Too Long Delay Passig Answered Call to Agent Live Call

[marcelo]

Hi everyone

Marcelo here.

- ViciBox v.9.0.0 190913-1108 * Released on Friday the 13th during a full moon. So spooky, much wow! |Vicidial
2.14-588c BUILD 190925-1346 | Asterisk 13.27.0-vici | Linux version 4.12.14-lp151.28.16-default | Single Se
rver | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel(R) Xeon(R) CPU E5-2450 0 @ 2.10GHz

On a Test campaign I have notice a huge (5+s) delay on passing an answered call to the agent, while the call is already answer by the customer, it is taking over 5 seconds for this call to be passed as Live Call to a agent.

Please note that the test had at least one agent complete idle, but customer would have answer the call, and it is taking over 5+ seconds to connect to an agent "Live Call."

My expectation is if we have an agent available an answered call "must" be placed immediately on agent hands as "Live Call."

[Oct 18 14:35:44] -- SIP/es1.langineers.com:5060-00000126 answered SIP/201-00000125

Instead it takes 5+ seconds for the agent pick up the call at the conference room, after many "hello, hello, hello?" from the customer.

We expect the agent to have the call at the first "he.." of "hello....," and NOT having dead time with the customer waiting on line with NO Live agent.

> > > > > > > > >> Is above possible? Have the call passed to agent at the first "he.." of "hello....," and NOT having dead time with the customer waiting on line with NO Live agent.

I am using predictive dialing, and would change to an automated dialing if exists and solve the problem. One call per free agent, but need the immediate passing of an answered call to the Agent "Live" room.

Please advise.

[ambiorixg12]

This delay most of the time it is relate to RTP issue, have you verifiedd on the console if any rtp traffic while this occur, also I saw a post related to this issue where the problem was the media was arriving to Asterisk using to diffent address, You can tweak this on the /etc/asterisk/rtp.conf also you can discard this issue testing also testing with a differrent provider.


https://asteriskfaqs.org/2014/01/10/ast ... audio.html

[blackbird2306]

> > > > > > > > >> Is above possible? Have the call passed to agent at the first "he.." of "hello....," and NOT having dead time with the customer waiting on line with NO Live agent.

Yes that's almost possible!

What is your routing extension in campaign settings? How is the server load and how many agents are working, when this happens? Post the asterisk cli messages for one call.

[marcelo]

*************** After Submitting Post **********************
I got the delay between customer "Hello" and "Live Call" to a minimum using Dial Method: "RATIO" and Routing Extension: 8368 no AMD Answering Machine Detection.

> > > > > > > Does anybody have better configuration? < < < < < < <

*******************************************************************

> > > > > > > > >> Is above possible? Have the call passed to agent at the first "he.." of "hello....," and NOT having dead time with the customer waiting on line with NO Live agent.

Yes that's almost possible!

Is that possible with predictive dialing on? I mean, before or about the person says "He..." does the system have already figured out that is not a answering machine, fax, or other undesired call?

What is your routing extension in campaign settings? How is the server load and how many agents are working, when this happens? Post the asterisk cli messages for one call.

Server is running idle, only one test campaign running, not too many leads (half a dozen at a time.)

Maximum Adapt Dial Level: 1.0

Dial Method: ADAPT_TAPERAD
Auto Dial Level: 1.0
Adapt Intensity Modifier: 0-Balanced


At the time I witnessing the delay, with one Agent available, free of calls, campaign running on predictive dialing was doing a SINGLE call to my cell, call came into my cell, I say "Hello?...hello?...hello?..." and silent. It takes 3 to 5 seconds to bridge this call to conference to "Live connection" the Agent, for agent and "customer" in this case myself to be connected talking.

On MANUAL call we get immediate bridge connection for agent and customer call.
[Oct 21 10:31:30] -- Channel SIP/es1.langineers.com:5060-00000143 joined 'simple_bridge' basic-bridge <8bb6e311-cd9f-4a81-9ff1-5d904f30bfd2>
[Oct 21 10:31:30] -- Channel Local/8600051@default-0000014f;1 joined 'simple_bridge' basic-bridge <8bb6e311-cd9f-4a81-9ff1-5d904f30bfd2>
Far below we have the log for this call

Still at the predictive call below try, connection took from second 48 to 49, what would be max 2000 milliseconds but still felt like an eternity.

> > > > > >> > > ASTERISK LOG Using Predictive Dialing Dial Method: ADAPT_TAPERAD < < < < < < < < < < < < <
[Oct 21 10:09:38] Asterisk 13.27.0-vici, Copyright (C) 1999 - 2014, Digium, Inc. and others.
[Oct 21 10:09:38] Connected to Asterisk 13.27.0-vici currently running on vicibox9 (pid = 2426)
[Oct 21 10:09:43] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 21 10:09:43] -- Called 916666666666@default
[Oct 21 10:09:43] -- Executing [916666666666@default:1] AGI("Local/916666666666@default-0000014d;2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 21 10:09:43] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Oct 21 10:09:43] -- <Local/916666666666@default-0000014d;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 21 10:09:43] -- Executing [916666666666@default:2] Dial("Local/916666666666@default-0000014d;2", "SIP/es1.langineers.com:5060/6666666666") in new stack
[Oct 21 10:09:43] == Using SIP RTP CoS mark 5
[Oct 21 10:09:44] -- Called SIP/es1.langineers.com:5060/6666666666
[Oct 21 10:09:44] -- SIP/es1.langineers.com:5060-00000141 is ringing
[Oct 21 10:09:44] -- Local/916666666666@default-0000014d;1 is ringing
[Oct 21 10:09:48] > 0x7fc8e4020960 -- Strict RTP learning after remote address set to: 64.124.219.133:15858
[Oct 21 10:09:48] -- SIP/es1.langineers.com:5060-00000141 answered Local/916666666666@default-0000014d;2
[Oct 21 10:09:48] -- Local/916666666666@default-0000014d;1 answered
[Oct 21 10:09:48] -- Executing [138369@default:1] AGI("Local/916666666666@default-0000014d;1", "agi-VDAD_local_optimize.agi,V0211009430000000054") in new stack
[Oct 21 10:09:48] -- Channel SIP/es1.langineers.com:5060-00000141 joined 'simple_bridge' basic-bridge <8a8ad106-5d93-42cd-8b57-fd5e6ea754d1>
[Oct 21 10:09:48] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi
[Oct 21 10:09:48] -- Channel Local/916666666666@default-0000014d;2 joined 'simple_bridge' basic-bridge <8a8ad106-5d93-42cd-8b57-fd5e6ea754d1>
[Oct 21 10:09:48] > 0x7fc8e4020960 -- Strict RTP switching to RTP target address 64.124.219.133:15858 as source
[Oct 21 10:09:48] -- <Local/916666666666@default-0000014d;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
[Oct 21 10:09:48] -- Executing [138369@default:2] Wait("Local/916666666666@default-0000014d;1", "2") in new stack
[Oct 21 10:09:49] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 21 10:09:49] -- Channel SIP/es1.langineers.com:5060-00000141 left 'simple_bridge' basic-bridge <8a8ad106-5d93-42cd-8b57-fd5e6ea754d1>
[Oct 21 10:09:49] -- Executing [8369@default:1] AGI("SIP/es1.langineers.com:5060-00000141", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 21 10:09:49] -- Channel Local/916666666666@default-0000014d;2 left 'simple_bridge' basic-bridge <8a8ad106-5d93-42cd-8b57-fd5e6ea754d1>
[Oct 21 10:09:49] == Spawn extension (default, 916666666666, 2) exited non-zero on 'Local/916666666666@default-0000014d;2'
[Oct 21 10:09:49] -- Executing [h@default:1] AGI("Local/916666666666@default-0000014d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0-----SIP 200 OK)") in new stack
[Oct 21 10:09:49] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Oct 21 10:09:49] -- <SIP/es1.langineers.com:5060-00000141>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 21 10:09:49] -- Executing [8369@default:2] Playback("SIP/es1.langineers.com:5060-00000141", "sip-silence") in new stack
[Oct 21 10:09:49] -- <SIP/es1.langineers.com:5060-00000141> Playing 'sip-silence.gsm' (language 'en')
[Oct 21 10:09:49] -- Executing [8369@default:3] AMD("SIP/es1.langineers.com:5060-00000141", "2000,2000,1000,5000,120,50,4,256") in new stack
[Oct 21 10:09:49] -- AMD: SIP/es1.langineers.com:5060-00000141 (N/A) (N/A) (Fmt: slin)
[Oct 21 10:09:49] -- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256] maximumWordLength [5000]
> > > > > > > > > > END of Predictive dialing test Dial Method: ADAPT_TAPERAD < < < < < < < < < < < < < <<


> > > > > > > > >> > > > > Asterisk LOG using MANUAL calls < < < < < < < < < < < <
vicibox9*CLI>
[Oct 21 10:31:26] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 21 10:31:26] -- Called 8600051@default/n
[Oct 21 10:31:26] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-0000014f;2", "8600051,F") in new stack
[Oct 21 10:31:26] -- Local/8600051@default-0000014f;1 answered
[Oct 21 10:31:26] -- Executing [916666666666@default:1] AGI("Local/8600051@default-0000014f;1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 21 10:31:26] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Oct 21 10:31:26] -- <Local/8600051@default-0000014f;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 21 10:31:26] -- Executing [916666666666@default:2] Dial("Local/8600051@default-0000014f;1", "SIP/es1.langineers.com:5060/6666666666") in new stack
[Oct 21 10:31:26] == Using SIP RTP CoS mark 5
[Oct 21 10:31:26] -- Called SIP/es1.langineers.com:5060/6666666666
[Oct 21 10:31:27] -- SIP/es1.langineers.com:5060-00000143 is ringing
[Oct 21 10:31:27] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 21 10:31:30] > 0x7fc8940144c0 -- Strict RTP learning after remote address set to: 64.124.219.136:11800
[Oct 21 10:31:30] -- SIP/es1.langineers.com:5060-00000143 answered Local/8600051@default-0000014f;1
[Oct 21 10:31:30] -- Channel SIP/es1.langineers.com:5060-00000143 joined 'simple_bridge' basic-bridge <8bb6e311-cd9f-4a81-9ff1-5d904f30bfd2>
[Oct 21 10:31:30] -- Channel Local/8600051@default-0000014f;1 joined 'simple_bridge' basic-bridge <8bb6e311-cd9f-4a81-9ff1-5d904f30bfd2>
[Oct 21 10:31:31] > 0x7fc8940144c0 -- Strict RTP switching to RTP target address 64.124.219.136:11800 as source
[Oct 21 10:31:35] -- Channel SIP/es1.langineers.com:5060-00000143 left 'simple_bridge' basic-bridge <8bb6e311-cd9f-4a81-9ff1-5d904f30bfd2>
[Oct 21 10:31:35] -- Channel Local/8600051@default-0000014f;1 left 'simple_bridge' basic-bridge <8bb6e311-cd9f-4a81-9ff1-5d904f30bfd2>
[Oct 21 10:31:35] == Spawn extension (default, 916666666666, 2) exited non-zero on 'Local/8600051@default-0000014f;1'
[Oct 21 10:31:35] -- Executing [h@default:1] AGI("Local/8600051@default-0000014f;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----9-----SIP 200 OK)") in new stack
[Oct 21 10:31:35] -- <Local/8600051@default-0000014f;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -9-----SIP 200 OK) completed, returning 0
[Oct 21 10:31:35] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0000014f;2'
[Oct 21 10:31:35] WARNING[7147][C-000002fb]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Oct 21 10:31:35] -- Executing [h@default:1] AGI("Local/8600051@default-0000014f;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Oct 21 10:31:36] -- <Local/8600051@default-0000014f;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
vicibox9*CLI> quit
[Oct 21 10:31:43] Asterisk cleanly ending (0).
[Oct 21 10:31:43] Executing last minute cleanups
> > > > > >> > > > END of LOG using MANUAL call..< < < < < < < <