Регистрация есть - звонки не ходят

Доброго времени суток. Установлен Elastix на VirtualBox.

Есть телефон от МТТ, который через какое-то время перестает работать, НО при этом регистрация есть. Стоит сделать sip reload и опять все в порядке.

Вот дебаг когда я делаю входящий звонок на номер МТТ:

CLI> sip show  registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
voip.mtt.ru:5060                        N      888888888888       105 Registered           Wed, 14 May 2014 23:10:27


<--- SIP read from UDP:80.75.132.66:5060 --->
INVITE sip:888888888888@86.110.174.46:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NzUyNDc1MmZjNjYwNTk1YzlmNzU4Y2Y4MjhkYzg2Nzc.--d934b7642fcbd749;rport
Via: SIP/2.0/UDP 80.75.132.66:5072;branch=z9hG4bK-xzflk2rjcffcmjzj;rport=5072
Max-Forwards: 69
Record-Route: <sip:80.75.132.66:5060;lr;transport=UDP>
Contact: "Anonymous"<sip:80.75.132.66:5072>
To: <sip:888888888888@80.75.132.66>
From: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;tag=jl6qqylktufi4ywt.o
Call-ID: 1400-87504-260020
CSeq: 381 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
P-Asserted-Identity: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>
Remote-Party-ID: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;party=calling
h323-conf-id: 2995968909-972723075-3869309460-2606746716
Portasip-3264-action: offer 1
cisco-GUID: 2995968909-972723075-3869309460-2606746716
Content-Length: 291

v=0
o=Sippy 1583774704421374595 0 IN IP4 80.75.132.66
s=SMG SIP session
t=0 0
m=audio 51262 RTP/AVP 0 8 18 101
c=IN IP4 80.75.132.66
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (20 headers 13 lines) ---
Sending to 80.75.132.66:5060 (NAT)
Using INVITE request as basis request - 1400-87504-260020
No matching peer for '+7918xxxxxxx' from '80.75.132.66:5060'

<--- Reliably Transmitting (NAT) to 80.75.132.66:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NzUyNDc1MmZjNjYwNTk1YzlmNzU4Y2Y4MjhkYzg2Nzc.--d934b7642fcbd749;received=80.75.132.66;rport=5060
Via: SIP/2.0/UDP 80.75.132.66:5072;branch=z9hG4bK-xzflk2rjcffcmjzj;rport=5072
From: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;tag=jl6qqylktufi4ywt.o
To: <sip:888888888888@80.75.132.66>;tag=as2f35a509
Call-ID: 1400-87504-260020
CSeq: 381 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55b23950"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1400-87504-260020' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog '1400-87504-260020' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:80.75.132.66:5060 --->
ACK sip:888888888888@86.110.174.46:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NzUyNDc1MmZjNjYwNTk1YzlmNzU4Y2Y4MjhkYzg2Nzc.--d934b7642fcbd749;rport
Max-Forwards: 70
To: <sip:888888888888@80.75.132.66>;tag=as2f35a509
From: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;tag=jl6qqylktufi4ywt.o
Call-ID: 1400-87504-260020
CSeq: 381 ACK
Content-Length: 0

<------------->

делаю sip reload и снова делаю входящий звонок на номер МТТ:

*CLI> sip show  registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
voip.mtt.ru:5060                        N      888888888888       105 Registered           Wed, 14 May 2014 23:33:37

<--- SIP read from UDP:80.75.132.66:5060 --->
INVITE sip:888888888888@86.110.174.46:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NTY5NmFhODYwNDVjNDFlZjBmNTQ1M2NmYWRjMmJjMjA.--bc0db2650e96b24e;rport
Via: SIP/2.0/UDP 80.75.132.66:5071;branch=z9hG4bK-76d76u6zrywngrzc;rport=5071
Max-Forwards: 69
Record-Route: <sip:80.75.132.66:5060;lr;transport=UDP>
Contact: "Anonymous"<sip:80.75.132.66:5071>
To: <sip:888888888888@80.75.132.66>
From: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;tag=juzhsmysauuywol6.o
Call-ID: 1400-88696-954501
CSeq: 183 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
P-Asserted-Identity: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>
Remote-Party-ID: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;party=calling
h323-conf-id: 2302055573-1697763932-2249226389-2620900369
Portasip-3264-action: offer 1
cisco-GUID: 2302055573-1697763932-2249226389-2620900369
Content-Length: 291

v=0
o=Sippy 2659277483483481539 0 IN IP4 80.75.132.66
s=SMG SIP session
t=0 0
m=audio 35552 RTP/AVP 0 8 18 101
c=IN IP4 80.75.132.66
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (20 headers 13 lines) ---
Sending to 80.75.132.66:5060 (NAT)
Using INVITE request as basis request - 1400-88696-954501
Found peer 'MTT' for '+7918xxxxxxx' from 80.75.132.66:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.75.132.66:35552
Looking for 888888888888 in from-trunk-sip-MTT (domain 86.110.174.46)
list_route: hop: <sip:80.75.132.66:5060;lr;transport=UDP>

<--- Transmitting (NAT) to 80.75.132.66:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 80.75.132.66:5060;branch=z9hG4bK-524287-1-NTY5NmFhODYwNDVjNDFlZjBmNTQ1M2NmYWRjMmJjMjA.--bc0db2650e96b24e;received=80.75.132.66;rport=5060
Via: SIP/2.0/UDP 80.75.132.66:5071;branch=z9hG4bK-76d76u6zrywngrzc;rport=5071
Record-Route: <sip:80.75.132.66:5060;lr;transport=UDP>
From: 7918xxxxxxx <sip:+7918xxxxxxx@80.75.132.66>;tag=juzhsmysauuywol6.o
To: <sip:888888888888@80.75.132.66>
Call-ID: 1400-88696-954501
CSeq: 183 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:888888888888@192.168.0.200:5060>
Content-Length: 0


<------------>
    -- Executing [888888888888@from-trunk-sip-MTT:1] Set("SIP/MTT-0000000a", "GROUP()=OUT_3") in new stack
    -- Executing [888888888888@from-trunk-sip-MTT:2] Goto("SIP/MTT-0000000a", "from-trunk,888888888888,1") in new stack

Вот настройки пира МТТ

type=peer
defaultuser=888888888888
fromuser=888888888888
secret=<pass>
nat=yes
qualify=yes
insecure=invite,port
host=voip.mtt.ru
fromdomain=voip.mtt.ru
dtmfmode=rfc2833
directmedia=no
disallow=all
allow=alaw,ulaw,g729

regyster => 888888888888:<pass>@voip.mtt.ru/888888888888

При чем с исходящей связью такая же фигня... он не находит пира MTT чтоб выполнить dial и выдает что все линии заняты... НО после sip reload какое-то время работает нормально...