[Asterisk-Users] Problem using Queue and Sip Soft

[Julien SIRBU]

Hi,

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 (see the log below...)

##
Dec 19 14:37:17 ERROR[31234]: chan_sip.c:2238 update_call_counter: Call 
to user '2788' rejected due to usage limit of 1
    -- Couldn't call SIP/2788
    -- Called SIP/2788
    -- SIP/2788-7442 is ringing
##

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