I have 2 Queues Member setup via queues.conf
Its routed to their cell phone, when the queue members ignores the calls and let it ring
after the timeout, it goes to the next member like its suppose to.
The problem is if the queue members literally Cancel the call,
its goes to their cell phone voicemail.
Is there a way to prevent this and instead just go to the next member?
The cancelation on the cell phone user side it is not consired by Asterisk like a busy reponse, instead it is answered by the voicemail or answerd machine
Ah so basically Asterisk has no control over that and its on the user end.
Is there anything the user can do?
What thing you could do on the Asteisk side is reduce the timeout before it reach the voicemail
Thanks! I appreciate all the info. 
You could use the āpā option when you dial your phones to run it through the privacy manager.
That would require your agents to press 1 to accept the call after they answer.
Since voicemail canāt send a ā1ā to connect the call the callers wonāt end up on the cell phones voicemail.
Thanks for the tip, but can you elaborate more? I tried google what you said but nothing tangible came up
The only thing I found was the followingā¦
ackcall is probably a better solution. The reference to privacy manager was for a Dial application option.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial
Iām assuming you are using local channels to route your call to the cellphones since you canāt usually add a cell phone as a queue member otherwise.
In the Dial() Command that you use to reach your cell phones add the āpā flag Which āenables screening modeā where the caller will be asked to say their name and the callee will be given the option to accept the call from the caller.
unfortunately I wont be able to use the dial option p because Im adding the cell phones directly to the queues. Iād like to take advantage of the queues hunt group functions which is awesome.
im adding cell phone like soā¦
member => PJSIP/4075558888@anveodirect_queue, , , PJSIP/anveodirect_queue
and it works fine plus it actually recognize when the user is on a call and will go to next member if so.
I didnt do much research into it yet hoping there is a better solution so far ackcall may be the solution, I wonder how well it will do in my application especially since im using Asterisk 13 and voip-info appears to have old info.
Interesting. I figured you were looping the call through a local channel like this:
[myqueue]
member => LOCAL/4075558888@queue-phones,20,Bob,hint:4075558888@queue-phones
The Dialplan Context:
[queue-phones]
exten => 4075558888,1,Dial(PSIP/${EXTEN@anveodirect_queue,20,p)
exten => 4075558888,hint,PSIP/${EXTEN}@anveodirect_queue
pretty cool alternative, but the only down side I see is the 1st member will keep getting the most calls
Thanks for the tip though, it a good technique that will be useful in other applications
That could be solved by using ārrmemoryā or ārandomā queue ring strategy.