict RTP switching to RTP remote address 23.253.253.225:22160 as source
[Sep 13 09:30:49] > 0x7fbeb0141ca0 -- Strict RTP learning complete - Lock ing on source address 23.253.253.225:22160
[Sep 13 09:30:56] DTMF[3310][C-00000121]: channel.c:4129 __ast_read: DTMF end '9' received on SIP/SkyetelOR-00000002, duration 250 ms
[Sep 13 09:30:56] DTMF[3310][C-00000121]: channel.c:4199 __ast_read: DTMF end passthrough '9' on SIP/SkyetelOR-00000002
[Sep 13 09:30:56] DTMF[3310][C-00000121]: channel.c:4129 __ast_read: DTMF end '9' received on SIP/SkyetelOR-00000002, duration 250 ms
[Sep 13 09:30:56] DTMF[3310][C-00000121]: channel.c:4199 __ast_read: DTMF end passthrough '9' on SIP/SkyetelOR-00000002
[Sep 13 09:30:57] DTMF[3310][C-00000121]: channel.c:4129 __ast_read: DTMF end '9' received on SIP/SkyetelOR-00000002, duration 250 ms
[Sep 13 09:30:57] DTMF[3310][C-00000121]: channel.c:4199 __ast_read: DTMF end passthrough '9' on SIP/SkyetelOR-00000002
[Sep 13 09:30:57] DTMF[3310][C-00000121]: channel.c:4129 __ast_read: DTMF end '9' received on SIP/SkyetelOR-00000002, duration 250 ms
[Sep 13 09:30:57] DTMF[3310][C-00000121]: channel.c:4199 __ast_read: DTMF end passthrough '9' on SIP/SkyetelOR-00000002
[Sep 13 09:30:57] == CDR updated on SIP/SkyetelOR-00000002
[Sep 13 09:30:57] -- Executing [9999@IVRMAIN:1] AGI("SIP/SkyetelOR-00000002", "agi-AGENT_route.agi,default---AGENTDIRECT---ACTIVE") in new stack
[Sep 13 09:30:57] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-AGENT_route.agi
[Sep 13 09:30:57] -- <SIP/SkyetelOR-00000002>AGI Script agi-AGENT_route.agi completed, returning 0
[Sep 13 09:30:57] -- Executing [99909**AGENTDIRECT*9999*@default:1] Answer("SIP/SkyetelOR-00000002", "") in new stack
[Sep 13 09:30:57] -- Executing [99909**AGENTDIRECT*9999*@default:2] AGI("SIP/SkyetelOR-00000002", "agi-VDAD_ALL_inbound.agi") in new stack
[Sep 13 09:30:57] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Sep 13 09:30:57] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:30:57] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:30:57] -- <SIP/SkyetelOR-00000002> Playing 'aaa7-secsil.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:01] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 13 09:31:01] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 13 09:31:01] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 13 09:31:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 13 09:31:04] == Using SIP RTP CoS mark 5
[Sep 13 09:31:04] NOTICE[1847][C-00000122]: chan_sip.c:10125 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 25282 RTP/AVP 0 101
[Sep 13 09:31:04] WARNING[1847][C-00000122]: chan_sip.c:10519 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
[Sep 13 09:31:06] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 13 09:31:06] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 13 09:31:06] -- Started music on hold, class 'default', on SIP/SkyetelOR-00000002
[Sep 13 09:31:09] -- Stopped music on hold on SIP/SkyetelOR-00000002
[Sep 13 09:31:09] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:09] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:09] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:09] WARNING[3310][C-00000121]: file.c:701 ast_openstream_full: File generic_hold does not exist in any format
[Sep 13 09:31:10] -- Started music on hold, class 'default', on SIP/SkyetelOR-00000002
[Sep 13 09:31:58] -- Stopped music on hold on SIP/SkyetelOR-00000002
[Sep 13 09:31:58] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:58] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:59] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:59] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:59] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:59] -- <SIP/SkyetelOR-00000002> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Sep 13 09:31:59] -- <SIP/SkyetelOR-00000002>AGI Script agi-VDAD_ALL_inbound.agi completed, returning -1
[Sep 13 09:31:59] -- Executing [8307@default:1] Answer("SIP/SkyetelOR-00000002", "") in new stack
[Sep 13 09:31:59] -- Executing [8307@default:2] Playback("SIP/SkyetelOR-00000002", "vm-goodbye") in new stack
[Sep 13 09:31:59] -- <SIP/SkyetelOR-00000002> Playing 'vm-goodbye.gsm' (language 'en')
[Sep 13 09:32:00] -- Executing [8307@default:3] Hangup("SIP/SkyetelOR-00000002", "") in new stack
[Sep 13 09:32:00] == Spawn extension (default, 8307, 3) exited non-zero on 'SIP/SkyetelOR-00000002'
[Sep 13 09:32:00] -- Executing [h@default:1] AGI("SIP/SkyetelOR-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Sep 13 09:32:00] -- <SIP/SkyetelOR-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0




Appreciate your help.

Thanks
jheengoo
 
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Re: Call goes to queue but never connects to agent

Postby williamconley » Fri Sep 14, 2018 9:39 am

Have you gone through the Vicidial Manager's Manual setup for AGENTDIRECT? If you haven't set that up and tested it before attempting to route a call through a call menu to it, you've got your horse behind your buggy.

Also, what's "aaa7-secsil"?
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Re: Call goes to queue but never connects to agent

Postby jheengoo » Fri Sep 14, 2018 1:00 pm

yes, that's setup exactly as mentioned in the Manual.

aaa7-secsil is an audio file we have playing at beginning of calls preceded by sip-silence
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Re: Call goes to queue but never connects to agent

Postby williamconley » Fri Sep 14, 2018 1:07 pm

And when you set up AGENTDIRECT ... did it work? Did you successfully transfer a prospect to an agent through it?

How is this file configured to play? Did you alter the AGENTDIRECT campaign or alter something else?
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Re: Call goes to queue but never connects to agent

Postby jheengoo » Fri Sep 14, 2018 1:28 pm

Never been able to transfer a call through Call Menu directly to an agent. That file aaa7-secsil, is set as Menu Timeout Prompt under Call Menu.
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Re: Call goes to queue but never connects to agent

Postby williamconley » Fri Sep 14, 2018 1:54 pm

Revert to the original AGENTDIRECT and follow the instructions in the manual to configure it. Textbook. Provide the page/line where you hit your first snag. As with all "first time use" Vicidial Modules, this is always the best method to get up and running. This requires knowing everything else works. All the exercises before that one are required.

This is why I *always* advise everyone to start at page one of the manual and not skip anything. Just keep going (without skipping) until everything you need works.

Otherwise you can chase your tail for weeks because you missed one tiny detail. For the record: ALL those details are necessary. Every. Single. One. 8-)
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