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NO LIVE CALL

PostPosted: Thu Dec 08, 2011 12:40 pm
by sasukael
hi

right now i have this issue :

all of our agents can call (actually they made contact to the customer) but the web gui client shows "NO LIVE CALL" .. and that so .. the dialed time out will appear contact your system administrator
the agents get annoyed everytime they made contact to the customer.

Why is this happening?

Can someone fix this?

PostPosted: Mon Dec 12, 2011 9:25 pm
by sniper888
Hi,

Try to check your campaign settings for dialed time out setting.

PostPosted: Wed Dec 14, 2011 8:56 am
by sasukael
I changed it to 30 seconds since i recycle answering machine leads ... but its still the same ....


any another suggestion?

PostPosted: Thu Dec 22, 2011 5:21 pm
by williamconley
are these calls manual dials?

show your carrier setup (particularly your "dial plan entry" as this can cause failure if not standard)

PostPosted: Fri Dec 23, 2011 12:17 pm
by sasukael
The dial method we use is MANUAL DIAL

Here is the Account entry:

[AccelaSansay1]
type=friend
host=xx.xx.xxx.xxx
canreinvite=no
disallow=all
allow=g729
allow=ulaw
dtmfmode=rfc2833

Global String : VOIPTRUNK = SIP/AccelaSansay1

Dial plan entry:

exten => _011.,1,Dial(SIP/${EXTEN}@AccelaSansay1)
exten => _011.,2,Dial(SIP/${EXTEN}@AccelaSansay2)
exten => _011.,3,Congestion
exten => _011.,103,Congestion

PostPosted: Fri Dec 23, 2011 12:33 pm
by williamconley
your dialplan entry does not contain the AGI line (required, as line 1, not optional, check the vicidial manager's manual and all the "sample carrier" setups available in admin->carriers) nor does it contain the hangup line.

dialplan entries supplied by the carrier are "suggestions" that should result in the correct signalling from your server to theirs, but they do not dictate the entire dialplan, only the proper format of the "dial" command itself any anything that may need to happen before it to be sure it contains the proper information and format.

Try:

Code: Select all
exten =>_011XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_011XXX.,n,Dial(${VOIPTRUNK}/${EXTEN},,tTor)
exten =>_011XXX.,n,Hangup
for starters. you should also look setting up multiple carriers (which ordinarily requires the use of a dial prefix in your campaign).

you should be using auto-dial to get the most of your dialer (you have a maserati in your garage and you're driving a bicycle instead ... for long trips? LOL)

PostPosted: Fri Dec 23, 2011 3:57 pm
by sasukael
yes thank you it works .... :)


Can u give me a hint why there is 3 XXX in the dial plan entry? ... or it should be three...

PostPosted: Fri Dec 23, 2011 4:33 pm
by williamconley
to make the minimum length of a "match" at least 7 digits (ie: the minimum length of a phone number ...) which can avoid a collision with other vicidial extensions. :)

Re: NO LIVE CALL

PostPosted: Mon Mar 17, 2014 5:10 pm
by arf
hi, i have the same issue.

VERSION: 2.7-401a
BUILD: 130508-2256
© 2013 ViciDial Group

Dial timeout = 60

my carrier setup

[aaa]
type=friend
username=****
secret=****
host=***.***.***.***
canreinvite=no
fromuser=****
dtmfmode=rfc2833

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Set(CALLERID(name)=8774539055)
exten => _91NXXNXXXXXX,3,Dial(sip/1${EXTEN:2}@aaa,,tTo)
exten => _91NXXNXXXXXX,4,Hangup

Re: NO LIVE CALL

PostPosted: Thu May 01, 2014 7:55 pm
by williamconley
arf wrote:hi, i have the same issue.

VERSION: 2.7-401a
BUILD: 130508-2256
© 2013 ViciDial Group

Dial timeout = 60

my carrier setup

[aaa]
type=friend
username=****
secret=****
host=***.***.***.***
canreinvite=no
fromuser=****
dtmfmode=rfc2833

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Set(CALLERID(name)=8774539055)
exten => _91NXXNXXXXXX,3,Dial(sip/1${EXTEN:2}@aaa,,tTo)
exten => _91NXXNXXXXXX,4,Hangup

Multiple issues were mentioned, and you did not state which one(s) you have. Possible you just haven't loaded any leads. Or your agents logged in as closers. Or your campaign has no "Dial Statuses". Lots of things. Some more details would be helpful. :)

For instance: Are there leads available to be dialed in the campaign (this is actually a value shown below the thick black line under the campaign settings)? Are there leads in the hopper (also a shown value)?

Re: NO LIVE CALL

PostPosted: Thu Jan 21, 2016 5:35 pm
by bondkmf
Having same issue. No one can make manual dial or callbacks (it appears to be happening from only 1 of the sip servers not the other)

VERSION: 2.12-497a
BUILD: 150717-1333

1 DB
2 vicibox sip servers

[iplinkOUT65]
type=peer
host= x.x.x.x
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=trunkinbound

----------------------------------------

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN:1},,To)
exten => _91XXXXXXXXXX,3,Hangup

exten => _1XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1XXXXXXXXXX,2,Dial(${SIPTRUNK}/${EXTEN},,To)
exten => _1XXXXXXXXXX,3,Hangup

exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(${SIPTRUNK}/1${EXTEN},,To)
exten => _XXXXXXXXXX,3,Hangup









Please help. Since vici isn't registering as live call we can not transfer to verif and are 100% shut down right now.
If I left anything out please let me know and I will include it.

Re: NO LIVE CALL

PostPosted: Fri Feb 12, 2016 5:46 pm
by bondkmf
Bump. Still having NO LIVE CALL problem. It is intermittent. Please see my previous comment for specs.

Re: NO LIVE CALL

PostPosted: Fri Apr 22, 2016 11:21 am
by bondkmf
Anyone? It randomly happens. Maybe a faulty build?
Internet and all mtr's and pathpings are fine.
auto dialing registers live calls.

ONLY manual dial and callbacks does not register as a live call.

I am at a loss.

Re: NO LIVE CALL

PostPosted: Fri Apr 22, 2016 11:47 am
by williamconley
1) Welcome to the Party! 8-)

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

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)

You should also post: 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. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Asterisk CLI of a single example would be quite useful. Troubleshooting. 8-)