vicidial.org

Hi every one,

I am new bee to this. i have installd the asterisk 1.2.17 with SIP softphone (xlite)and astguiclient 2.0.2 i have succesfully created the database and also logged with admin account.

i have created the phones, users ,group and campaign.
Now i want to make call how actually i have do iam not understanding..any body possible plse make me understand how actually i can make outgoing calls through the astguiclient with sip softphone xlite..and how to record the calls.


Thanks in advance

Regards,
Sandro

I highly suggest you buy the manual from www.eflo.net

best regards,

Hello Sandro

1) Test that you have working extensions , try calling from xlite to xlite on ur local network

2) configure your sip provider in sip.conf and the dialplan in extensions.conf and make a manual call from xlite

3) Create a campaign in vicidial admin and load leads, run the campaign in manual dial mode, read the manual on how to do this,

4) login to vicidial with the agent phone and user login, and click next to dial a lead, if u can place a call sucessfully change the campaign to autodial mode RATIO and login with the agent again click on resume , monitor the realtime screen as well as asterisk console (asterisk -r)

5) go through the scratch install again and again and do RTFM (read the friendly manual) search on the forums for problems and post after you have tried enough

Regards

Diyanat

Thank Q Diyanat for your response.

I have configured the nodes in sip.conf and extension.conf .and tried calling with xlite to xlite.i t is working fine.

should the users added in astguiclient are should be added in sip.conf and extension.conf also. can u plse clear this to me..!

i have read the scratch install on astguiclient and installed sucessfully on redhat AS 4

Regards,
Sandro

phones and users are 2 different things in vicidial, the phones will be your xlite/ip-phone extension and the users will be the agents login, you will need to add the phones in vicidial admin under phones, same as u created in sip.conf, The users need to be added in the vicdial users only and not in the sip.conf

for example agent Mike will have 1 phone login and 1 user login

phone :1001
user: agentmike

on the vicidial /agc/vicidial.php login page, Mike enters 1001 as the phone and agentmike as the user

Regards

Diyanat

Hi

i recomend to just download
admin and user manual free version and read
so you will understand more on the how you can use vici
dial for making calls

ram

Hi diyanat

can u plse give me a example config files of sip.conf and extension.conf dial plans
and also the how to add leads, users phones and campaign etc overall to make a call and test...! b/w nodes in local and also the recording option..

even iam also going through the manuals.....but it would be appriciated and would be help ful to me if u can help me providing the above..

Thanks
Regards,
Sandro

Hi sandro

sip.conf have different configuration depends on provider to provider

first check with your provider, does he support multi channel calling.

some providers does not support multi channel support, so after 2 agents you will see the message, all circuits busy.

general sip.conf examples are there in asterisk source. ( after that also you have any problem with sip.conf, post your problem to group)

extension.conf dial plan there in scratch installation


in the source directory you can see the lead file in text and as well as excel

you can load this lead either from command level or from admin.php.


once you load the leads, create list, create user and phone ( follow admin manual)

login using x-lite to your asterisk, then,login vicidial

user and phone, then you get a call on x-lite on your pc

accept that call, then you hear you are first person in the conference.

then use vicidial web to call the leads.

recordings you can check in campaign settings.

ram

Hi Ram,

Thank Q very much.

sandro