I have a problem with the Queue function. I am using asterisk 1.8.13.0 version. And the clients have eyeBeam Sip Phone.
I am receiving about 120 incoming calls at the same time. There is no problem. It works good. But when on each member is on the line, another incoming calls are ringing. And 6 incoming calls are ringing on each member. I have tried to set limitation on each member to have only one line but i can not do it. Is there anyway to limit incoming calls on each member ?
I am using realtime users, queues and queue_members table.
you may want to try setting “autofill” to “no” in the queues config, that tells the queue to just send the next waiting caller to the phones (and only the next waiting caller).
to eliminate call waiting, you may want to set “ringinuse” to no as well, otherwise phones that are on a call still get 1 waiting caller.
I had the same problem with too many callers at once, and the phones were denying calls after the 4th one (I have 4-line phones). Solution was to have a wallboard displaying the number of waiting callers, set autofill to no, and also have custom hints lighting x amount of BLF keys according to the number of waiting callers for those who couldn’t see the wallboard. Works great, and you eliminate the problem where agents don’t answer the next waiting caller, but maybe the 3rd or fourth in line because they’re using the line keys instead of the answer button.
I’m also using realtime, but not the SIP extensions, just queues.
Thanks for your reply. It works very well on the SoftPhones.
When the calls were coming, they are all quering in the queue. And one by one they are
distrubuting to the agents. But one problem occurs. When an agent is talking, is in use on the line,
another call is ringingin also. I have tried it many times. I set “no” ringinuse and autofill.
Eyebeam has 6 lines. When an agent is talking on the line the second line is ringing.
I set the call-limit=1. It works. But i want the agent to use the first line for holding and the second line for talking.
i dont want to use the call-limit = 1. I give the test example below.
I register only one agent that is using eyebeam to asterisk. And call it from two cell phones. It is quering very good.
But when i answer the first call, and talk with the first line, the second line is ringing. I dont want the second line ringing.
i can solve it with call-limit=1. But i dont want to use call-limit=1.
my sip.conf user details is below :
[201]
type = friend
username = 201
secret = 201
host = dynamic
context = context-out
nat = yes
qualify = yes
dtmfmode = rfc2833
insecure = port,invite
canreinvite = yes
disallow = all
allow = g729
allow = ulaw
allow = alaw
allow = gsm
;call-limit = 1 //i dont want to use this feature