[asterisk-users] calls get stuck in the asterisk box

[Fons van der Beek]

At the end of the day SIP calles keep stuck in asterisk, is there any 
way to prevent this or debug this?
The sip calls which get stuck all are calles on a  krik IP600v3 dect 
gateway,
I cant tell if they originate of the ip600v3, probably this are calls TO 
the IP600v3


10.0.0.71        240         2c2cfcc47ca  05593/103700  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        238         d4b2f570e90  00105/103150  0x0 (nothing)    
No       Rx: BYE
10.0.0.71        240         5d02b0d503e  06353/102998  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        240         4b303fed159  16797/93872  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        240         181151d9010  16819/93839  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        240         4abf61ec5ee  18318/92482  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        240         43a74c2f08d  19014/91859  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        240         672a3a624b5  19237/91616  0x0 (nothing)    
No       Tx: BYE         Done
10.0.0.71        240         4ede9bb258e  19332/91525  0x0 (nothing)    
No       Tx: BYE         Done

9 active SIP channels
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71
    -- Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.71


the sip.conf for the phones on the IP600v3 all have this settings in 
sip.conf
[239]
type=friend
username = 239
callerid="name" <239>
host = dynamic
secret = 239
context = default
qualify = yes
login = 239
callgroup = 3
pickupgroup = 3
disallow = all
allow = alaw
call-limit = 6

setting of call-limit  to 1 doesn't prevent the above mentioned problem.