входящая маршрутизация sip транк [закрыт]

Привет. При входящем вызове у меня в логах появляется

[Jun 14 12:10:10] VERBOSE[22998] config.c:   == Parsing '/etc/asterisk/logger.conf': [Jun 14 12:10:10] VERBOSE[22998] config.c:   == Found
[Jun 14 12:10:10] VERBOSE[22998] logger.c:  Asterisk Queue Logger restarted
[Jun 14 12:10:18] VERBOSE[21917] netsock2.c:   == Using SIP RTP TOS bits 184
[Jun 14 12:10:18] VERBOSE[21917] netsock2.c:   == Using SIP RTP CoS mark 5
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:1] GotoIf("SIP/217.15.55.203:5061-0000003b", "0?checklang:noanonymous") in new stack
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Goto (from-sip-external,s,5)
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:5] Set("SIP/217.15.55.203:5061-0000003b", "TIMEOUT(absolute)=15") in new stack
[Jun 14 12:10:18] VERBOSE[22999] func_timeout.c: Channel will hangup at 2012-06-14 12:10:33.975 MSD.
[Jun 14 12:10:18] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:6] Answer("SIP/217.15.55.203:5061-0000003b", "") in new stack
[Jun 14 12:10:19] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:7] Wait("SIP/217.15.55.203:5061-0000003b", "2") in new stack
[Jun 14 12:10:21] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:8] Playback("SIP/217.15.55.203:5061-0000003b", "ss-noservice") in new stack
[Jun 14 12:10:21] VERBOSE[22999] file.c:     -- <SIP/217.15.55.203:5061-0000003b> Playing 'ss-noservice.gsm' (language 'en')
[Jun 14 12:10:26] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:9] PlayTones("SIP/217.15.55.203:5061-0000003b", "congestion") in new stack
[Jun 14 12:10:26] VERBOSE[22999] pbx.c:     -- Executing [s@from-sip-external:10] Congestion("SIP/217.15.55.203:5061-0000003b", "5") in new stack
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:   == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/217.15.55.203:5061-0000003b'
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:     -- Executing [h@from-sip-external:1] Hangup("SIP/217.15.55.203:5061-0000003b", "") in new stack
[Jun 14 12:10:29] VERBOSE[22999] pbx.c:   == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/217.15.55.203:5061-0000003b'
[Jun 14 12:10:29] VERBOSE[21917] dnsmgr.c:        > doing dnsmgr_lookup for 'sipnet.ru'
[Jun 14 12:10:32] VERBOSE[22998] asterisk.c:     -- Remote UNIX connection disconnected

Что за IP такой 217.15.55.203, без понятия. Точно не наш. Транк настроен как обычно

dtmfmode=rfc2833
type=friend
host=217.15.55.200
fromuser=6XXX
username=495YYYYYYY
secret=Secr
insecure=port,invite
conext=contex-internal
nat=no
allow=all

Строка регистрации 6XXX:secr:495YYYYYYY@217.15.55.200 Входящая маршрутизация на внутренний номер.

И еще замечена странная штука, при звонке на наш номер с мобильного, когда астериск начинает говорить что номер не существует, на экране мобильного номер меняется на другой 4959269211, т.е не на тот на который звоним. FreePBX 2.8, Centos 5.7 Спасибо.