vicidial.org

everytime I try to add a carrier or even modify the test sip account info, i always get:

CARRIER NOT MODIFIED - Please go back and look at the data you entered

How do I get this to work, I mean I can add and delete users but I cant modify carrier info, and this is logged in as admin and setting all values for this user to 1

VERSION: 2.2.1-237
BUILD: 100510-2015

1)

when you post, please post your entire configuration including (but not limited to) your installation method and OS with kernel or version, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

2)

what did you enter? It's telling you to check what you entered ... so you show me what you entered in the fields and I may be able to tell you what to change. (replace passwords and user/domain names and ip's with same # of x's)

3)

Please do not double post.

http://www.vicidial.org/VICIDIALforum/v ... ght=#52185

ViciBox_Redux.i686-2.0.2 from .iso | Vicidial 2.2.1-237 Build 100510-2015 | Asterisk 1.4.21.2 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation |

I got a flowroute account put in this information like this:

Carrier ID: Flowroute
Carrier Name: Flowroute
Carrier Description: none
Registration String: register => 14512814:xxxxxxxxxx@sip.flowroute.com
Template ID:
Account Entry:

[flowroute]
type=friend
secret=xxxxxxxxxx
username=14512814
host=sip.flowroute.com
dtmfmode=inband
rfc2833compensate=yes
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=sip.flowroute.com

Protocol: SIP

Globals String:TESTSIPTRUNK = SIP/testcarrier

Dialplan Entry:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@flowroute,55,tTo)
exten => _91NXXNXXXXXX,3,Hangup

Server IP:Test install of Asterisk server


For the server ip I selected the option it provided, yet when I click on the server: test install, it has no server ip

I got the carrier info in, once I assigned the server ip, I created the campaign, list, and user.

now it is not loading leads into the hopper, and when i try to login with my new user per the agent interface it will not log in and everything is active

I deleted the campaign and recreated it, recreated the lead list, which only has 10 leads.

its on 24 hour dial plan

and still no leads in hopper

the dial status is at new

Dial Status 1: NEW - New Lead

This campaign has 1 active lists and 0 inactive lists

|| This campaign has 10 leads to be dialed in those lists - HIDE

This campaign has 0 leads in the dial hopper


even if i change the dial status to something else it still loads no leads in the hopper

useing:

/usr/share/astguiclient/AST_VDhopper.pl --debugX

it says:


----- DEBUG -----


----- SUPER DEBUG -----

Server IP does not have an entry in the servers table


Also I changed my dial prefix to x because I do not use 9 to dial out

I find this strange becuase if the debug for the hopper says that why is there info for the server in the mysql db under the servers table