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Vicidial and Sip phone
Posted:
Wed Nov 15, 2006 12:47 pm
by Aminovsky
Hi Dear,
I try to install vicidial on trixbox but now it doesn't work.
I ask about how can I verify if my sip phone when I confugured it whith Astguiclient work fine?
Thincks
Posted:
Wed Nov 15, 2006 1:57 pm
by mflorell
Try not using Trixbox, and install VICIDIAL using the SCRATCH_INSTALL document in the docs folder of the release.
Posted:
Wed Nov 15, 2006 11:25 pm
by Op3r
whats the gain when u use trixbox with a dialer?
if you want a dialer install it on its dedicated server. that way your confs are clean and when some problems come the troubleshooting wont be complicated.
simple solutions can really help an adminstrator's life easier.
Posted:
Thu Nov 16, 2006 5:42 am
by Aminovsky
thank-you for your response,
which system UNIX could you advise me ?
i used FADORA 3 and i couldn't fix up asterisk
THANKS IN ADVANCE
Posted:
Thu Nov 16, 2006 5:55 am
by gerski
use slackware 10.2 or higher version as stated on scratch install
Posted:
Fri Dec 01, 2006 9:11 am
by Aminovsky
Hello
i have well followed all instructions of "Scrach Install" and i succeeded planting vicidial,with opening logon "/agc/asteguiclient.php my soft phone call lasts only 3 seconds then it Hung up.
what did i should do?
thank you
Posted:
Fri Dec 01, 2006 11:49 am
by Delta239
did you try /agc/vicidial/php ????
Posted:
Sat Dec 02, 2006 6:08 am
by Aminovsky
yes i had tryed it but the phone don't work
Posted:
Sat Dec 02, 2006 11:14 am
by Aminovsky
thank-you,
I have reussi to install and vicidial works correctly
Posted:
Sun Dec 03, 2006 4:53 pm
by Aminovsky
Hi,
I succeeded in the installation of VICIDIAL as you tlod me and the progma worked in the manual way "/agc/vicidial.php". But when I tried to do it automaticically ""/agc/asteguiclient.php", the program didn't work.
May be because I didn't understand the mechanism.
My situation is the following :
I have a list of number of 8 character each in campaign could you explaine to me the functioning of vicidial so that I can configure the extension.conf
Thank you very much
extension.conf
Posted:
Tue Dec 05, 2006 5:58 am
by Aminovsky
if somebody can help me to understand the file extension.conf
or there is doc where is described?
thanks
Posted:
Tue Dec 05, 2006 6:58 am
by gerski
have you tried to type "make samples" after compiling asterisk?
in there you can see the explanation how the commands work.
Posted:
Tue Dec 05, 2006 6:59 am
by Aminovsky
hi
somebody know why in the logs (CLI) I see that the system added 91 for the leads numbre that I entered in the list.
thanks
Posted:
Tue Dec 05, 2006 7:04 am
by Aminovsky
I done "make samples", but it just for USA, and i cant use this one for my case.
Asterisk is working fine, vicidial seem to working fine too, but I got problem only with autodialing, and I try to look of the problem, I find that the leads nbre is called with 91 and than the nbre. I dont find from where is the 91 comming.
thanks
Posted:
Tue Dec 05, 2006 7:07 am
by gerski
what country are you dialing? are you using autodialing?
if that's the case have you check in the campaign detail, DIAL PREFIX?
Posted:
Tue Dec 05, 2006 9:05 am
by mflorell
The 9 comes from the campaign setting for dial prefix
the 1 comes from the phone_code field of vicidial_list table
(both can be disabled using campaign settings)
]
Posted:
Tue Dec 05, 2006 11:02 am
by Aminovsky
thanks for your helps,
the problem with the 9 and 1 is solved now, but I dont get the (auto-) dail working.
Can you help me to now why I got this message, when I start autodial from vicidial:
Posted:
Tue Dec 05, 2006 11:09 am
by Aminovsky
thanks for your helps,
the problem with the 9 and 1 is solved now
thanks
Posted:
Fri Dec 08, 2006 8:43 am
by Aminovsky
Hi ,
I can’t establish outcoming calls ;is there any one who can help me?
thunks
Posted:
Fri Dec 08, 2006 10:44 am
by mflorell
You need to post some output of some kind.
screen -r
output from Asterisk CLI when you are trying to call.
output from /var/log/astguiclient/agiout.
Posted:
Fri Dec 08, 2006 12:10 pm
by Aminovsky
Hi Matt,
Excuse me I make mistake not oucoming but INcoming
.
when anyone call me and hungup my asterisk still connected ( with out Hanguping )
witch parmetre can I add for resolve this problem ?
thunks