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


 
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fons.vanderbeek at 84-...
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PostPosted: Wed Jan 30, 2008 3:18 pm    Post subject: [asterisk-users] calls get stuck in the asterisk box Reply with quote

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.
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