vicidial.org

Vicibox Install: ViciBox_v8.x86_64-8.0.1
Build: 4223:$build = '171224-1220'

Asterisk 11.25.3-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.

Path: .
Working Copy Root Path: /usr/src/astguiclient/trunk
URL: svn://svn.eflo.net/agc_2-X/trunk
Relative URL: ^/agc_2-X/trunk
Repository Root: svn://svn.eflo.net
Repository UUID: 3d104415-ff17-0410-8863-d5cf3c621b8a
Revision: 2876
Node Kind: directory
Schedule: normal
Last Changed Author: mattf
Last Changed Rev: 2876
Last Changed Date: 2017-12-27 12:47:35 +0000 (Wed, 27 Dec 2017)

I am new to vicidial. My question is when I am logged into the inbound campaign, calls that come in do not show in the agent web application and do not show in the Real-Time Main Report. My on hook phone does ring and shows the caller Id. Outbound calls working great.

I have CLI Output of the call to my phone from the server and the inbound call to my phone.

[Sep 12 16:56:09] > 0x7f78e4015970 -- Strict RTP learning after remote address set to: 172.16.100.157:2232
[Sep 12 16:56:09] > Channel SIP/6001-000000a6 was answered
[Sep 12 16:56:09] -- Executing [8600051@default:1] MeetMe("SIP/6001-000000a6", "8600051,F") in new stack
[Sep 12 16:56:09] == Parsing '/etc/asterisk/meetme.conf': Found
[Sep 12 16:56:09] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Sep 12 16:56:09] -- Created MeetMe conference 1023 for conference '8600051'
[Sep 12 16:56:09] -- <SIP/6001-000000a6> Playing 'conf-onlyperson.gsm' (language 'en')
[Sep 12 16:56:09] > 0x7f78e4015970 -- Strict RTP switching to RTP remote address 172.16.100.157:2232 as source
[Sep 12 16:56:10] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 12 16:56:11] > 0x7f78e4015970 -- Strict RTP learning complete - Locking on source address 172.16.100.157:2232
[Sep 12 16:56:33] == Using SIP RTP CoS mark 5
[Sep 12 16:56:33] > 0x7f78f40c1950 -- Strict RTP learning after remote address set to: 54.172.60.156:10950
[Sep 12 16:56:33] -- Executing [+16363031785@trunkinbound:1] Dial("SIP/twilio1-000000a7", "SIP/6001") in new stack
[Sep 12 16:56:33] == Using SIP RTP CoS mark 5
[Sep 12 16:56:33] -- Called SIP/6001
[Sep 12 16:56:33] -- SIP/6001-000000a8 is ringing
[Sep 12 16:56:41] > 0x7f78f002b130 -- Strict RTP learning after remote address set to: 172.16.100.157:2230
[Sep 12 16:56:41] -- SIP/6001-000000a8 answered SIP/twilio1-000000a7
[Sep 12 16:56:41] > 0x7f78f002b130 -- Strict RTP switching to RTP remote address 172.16.100.157:2230 as source
[Sep 12 16:56:41] > 0x7f78f40c1950 -- Strict RTP switching to RTP remote address 54.172.60.156:10950 as source
[Sep 12 16:56:41] > 0x7f78f40c1950 -- Strict RTP learning complete - Locking on source address 54.172.60.156:10950
[Sep 12 16:56:43] > 0x7f78f002b130 -- Strict RTP learning complete - Locking on source address 172.16.100.157:2230
[Sep 12 16:57:06] -- Executing [h@trunkinbound:1] AGI("SIP/twilio1-000000a7", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----33-----25") in new stack
[Sep 12 16:57:06] -- <SIP/twilio1-000000a7>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -33-----25 completed, returning 0
[Sep 12 16:57:06] == Spawn extension (trunkinbound, +16363031785, 1) exited non-zero on 'SIP/twilio1-000000a7'

I hope that this is enough info to give you an idea of my dilemma.

You tried so hard. But you left off the Vicidial Version. 2.14.x.xxx?

Your asterisk cli only shows the hangup from the inbound call. Not entirely useful. Like showing the flat tire instead of the video showing the blowout.

You should also share the DID configuration and the configuration of wherever the DID's route is set to go (and so-on until it gets to either an ingroup or a phone, and if it's an ingroup share the campaign configuration demonstrating that the Ingroup is allowed in the campaign the agent is logged in to.

And a technical note: Campaigns are not inbound. Campaigns are outbound. Ingroups are Inbound. Ingroups can be 'allowed' on a campaign. So we don't know if your agent was actually logged in to a campaign AND that campaign allowed the ingroup, AND the agent selected that ingroup during login to accept calls from it. We also don't know if the inbound DID is configured to ring to the agent's phone or an ingroup or a campaign.

Next up: Test with logged in agent NOT using an "OnHook" phone before you jump into the OnHook configuration options. You should be following the instructions in the Vicidial Manager's Manual from beginning to end until everything works that you need. If you get all the way to the end of the Manual and you still have a feature (like OnHook), enable that after everything else is tested and functional. For instance: Have you confirmed Outbound calling works?

Here is the Vicidial Version: 2.14b0.5

DID route

[trunkinbound]
exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => +16363031785,1,Dial(SIP/6001)

Should this be pointed to a ingroup? If so what should be the correct entry?

I tried creating Add A New DID, New In-Group, and a New campaign to associate the New In-Group. Still the same result. Please excuse me in advance for any ignorance.

I forgot to mention that outbound calling is working both manual and through the campaign lists. All customer information and call statuses are displayed on the agent web application.

williamconley wrote:

williamconley wrote:Your asterisk cli only shows the hangup from the inbound call. Not entirely useful. Like showing the flat tire instead of the video showing the blowout.

It appears that the Inbound call is not even using the agi-DID_route.agi

I am getting the default message.
Here is the CLI.

[Sep 21 19:20:27] == Using SIP RTP CoS mark 5
[Sep 21 19:20:27] > 0x7f5758021a50 -- Strict RTP learning after remote address set to: 34.203.251.14:17744
[Sep 21 19:20:27] -- Executing [+16363894670@trunkinbound:1] AGI("SIP/twilio4-00000006", "agi-DID_route.agi") in new stack
[Sep 21 19:20:27] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Sep 21 19:20:27] -- <SIP/twilio4-00000006>AGI Script agi-DID_route.agi completed, returning 0
[Sep 21 19:20:27] -- Executing [9998811112@default:1] Wait("SIP/twilio4-00000006", "2") in new stack
[Sep 21 19:20:29] -- Executing [9998811112@default:2] Answer("SIP/twilio4-00000006", "") in new stack
[Sep 21 19:20:29] > 0x7f5758021a50 -- Strict RTP switching to RTP remote address 34.203.251.14:17744 as source
[Sep 21 19:20:29] -- Executing [9998811112@default:3] Playback("SIP/twilio4-00000006", "ss-noservice") in new stack
[Sep 21 19:20:29] -- <SIP/twilio4-00000006> Playing 'ss-noservice.gsm' (language 'en')
[Sep 21 19:20:29] > 0x7f5758021a50 -- Strict RTP learning complete - Locking on source address 34.203.251.14:17744
[Sep 21 19:20:32] == Spawn extension (default, 9998811112, 3) exited non-zero on 'SIP/twilio4-00000006'
[Sep 21 19:20:32] -- Executing [h@default:1] AGI("SIP/twilio4-00000006", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 21 19:20:32] -- <SIP/twilio4-00000006>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

I have the Sip Debug also of this call.

Add to [trunkinbound] context in /etc/asterisk/extensions.conf:

Code: Select all

; removal of Leading Plus Sign from the phone
exten => _+X!,1,Noop(Stripping + from start of number, for annoying carriers who insist)
exten => _+X!,n,Goto(trunkinbound,${EXTEN:1},1)
; removal of Leading 1 from the the phone (US numbers only, for annoying carriers who insist)
exten => _1!,1,Noop(Stripping 1 from start of number|exten${EXTEN}|len${LEN(${EXTEN})})
exten => _1!,n,Goto(trunkinbound,${EXTEN:1},1)


This will catch and kill the +1 and then send the call to vicidial scripting as a 10 digit US DID.