Issue with agent monitoring

[wadza]

Hi I am having an issue with agent monitoring. When I view the campaign report, set the options to show monitoring and click 'Listen' I get the message 'Success: Dialing 102' which is the extension number I am on, but nothing happens.

Asterisk CLI shows this:

[Apr 17 11:42:21] -- Executing [010*001*001*100*08600052@default:1] Goto("Local/010*001*001*100*08600052@default-000006ad;2", "default,08600052,1") in new stack
[Apr 17 11:42:21] -- Goto (default,08600052,1)
[Apr 17 11:42:21] -- Executing [08600052@default:1] Dial("Local/010*001*001*100*08600052@default-000006ad;2", "IAX2/ASTblind:GidX57Caret@127.0.0.1:41569/68600052,55,To") in new stack
[Apr 17 11:42:21] -- Called IAX2/ASTblind:GidXXXXXXXX@127.0.0.1:41569/68600052
[Apr 17 11:42:25] NOTICE[1583]: chan_iax2.c:4676 __auto_congest: Auto-congesting call due to slow response
[Apr 17 11:42:25] -- IAX2/127.0.0.1:41569-14321 is circuit-busy
[Apr 17 11:42:25] -- Hungup 'IAX2/127.0.0.1:41569-14321'
[Apr 17 11:42:25] == Everyone is busy/congested at this time (1:0/1/0)
[Apr 17 11:42:25] -- Executing [08600052@default:2] Hangup("Local/010*001*001*100*08600052@default-000006ad;2", "") in new stack
[Apr 17 11:42:25] == Spawn extension (default, 08600052, 2) exited non-zero on 'Local/010*001*001*100*08600052@default-000006ad;2'
[Apr 17 11:42:25] -- Executing [h@default:1] AGI("Local/010*001*001*100*08600052@default-000006ad;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack
[Apr 17 11:42:25] -- <Local/010*001*001*100*08600052@default-000006ad;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


It looks like it's dialing out through a carrier that is not even active... We only use SIP carriers anyway... Any ideas?

We are running Asterisk 1.8.19.0-vici-beta with Vicidial VERSION: 2.6-396a BUILD: 130402-2322

[williamconley]

it is dialing an inter-server number trying to reach you, but not getting a response and failing. perhaps it would help if you mentioned your install information. is this a multi-server installation? are you at 102 or cc102 or is there a phone extension 102 such as gs102 interfering with "your" 102? can you dial 102 from another phone in the office and make your phone ring?

newbie suggestions:
when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with 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 "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. 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

Happy Hunting!

[wadza]

Install from .iso | Vicidial VERSION: 2.6-396a BUILD: 130402-2322 | Asterisk 1.8.19.0-vici-beta | Single Server | No Extra Software After Installation | 8Gb ram, 500gb hdd| core 2 duo 3.0ghz

I can call from one extension to another no problem. The extension is named cc102. There are no other phones on 102 only this one..

Here is Asterisk CLI when I dial 102 from another extension:

[Apr 17 15:17:40] == Using SIP RTP CoS mark 5
[Apr 17 15:17:40] -- Executing [102@default:1] Dial("SIP/cc106-00000803", "SIP/cc102,60,") in new stack
[Apr 17 15:17:40] == Using SIP RTP CoS mark 5
[Apr 17 15:17:40] -- Called SIP/cc102
[Apr 17 15:17:40] -- SIP/cc102-00000804 is ringing
[Apr 17 15:17:40] -- SIP/cc102-00000804 is ringing
[Apr 17 15:17:51] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 17 15:17:51] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/optus-00000802'
[Apr 17 15:17:51] -- Executing [h@default:1] AGI("SIP/optus-00000802", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 17 15:17:51] -- <SIP/optus-00000802>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[williamconley]

Hm. Have you tried the manual for monitoring yet? (Often when utilizing a feature like this, the manual is actually helpful ...)

[wadza]

Manual only provides instruction telling me to do what i'm already doing.

I will try setting up an external phone with my cell phone.

The manual also has information on dialing 8162 from any vici connected SIP phone which gives me the ability to monitor conferences. Not sure if one of my dialplans which starts with an 8 will interfere with this. may need to change the dialplan?

[williamconley]

Manual only provides instruction telling me to do what i'm already doing.

I will try setting up an external phone with my cell phone.

The manual also has information on dialing 8162 from any vici connected SIP phone which gives me the ability to monitor conferences. Not sure if one of my dialplans which starts with an 8 will interfere with this. may need to change the dialplan?

you can also monitor by dialing the conference number (noisy) or add a 6 or 0 (i forget which) which will enter silently.

[mflorell]

You can also dial by entering the agent's ID or phone extension(if they are numbers only). If your code version information is correct it will work on your system:


# agi-phone_monitor.agi version 2.6
#
# This script is designed to allow managers to listen to agent sessions by
# dialing a phone exten number or user ID number. For this to work, the user ID
# or phone extension must be only digits. If you are using the phone extension,
# then all phone extensions on the system must be unique.
#
# user is prompted for a user ID or phone login then press #
# then the user is forwarded to monitor the active vicidial agent session
#
# ; settings for this script:
# ; 1. USER or PHONE, lookup by user id or phone extension (default is USER)
# ; 2. Y or N, log in api log (default is N)
# ; 3. BARGE or LISTEN, (default is LISTEN)
#
# can be added as an AGI to a Call Menu, or just a direct extension transfer:
#
# ; call menu AGI with options:
# agi-phone_monitor.agi,USER---N---LISTEN
#
# ; or as extension transfer to "up_monitor_exit" context and "s" extension
#

(and from the UPGRADE doc):

For this to work you need to add these lines to your extensions.conf:
(which can be added to the "System Settings -> Custom Dialplan" instead)

; quiet entry, listen-only, exit-on-dtmf conferences for VICIDIAL (listen)
exten => _568600XXX,1,Set(MEETME_EXIT_CONTEXT=up_monitor_exit)
exten => _568600XXX,n,Meetme(${EXTEN:2},mqX)
exten => _568600XXX,n,Hangup()

; barge, exit-on-dtmf conferences for VICIDIAL (barge)
exten => _578600XXX,1,Set(MEETME_EXIT_CONTEXT=up_monitor_exit)
exten => _578600XXX,n,Meetme(${EXTEN:2},X)
exten => _578600XXX,n,Hangup()

[up_monitor]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => s,1,Set(MEETME_EXIT_CONTEXT=up_monitor_exit)
exten => s,n,AGI(agi-phone_monitor.agi,USER---N---LISTEN)
exten => s,n,Hangup()

exten => i,1,Goto(up_monitor_exit,s,1)
exten => i,n,Hangup()
exten => #,1,Goto(up_monitor_exit,s,1)
exten => #,n,Hangup()
exten => t,1,Goto(up_monitor_exit,s,1)
exten => t,n,Hangup()

[up_monitor_exit]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _X,1,Goto(up_monitor,s,1)
exten => _X,n,Hangup()

exten => i,1,Goto(up_monitor,s,1)
exten => i,n,Hangup()
exten => #,1,Goto(up_monitor,s,1)
exten => #,n,Hangup()
exten => t,1,Goto(up_monitor,s,1)
exten => t,n,Hangup()

[rrb555]

Code: Select all

For this to work you need to add these lines to your extensions.conf:
(which can be added to the "System Settings -> Custom Dialplan" instead)

; quiet entry, listen-only, exit-on-dtmf conferences for VICIDIAL (listen)
exten => _568600XXX,1,Set(MEETME_EXIT_CONTEXT=up_monitor_exit)
exten => _568600XXX,n,Meetme(${EXTEN:2},mqX)
exten => _568600XXX,n,Hangup()

; barge, exit-on-dtmf conferences for VICIDIAL (barge)
exten => _578600XXX,1,Set(MEETME_EXIT_CONTEXT=up_monitor_exit)
exten => _578600XXX,n,Meetme(${EXTEN:2},X)
exten => _578600XXX,n,Hangup()

[up_monitor]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => s,1,Set(MEETME_EXIT_CONTEXT=up_monitor_exit)
exten => s,n,AGI(agi-phone_monitor.agi,USER---N---LISTEN)
exten => s,n,Hangup()

exten => i,1,Goto(up_monitor_exit,s,1)
exten => i,n,Hangup()
exten => #,1,Goto(up_monitor_exit,s,1)
exten => #,n,Hangup()
exten => t,1,Goto(up_monitor_exit,s,1)
exten => t,n,Hangup()

[up_monitor_exit]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _X,1,Goto(up_monitor,s,1)
exten => _X,n,Hangup()

exten => i,1,Goto(up_monitor,s,1)
exten => i,n,Hangup()
exten => #,1,Goto(up_monitor,s,1)
exten => #,n,Hangup()
exten => t,1,Goto(up_monitor,s,1)
exten => t,n,Hangup()

I've added this lines on the Custom Dial plan under system settings. But i am not sure how to operate this. I tried calling the phone extension (102) but the phone extension is just ringing.

Any help

Thanks

[mflorell]

If your version information in your signature is correct, you would need to upgrade to a newer svn/trunk version before this would work for you.

[rrb555]

upgrade to
SVN Version:1985
VERSION: 2.8-403a
BUILD: 130510-1350

I was still unable to monitor. When I dial the phone extension the softphone of the agent rings

[mflorell]

How are you getting to the up_monitor context?

[ZoVoS]

go to the asterisk CLI
type;

"iax2 show peers"

Name/Username Host Mask Port Status
ASTblind (null) (D) 255.255.255.255 0 UNKNOWN
ASTloop (null) (D) 255.255.255.255 0 UNKNOWN


"iax2 show registry"

Host dnsmgr Username Perceived Refresh State
127.0.0.1:41569 N ASTblind <Unregistered> 60 Request
127.0.0.1:40569 N ASTloop <Unregistered> 60 Request

If both of your outputs are the same as above there is a temperary solution to use untill i find the route cause of this issue

nano /etc/asterisk/iax.conf

change
bindport=4569
to
bindport=41569

DISCLAIMER - THIS WILL NOT FIX THE ROOT CAUSE OF THE PROBLEM, I'm working on that now, what it will do is let the ASTblind bind to a port so that it becomes available for use in agent monitoring IF you have the same issues I am having. I have been mulling over many different possibilities such as the new res_timers etc, but i'm still not 100% sure the root cause.

I have modified the source code for chan_iax2.so so a reload will rebind ports for testing and that's speeding everything up. Hopefully i'll have a answer as to why this is happening soon enough. Hope this helps

EDIT
after the change type "core restart now" in the asterisk CLI

this should be the new output of the previouse commands

Code: Select all

X*CLI> iax2 show peers
Name/Username    Host                 Mask             Port          Status
ASTblind         127.0.0.1       (D)  255.255.255.255  41569         OK (3 ms)
ASTloop          (null)          (D)  255.255.255.255  0             UNKNOWN
2 iax2 peers [1 online, 1 offline, 0 unmonitored]


Code: Select all

X*CLI> iax2 show registry
Host                  dnsmgr  Username    Perceived             Refresh  State
127.0.0.1:41569       N       ASTblind    127.0.0.1:41569            60  Registered
127.0.0.1:40569       N       ASTloop     <Unregistered>             60  Request Sent
2 IAX2 registrations.