vicidial.org

Hello i seeking for help because i am stuck with a installation having several random problems. ( I will open separate threads for each)

INSTALLATION METHOD - Vicibox 6.0 - Single Server Install - VICIDIAL VERSION: 2.10-448a BUILD: 140706-0927. Asterisk: 1.8.28.2-vici
HARDWARE: Not Sure Its a HP Server, cpuinfo show 8 x CPU E3-1240, 8GB ECC RAM and RAID 5 with 3 SCSI disks over a HP controller

As stated in the title, Vicidial is not recording.

Campaign Set to ALL_FORCE, User set to ALL_FORCE, Recording Extension 8309 (already tried with 8310)

If i set campaign and User to ONDEMAND, the button is always disabled.

Same happened with ViCIBOX 5.0.3 using VICIDIAL 2.8. i upgraded to VICIBOX 6.0 and just migrate the Database via ADMIN_backup.pl --db-only, still the same behavior

My asterisk logs, never shows the attempt to record the calls. BTW My asterisk version match with the one under server in vicidial Admin

I though there was some option trough system setting or something, to disable call recording, but already update my Managers Manual and i couldn’t find any reference.

Is there any options to disable recordings system wide or do you have any suggestions on why my system is not recording?

When you upgrade, you do not just "migrate the DB". There is an UPGRADE document. Read it!

You'll need to upgrade the DB to match the code in the system. Since the Vicidial system is fresh ... but the DB was old ... the DB must now be upgraded to match the installed code. Instructions are in /usr/src/astguiclient/trunk (most likely, if you used a Vicibox install .iso).

Both system (the old vicibox and the new) was on the same SVN revision and using the same DB Schema.
And the problem exists before the upgrade, what kind of data may i provide for this scenario in order to find a solution for this?

Does the asterisk even attempt to start recording? When you go into the asterisk console and set a verbose level over 5..., do you see something like this:

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Monitor("Local/58600053@default-000007e8;1", "wav,CAMPAIGN_20140712-130423_89516754_10357")

GaD wrote:Does the asterisk even attempt to start recording? When you go into the asterisk console and set a verbose level over 5..., do you see something like this:

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Monitor("Local/58600053@default-000007e8;1", "wav,CAMPAIGN_20140712-130423_89516754_10357")

excellent troubleshooting. ie: are you sure it's not recording, or is it just that the recordings are not "visible" to you anywhere ... in which case where have you looked?

check the /var/spool/asterisk/monitor and monitorDONE folders and (if they are there) run the recording management scripts manually with --debugX (assuming you find that it is actually recording.) if you do NOT find that it is recording ... try pushing RECORD during a session and see if that at least works.

GaD wrote:Does the asterisk even attempt to start recording? When you go into the asterisk console and set a verbose level over 5..., do you see something like this:

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Monitor("Local/58600053@default-000007e8;1", "wav,CAMPAIGN_20140712-130423_89516754_10357")

Tnx for responde, and sorry for the delay.

No, it is not even trying to record. In the asterisk console i cant never see the Monitor command. That is why i though that the recordings are disabled.

can you check whether the recording dialplan exists in ur server
type the below command in asterisk cli.

dialplan show 8309@default

do you have sox installed?

striker wrote:can you check whether the recording dialplan exists in ur server
type the below command in asterisk cli.

dialplan show 8309@default

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cc-ippbx*CLI> dialplan show 8309@default
[ Included context 'vicidial-auto-internal' created by 'pbx_config' ]
  '8309' =>         1. Answer()                                   [pbx_config]
                        2. Monitor(wav,${CALLERID(name)})             [pbx_config]
                        3. Wait(3600)                                 [pbx_config]
                        4. Hangup()                                   [pbx_config]

boybawang wrote:do you have sox installed?

Yes, i have sox installed, but anyway, the calls are not even trying to being recorded

make a test call and post the full cli log here

Do this on your DB server

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SELECT campaign_id,campaign_rec_exten,campaign_recording,campaign_rec_filename  FROM vicidial_campaigns;

Just to check your recording settings

Also run this on your asterisk CLI

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dialplan show 8309@default

And also

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dialplan show 8310@default

Also enter this on your linux CLI, your bash/sh session, the regular command prompt

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ls -la /var/spool/asterisk; ls -la /var/spool/asterisk/monitorDONE/

Post your results to all of these outcomes.

Awesome troubleshooting here the results of the commands.

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MariaDB [asterisk]>     SELECT campaign_id,campaign_rec_exten,campaign_recording,campaign_rec_filename  FROM vicidial_campaigns;
+-------------+--------------------+--------------------+-----------------------+
| campaign_id | campaign_rec_exten | campaign_recording | campaign_rec_filename |
+-------------+--------------------+--------------------+-----------------------+
| GPOARIES    | 8309               | ALLFORCE           | FULLDATE_CUSTPHONE    |
| MDIRECTA    | 8309               | ALLFORCE           | FULLDATE_CUSTPHONE    |
| INFOLA      | 8309               | ALLFORCE           | FULLDATE_CUSTPHONE    |
| INFOMXLI    | 8309               | ALLFORCE           | FULLDATE_CUSTPHONE    |
| C4TEST      | 8310               | ALLFORCE           | FULLDATE_CUSTPHONE    |
| CAPACITA    | 8309               | ALLFORCE           | FULLDATE_CUSTPHONE    |
| NUEVA       | 8309               | ONDEMAND           | FULLDATE_CUSTPHONE   

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cc-ippbx:~ # asterisk -rx "dialplan show 8309@default"
[ Included context 'vicidial-auto-internal' created by 'pbx_config' ]
  '8309' =>         1. Answer()                                   [pbx_config]
                    2. Monitor(wav,${CALLERID(name)})             [pbx_config]
                    3. Wait(3600)                                 [pbx_config]
                    4. Hangup()                                   [pbx_config]

-= 1 extension (4 priorities) in 1 context. =-

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cc-ippbx:~ # asterisk -rx "dialplan show 8310@default"
[ Included context 'vicidial-auto-internal' created by 'pbx_config' ]
  '8310' =>         1. Answer()                                   [pbx_config]
                    2. Monitor(gsm,${CALLERID(name)})             [pbx_config]
                    3. Wait(3600)                                 [pbx_config]
                    4. Hangup()                                   [pbx_config]

-= 1 extension (4 priorities) in 1 context. =-

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cc-ippbx:~ #     ls -la /var/spool/asterisk; ls -la /var/spool/asterisk/monitorDONE/
total 40
drwxr-x--- 10 asterisk asterisk 4096 Jul  8 01:45 .
drwxr-xr-x 12 root     root     4096 Jun 19 08:55 ..
drwxr-x---  2 asterisk asterisk 4096 Jun 25 14:18 dictate
drwxr-x---  2 asterisk asterisk 4096 Jun 25 14:18 meetme
drwxr-x---  3 asterisk asterisk 4096 Jul 20 11:24 monitor
drwxr-xr-x  9 wwwrun   root     4096 Jul 21 11:46 monitorDONE
drwxr-xr-x  2 root     root     4096 Jul  8 01:45 outgoing
drwxr-x---  2 asterisk asterisk 4096 Jun 25 14:18 system
drwxr-x---  2 asterisk asterisk 4096 Jun 25 14:18 tmp
drwxr-x---  3 asterisk asterisk 4096 Jun 25 14:18 voicemail
total 164
drwxr-xr-x  9 wwwrun   root      4096 Jul 21 11:46 .
drwxr-x--- 10 asterisk asterisk  4096 Jul  8 01:45 ..
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 FTP
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 FTP2
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 GSM
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 GSW
drwxr-xr-x  2 root     root     53248 Jul 21 11:46 MP3
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 OGG
drwxr-xr-x  2 root     root     77824 Jul 21 11:46 ORIG

Notice that you have

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drwxr-xr-x  2 root     root      4096 Jul  8 01:31 FTP
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 FTP2
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 GSM
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 GSW
drwxr-xr-x  2 root     root     53248 Jul 21 11:46 MP3
drwxr-xr-x  2 root     root      4096 Jul  8 01:31 OGG
drwxr-xr-x  2 root     root     77824 Jul 21 11:46 ORIG

Your asterisk cannot write to ths directory. Do the following:

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chown -R asterisk /var/spool/asterisk/

Wait for a couple of minutes to have some calls to get recorded and then enter this in your CLI (Bash, not asterisk)

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ls /var/spool/asterisk/monitorDONE/ORIG

Lets see if this makes any difference.

O.K. i guess i give up.

I made a fresh install of Vicibox 6.0, configure the basic and recordings was working like a charm.

after that i restore my database and recordings stop working, please note that even ONDEMAND is not letting me to even press the button.

So my database is broken..... this is a relative new server (4 months old) but it will hurt to lose all the data with a fresh database but i think it is my only way out of this.

Thank all of you guys

I am experiencing the same issue however I am not giving up since I made reinstallation several times following http://ray-solomon.com/vicidial.html

I am getting below:

[ Included context 'vicidial-auto-phones' created by 'pbx_config' ]
'1111' => 1. Dial(SIP/1111,60,) [pbx_config]
2. Goto(default,850266666666661111,1) [pbx_config]
3. Hangup() [pbx_config]

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s1*CLI> dialplan show 1111@default
[ Included context 'vicidial-auto-phones' created by 'pbx_config' ]
  '1111' =>         1. Dial(SIP/1111,60,)                         [pbx_config]
                    2. Goto(default,850266666666661111,1)         [pbx_config]
                    3. Hangup()                                   [pbx_config]



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[root@s1 log]# ls -las -d asterisk/
4 drwxr-xr-x 5 asterisk asterisk 4096 Feb 22 22:39 asterisk/


I believe the problem is with a dialplan to record all calls. Can anyone help me with this?

Here my config:

VERSION: 2.12-538a
BUILD: 160122-1401

Version: 2.12b0.5
SVN Version: 2480
DB Schema Version: 1452
DB Schema Update Date: 2016-02-22 08:32:37

extensions.conf

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[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             
;TRUNK=DAHDI/r1                                   
;TRUNKX=DAHDI/r2               
;TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569   
;TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569   
;TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com   
;SIPtrunk=SIP/1234:PASSWORD@sip.provider.net   

[trunkinbound]

exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


#include extensions-vicidial.conf

[default]
include => vicidial-auto
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _999XXXXXX1,1,Answer()
exten => _999XXXXXX1,n,Wait(2)
exten => _999XXXXXX1,n,Playback(vicidial-welcome)
exten => _999XXXXXX1,n,Hangup()

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,n,Answer()
exten => _999XX11112,n,Playback(ss-noservice)
exten => _999XX11112,n,Playback(vm-goodbye)
exten => _999XX11112,n,Hangup()

exten => _999XX18112,1,Wait(2)
exten => _999XX18112,n,Answer()
exten => _999XX18112,n,Playback(vtiger-fax)
exten => _999XX18112,n,Playback(vtiger-fax)
exten => _999XX18112,n,Hangup()

exten => _999XX19112,1,Wait(2)
exten => _999XX19112,n,Answer()
exten => _999XX19112,n,Playback(vtiger-email)
exten => _999XX19112,n,Playback(vtiger-email)
exten => _999XX19112,n,Hangup()

exten => _999XXXX112,1,Wait(5)
exten => _999XXXX112,n,Answer()
exten => _999XXXX112,n,Playback(demo-instruct)
exten => _999XXXX112,n,Playback(demo-instruct)
exten => _999XXXX112,n,Hangup()

exten => _999XXXXXX2,1,Wait(8)
exten => _999XXXXXX2,n,Answer()
exten => _999XXXXXX2,n,Playback(demo-instruct)
exten => _999XXXXXX2,n,Hangup()

exten => _999XXXXXX3,1,Set(PRI_CAUSE=1)
exten => _999XXXXXX3,n,Hangup()

exten => _999XXXXXX4,1,Set(PRI_CAUSE=27)
exten => _999XXXXXX4,n,Hangup()

exten => _999XXXXXX5,1,Ringing
exten => _999XXXXXX5,n,Wait(120)
exten => _999XXXXXX5,n,Hangup()

exten => _999XXXXXX6,1,Wait(10)
exten => _999XXXXXX6,n,Answer()
exten => _999XXXXXX6,n,Playback(demo-instruct)
exten => _999XXXXXX6,n,Hangup()

exten => _999XXXXXX7,1,Wait(12)
exten => _999XXXXXX7,n,Answer()
exten => _999XXXXXX7,n,Playback(demo-enterkeywords)
exten => _999XXXXXX7,n,Hangup()

exten => _999XXXXXX8,1,Set(PRI_CAUSE=17)
exten => _999XXXXXX8,n,Hangup()

exten => _999XXXXXX9,1,Wait(6)
exten => _999XXXXXX9,n,Answer()
exten => _999XXXXXX9,n,Playback(demo-abouttotry)
exten => _999XXXXXX9,n,Hangup()

exten => _999XXXXXX0,1,Wait(5)
exten => _999XXXXXX0,n,Answer()
exten => _999XXXXXX0,n,Playback(vm-goodbye)
exten => _999XXXXXX0,n,Hangup()

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

exten => s,1,Set(TIMEOUT(digit)=10)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Set(MEETME_EXIT_CONTEXT=monitor_exit)
exten => s,n,Background(vm-extension)
exten => s,n,WaitExten(10)
exten => s,n,Hangup()

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

exten => _8[0-2]XX,1,Meetme(8600${EXTEN:1},mqX)
exten => _8[0-2]XX,n,Hangup()
exten => _99[0-2]XX,1,Meetme(8600${EXTEN:2},X) 
exten => _99[0-2]XX,n,Hangup()

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

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

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


sip.conf

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[general]
context=trunkinbound           
allowguest=no                 
allowoverlap=no                 
bindport=5060                   
bindaddr=0.0.0.0
udpbindaddr=0.0.0.0               
srvlookup=yes                   
disallow=all                   
allow=ulaw                     
allow=gsm
mohinterpret=default
mohsuggest=default
language=en                     
relaxdtmf=yes                   
trustrpid = no                 
sendrpid = yes                 
progressinband=no               
dtmfmode = rfc2833             
videosupport=no                 
callevents=yes                 
rtptimeout=60                   
notifyringing = yes             
notifyhold = yes               
limitonpeers = yes   
externip=123.223.125.121           
;localnet=192.168.0.0/255.255.0.0
;localnet=10.0.0.0/255.0.0.0     
;localnet=172.16.0.0/12         
;localnet=169.254.0.0/255.255.0.0
nat=yes                         
canreinvite=no         
jbenable = yes             
jbforce = no               
jbmaxsize = 100             
jbresyncthreshold = 1000   
jbimpl = fixed             
jblog = no                 
qualify=yes             
limitonpeer = yes       
#include sip-vicidial.conf

I have SOX installed also.

Anyone can help me with this please?

tom wrote:Anyone can help me with this please?

The previous request was from someone who installed with a standard Vicibox .iso installer. You apparently did not install with this method.

You also did not show your CLI output from a single event or even state the type of call (inbound? outbound? DID?) that is being recorded.

I would start by explaining why you did not use the Vicibox installer (often quite useful to get an understanding of your situation) and next describe the call path and show CLI from a single occurrence ... and what was missing or wrong about that call path (such as "the 'record' never happened and I don't understand why").

williamconley wrote:

tom wrote:Anyone can help me with this please?

The previous request was from someone who installed with a standard Vicibox .iso installer. You apparently did not install with this method.

You also did not show your CLI output from a single event or even state the type of call (inbound? outbound? DID?) that is being recorded.

I would start by explaining why you did not use the Vicibox installer (often quite useful to get an understanding of your situation) and next describe the call path and show CLI from a single occurrence ... and what was missing or wrong about that call path (such as "the 'record' never happened and I don't understand why").

Thank you William for checking on this. I figured that I should continue this post since the problem I am encountering is the same even if I use the new install of Vicibox using vmware or virtualbox just to simulate what will i get compared to the set up I am doing remotely in a cloud base server.

We originally have Vicibox installed in our local office and we have a client who wishes to have the same set up. Unfortunately, the vicibox that we have is pretty old so we needed a newer version installed in the cloud reason now I am following instructions by ray-solomon link.

The situation is, this is a new install and everything should work based from the instructions but I have tried several versions and combinations of vicidial and asterisk installation to no avail. The latest is what I got above and that can make calls outside (call US numbers etc. using the sip.conf and extensions.conf configs). However, there were no recordings and this is my main concern or a feature required by our client.

Here is the CLI output. I have read and finding dialplans using the free manager manual to no avail and still can't get the recordings works. No Monitor recording when calling, no link generated, no MP3 or wav being created in the directories.

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    -- Executing [99918004563355@default:1] AGI("SIP/1111-00000039", "agi://127.0.0.1:4577/call_log") in new stack
AGI Tx >> agi_network: yes
AGI Tx >> agi_network_script: call_log
<SIP/1111-00000039>AGI Tx >> agi_request: agi://127.0.0.1:4577/call_log
<SIP/1111-00000039>AGI Tx >> agi_channel: SIP/1111-00000039
<SIP/1111-00000039>AGI Tx >> agi_language: en
<SIP/1111-00000039>AGI Tx >> agi_type: SIP
<SIP/1111-00000039>AGI Tx >> agi_uniqueid: 1456263273.57
<SIP/1111-00000039>AGI Tx >> agi_version: 1.8.32.3
<SIP/1111-00000039>AGI Tx >> agi_callerid: 1111
<SIP/1111-00000039>AGI Tx >> agi_calleridname: 1111
<SIP/1111-00000039>AGI Tx >> agi_callingpres: 0
<SIP/1111-00000039>AGI Tx >> agi_callingani2: 0
<SIP/1111-00000039>AGI Tx >> agi_callington: 0
<SIP/1111-00000039>AGI Tx >> agi_callingtns: 0
<SIP/1111-00000039>AGI Tx >> agi_dnid: 99918004563355
<SIP/1111-00000039>AGI Tx >> agi_rdnis: unknown
<SIP/1111-00000039>AGI Tx >> agi_context: default
<SIP/1111-00000039>AGI Tx >> agi_extension: 99918004563355
<SIP/1111-00000039>AGI Tx >> agi_priority: 1
<SIP/1111-00000039>AGI Tx >> agi_enhanced: 0.0
<SIP/1111-00000039>AGI Tx >> agi_accountcode: 1111
<SIP/1111-00000039>AGI Tx >> agi_threadid: 139959067891456
<SIP/1111-00000039>AGI Tx >>
    -- <SIP/1111-00000039>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
<SIP/1111-00000039>AGI Tx >> HANGUP
    -- Executing [99918004563355@default:2] Dial("SIP/1111-00000039", "SIP/570818004563355@APN,60,tToR") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/570818004563355@APN
    -- SIP/APN-0000003a is making progress passing it to SIP/1111-00000039
    -- SIP/APN-0000003a is ringing
    -- SIP/APN-0000003a answered SIP/1111-00000039
    -- Executing [h@default:1] AGI("SIP/1111-00000039", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----15") in new stack
AGI Tx >> agi_network: yes
AGI Tx >> agi_network_script: call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----15
<SIP/1111-00000039>AGI Tx >> agi_request: agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----15
<SIP/1111-00000039>AGI Tx >> agi_channel: SIP/1111-00000039
<SIP/1111-00000039>AGI Tx >> agi_language: en
<SIP/1111-00000039>AGI Tx >> agi_type: SIP
<SIP/1111-00000039>AGI Tx >> agi_uniqueid: 1456263273.57
<SIP/1111-00000039>AGI Tx >> agi_version: 1.8.32.3
<SIP/1111-00000039>AGI Tx >> agi_callerid: 1111
<SIP/1111-00000039>AGI Tx >> agi_calleridname: 1111
<SIP/1111-00000039>AGI Tx >> agi_callingpres: 0
<SIP/1111-00000039>AGI Tx >> agi_callingani2: 0
<SIP/1111-00000039>AGI Tx >> agi_callington: 0
<SIP/1111-00000039>AGI Tx >> agi_callingtns: 0
<SIP/1111-00000039>AGI Tx >> agi_dnid: 99918004563355
<SIP/1111-00000039>AGI Tx >> agi_rdnis: unknown
<SIP/1111-00000039>AGI Tx >> agi_context: default
<SIP/1111-00000039>AGI Tx >> agi_extension: h
<SIP/1111-00000039>AGI Tx >> agi_priority: 1
<SIP/1111-00000039>AGI Tx >> agi_enhanced: 0.0
<SIP/1111-00000039>AGI Tx >> agi_accountcode: 1111
<SIP/1111-00000039>AGI Tx >> agi_threadid: 139959067891456
<SIP/1111-00000039>AGI Tx >>
    -- <SIP/1111-00000039>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----15 completed, returning 0
<SIP/1111-00000039>AGI Tx >> HANGUP
  == Spawn extension (default, 99918004563355, 2) exited non-zero on 'SIP/1111-00000039'

Gleaned:
You have several installations ... this call is outbound.

1) You did not state which intallation (Vicibox 6.0.4? Vicibox 7.0.0? Scratch install from Solomon?) this output was from.

2) You have agi debugging on. No need for this as there is no agi script in this example that would be expected to activate recording that would make agi debugging useful.

3) You have not described the call path of this call and how you expect it to be recorded. (IE: This call was from a campaign with recording set to ...XXXX?)

4) What settings have you modified to cause this particular call to be recorded?

5) What generated this call?

williamconley wrote:Gleaned:
You have several installations ... this call is outbound.

1) You did not state which intallation (Vicibox 6.0.4? Vicibox 7.0.0? Scratch install from Solomon?) this output was from.

2) You have agi debugging on. No need for this as there is no agi script in this example that would be expected to activate recording that would make agi debugging useful.

3) You have not described the call path of this call and how you expect it to be recorded. (IE: This call was from a campaign with recording set to ...XXXX?)

4) What settings have you modified to cause this particular call to be recorded?

5) What generated this call?

Thanks William.

1. I just have one installation in the cloud. I just did several test using those virtual environments to see if setting up vicibox as default would make it easier for me because if I can make it work there I can somehow replicate the configuration to the cloud. The output was from the cloud where I used Solomon's instructions.

2. I just followed a instruction from one of the forums here so i thought this would help.

3. I created a sample campaign 12345678 and had set the required recording settings to ALLCALLS. and has created an xlite extension of 1111. I wanted it to record all calls made by 1111 using campaign i provided earlier. Surely I am missing something here which I cannot find due to lack of knowledge about setting this up.

4. I have enabled campaign and phone recordings.

5. Not sure if I understood you correctly but forgive if I don't. The call was generated from xlite dialling a US number.

Please let me know what you need and I will be happy to provide the information as soon as possible while you are also online.

Thanks again.

I think we found our problem.

3. I created a sample campaign 12345678 and had set the required recording settings to ALLCALLS. and has created an xlite extension of 1111. I wanted it to record all calls made by 1111 using campaign i provided earlier. Surely I am missing something here which I cannot find due to lack of knowledge about setting this up.

The call was generated from xlite dialling a US number.

Configuring a campaign to record calls and then making a call outside the campaign will not cause the call to be recorded. Also: Since you made the call outside the Vicidial system completely (basically using the SIP phone like a desk phone, no "agent" involvement or software interaction at all ...), you have not captured any client/prospect data. You have nowhere to store the client name, demographics, notes, address ... nothing. And no disposition at the end of the call.

No previous call history to this lead ... how is that better than just using the soft phone without the dialer? (Pretty sure there are soft phones out there that can be set to autorecord ...,. why bother with a dialer?)

So you make the call with your SIP phone, store all your client information in a spreadsheet or a google doc (or on a sheet of paper? LOL), and then ... how do you know when (or if) to call this prospect again? Or track the statistical outcome of this call?

Yep, I'm giving you a hard time for buying a car and then pulling it to the corner with a team of horses.

How about this: Load leads into a list which is assigned to a campaign. Log in to the campaign as an agent. Unpause the agent (older systems say "Resume"). Then the Campaign will generate the outbound call. The call will only be passed to the agent if someone answers the phone. Then you'll have a place to store all the client information (including the disposition at the end of the call). And if the campaign is set to record... recording will happen. THIS process is described in minute detail in the Vicidial Manager's Manual. Start at page one. Don't skip anything. EFLO.net ... download free version (buy the paid version as soon as you have tested the system working and realize how much $$ you just saved by not purchasing $25k worth of software to do this!)

Of course, if the campaign is manual dial (or inbound-man, which is Manual Dial with Inbound), you would push the "dial next lead" button instead of resume or "unpause". But the campaign would still be in charge and the recording would still happen.

And last but least, it IS possible to record the calls made outside the system ... I just hesitate to give you that without either making you work for it or strongly suggesting you make the agents log in to do their dialing. If these are agents, they'll steal those leads and you'll never know. LOL