Hello everyone, I have a question regard this command:
I have set a channel SIP “200”, from a patton bri with a call-limit 2 (for 1 line ISDN)
then i have another voip number SIP/0XXXXXXXXX7
In my dialplan the extension is placed in this mode:
exten => _XXXXXXX.,1,ChanIsAvail(SIP/200&SIP/0XXXXXXXXX7|j)
exten => _XXXXXXX.,n,NoOp(availstatus: ${AVAILSTATUS})
exten => _XXXXXXX.,n,Dial(${CUT(AVAILCHAN,1)}/${EXTEN},60,TtRr)
exten => _XXXXXXX.,n,Hangup
exten => _XXXXXXX.,102,Congestion()
exten => _XXXXXXX.,103,Hangup
For a first two call all function ok, the call are directed at SIP/200, but a third call asterisk output this:
chan_sip.c:3192 update_call_counter: Call to peer ‘200’ rejected due to usage limit of 2
– Couldn’t call 200/XXXXXXX…
The phone are in same context.
With option “s” after a first call in channel 200 the call are redirected al SIP/0XXXXXXXXX7!!! 
Anyone had ideas?
The asterisk is the latest SVN-branch-1.4-r81455, the same problem was in previous version.
Thanx for response, bye! 
I have find this mythic command:
RetryDial(announce|sleep|loops|Technology/resource[&Technology2/resource2…][|[timeout][|[options][|URL]]])
I hope to help someone 8) .
Bye!