We’re working with asterisk 1.2.0, hardware sip phone (Thomson st2020 by example), and sip soft like “x-ten” or “snom 360” (who can both manage many lines). We are also using the queue with round-robin strategy and dynamic members.
When the hardware phone is busy, the call is redirected to another phone within the queue members (I think it’s normal). But when using a sip soft, it always receive the calls on the others lines even if he is busy and other members are free… Parameters like call-limit or incominglimit have no effects
How could we arrange this problem ? We want to use a sip soft and have the possibility to do attended transfer
Thanks for the help. It’s very important