SIP Channel Stuck on Asterisk

In some case we need to set call-limit=1 in sip config but after a while a SIP channel stuck and we got problem on sending new call to that SIP number.
Logger save this error :
“Call from peer XXX rejected due to usage limit of 1”
Asterisk*CLI> sip show channels 111 1e436b221c0 00102/00000 0x0 (nothing) No (d) Tx: ACK

Even when turn of that SIP, we got that error !
I always try to fixed that but every time this event happen we forced to fix this problem.

Now it seems this problem solved in Asterisk 1.4.26 !
If you have problem like me check new version of asterisk