Some issues while testing VICIDIAL

[dev_4901]

Hi all,
During the testing of VICIDial there are a few issues that I'm facing.

1) Answering Machine Detection : I'm not sure whether VICIDial is working as a predictive dialer or not. I have placed the amd.conf in /etc/asterisk and the lines for AMD in extensions.conf. How is this to be checked?

2) Load on server : Although I've installed Ploticus in /usr/local, I'm still not able view the load on server through he VICIDial Admin interface.

3) Transfer - Conf : I'm not able to transfer the calls to another extension. I get a message stating the 'originating call' but the call does not get tranferred nor I'm able to put the customer line onto a 3 way conference with any IVR.

4) Call Barge/Snoop : when I click on 'Listen' in the 'ACTIVE LINES' of 'agc/astguiclient.php', I get a message that 'YOU CAN ONLY LISTEN TO ZAP CHANNELS'. I'm not using any ZAP cards. I'm using pure SIP setup.

5) Call Recording Download : How can I download the Voice Logs(Recordings) from the ViciDial Interface only?

6) Manual Dial : While dialing a number manual through the agent login, after the call connects, all the command buttons of 'START RECORDING', TRANSFER-CONF', 'HANGUP CUSTOMER' etc are disabled. How do i get over this?

Plz advice and thanx in advance.

Dev Singhal.

[mflorell]

Double post from mailing list. answered on mailing list:

> 1) Answering Machine Detection : I'm not sure whether VICIDial is working as
> a predictive dialer or not. I have placed the amd.conf in /etc/asterisk and
> the lines for AMD in extensions.conf. How is this to be checked?

Check your /home/cron/AMD_log.txt file. If there is none after calling
with AMD, then it is not set up correctly


> 2) Load on server : Although I've installed Ploticus in /usr/local, I'm
> still not able view the load on server through he VICIDial Admin interface.

Collecting of system load stats is not on by default. You need to set
the $SYSPERF variable to "1" at the top of the AST_update.pl script
and restart it for the status to be collected.


> 3) Transfer - Conf : I'm not able to transfer the calls to another
> extension. I get a message stating the 'originating call' but the call does
> not get tranferred nor I'm able to put the customer line onto a 3 way
> conference with any IVR.

Please post the Asterisk CLI output when you try to do this.


> 4) Call Barge/Snoop : when I click on 'Listen' in the 'ACTIVE LINES' of
> 'agc/astguiclient.php', I get a message that 'YOU CAN ONLY LISTEN TO ZAP
> CHANNELS'. I'm not using any ZAP cards. I'm using pure SIP setup.

To quietly listen in on agent you can dial 6 + session_id (68600051)
To barge in on an agent dial their session_id (8600051)


> 5) Call Recording Download : How can I download the Voice Logs(Recordings)
> from the ViciDial Interface only?

You cannot. You need to setup a method of doing that yourself. We do
not include the option from the VICIDIAL server since it would greatly
reduce performance and most systems use an archive FTP server for
recordings.


> 6) Manual Dial : While dialing a number manual through the agent login,
> after the call connects, all the command buttons of 'START RECORDING',
> TRANSFER-CONF', 'HANGUP CUSTOMER' etc are disabled. How do i get over this?

If they are not showing up then the system is not set up properly.
Are you altering CallerID in any way in your dial plan?
Are you using the "o" flag in your external dialout Dial function?