BAZA IP не проходят исходящие и входящие звонки

Приветствую!

Помогите разобраться в чем проблема, не могу никак настроить в Trixbox Исходящие и Входящие звонки с BAZA Офис 495.

Конфигурация:

Транк:

Trunk - Baza (sip)
Trunk Name: Baza 
Outbound Caller ID: 8495XXXXXXX 
Dialed Number Manipulation Rules: . 
Trunk Name: Baza 
*Peer Details:*
disallow=all 
allow=alaw&ulaw&gsm 
canredirect=no 
canreinvite=no 
host=qwerty.cnt.ru 
insecure=port,invite 
secret=PASSWD 
type=peer 
username=8495XXXXXXX
fromuser=8495XXXXXXX 
fromdomain=qwerty.cnt.ru 
context=from-trunk 

User Context: 8495XXXXXXX

*User Details:* 

secret=PASSWD 
type=user 
context=from-trunk 

Register String: 8495XXXXXXX:PASSWD@qwerty.cnt.ru/8495XXXXXXX

Исходящий маршрут:

Outband Routes 

Route Name: BazaOut
Dial Patterns: 
8916XXXXXXX 
Trunk Sequence 
0 baza

Входящий маршрут:

Description: BazaOut
DID Number: 8495XXXXXXX 
Extensions: <101> Alex

Логи:

trixbox1*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
qwerty.cnt.ru:5060              84956693455        120 Request Sent
1 SIP registrations.

Исходящий звонок на мобильный 8916XXXXXXX:

       -- Called Baza/89161111111
    -- ast_get_srv: SRV lookup for '_sip._UDP.qwerty.cnt.ru' mapped to host sip.qwerty.cnt.ru, port 5060
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 213.85.168.52:5060:
REGISTER sip:qwerty.cnt.ru SIP/2.0
Via: SIP/2.0/UDP 89.XXX.XXX.XXX:5060;branch=z9hG4bK3dcfada6;rport
Max-Forwards: 70
From: <sip:84956693455@qwerty.cnt.ru>;tag=as392d8517
To: <sip:84956693455@qwerty.cnt.ru>
Call-ID: 4152177e200c66e670fbc08503521803@127.0.0.1
CSeq: 195 REGISTER
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Expires: 120
Contact: <sip:84956693455@89.XXX.XXX.XXX>
Event: registration
Content-Length: 0


---
Really destroying SIP dialog '4152177e200c66e670fbc08503521803@127.0.0.1' Method: REGISTER
trixbox1*CLI>
<--- SIP read from UDP://192.168.1.4:53060 --->



<------------->
Scheduling destruction of SIP dialog '4525185d746db5a4367db8686cdbdb79@qwerty.cnt.ru' in 32000 ms (Method: INVITE)
    -- SIP/Baza-00000007 is circuit-busy
Scheduling destruction of SIP dialog '4525185d746db5a4367db8686cdbdb79@qwerty.cnt.ru' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/101-00000006", "s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/101-00000006", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/101-00000006", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing [89161111111@from-internal:5] Macro("SIP/101-00000006", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/101-00000006", "all-circuits-busy-now,noanswer") in new stack
Audio is at 192.168.1.197 port 13088
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
trixbox1*CLI>
<--- Transmitting (no NAT) to 192.168.1.4:53060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.4:53060;branch=z9hG4bK-d8754z-51dfe2673b88bc76-1---d8754z-;received=192.168.1.4;rport=53060
From: "п-п¦п¦п¦я¦п¦п¦"<sip:101@192.168.1.197:5060>;tag=4f0d8891
To: <sip:89161111111@192.168.1.197:5060>;tag=as475dc30c
Call-ID: MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:89161111111@192.168.1.197>
Content-Type: application/sdp
Content-Length: 302

v=0
o=root 1952683458 1952683458 IN IP4 192.168.1.197
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 192.168.1.197
t=0 0
m=audio 13088 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- <SIP/101-00000006> Playing 'all-circuits-busy-now.ulaw' (language 'en')
    -- Executing [s@macro-outisbusy:2] Playback("SIP/101-00000006", "pls-try-call-later,noanswer") in new stack
    -- <SIP/101-00000006> Playing 'pls-try-call-later.ulaw' (language 'en')
    -- Executing [s@macro-outisbusy:3] Macro("SIP/101-00000006", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000006", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000006", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000006", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/101-00000006", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/101-00000006' in macro 'hangupcall'
  == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/101-00000006' in macro 'outisbusy'
  == Spawn extension (from-internal, 89161111111, 5) exited non-zero on 'SIP/101-00000006'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000006", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000006", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000006", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000006", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/101-00000006", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/101-00000006' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000006'
Scheduling destruction of SIP dialog 'MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 192.168.1.4:53060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.4:53060;branch=z9hG4bK-d8754z-51dfe2673b88bc76-1---d8754z-;received=192.168.1.4;rport=53060
From: "п-п¦п¦п¦я¦п¦п¦"<sip:101@192.168.1.197:5060>;tag=4f0d8891
To: <sip:89161111111@192.168.1.197:5060>;tag=as475dc30c
Call-ID: MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
trixbox1*CLI>
<--- SIP read from UDP://192.168.1.4:53060 --->
ACK sip:89161111111@192.168.1.197:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:53060;branch=z9hG4bK-d8754z-51dfe2673b88bc76-1---d8754z-;rport
Max-Forwards: 70
To: <sip:89161111111@192.168.1.197:5060>;tag=as475dc30c
From: "п-п¦п¦п¦я¦п¦п¦"<sip:101@192.168.1.197:5060>;tag=4f0d8891
Call-ID: MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc.
CSeq: 2 ACK
Content-Length: 0

Соответственно всем время выдает сигнал занято. При входящем звонке в соответствии с правилами выставлена переадресация на номер 880308495XXXXXXX (по правилам Базы), однако никакой реакции на входящий звонок нет.

При настройке софтфона напрямую через Baza IP и исходящие и входящие звонки работают нормально.