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
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