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Call dropped once agent recieved

PostPosted: Thu Apr 07, 2011 9:52 pm
by rtabigue
Hello,

We changed our internet provider with the same speed in our current ISP (E1 2Mbps).We got problem when the agent received the call, agent can't hear the customer and call will dropped. Below the setting of dialing method

Call per second: 2
Dial Method: Ratio
Auto dial Level: 2.5
Drop rate: 3%
Dial level difference: 0 --- Balanced
Drop call seconds: 5
Dial time out: 15

But when we tried manual dial its working fine. Is this an issue with our new ISP provider? could anyone from here help me to resolve this issue. Thanks!

Also i found this error messages below in the asterisk cli during the call testing with one agent logged!..Is this possible cause why the call will drop?Please advise. thank!


[WARNING[5239]: app_meetme.c:2985 admin_exec: Conference number '8600051' not found!]


VERSION: 2.2.1-237
BUILD: 100510-2015
Vicidial 2.2.1
Asterisk 1.4.27.1-vici
Dahdi 2.3
VtigerCRM 5.1.0
Sangoma 3.5.14
CentOS 5.5

PostPosted: Fri Apr 08, 2011 4:07 am
by mflorell
What kind of zaptel/dahdi timer are you using?

PostPosted: Fri Apr 08, 2011 9:57 am
by williamconley
and (just to be sure): when your agents log in, do they hear "You are the only person in this conference"?

PostPosted: Fri Apr 08, 2011 4:01 pm
by rtabigue
mflorell wrote:What kind of zaptel/dahdi timer are you using?


Thank you for your reply..Sorry but i dont know how to check the zaptel/dahdi timer. Can you please guide me?.

PostPosted: Fri Apr 08, 2011 4:06 pm
by rtabigue
williamconley wrote:and (just to be sure): when your agents log in, do they hear "You are the only person in this conference"?


Hello William, thank you for your reply.. Yes, they can hear the said voice prompt "You are the only person in this conference" upon logging in but when the call arrived to the agent live call indicator turned to green and after 2 to 3 seconds it will turned to grayed out. the agent did not yet talk to customer

PostPosted: Fri Apr 08, 2011 8:14 pm
by williamconley
One step at a time.

We just covered the agent logging in and hearing "You are the only person in this conference". Then the agent presses "Resume" and does NOT hang up the phone.

At this moment, you should be watching the asterisk CLI (available via "asterisk -R" in the ssh console). I mean at the moment that the agent presses "Resume".

Right then: Does the system initiate a phone call outbound? (Please show the output from that moment ... or the next 2 or 3 seconds at most, until the "Hangup" line if there is a call).

For this to work, you'll need to have a Campaign set up with a list assigned and leads in the list, etc. Just seeing where we are. 8)