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Vicidial not dialing out numbers

PostPosted: Tue Aug 18, 2009 10:31 am
by vicidial0
hello


the probelm is my vicidial is not autodialing any customer.

I have done everything accordingly with getting started Guide.

I am trying to dial customers in canada.

I just Did config mentioned in getting started guide..

tell me how may I know if vicidial is dialing voip provier?

please help its very urgent.

thanks in advance.

tell me if you need any output . do tell me what do I have to do

PostPosted: Tue Aug 18, 2009 12:07 pm
by gardo
Please read the stickies first before posting your issues.

stickeies read

PostPosted: Tue Aug 18, 2009 1:19 pm
by mg1
I have already read the getting started stickies and following a guide given in this forum ..

PostPosted: Tue Aug 18, 2009 2:26 pm
by gardo
Then you should have posted the details of your system if you have read the stickies. Asterisk version, Astguiclient/Vicidial version, Linux distro, and etc.

I am using Vicidial now (current)

PostPosted: Wed Aug 19, 2009 4:06 pm
by mg1
vicidial now community edition 1.2
kernet 2.6.18-92.e15.vnow on i686

Confusion

PostPosted: Wed Aug 19, 2009 4:20 pm
by mg1
IS there any Free SIP provider so that I may know if there is something wrong with the SIP provider or Vicidial. how may I check if SIP is supported on my ISP.

I have two IDS i.e Vicidial0 and MG1 .. I am the same person

PostPosted: Wed Aug 19, 2009 4:20 pm
by mg1
given below are my screen shots of vicibox

screen outputs

PostPosted: Wed Aug 19, 2009 4:21 pm
by mg1
I have done this config in Carrier

Image


When I go on agent side and click dial I dot get green live call sign on agent page neither

Image


at the end I get this thing



Image

My gateway is set further more I have configured internet properly over VICIDIALnow

Image



host being resolved



Image


Vicidial not being registered


Image

and using different Ip


Image


what may be the possibility ??

following are the information provided by my SIP provider

User ID: xxxx
Password: xxxxx
SIP Proxies: 69.1.224.14 or sip-mytalktel.borderproxy.com or 208.76.18.241
SIP Alternate ports: 5744 - 20051 - 5566 - 6070
Voice Codec: G.729 20ms
DTMF Method: RFC 2833 Out of Band
Signalling Protocol: SIP
SIP Compliance: RFC 3261 (SIP v2)


Any help for the obove mentioned case will be highly appriciated

thanks in advance

say

PostPosted: Wed Aug 19, 2009 4:41 pm
by brett05
say me what number you write with xlite directly to canada
because i think you have a mistake in your dial plan
and then i can help you

Confusion

PostPosted: Wed Aug 19, 2009 4:50 pm
by mg1
I had heard that asterisk it self dial the numbers mentioned in lists loaded in campaign hover. do I have to manually dial the number using Xlite? , vicidial actually call my soft phone and after that vicidial suppose to dial the number automatically when I click dial number in Agent web page.

xlite dialing

PostPosted: Wed Aug 19, 2009 5:03 pm
by mg1
I dial (without Vicidial) using Xlite +1 then the number for canada

E.g : +1 403-269-38xx

sample

PostPosted: Wed Aug 19, 2009 5:06 pm
by mg1
following are the entries I made in xls file


PHONE CODE PHONE NUMBER FIRST NAME LAST NAME ADDRESS1 CITY STATE POSTAL CODE COUNTRY CODE
1 403-269-38xx A Singh xxx 6 Ave SW 804 Calgary
1 403-569-84xx A Singh xxx Saddlebrook Way NE Calgary

PostPosted: Wed Aug 19, 2009 5:07 pm
by chill_master
It seems to me that your asterisk can't register to your VOIP provider. Check with your provider. What I do is I have this list of servers available from my provider and everytime I encounter this kind of problem, I just switch to another server.

Finding your provider's server through nslookup will only tell you that the machine is up, but it doesn't tell you if the SIP is up in this machine.

say

PostPosted: Wed Aug 19, 2009 5:17 pm
by brett05
ok we need to make same test :
first in your lead file you have write PHONE CODE =1
and your number is +1 403-269-38xx
also you have write your dial plan as this:_91XXNNXXXXXX so as i see you have triple the code phone ,
try this :
let your lead the same ,then change your dial plan as this
_9XXXXXXXXXX
and do you have install codec G729 ?
have you try in cli asterisk "sip show registry and "sip show peers" to verfiy your voip provider if he is registred?
have you verify if you have G729 installed good with cli asterisk "show translations"?
have you verify if your route and internet connection work good ?
try to ping to your voip or try to add your gateway to your network device?

PostPosted: Wed Aug 19, 2009 5:23 pm
by chill_master
I don't think it's even the codec... it's clear to see from the CLI screenshots that his server cannot register to his provider.

Settle that first. You cannot do anything if it's not registering.

PostPosted: Wed Aug 19, 2009 5:32 pm
by brett05
yes true i share your this opinion
he need verify this with his voip provider because asterisk can not register

Vicidial

PostPosted: Wed Aug 19, 2009 6:27 pm
by mg1
As you can see I can ping and resolve the Sip provider.
actually I took this sip account from another guy having .5$ balance. He is saying that you may check it and the account has 30 mins for talking.

he is right.

is there any other way I can check the registration with the sip provider. Can anybody tell me some solution or just let me use some sort of trial.


do anybody offer trial SIP service just for checking.

brett05
Will I be able to register vicidial with sip provider after

let my lead the same ,then change my dial plan as this
_9XXXXXXXXXX

PostPosted: Wed Jan 20, 2010 4:10 am
by albokos
hi

i have the same issue, it seems asterisk does not dial any leads, i checked my sip provider and my boc register succesfully but there is no way for me to pass calls (manual an auto) asterisk CLI doesn't log any occurring call. i'm on production now and all my seats are out of service. please help

PostPosted: Wed Jan 20, 2010 4:28 am
by albokos
i've found out that when i place call manualy asterisk executes the meetme application not dial. what is going on

PostPosted: Wed Jan 20, 2010 11:08 am
by gardo
Please post the Asterisk CLI when dialing.

PostPosted: Thu Jan 21, 2010 4:07 am
by albokos
thank for the reply;

Iwas in a hurry so i reinstall asterisk and vicidial and the problem was fixed i still don't know what happened but to answer your question in manual dial my CLI just do not show anything but the meetme application and in auto dial calls are passed and answered but no agent receive them.

Fortunatly it works now maybe i did something wrong in my first install thanks for watching my case.