vicidial.org

This is a brand new install and I cannot place an outbound manual call from the vici agent screen but can dial out straight from Xlite. Here is my dialplan and cli output for the call from vicidial agent screen. This is a dual server cluster setup and I have another 2 single server setups that have these same settings exactly with no issues. So I am lost trying to figure out why it will not dial out. When I dial from the vici agent screen it just hangs up right away and I get this error.

DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 1 - Unallocated (unassigned) number.
SIP: 404 - Not Found

------------------------------------------------------------

[carrier]
host=xx.xx.xxx.xx
type=peer
port=5060
disallow=all
allow=ulaw
allow=alaw
nat=force_rport,comedia
directmedia=no
context=trunkinbound
dtmfmode=rfc2833
trustrpid=yes
sendrpid=yes
qualify=yes
insecure=port,invite

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(SIP/carrier/XXXX#${EXTEN},,To)
exten => _1NXXNXXXXXX,3,Hangup

-----------------------------------------------------------------------------------

Manual Dial from Vici Agent Screen Doesn;t work

[May 8 07:30:45] > Channel SIP/1234-00000002 was answered
[May 8 07:30:45] -- Executing [8600051@default:1] MeetMe("SIP/1234-00000002", "8600051,F") in new stack
[May 8 07:30:45] == Parsing '/etc/asterisk/meetme.conf': Found
[May 8 07:30:45] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[May 8 07:30:45] -- Created MeetMe conference 1023 for conference '8600051'
[May 8 07:30:45] -- <SIP/1234-00000002> Playing 'conf-onlyperson.gsm' (language 'en')
[May 8 07:30:45] > 0x7f2d44019f20 -- Strict RTP switching to RTP remote address 192.168.1.141:62958 as source
[May 8 07:30:46] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 07:30:46] > 0x7f2d44019f20 -- Strict RTP learning complete - Locking on source address 192.168.1.141:62958
[May 8 07:30:53] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 07:30:53] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000002;2", "8600051,F") in new stack
[May 8 07:30:53] > Channel Local/8600051@default-00000002;1 was answered
[May 8 07:30:53] -- Executing [1651xxxxxxx@default:1] AGI("Local/8600051@default-00000002;1", "agi://127.0.0.1:4577/call_log") in new stack
[May 8 07:30:53] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=NCP_SALE))
[May 8 07:30:53] -- <Local/8600051@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 8 07:30:53] -- Executing [1651xxxxxxx@default:2] Dial("Local/8600051@default-00000002;1", "SIP/carrier/xxxx#1651xxxxxxx,,To") in new stack
[May 8 07:30:53] == Using SIP RTP CoS mark 5
[May 8 07:30:53] -- Called SIP/carrier/xxxx#1651xxxxxxx
[May 8 07:30:54] == Everyone is busy/congested at this time (1:0/0/1)
[May 8 07:30:54] -- Executing [1651xxxxxxx@default:3] Hangup("Local/8600051@default-00000002;1", "") in new stack
[May 8 07:30:54] == Spawn extension (default, 1651xxxxxxx, 3) exited non-zero on 'Local/8600051@default-00000002;1'
[May 8 07:30:54] -- Executing [h@default:1] AGI("Local/8600051@default-00000002;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL----------") in new stack
[May 8 07:30:54] -- <Local/8600051@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CHANUNAVAIL---------- completed, returning 0
[May 8 07:30:54] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000002;2'
[May 8 07:30:54] -- Executing [h@default:1] AGI("Local/8600051@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 8 07:30:54] -- <Local/8600051@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0

-------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

Manual Dial from Xlite Works

[May 8 07:47:02] > 0x7f2e3c012a50 -- Strict RTP learning after remote address set to: 192.168.1.141:63504
[May 8 07:47:02] -- Executing [1651xxxxxxx@default:1] AGI("SIP/1234-00000004", "agi://127.0.0.1:4577/call_log") in new stack
[May 8 07:47:02] -- <SIP/1234-00000004>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 8 07:47:02] -- Executing [1651xxxxxxx@default:2] Dial("SIP/1234-00000004", "SIP/carrier/xxxx#1651xxxxxxx,,To") in new stack
[May 8 07:47:02] == Using SIP RTP CoS mark 5
[May 8 07:47:02] -- Called SIP/carrier/xxxx#1651xxxxxxx
[May 8 07:47:05] > 0x7f2d38014940 -- Strict RTP learning after remote address set to: 64.206.43.107:20050
[May 8 07:47:05] -- SIP/teleinx-00000005 is making progress passing it to SIP/1234-00000004
[May 8 07:47:05] > 0x7f2e3c012a50 -- Strict RTP switching to RTP remote address 192.168.1.141:63504 as source
[May 8 07:47:05] > 0x7f2d38014940 -- Strict RTP switching to RTP remote address 64.206.43.107:20050 as source
[May 8 07:47:05] > 0x7f2e3c012a50 -- Strict RTP learning complete - Locking on source address 192.168.1.141:63504
[May 8 07:47:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 07:47:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 07:47:07] > 0x7f2d38014940 -- Strict RTP learning complete - Locking on source address 64.206.43.107:20050
[May 8 07:47:10] -- SIP/carrier-00000005 answered SIP/1234-00000004
[May 8 07:47:17] -- Executing [h@default:1] AGI("SIP/1234-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----7") in new stack
[May 8 07:47:17] -- <SIP/1234-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----7 completed, returning 0
[May 8 07:47:17] == Spawn extension (default, 1651xxxxxxx, 2) exited non-zero on 'SIP/1234-00000004'



Vicibox 8.0.1 from .iso | Vicidial 2.14-669a Build 180411-1647 | Asterisk 11.25.3-vici | Cluster 2 Servers | No Digium/Sangoma Hardware | No Extra Software After Installation | 2 x 2.4 GHz Quad-Core Intel Xeon Processor with 12MB Cache -- E5620

Code: Select all

[May 8 07:30:53] -- Called SIP/carrier/xxxx#1651xxxxxxx
[May 8 07:30:54] == Everyone is busy/congested at this time (1:0/0/1)

1) Show a phone number (use one that doesn't matter). Remove the x's from the above sample, they could be relevant. Call a Hilton or Prayer line.

2) Ensure your campaign has a callerid. Provide your manual dial callerID settings from the campaign.

3) The carrier rejected the call. Ask them why. Surprisingly enough that often turns out to be very easy and enlightening.

Okay so the CallerID was set to 0000000000 and I changed that to just a generic fake number temporarily and it dials out now. I can't believe it's something that simple lol

Has this always been an issue because I could swear I've dialed on my other dialer's without a CallerID number populated when testing.

Either way it works so thank you yet again sir!

Recent government rules have changed. Carriers are no longer required to complete calls that are potentially illegal robodialers. Using 0000000000 as a CID, for instance, are an indication that the call is fake. Unassigned CIDs are often now using the CID Name "Possible SCAM". Awkward. So using your own DID is now a wise choice.

yeah interesting thanks.

Now on my other dialer with the exact same settings and dialer configuration except its a single server dialer. I can dial manually but autodial is not working for some reason. There are leads in the hopper and its set to ratio 2 but no calls are dialing per the real-time report. The agent shows ready but NO LIVE CALLS WAITING

any ideas on that?

kjburto wrote:yeah interesting thanks.

Now on my other dialer with the exact same settings and dialer configuration except its a single server dialer. I can dial manually but autodial is not working for some reason. There are leads in the hopper and its set to ratio 2 but no calls are dialing per the real-time report. The agent shows ready but NO LIVE CALLS WAITING

any ideas on that?

On your previous request, you showed asterisk CLI output. Now you just use the real-time report as your source. You could have any one of several hundred problems. Wow.

Screens running? Database not crashed? One of those screens controls dialing ... but if it's already showing dialing in the asterisk CLI, then the CLI may have a hint. Of course it could also be something as simple as a bad dial prefix (we always put an "s" extension in a Default carrier to capture mismatched dialplans).

sorry my bad it's been a long day. The CLI isn't showing anything really. It's not even trying to dial.

[May 8 18:20:57] == Using SIP RTP CoS mark 5
[May 8 18:21:00] > 0x7fa1c800d1e0 -- Strict RTP learning after remote address set to: 192.168.1.141:50788
[May 8 18:21:00] > Channel SIP/1234-00000006 was answered
[May 8 18:21:00] -- Executing [8600051@default:1] MeetMe("SIP/1234-00000006", "8600051,F") in new stack
[May 8 18:21:00] == Parsing '/etc/asterisk/meetme.conf': Found
[May 8 18:21:00] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[May 8 18:21:00] -- Created MeetMe conference 1023 for conference '8600051'
[May 8 18:21:00] -- <SIP/1234-00000006> Playing 'conf-onlyperson.gsm' (language 'en')
[May 8 18:21:00] > 0x7fa1c800d1e0 -- Strict RTP switching to RTP remote address 192.168.1.141:50788 as source
[May 8 18:21:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:21:01] > 0x7fa1c800d1e0 -- Strict RTP learning complete - Locking on source address 192.168.1.141:50788
[May 8 18:21:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:21:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:21:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:21:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:21:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:21:06] == Manager 'sendcron' logged off from 127.0.0.1

[carrier]
host=xx.27.xxx.xx
type=peer
port=5060
disallow=all
allow=ulaw
allow=alaw
nat=force_rport,comedia
directmedia=no
context=trunkinbound
dtmfmode=rfc2833
trustrpid=yes
sendrpid=yes
qualify=yes
insecure=port,invite

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(SIP/carrier/xxxx#${EXTEN},,To)
exten => _1NXXNXXXXXX,3,Hangup

and This is what it shows sitting on ready then making a manual dial call which worked.


[May 8 18:22:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:22:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:22:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:22:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:22:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:22:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:23:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:23:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:23:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:23:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:23:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:23:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:24:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:24:02] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:24:02] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:24:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:24:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:24:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:25:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:25:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:25:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:25:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:25:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:25:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:26:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:26:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:26:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:26:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:26:14] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:26:14] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000006;2", "8600051,F") in new stack
[May 8 18:26:14] > Channel Local/8600051@default-00000006;1 was answered
[May 8 18:26:14] -- Executing [18008693557@default:1] AGI("Local/8600051@default-00000006;1", "agi://127.0.0.1:4577/call_log") in new stack
[May 8 18:26:14] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=900_NG_S))
[May 8 18:26:14] -- <Local/8600051@default-00000006;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 8 18:26:14] -- Executing [18008693557@default:2] Dial("Local/8600051@default-00000006;1", "SIP/carrier/xxxx#18008693557,,To") in new stack
[May 8 18:26:14] == Using SIP RTP CoS mark 5
[May 8 18:26:14] -- Called SIP/carrier/xxxx#18008693557
[May 8 18:26:15] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:26:19] > 0x7fa1c0014620 -- Strict RTP learning after remote address set to: 8.20.212.170:28128
[May 8 18:26:19] -- SIP/carrier-00000007 is making progress passing it to Local/8600051@default-00000006;1
[May 8 18:26:19] > 0x7fa1c0014620 -- Strict RTP switching to RTP remote address 8.20.212.170:28128 as source
[May 8 18:26:19] -- SIP/carrier-00000007 answered Local/8600051@default-00000006;1
[May 8 18:26:20] > 0x7fa1c0014620 -- Strict RTP learning complete - Locking on source address 8.20.212.170:28128
[May 8 18:26:36] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:26:36] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600051@default-00000006;1
[May 8 18:26:36] -- Executing [h@default:1] AGI("Local/8600051@default-00000006;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----22-----17") in new stack
[May 8 18:26:36] -- <Local/8600051@default-00000006;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -22-----17 completed, returning 0
[May 8 18:26:36] == Spawn extension (default, 18008693557, 2) exited non-zero on 'Local/8600051@default-00000006;1'
[May 8 18:26:36] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000006;2'
[May 8 18:26:36] -- Executing [h@default:1] AGI("Local/8600051@default-00000006;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 8 18:26:36] -- <Local/8600051@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 8 18:26:36] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 18:26:36] NOTICE[27596]: manager.c:3407 action_hangup: Request to hangup non-existent channel: Local/8600051@default-00000006;2
[May 8 18:26:37] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 18:26:37] == Manager 'sendcron' logged off from 127.0.0.1

_1NXXNXXXXXX

If your dialplan has ONLY this dial pattern, add an "s" dialplan entry in a default carrier. Don't need anything else in this carrier.

Code: Select all

exten=> s,1,AGI(agi://127.0.0.1:4577/call_log)
exten=> s,n,NoOp(No Dial Pattern Matches This Extension)
exten=> s,n,Hangup

If the system is trying to dial, this will catch and kill and we'll see why _1NXXNXXXXXX wasn't enough. Dial prefix, Manual Dial prefix, phone_number and dial code ... vs dialplan entry.

I'm not sure what you are saying? I don't have a lot of experience or knowledge on dialplans as of yet. I have been using this dialplan on almost every server I've worked with. Do I use this:

exten=> s,1,AGI(agi://127.0.0.1:4577/call_log)
exten=> s,n,NoOp(No Dial Pattern Matches This Extension)
exten=> s,n,Hangup

Instead of my current dialplan in my current carrier setup?

NEW carrier with JUST this entry in the dialplan entries field. NEW. Not a change to the old one. This will capture and kill (visibly, in the asterisk ClI) impossible numbers. Otherwise, those that don't fit your existing dialplan entry may remain invisible and give no hint of why they fail since they never appeared in the first place.

Here is dialing manually


[May 8 20:50:41] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000008;2", "8600051,F") in new stack
[May 8 20:50:41] > Channel Local/8600051@default-00000008;1 was answered
[May 8 20:50:41] -- Executing [18008693557@default:1] AGI("Local/8600051@default-00000008;1", "agi://127.0.0.1:4577/call_log") in new stack
[May 8 20:50:41] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=900_NG_S))
[May 8 20:50:41] -- <Local/8600051@default-00000008;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 8 20:50:41] -- Executing [18008693557@default:2] Dial("Local/8600051@default-00000008;1", "SIP/carrier/xxxx#18008693557,,To") in new stack
[May 8 20:50:41] == Using SIP RTP CoS mark 5
[May 8 20:50:41] -- Called SIP/carrier/xxxx#18008693557
[May 8 20:50:42] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:50:45] > 0x7fa1c00157b0 -- Strict RTP learning after remote address set to: 8.28.85.29:20912
[May 8 20:50:45] -- SIP/carrier-00000009 is making progress passing it to Local/8600051@default-00000008;1
[May 8 20:50:45] > 0x7fa1c00157b0 -- Strict RTP switching to RTP remote address 8.28.85.29:20912 as source
[May 8 20:50:45] -- SIP/carrier-00000009 answered Local/8600051@default-00000008;1
[May 8 20:50:47] > 0x7fa1c00157b0 -- Strict RTP learning complete - Locking on source address 8.28.85.29:20912
[May 8 20:51:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:51:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:51:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:51:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:07] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600051@default-00000008;1
[May 8 20:51:07] -- Executing [h@default:1] AGI("Local/8600051@default-00000008;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----22") in new stack
[May 8 20:51:07] -- <Local/8600051@default-00000008;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----22 completed, returning 0
[May 8 20:51:07] == Spawn extension (default, 18008693557, 2) exited non-zero on 'Local/8600051@default-00000008;1'
[May 8 20:51:07] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000008;2'
[May 8 20:51:07] -- Executing [h@default:1] AGI("Local/8600051@default-00000008;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[May 8 20:51:07] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:07] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600051@default-00000008;2
[May 8 20:51:07] -- <Local/8600051@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[May 8 20:51:08] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:51:08] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:51:15] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:15] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/1234-00000008
[May 8 20:51:15] -- Hungup 'DAHDI/pseudo-744064104'
[May 8 20:51:15] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/1234-00000008'
[May 8 20:51:15] -- Executing [h@default:1] AGI("SIP/1234-00000008", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 8 20:51:15] -- <SIP/1234-00000008>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[May 8 20:51:15] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:51:15] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000009;2", "8600051,K") in new stack
[May 8 20:51:15] WARNING[17523][C-00000010]: app_meetme.c:5053 admin_exec: Conference number '8600051' not found!
[May 8 20:51:15] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000009;2", "") in new stack
[May 8 20:51:15] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000009;2'
[May 8 20:51:15] -- Executing [h@default:1] AGI("Local/55558600051@default-00000009;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 8 20:51:15] -- <Local/55558600051@default-00000009;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[May 8 20:51:16] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:51:16] == Manager 'sendcron' logged off from 127.0.0.1


and here is autodial

[May 8 20:52:12] == Using SIP RTP CoS mark 5
[May 8 20:52:14] > 0x7fa1c0014100 -- Strict RTP learning after remote address set to: 192.168.1.141:61754
[May 8 20:52:14] > Channel SIP/1234-0000000a was answered
[May 8 20:52:14] -- Executing [8600051@default:1] MeetMe("SIP/1234-0000000a", "8600051,F") in new stack
[May 8 20:52:14] == Parsing '/etc/asterisk/meetme.conf': Found
[May 8 20:52:14] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[May 8 20:52:14] -- Created MeetMe conference 1023 for conference '8600051'
[May 8 20:52:14] -- <SIP/1234-0000000a> Playing 'conf-onlyperson.gsm' (language 'en')
[May 8 20:52:14] > 0x7fa1c0014100 -- Strict RTP switching to RTP remote address 192.168.1.141:61754 as source
[May 8 20:52:15] == Manager 'sendcron' logged off from 127.0.0.1
[May 8 20:52:16] > 0x7fa1c0014100 -- Strict RTP learning complete - Locking on source address 192.168.1.141:61754
[May 8 20:52:44] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:52:44] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/1234-0000000a
[May 8 20:52:44] -- Hungup 'DAHDI/pseudo-1865447446'
[May 8 20:52:44] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/1234-0000000a'
[May 8 20:52:44] -- Executing [h@default:1] AGI("SIP/1234-0000000a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 8 20:52:44] -- <SIP/1234-0000000a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[May 8 20:52:44] == Manager 'sendcron' logged on from 127.0.0.1
[May 8 20:52:44] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-0000000a;2", "8600051,K") in new stack
[May 8 20:52:44] WARNING[23930][C-00000012]: app_meetme.c:5053 admin_exec: Conference number '8600051' not found!
[May 8 20:52:44] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-0000000a;2", "") in new stack
[May 8 20:52:44] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-0000000a;2'
[May 8 20:52:44] -- Executing [h@default:1] AGI("Local/55558600051@default-0000000a;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 8 20:52:44] -- <Local/55558600051@default-0000000a;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0


It's just not doing anything at all, not even trying to dial

Assuming of course, you have leads available to be dialed and leads in the hopper and at least one READY agent in that campaign ... (verify each of these on that specific campaign)

Check your autodial "screen" (find with screen -list) and find out why that campaign's not making calls.

Yep leads available to be dialed and in the hopper with an open agent ready to take calls and it's not dialing anything at all. I am lost not sure what it could be, but I am sure it is just something simple and small I am overlooking.

NewGen1:~ # screen -list
There are screens on:
8279.ASTVDremote (Detached)
8276.ASTVDauto (Detached)
8273.ASTlisten (Detached)
8270.ASTsend (Detached)
8267.ASTupdate (Detached)
3777.ASTemail (Detached)
3774.ASTVDadFILL (Detached)
3770.ASTfastlog (Detached)
3767.ASTVDadapt (Detached)
3701.asterisk (Detached)
3474.astshell20180508211959 (Detached)
11 Sockets in /var/run/screens/S-root.

Does the campaign's real-time screen show any trunk "short/fill" values?

If not:

For a "live" view showing the logic involved in dialing:

Code: Select all

screen -r ASTVDauto

ctrl-a - ctrl-d to get out of it.

Beware: If you kill this process, it can stall the dialer (so control C is a bad reflex, and there is NO "exit" command available)

Well its too early to make any calls but I was looking at that live view and it's showing no live agents. Shouldn't this still show there is a live agent when Im logged in and ready, even though there are no leads to dial because it's too early?


loop counter: |5747427|
TIME DEBUG: -6.00|-5|1| GMT: 11:07
2018-05-09 06:07:26|SERVER CALLS PER SECOND MAXIMUM SET TO: 20 |50||
2018-05-09 06:07:26|LIVE AGENTS LOGGED IN: 0 ACTIVE CALLS: 0|
2018-05-09 06:07:26| : agents: 0 (READY: 0) dial_level: 0 (||) |
2018-05-09 06:07:26| : Calls to place: 0 (0 - 0 [0 + 0|0|0|0]) 0 |
2018-05-09 06:07:26|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2018-05-09 06:07:26|LOCAL TRUNK SHORTAGE: 0|0 (0 - 250)|
2018-05-09 06:07:26| : CALLING|
TIME DEBUG: -6.00|-5|1| GMT: 11:07
TIME DEBUG: -6.00|-5|1| GMT: 11:07

check for crashed tables.

Code: Select all

mysqlcheck --all-databases --auto-repair --user=cron -p1234

Verify you're really logged in to the campaign in question (campaign ID is shown in the agent screen).

I ran that and everything came back ok and I doubled checked the campaign i'm logged into and it is correct. I have dialable leads and the real time report shows me logged in and ready but it still shows zero's across the board in ASTVauto.

TRUNK SHORT/FILL: / 0

loop counter: |5744047|
TIME DEBUG: -6.00|-5|1| GMT: 22:43
2018-05-09 17:43:59|SERVER CALLS PER SECOND MAXIMUM SET TO: 20 |50||
2018-05-09 17:43:59|LIVE AGENTS LOGGED IN: 0 ACTIVE CALLS: 0|
2018-05-09 17:43:59| : agents: 0 (READY: 0) dial_level: 0 (||) |
2018-05-09 17:43:59| : Calls to place: 0 (0 - 0 [0 + 0|0|0|0]) 0 |
2018-05-09 17:43:59|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2018-05-09 17:43:59|LOCAL TRUNK SHORTAGE: 0|0 (0 - 250)|
2018-05-09 17:43:59| : CALLING|
TIME DEBUG: -6.00|-5|1| GMT: 22:43
TIME DEBUG: -6.00|-5|1| GMT: 22:43

According to this the dial level isn't showing correctly either as it should show 3

I don't see the trunk short value, just the fill value.

I also don't see the campaign ID in your campaign output. Change all the other campaigns to inactive so the only one in the screen will be your specific campaign and the campaign ID will show. (and verify that this is in fact the campaign the agent is logged in to).

Also verify that the server's IP has never changed. If it has, pay attention to the ip update script mentioned in the splash screen during login to the console of the server. running that script with *all* prior IPs of the server never hurts if the system is a single server and the "new ip" is always the correct IP for the server.

Okay I just changed closer default blended under the user profile I was using to test to 1 from 0 and now it's dialing now.

I didn't realize that had to be set at 1 for it to work.

loop counter: |5743803|
TIME DEBUG: -6.00|-5|1| GMT: 22:54
2018-05-09 17:54:13|SERVER CALLS PER SECOND MAXIMUM SET TO: 20 |50||
2018-05-09 17:54:13|LIVE AGENTS LOGGED IN: 0 ACTIVE CALLS: 2|
2018-05-09 17:54:13|OLD TRUNK SHORTS CLEARED: 1 |''||
2018-05-09 17:54:13| : agents: 0 (READY: 0) dial_level: 0 (||) |
2018-05-09 17:54:13| : Calls to place: 0 (0 - 0 [0 + 0|0|0|0]) 2 |
2018-05-09 17:54:13|CAMPAIGN DIFFERENTIAL: 0.75 0.45 (0.45 - 0)|
2018-05-09 17:54:13|LOCAL TRUNK SHORTAGE: 0|0 (2 - 250)|
2018-05-09 17:54:13| : CALLING|
TIME DEBUG: -6.00|-5|1| GMT: 22:54
2018-05-09 17:54:13|| dead call vac deleted|8|SENT|18|V5091753420000000018|1525906442|1||OUT|19|3|22|19|23|0E0||
2018-05-09 17:54:13||SELECT result FROM vicidial_cpd_log where callerid='V5091753420000000018' order by cpd_id desc limit 1;||
2018-05-09 17:54:13||SELECT list_id,called_count from vicidial_list where lead_id='18' limit 1;||
2018-05-09 17:54:13|| dead NA call added to log 1525906423.46|18|8177189523|SENT|NA|1|1|0|
2018-05-09 17:54:13|| dead call vac lead marked NA|18|8177189523|SENT||
2018-05-09 17:54:13||SELECT auto_alt_dial,auto_alt_dial_statuses,use_internal_dnc,use_campaign_dnc,use_other_campaign_dnc FROM vicidial_campaigns where campaign_id='900_NG_S';|NONE| B N NA DC -|NA|18|NONE|
2018-05-09 17:54:13|| dead call vac deleted|7|SENT|17|V5091753420000000017|1525906442|1||OUT|19|3|22|20|24|0E0||
2018-05-09 17:54:13||SELECT result FROM vicidial_cpd_log where callerid='V5091753420000000017' order by cpd_id desc limit 1;||
2018-05-09 17:54:13||SELECT list_id,called_count from vicidial_list where lead_id='17' limit 1;||
2018-05-09 17:54:13|| dead NA call added to log 1525906422.43|17|2098948118|SENT|NA|1|1|0|
2018-05-09 17:54:13|| dead call vac lead marked NA|17|2098948118|SENT||
2018-05-09 17:54:13||SELECT auto_alt_dial,auto_alt_dial_statuses,use_internal_dnc,use_campaign_dnc,use_other_campaign_dnc FROM vicidial_campaigns where campaign_id='900_NG_S';|NONE| B N NA DC -|NA|17|NONE|
TIME DEBUG: -6.00|-5|1| GMT: 22:54

williamconley wrote:I don't see the trunk short value, just the fill value.

I also don't see the campaign ID in your campaign output. Change all the other campaigns to inactive so the only one in the screen will be your specific campaign and the campaign ID will show. (and verify that this is in fact the campaign the agent is logged in to).

Also verify that the server's IP has never changed. If it has, pay attention to the ip update script mentioned in the splash screen during login to the console of the server. running that script with *all* prior IPs of the server never hurts if the system is a single server and the "new ip" is always the correct IP for the server.

Yeah this is a brand new install so the IP address was never changed.

It's definitely that closer default blended option under user profile because I just went and changed it back to 0 and it doesn't work then back to 1 and it works fine.