No Recording. :(

[Natasha]

Hi, installed ViciBox 5.0.3 Preload ISO. Campaign Recording set to ALLFORCE but Vicidial is not recording anything.

[tick]

hi,
did you check /var/spool/asterisk/monitorDONE/
And please reboot your server..1'st

[Natasha]

All folders under monotorDONE are empty, and the server has been rebooted several times ... no luck

[boybawang]

take a look at the command line when your trying to do a test call, it helps

[Natasha]

Here is the CLI, but I don't think I understand all these Asterisk language.

Code: Select all

[Oct  1 13:24:39]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  1 13:24:39]     -- Executing [8368@default:1] Playback("Local/9414137839565@default-0000020c;1", "sip-silence") in new stack
[Oct  1 13:24:39]     -- <Local/9414137839565@default-0000020c;1> Playing 'sip-silence.gsm' (language 'en')
[Oct  1 13:24:39]     -- Executing [h@default:1] AGI("Local/9414137839565@default-0000020c;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----0") in new stack
[Oct  1 13:24:39]     -- Executing [8368@default:2] AGI("SIP/kashcarrier-00000205", "agi://127.0.0.1:4577/call_log") in new stack
[Oct  1 13:24:39]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=KASH13))
[Oct  1 13:24:39]     -- <SIP/kashcarrier-00000205>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct  1 13:24:39]     -- Executing [8368@default:3] AGI("SIP/kashcarrier-00000205", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Oct  1 13:24:39]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct  1 13:24:39]     -- SIP/kashcarrier-0000020b is ringing
[Oct  1 13:24:39]     -- SIP/kashcarrier-0000020b is making progress passing it to Local/9419783199604@default-00000212;2
[Oct  1 13:24:39]     -- <SIP/kashcarrier-00000205>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct  1 13:24:39]     -- Executing [192*168*000*250*8600052@default:1] Goto("SIP/kashcarrier-00000205", "default,8600052,1") in new stack
[Oct  1 13:24:39]     -- Goto (default,8600052,1)
[Oct  1 13:24:39]     -- Executing [8600052@default:1] MeetMe("SIP/kashcarrier-00000205", "8600052,F") in new stack
[Oct  1 13:24:39]     -- <Local/9415087560177@default-00000217;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------- completed, returning 0
[Oct  1 13:24:40]     -- <Local/9414137839565@default-0000020c;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----26-----0 completed, returning 0
[Oct  1 13:24:40]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  1 13:24:40]   == Spawn extension (default, 8600056, 1) exited non-zero on 'SIP/kashcarrier-00000208'
[Oct  1 13:24:40]     -- Executing [h@default:1] AGI("SIP/kashcarrier-00000208", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct  1 13:24:40]     -- <SIP/kashcarrier-00000208>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0

[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 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
__________________________________
3) Please check your settings in crontab -e for recording mix/move/ftp
4) You posted CLI details that show more than one call but only last for 1 second. Obviously there would not have been any recording if there was no call (congestion?). So try posting output when there are NO other calls except the one test call you want to see "record". Make the number that is dialed your cell phone and no other calls active. Keep the call "live" for at least 30 seconds and Say Something that could be recorded. LOL (and be sure it is an outbound autodialed call by the campaign so it can be recorded by the campaign settings).
5) During operation, check the /var/spool/asterisk/monitor folder to see if there are any recordings that have not yet been moved to the hard drive. (This is usually a virtual drive which would be lost at reboot if the files are not copied via the crontab -e scripts).
6) Happy Hunting!

[Natasha]

1. Thank you williamconley
2. Vicibox 5.0.3 from ViciBox5.i686-5.0.3.preload.iso | Vicidial 2.8-412a Build 130926-1731 | CONFUSING!!! Asterisk 1.8.23.0-vici (from asterisk CLI), 1.4.21.2 (as it reads in Servers>Asterisk Version in Admin panel) | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel Core2Quad ( We have been using this server as GoAutoDial dialer for last 4 months)
3. crontab -e
It won't let me write my crontab (Your post looks too spamy for a new user, please remove off-site URLs. )

4. Tough one! (Single outbound autodialed call by the campaign so it can be recorded by the campaign settings)

Code: Select all

[Oct  2 09:49:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  2 09:49:08]     -- Executing [9912127773456@default:1] AGI("Local/9912127773456@default-00000004;2", "agi://127.0.0.1:4577/call_log") in new stack
[Oct  2 09:49:08]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=SONIKA12))
[Oct  2 09:49:08]     -- <Local/9912127773456@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct  2 09:49:08]     -- Executing [9912127773456@default:2] Dial("Local/9912127773456@default-00000004;2", "SIP/rajucarrier/12127773456,,tTor") in new stack
[Oct  2 09:49:08]   == Using SIP RTP CoS mark 5
[Oct  2 09:49:08]     -- Called SIP/rajucarrier/12127773456
[Oct  2 09:49:11]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  2 09:49:12]     -- SIP/rajucarrier-00000007 is making progress passing it to Local/9912127773456@default-00000004;2
[Oct  2 09:49:12]     -- SIP/rajucarrier-00000007 answered Local/9912127773456@default-00000004;2
[Oct  2 09:49:12]        > Channel Local/9912127773456@default-00000004;1 was answered.
[Oct  2 09:49:12]     -- Executing [8368@default:1] Playback("Local/9912127773456@default-00000004;1", "sip-silence") in new stack
[Oct  2 09:49:13]     -- <Local/9912127773456@default-00000004;1> Playing 'sip-silence.gsm' (language 'en')
[Oct  2 09:49:13]     -- Executing [h@default:1] AGI("Local/9912127773456@default-00000004;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----1") in new stack
[Oct  2 09:49:13]     -- Executing [8368@default:2] AGI("SIP/rajucarrier-00000007", "agi://127.0.0.1:4577/call_log") in new stack
[Oct  2 09:49:13]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=SONIKA12))
[Oct  2 09:49:13]     -- <SIP/rajucarrier-00000007>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct  2 09:49:13]     -- Executing [8368@default:3] AGI("SIP/rajucarrier-00000007", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Oct  2 09:49:13]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct  2 09:49:13]     -- <SIP/rajucarrier-00000007>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct  2 09:49:13]     -- Executing [192*168*000*250*8600051@default:1] Goto("SIP/rajucarrier-00000007", "default,8600051,1") in new stack
[Oct  2 09:49:13]     -- Goto (default,8600051,1)
[Oct  2 09:49:13]     -- Executing [8600051@default:1] MeetMe("SIP/rajucarrier-00000007", "8600051,F") in new stack
[Oct  2 09:49:13]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  2 09:49:14]     -- <Local/9912127773456@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----1 completed, returning 0
[Oct  2 09:49:14]   == Spawn extension (default, 9912127773456, 2) exited non-zero on 'Local/9912127773456@default-00000004;2'
[Oct  2 09:50:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  2 09:50:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  2 09:50:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  2 09:50:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  2 09:50:06]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/rajucarrier-00000007'
[Oct  2 09:50:06]     -- Executing [h@default:1] AGI("SIP/rajucarrier-00000007", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct  2 09:50:06]     -- <SIP/rajucarrier-00000007>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Oct  2 09:50:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  2 09:50:06]   == Manager 'sendcron' logged on from 127.0.0.1


5. /var/spool/asterisk/monitor remains empty during calling. How did I check it? 'ls' command @ 3/sec. for 5 minutes during 4 agents in calls.

6. I was reading the follwing thread
ViciBox v.5.0 Bug Fixes - Updated Aug 21 2013 (It won't let me write the URL as a new user)
Kumba wrote,

The bugfix for the recordings was already addressed in the first post. It's a permission issue not a user/group issue.

Do I need to run patches mentioned in that thread? Did I miss it?

Finally my screenlist if that helps

Code: Select all

screen -list
There are screens on:
   7910.ASTemail   (Detached)
   24367.ASTVDadapt   (Detached)
   13108.ASTVDremote   (Detached)
   2578.ASTVDautoFILL   (Detached)
   2575.ASTfastlog   (Detached)
   2566.ASTVDauto   (Detached)
   2563.ASTlisten   (Detached)
   2560.ASTsend   (Detached)
   2557.ASTupdate   (Detached)
   2425.asterisk   (Detached)
   2420.astshell20130930221923   (Detached)
11 Sockets in /var/run/screens/S-root.

[Natasha]

Well, here is my crontab -e . I had to replace the 'c' in the word compress to 'k' ( marked in red) to make is valid for posting.

### keepalive script for astguiclient processes
* * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl

### remove old vicidial logs and asterisk logs more than 2 days old
28 0 * * * /usr/bin/find /var/log/astguiclient -maxdepth 1 -type f -mtime +2 -print | xargs rm -f
29 0 * * * /usr/bin/find /var/log/asterisk -maxdepth 3 -type f -mtime +2 -print | xargs rm -f
30 0 * * * /usr/bin/find / -maxdepth 1 -name "screenlog.0*" -mtime +4 -print | xargs rm -f

### fix the vicidial_agent_log once every hour and the full day run at night
33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl
50 0 * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --last-24hours
## uncomment below if using QueueMetrics
#*/5 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl --only-qm-live-call-check

### updater for VICIDIAL hopper
* * * * * /usr/share/astguiclient/AST_VDhopper.pl -q

### adjust the GMT offset for the leads in the vicidial_list table
1 1,7 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug --list-settings

### optimize the database tables within the asterisk database
3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl

### VICIDIAL agent time log weekly and daily summary report generation
2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl
22 0 * * * /usr/share/astguiclient/AST_agent_day.pl

### VICIDIAL campaign export scripts (OPTIONAL)
#32 0 * * * /usr/share/astguiclient/AST_VDsales_export.pl
#42 0 * * * /usr/share/astguiclient/AST_sourceID_summary_export.pl

### inventory report optional
#1 7 * * * /usr/share/astguiclient/AST_dialer_inventory_snapshot.pl -q --override-24hours

### roll call_log and vicidial_log_extended daily on very high-volume dialing systems
#20 1 * * * /usr/share/astguiclient/ADMIN_archive_log_tables.pl --daily

## uncomment below if using Vtiger
#1 1 * * * /usr/share/astguiclient/Vtiger_optimize_all_tables.pl --quiet

# cleanup of the scheduled callback records
25 0 * * * /usr/share/astguiclient/AST_DB_dead_cb_purge.pl --purge-non-cb -q

### inbound email parser should only be active on a single server
* * * * * /usr/share/astguiclient/AST_inbound_email_parser.pl

### recording mixing/compressing/ftping scripts
#0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl --MIX
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl
1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_kompress.pl --MP3
#2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --MP3
#0 1 * * * /usr/share/astguiclient/AST_CRON_audio_4_ftp2.pl --ftp-server=server.ip --ftp-login=user --ftp-pass=pass --ftp-directory=/ --ftp-persistent --ftp-validate --transfer-limit=100000 --list-limit=100000

### kill Hangup script for Asterisk updaters
* * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl

### updater for voicemail
* * * * * /usr/share/astguiclient/AST_vm_update.pl

### updater for conference validator
* * * * * /usr/share/astguiclient/AST_conf_update.pl

### flush queue DB table every hour for entries older than 1 hour
11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q

### reset several temporary-info tables in the database
2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl

### remove old recordings more than 7 days old, and delete originals after 1 day
#24 0 * * * /usr/bin/find /var/spool/asterisk/monitorDONE -maxdepth 2 -type f -mtime +7 -print | xargs rm -f
24 1 * * * /usr/bin/find /var/spool/asterisk/monitorDONE/ORIG -maxdepth 2 -type f -mtime +1 -print | xargs rm -f

### Reboot nightly to manage asterisk issues and memory leaks - uncomment if issues arise
#30 6 * * * /sbin/reboot

### remove text to speech file more than 4 days old
#20 0 * * * /usr/bin/find /var/lib/asterisk/sounds/tts/ -maxdepth 2 -type f -mtime +4 -print | xargs rm -f

[DomeDan]

Good work posting your specs!

Start with to change the "Servers>Asterisk Version" to 1.8.23.0-vici
that might solve the problem, or williamconley might find something

[Natasha]

Thank you DomeDan. Changes made, rebooted but it did not help.

[DomeDan]

The patch you talked about is not really a patch, just a command to change permissions on a folder.
try that too, just run the command:
chmod 755 /var/spool/asterisk

[williamconley]

I don't see a call recording start in the CLI output. Perhaps we need to have a look at the campaign settings in detail.

[Natasha]

Thank you DomeDan it did not work.

Thank you willaimconley.

campaign settings in detail:

Active: Y
Admin User Group: ALL
Park Music-on-Hold: BLANK
Web Form: BLANK
Web Form Two: BLANK
Web Form Target: vdcwebform
Allow Closers: Y
Allow Inbound and Blended: N
Dial Status 1: B - Busy
Dial Status 2: DROP - Agent Not Available
Dial Status 3: NEW - New Lead
Add A Dial Status to Call: N
List Order: Down
List Order Randomize: N
List Order Secondary: LEAD_ASCEND
List Mix: DISABLED
Lead Filter: NONE
Drop Lockout Time: 0
Call Count Limit: 0
Call Count Target: 3
Minimum Hopper Level: 200
Automatic Hopper Level: Y
Automatic Hopper Multiplier: 1
Auto Trim Hopper: Y
Hopper VLC Dup Check: N
Force Reset of Hopper: N

Dial Method: RATIO
Auto Dial Level: 1.5 ADAPT OVERRIDE: not checked
Auto Dial Level Threshold: DISABLED
Available Only Tally: N
Available Only Tally Threshold: DISABLED
Drop Percentage Limit: 3%
Maximum Adapt Dial Level: 3
Latest Server Time: 2100
Adapt Intensity Modifier: 0 Balanced
Dial Level Difference Target: 0 Balanced
Dial Level Difference Target Method: ADAPT_CALC_ONLY
Concurrent Transfers: AUTO
Queue Priority: 50 Higher
Multiple Campaign Drop Rate Group: DISABLED
Inbound Queue No Dial: DISABLED

Auto Alt-Number Dialing: NONE
Next Agent Call: oldest_call_finish
Local Call Time: 24 hours
Dial Timeout: 60
Dial Prefix: 9
Manual Dial Prefix: BLANK
Omit Phone Code: N
Campaign CallerID: 0000000000

Custom CallerID: N
Routing Extension: 8368
Campaign Rec exten: 8309
Campaign Recording: ALLFORCE
Campaign Rec Filename: FULLDATE_CUSTPHONE
Recording Delay: 0
Call Notes Per Call: DISABLED
Agent Lead Search: DISABLED
Agent Lead Search Method: COMPLIST_ALL
Script: NONE
Get Call Launch: NONE

Answering Machine Message: vm-goodbye
WaitForSilence Options:
AMD send to Action: N
CPD AMD Action: DISABLED
AMD Inbound Group: DISABLED
AMD Call Menu: NONE

Transfer-Conf DTMF 1: BLANK
Transfer-Conf Number 1: BLANK
Transfer-Conf DTMF 2: BLANK
Transfer-Conf Number 2: BLANK
Transfer-Conf Number 3: BLANK
Transfer-Conf Number 4: BLANK
Transfer-Conf Number 5: BLANK
Enable Transfer Presets: DISABLED
Hide Transfer Number to Dial: DISABLED
Quick Transfer Button: N
Custom 3-Way Button Transfer: DISABLED
PrePopulate Transfer Preset: N
Park Call IVR: DISABLED
Park Call IVR AGI: BLANK

Timer Action: NONE
Timer Action Message: BLANK
Timer Action Seconds: 1
Timer Action Destination: BLANK

Alt Number Dialing: N
Scheduled Callbacks: N
Scheduled Callbacks Alert: NONE
Scheduled Callbacks Count: ALL ACTIVE
Scheduled Callbacks Days Limit: 0
Scheduled Callbacks Hours Block: 0
Scheduled Callbacks Calltime Block: DISABLED
My Callbacks Checkbox Default: UNCHECKED

Drop Call Seconds: 5
Drop Action: AUDIO
Safe Harbor Exten: 8307
Safe Harbor Audio: buzz
Safe Harbor Audio Field: DISABLED
Safe Harbor Call Menu: BLANK
Voicemail: BLANK
Drop Transfer Group: NONE

Disable Dispo Screen: DISPO_ENABLED
Disable Dispo Status: BLANK
Wrap Up Seconds: 0
Wrap Up Message: Wrapup Call

Use Internal DNC List: N
Use Campaign DNC List: Y
Other Campaign DNC: BLANK

Agent Pause Codes Active: N
Auto Pause Pre-Call Work: N
Auto Resume Pre-Call Work: N
Auto Pause Pre-Call Code: PRECAL
Campaign Stats Refresh: N
Real-Time Agent Time Stats: CALLS_WAIT_CUST_ACW_PAUSE

Disable Alter Customer Data: N
Disable Alter Customer Phone: Y

Allow No-Hopper-Leads Logins: N
No Hopper Dialing: N
Owner Only Dialing: NONE
Owner Populate: DISABLED

Agent Display Dialable Leads: N
Agent Screen Labels: SYSTEM-SETTINGS
Status Display Fields: CALLID
Agent Display Queue Count: Y
Agent View Calls in Queue: NONE
View Calls in Queue Launch: MANUAL
Agent Grab Calls in Queue: NONE

Agent Call Re-Queue Button: N
Agent Pause After Each Call: N
Agent Pause After Next Call Link: DISABLED

Manual Dial Override: NONE
Manual Dial List ID: 998
Manual Dial Filter: NONE
Manual Preview Dial: PREVIEW_AND_SKIP
Manual Call Time Check: DISABLED
Manual Dial API: STANDARD
Manual Dial CID: CAMPAIGN
Phone Post Time Difference Alert: DISABLED
In-Group Manual Dial: DISABLED
In-Group Manual Dial Select: CAMPAIGN_SELECTED

Agent Screen Clipboard Copy: NONE
Agent Screen Extended Alt Dial: N

3-Way Call Outbound CallerID: CAMPAIGN
3-Way Call Dial Prefix: BLANK
Customer 3-Way Hangup Logging: ENABLED
Customer 3-Way Hangup Seconds: 5
Customer 3-Way Hangup Action: NONE

Group Alias Allowed: N
CRM Popup Login: N
CRM Popup Address: BLANK
Start Call URL: BLANK
Dispo Call URL: BLANK
No Agent Call URL: BLANK
Extension Append CID: N

Blind Monitor Warning: DISABLED
Blind Monitor Notice: Someone is blind monitoring your session
Blind Monitor Filename: BLANK

Default Transfer Group: NONE
Allowed Transfer Groups: AGENTDIRECT - Single Agent Direct Queue (UNCHECKED)

[williamconley]

Asterisk 1.8.23.0-vici (from asterisk CLI), 1.4.21.2 (as it reads in Servers>Asterisk

Change the asterisk version in Vicidial (Admin->Servers->Asterisk version) to 1.8

Be sure the campaign you posted is the campaign you're logged in to and that a conversation actually took place. LOL

[Natasha]

Well, I changed Admin->Servers->Asterisk version to 1.8 . Earlier it was changed to 1.8.23.0-vici as recomended by DomeDan. OF COURSE it's the same campaign. !!!

[williamconley]

OF COURSE it's the same campaign. !!!

I had to ask. You'd be amazed at the "obvious" things that are not obvious to new users.

I did not see anything else obvious in the asterisk cli.

The next thing i'd check would be agi logging or debugging. In Admin->Servers you can turn on agi logging to the console (available via "screen -r asterisk") or to the log files (stored in /var/log/astguiclient) and you would access agi debugging via the asterisk console command to activate agi debugging (agi set debug or agi set debug on depending on version if I recall correctly).

Then you'll see the logic that kicks in when 8368 activates the outbound agi script which is expected to initiate recording. I believe it should activate immediately upon placing the call in the meetme room. You may want to "read along" in the outbound agi script and see if there is a problem with logic during execution (in /var/lib/asterisk/agi-bin). During which you may find an option turned off that I've overlooked or a bug/flaw in the system based on installation.

[Natasha]

Thank you William ... I suppose that's your name. I will try to understand that all you said and then I will give it a try. Do you think a reinstallation will be easier? may be to an earlier version? I don't see much post about 5.0.3 here.

[williamconley]

5.0.3 is the Vicibox version. That's the "installer" and would only be mentioned in the Viibox Installer Board.

Here in the Support Board we discuss the Vicidial Version (including Build) which is presently in version 2.8. As far as I know, recording has not been broken.

It could easily be the 1.8 version in Servers or perhaps an improper configuration file or customization or carrier dialplan ... many things can impact this process in unexpected ways.

It may be a good idea to post your carrier dialplan just to be safe, although ... that shouldn't be it if everything else is working. But a non-standard dialplan can have unexpected side-effects.

[estoque]

I can't see the recording extension invited in the meetme conference so it wont be recorded. By default it is 8309, have you tried searching extensions.conf or extensions-vicidial.conf if it exists. It's something like this:
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERID(name)})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup

[mav2287]

I did alot of fighting with the preload version last weekend and was having some wierd problems as well. For example I had the same issue with the asterisk version being wrong in the server admin page and still now my recording links give me a Forbidden error. Did you run the "Zypper Up" command before you did the vicibox express setup? I found out the hard way that if you did the preload and didn't do the "Zypper Up" Command before you do the vici express setup you get weird results and gremlins to chase. You didn't happen to run it after the fact or not at all did you?