Goautodial 4 - Campaign is not calling..

[paok1926]

Hello all,

I installed GoAutodial 4, at least 5-6 times the last week, but i still cant make calls..
Installation was from iso image, on a VM, enabled network, changed ip to kamailio, rtpeng, etc.. according to the manuals..
created an IAX carrier to connect to my existing pbx, when i create a sip account and connect a softphone, it works..

Then new campaign, new list, new script, agent logged in, no calls..
everything else is at the default settings.. Can anyone help me ?

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) If you were to explain why you were installing with Goautodial instead of Vicibox.com's .iso, that may be useful.

4) "I still can't make calls" is not something you prove by providing all your configuration information. You prove this by providing CLI output from asterisk after verifying that the system is trying to make calls. Stage one:

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screen -list

How many screens are running?

[paok1926]

hello,

I took your advice about posting my configuration very seriously, but i cant edit my profile and add it as a signature
Yesterday, when i stuck with GoAutodial, i tried Vicibox. But administration pages, aesthetically, look older than me (I'm 57) and i think it offers much-much more than i can use..

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[root@cc2 ~]# screen -r
There are several suitable screens on:
   14013.ASTfastlog   (Detached)
   14010.ASTVDadapt   (Detached)
   14007.ASTVDremote   (Detached)
   14004.ASTVDauto   (Detached)
   14001.ASTlisten   (Detached)
   13998.ASTsend   (Detached)
   13995.ASTupdate   (Detached)
   12490.asterisk   (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.


[williamconley]

I took your advice about posting my configuration very seriously, but i cant edit my profile and add it as a signature

Call me skeptical: You don't know how to edit your signature (which is a learning disability, not a fault with the system, IMHO) is not a reason to leave the information out of your post. Putting it directly in your post is preferable anyway so it won't change on your older posts when you update your system. But leaving it off is not viable even for the extremely lazy. lol

Preaching/flaming/picking on you session over. Back to our regularly scheduled programming:

Yesterday, when i stuck with GoAutodial, i tried Vicibox. But administration pages, aesthetically, look older than me (I'm 57) and i think it offers much-much more than i can use..

Goautodial and Vicibox install the same underlying application (Vicidial). But Goautodial installs an older version and breaks some deeper and/or newer features while supplying a prettier interface. I would suggest that if you want a race car to get your bank account as full as possible, you want Vicibox. If, however, you want to create a pretty call-center so you can sell it to someone OR you have a deep requirement for CentOS (which some facilities have) Goautodial is marvelous.

Cooler yet: Vicidial (stock, unaltered, but slightly old) is likely installed on this server. The screens aren't linked or mentioned during the install, but they ARE there. Thus you can always test to see if a feature is broken by the goautodial interface by merely using the Vicidial interface instead for the same purpose. Albeit a full generation behind, it's still a valid install (goautodial is very good at installing Vicidial).

If you really are just now building your system: My best advice is that you start over with Vicibox now. After installing with Vicibox (instructions in a PDF on Vicibox.com) switch to the Vicidial manager's manual available on EFLO.net.

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[root@cc2 ~]# screen -r
There are several suitable screens on:
   14013.ASTfastlog   (Detached)
   14010.ASTVDadapt   (Detached)
   14007.ASTVDremote   (Detached)
   14004.ASTVDauto   (Detached)
   14001.ASTlisten   (Detached)
   13998.ASTsend   (Detached)
   13995.ASTupdate   (Detached)
   12490.asterisk   (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.


Excellent. they are running. Now we go to "is it trying to dial". Asterisk is running in a screen and is likely writing to a log file (or several). You can either check the log file(s) or log in to the screen and see what happens when it tries to dial (IF it tries to dial).

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/var/log/asterisk
/var/log/astguiclient

or

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screen -r asterisk

Note that to get out of the screen, you do not type exit: You use 'ctrl-a' then 'ctrl-d' to "detach" from the screen and leave it running (important as asterisk is actually running in this console, exiting or 'ctrl-c' breaking would kill the pbx!)

[paok1926]

this is what i see when web client logins to dialer..

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[root@cc2 ~]# tail -f /var/log/asterisk/messages /var/log/astguiclient/action_process.2019-01-15

==> /var/log/astguiclient/action_process.2019-01-15 <==
2019-01-15 21:35:30|SQL_QUERY|SELECT man_id,uniqueid,entry_date,status,response,server_ip,channel,action,callerid,cmd_line_b,cmd_line_c,cmd_line_d,cmd_line_e,cmd_line_f,cmd_line_g,cmd_line_h,cmd_line_i,cmd_line_j,cmd_line_k FROM vicidial_manager where server_ip = '127.0.0.1' and status = 'QUEUE' order by man_id limit 1||
2019-01-15 21:35:30|2019-01-15 21:35:30|2019-01-16 03:35:29|29162|||Originate|S1901160335298600051|
2019-01-15 21:35:30|----BEGIN NEW COMMAND----
CallerID: S1901160335298600051
Action: Originate
Channel: Local/99998291286073@default
Context: default
Exten: 8600051
Priority: 1
Callerid: "S1901160335298600051" <5164536886>

----END NEW COMMAND----
|
2019-01-15 21:35:30|SQL_QUERY|UPDATE vicidial_manager set status='SENT' where man_id='29162'||

==> /var/log/asterisk/messages <==
[Jan 15 21:35:30] VERBOSE[14460] manager.c: Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 21:35:30] VERBOSE[14461] dial.c: Called 99998291286073@default
[Jan 15 21:35:30] VERBOSE[14462][C-00000006] pbx.c: Executing [99998291286073@default:1] Dial("Local/99998291286073@default-00000006;2", "SIP/8291286073@kamailio,,tTo") in new stack
[Jan 15 21:35:30] WARNING[14462][C-00000006] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Jan 15 21:35:30] VERBOSE[14462][C-00000006] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Jan 15 21:35:30] VERBOSE[14462][C-00000006] pbx.c: Executing [99998291286073@default:2] Hangup("Local/99998291286073@default-00000006;2", "") in new stack
[Jan 15 21:35:30] VERBOSE[14462][C-00000006] pbx.c: Spawn extension (default, 99998291286073, 2) exited non-zero on 'Local/99998291286073@default-00000006;2'
[Jan 15 21:35:30] WARNING[14462][C-00000006] func_hangupcause.c: Unable to find information for channel
[Jan 15 21:35:30] VERBOSE[14462][C-00000006] pbx.c: Executing [h@default:1] AGI("Local/99998291286073@default-00000006;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------)") in new stack
[Jan 15 21:35:30] VERBOSE[14462][C-00000006] res_agi.c: <Local/99998291286073@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------) completed, returning 0
[Jan 15 21:35:30] VERBOSE[14460] manager.c: Manager 'sendcron' logged off from 127.0.0.1



and this happens when he logs out..

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==> /var/log/astguiclient/action_process.2019-01-15 <==
2019-01-15 21:36:40|SQL_QUERY|SELECT man_id,uniqueid,entry_date,status,response,server_ip,channel,action,callerid,cmd_line_b,cmd_line_c,cmd_line_d,cmd_line_e,cmd_line_f,cmd_line_g,cmd_line_h,cmd_line_i,cmd_line_j,cmd_line_k FROM vicidial_manager where server_ip = '127.0.0.1' and status = 'QUEUE' order by man_id limit 1||
2019-01-15 21:36:40|2019-01-15 21:36:40|2019-01-16 03:36:40|29163|||Originate|ULGH34581547581000|
2019-01-15 21:36:40|----BEGIN NEW COMMAND----
CallerID: ULGH34581547581000
Action: Originate
Channel: Local/55558600051@default
Context: default
Exten: 8300
Priority: 1
Callerid: ULGH34581547581000
8600051

----END NEW COMMAND----
|
2019-01-15 21:36:40|SQL_QUERY|UPDATE vicidial_manager set status='SENT' where man_id='29163'||

==> /var/log/asterisk/messages <==
[Jan 15 21:36:40] VERBOSE[14642] manager.c: Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 21:36:40] VERBOSE[14643] dial.c: Called 55558600051@default
[Jan 15 21:36:40] VERBOSE[14644][C-00000007] pbx.c: Executing [55558600051@default:1] Hangup("Local/55558600051@default-00000007;2", "") in new stack
[Jan 15 21:36:40] VERBOSE[14644][C-00000007] pbx.c: Spawn extension (default, 55558600051, 1) exited non-zero on 'Local/55558600051@default-00000007;2'
[Jan 15 21:36:40] WARNING[14644][C-00000007] func_hangupcause.c: Unable to find information for channel
[Jan 15 21:36:40] VERBOSE[14644][C-00000007] pbx.c: Executing [h@default:1] AGI("Local/55558600051@default-00000007;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Jan 15 21:36:40] VERBOSE[14644][C-00000007] res_agi.c: <Local/55558600051@default-00000007;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Jan 15 21:36:40] VERBOSE[14642] manager.c: Manager 'sendcron' logged off from 127.0.0.1



[williamconley]

I don't care what happens when he logs in, except:

Does his phone ring? After his phone rings, does he hear "You are the only person in this conference"? If those things are true, login and logout no longer matter.

But when he pushes whatever button indicates he's "available" (not paused, unpaused, resumed, whatever the agent screen has in your version), what happens then is of utmost importance, along with the Carrier Dialplan Entry that causes it.

I've also taken to putting an "s" dialplan entry in (which is a 'catch-all') to show ANY attempts to dial. Without that, any attempt to dial that does not match a dialplan patter may fail invisibly, so the 's' dialplan entry is very handy if only to show an attempt occurred.

Adding this to ONE carrier (or making a generic carrier just for this) will expose all dial attempts, even those that fail, to the asterisk CLI output:

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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

[paok1926]

I created (copied) a carrier, and changed the dial plan according to your orders..

when the client logins to dialer, i see this:

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    -- Called 99998291286073@default
    -- Executing [99998291286073@default:1] Dial("Local/99998291286073@default-00000008;2", "SIP/8291286073@kamailio,,tTo") in new stack
[Jan 15 22:00:42] WARNING[17844][C-00000008]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [99998291286073@default:2] Hangup("Local/99998291286073@default-00000008;2", "") in new stack
  == Spawn extension (default, 99998291286073, 2) exited non-zero on 'Local/99998291286073@default-00000008;2'
[Jan 15 22:00:42] WARNING[17844][C-00000008]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/99998291286073@default-00000008;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------)") in new stack
    -- <Local/99998291286073@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------------) completed, returning 0


even if i try to dial a number manually, i see nothing on asterisk cli..

[williamconley]

SIP/8291286073@kamailio,,tTo") in new stack
[Jan 15 22:00:42] WARNING[17844][C-00000008]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

kamailio is not a valid sip account. what do you have in this carrier's "account entry" field? (change user/pass/ip to Xs of course)

[paok1926]

it was a misconfiguration in kamailio...

now, when the agent logs in i see:

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    -- Called 99998291286073@default
    -- Executing [99998291286073@default:1] Dial("Local/99998291286073@default-00000002;2", "SIP/8291286073@kamailio,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/8291286073@kamailio
    -- SIP/kamailio-00000001 is ringing
    -- Local/99998291286073@default-00000002;1 is ringing
       > 0x7fae60009c30 -- Strict RTP learning after remote address set to: 10.129.11.14:30050
    -- SIP/kamailio-00000001 answered Local/99998291286073@default-00000002;2
    -- Local/99998291286073@default-00000002;1 answered
    -- Executing [8600051@default:1] Konference("Local/99998291286073@default-00000002;1", "8600051,R") in new stack
    -- Channel SIP/kamailio-00000001 joined 'simple_bridge' basic-bridge <35a63db4-8736-43e0-b136-00b5f5f0b3de>
    -- Channel Local/99998291286073@default-00000002;2 joined 'simple_bridge' basic-bridge <35a63db4-8736-43e0-b136-00b5f5f0b3de>
  == Manager 'sendcron' logged off from 127.0.0.1
cc2*CLI>
[Jan 15 22:37:46] WARNING[14401][C-00000004]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format


and in a few moments it says RTP timeout:

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cc2*CLI>
    -- Channel SIP/kamailio-00000001 left 'simple_bridge' basic-bridge <35a63db4-8736-43e0-b136-00b5f5f0b3de>
    -- Channel Local/99998291286073@default-00000002;2 left 'simple_bridge' basic-bridge <35a63db4-8736-43e0-b136-00b5f5f0b3de>
  == Spawn extension (default, 99998291286073, 1) exited non-zero on 'Local/99998291286073@default-00000002;2'
    -- Executing [h@default:1] AGI("Local/99998291286073@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----14-----SIP 200 OK)") in new stack
    -- <Local/99998291286073@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----14-----SIP 200 OK) completed, returning 0
[Jan 15 22:37:58] WARNING[14401][C-00000004]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/99998291286073@default-00000002;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/99998291286073@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
  == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 22:37:58] NOTICE[14433]: manager.c:4309 action_hangup: Request to hangup non-existent channel: Local/99998291286073@default-00000002;2
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 55558600051@default
    -- Executing [55558600051@default:1] Hangup("Local/55558600051@default-00000003;2", "") in new stack
  == Spawn extension (default, 55558600051, 1) exited non-zero on 'Local/55558600051@default-00000003;2'
[Jan 15 22:37:58] WARNING[14436][C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/55558600051@default-00000003;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
    -- <Local/55558600051@default-00000003;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0



[williamconley]

rtp timeout means audio packets are failing. sip.conf value for externip or firewall blocking rports between kamailio and vici

[paok1926]

rtp timeout means audio packets are failing. sip.conf value for externip or firewall blocking rports between kamailio and vici

i have no firewall..

this is a part of my (default) sip.conf

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;register => 1234:password@mysipprovider.com
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
;externip = 192.168.1.1        ; Address that we're going to put in outbound SIP
;externhost=test.test.com     ; Alternatively you can specify a domain
;externrefresh=10               ; How often to refresh externhost if
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0     ; Also RFC1918
;localnet=172.16.0.0/12          ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=force_rport,comedia         ; Global NAT settings  (Affects all peers and users)
canreinvite=no                  ; Asterisk by default tries to redirect the
;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes              ; Save systemname in realtime database at registration
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4                 ; Add IP address as local domain
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld        ; When making outbound SIP INVITEs to



i use 10.10.100.108 at eth0 and 10.129.11.14 at ppp0 (so i can see the box remotely)

[paok1926]

kamailio is registered...

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cc2*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                     
kamailio                  127.0.0.1                                   Yes        Yes            5060     OK (1 ms)           


[williamconley]

is kamailio on the same server? your audio packets are failing (that's what the error means). why they are failing is entirely up to you.

um ... why are you using kamailio in the first place?

[paok1926]

yes, goautodial installs all them on the same server..

i still cant understand the architecture of vicidial... kamailio for the agents, asterisk for the carriers ? and rtpenc to connect webrtc to kamailio ?

p.s. i sent you a pm, please take a look

[williamconley]

there is NO need for kamailio to be involved. Just put in a real carrier. Agents and carriers both connect to Asterisk. Asterisk is controlled by Vicidial's scripts.

[cavagnaro]

I think we are fighting against Kamailio because GoAutodial 4 installs it by default. It is not our choice, it is there by some reason...
Have you tested V4 by the way?
I can't find a single post explaining exactly why but it is there. Also can't find a single post on showing how does kamailio sends the data to asterisk under GoAutoDial v4 kamailio conf files...

[williamconley]

because v4 was never fully released. this is goautodial, not vicidial. gardo gets busy.