Астериск падает

Рабочий FreePBX 12.0.76.2, Asterisk 13.0.1 Все работает - исходящие, входящие, все слышно. Поднят транк к lync-у. С астерисковских экстэншэнов нормально звонится на Линк. А вот при звонке с линка на астериск - астериск падает. Астериск тупо перезагружается (видно по длительности жизни процесса). Все активные звонки обрываются. В консоли видно:

Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

На стороне Линка в eventviewer'e нет никаких ошибок. Звонок сбрасывается через 2 секунды после поднятия трубки. Пользователь успевает услышать только первое "Алло". После этого ничего не слышно. Последнее что видно в дебаг логах это:

<------------->
--- (10 headers 0 lines) ---
Sending to lync-IP:55917 (no NAT)
Looking for s in from-sip-external (domain asterisk-IP)

<--- Transmitting (no NAT) to lync-IP:55917 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP lync-IP:55917;branch=z9hG4bK8adb6d9e;received=lync-IP
From: <sip:lync.org.monos.mn:5060;transport=Tcp;ms-opaque=d7994b2befb4e3e4>;epid=A81DB9A5BE;tag=1f53beeb4b
To: <sip:asterisk-IP>;tag=as47de0184
Call-ID: fed6c7ca3d074d95b9b82286b83fbfef
CSeq: 29385 OPTIONS
Server: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:asterisk-IP:5060;transport=TCP>
Accept: application/sdp
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3ed4d2b4595cb3d6440fc76842eb16ea@asterisk-IP:5060' Method: OPTIONS

<--- SIP read from TCP:lync-IP:55920 --->
INVITE sip:203@asterisk-IP;user=phone SIP/2.0
FROM: "Пользователь"<sip:+19001@domain.com;user=phone>;epid=A81DB9A5BE;tag=3c8343e62e
TO: <sip:203@asterisk-IP;user=phone>
CSEQ: 29386 INVITE
CALL-ID: 565e42a6-81d0-4502-b739-b2429190ed9b
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP lync-IP:55920;branch=z9hG4bK58afb89e
CONTACT: <sip:lync.org.monos.mn:5060;transport=Tcp;maddr=lync-IP;ms-opaque=d7994b2befb4e3e4>
CONTENT-LENGTH: 330
SUPPORTED: 100rel
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE

v=0
o=- 35 1 IN IP4 lync-IP
s=session
c=IN IP4 lync-IP
b=CT:1000
t=0 0
m=audio 54832 RTP/AVP 97 101 13 0 8
c=IN IP4 lync-IP
a=rtcp:54833
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (14 headers 18 lines) ---
Sending to lync-IP:55920 (no NAT)
Sending to lync-IP:55920 (no NAT)
Using INVITE request as basis request - 565e42a6-81d0-4502-b739-b2429190ed9b
Found peer 'Lync' for '+19001' from lync-IP:55920
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 13
Found RTP audio format 0
Found RTP audio format 8
Found unknown media description format RED for ID 97
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 13
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|h264|mpeg4), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port lync-IP:54832
Peer doesn't provide video
Looking for 203 in from-internal (domain asterisk-IP)
sip_route_dump: route/path hop: <sip:lync.org.monos.mn:5060;transport=Tcp;maddr=lync-IP;ms-opaque=d7994b2befb4e3e4>

<--- Transmitting (no NAT) to lync-IP:55920 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP lync-IP:55920;branch=z9hG4bK58afb89e;received=lync-IP
From: "Пользователь"<sip:+19001@domain.com;user=phone>;epid=A81DB9A5BE;tag=3c8343e62e
To: <sip:203@asterisk-IP;user=phone>
Call-ID: 565e42a6-81d0-4502-b739-b2429190ed9b
CSeq: 29386 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:203@asterisk-IP:5060;transport=TCP>
Content-Length: 0


<------------>
Audio is at 15776
Video is at asterisk-IP:15862
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to exten-IP:57519:
INVITE sip:203@exten-IP:57519;rinstance=326fead00d49a771 SIP/2.0
Via: SIP/2.0/UDP asterisk-IP:5060;branch=z9hG4bK1b847fc6
Max-Forwards: 70
From: "Пользователь" <sip:+19001@asterisk-IP>;tag=as2071f796
To: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>
Contact: <sip:+19001@asterisk-IP:5060>
Call-ID: 0fd24ab73500c75851d4c9125799a92f@asterisk-IP:5060
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Date: Thu, 26 Jan 2017 05:11:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Пользователь" <sip:+19001@asterisk-IP>
Content-Type: application/sdp
Content-Length: 359

v=0
o=root 592469055 592469055 IN IP4 asterisk-IP
s=Asterisk PBX 13.0.1
c=IN IP4 asterisk-IP
b=CT:384
t=0 0
m=audio 15776 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=maxptime:150
a=sendrecv
m=video 15862 RTP/AVP 99 104
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv

---

<--- Transmitting (no NAT) to lync-IP:55920 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP lync-IP:55920;branch=z9hG4bK58afb89e;received=lync-IP
From: "Пользователь"<sip:+19001@domain.com;user=phone>;epid=A81DB9A5BE;tag=3c8343e62e
To: <sip:203@asterisk-IP;user=phone>;tag=as5a78464a
Call-ID: 565e42a6-81d0-4502-b739-b2429190ed9b
CSeq: 29386 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:203@asterisk-IP:5060;transport=TCP>
Content-Length: 0


<------------>
Audio is at 13712
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to lync-IP:55920 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP lync-IP:55920;branch=z9hG4bK58afb89e;received=lync-IP
From: "Пользователь"<sip:+19001@domain.com;user=phone>;epid=A81DB9A5BE;tag=3c8343e62e
To: <sip:203@asterisk-IP;user=phone>;tag=as5a78464a
Call-ID: 565e42a6-81d0-4502-b739-b2429190ed9b
CSeq: 29386 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:203@asterisk-IP:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 927905744 927905744 IN IP4 asterisk-IP
s=Asterisk PBX 13.0.1
c=IN IP4 asterisk-IP
t=0 0
m=audio 13712 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=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:exten-IP:57519 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk-IP:5060;branch=z9hG4bK1b847fc6
To: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>
From: "Пользователь" <sip:+19001@asterisk-IP>;tag=as2071f796
Call-ID: 0fd24ab73500c75851d4c9125799a92f@asterisk-IP:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:exten-IP:57519 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP asterisk-IP:5060;branch=z9hG4bK1b847fc6
Contact: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>
To: "203"<sip:203@exten-IP:57519;rinstance=326fead00d49a771>;tag=109f8b31
From: "Пользователь" <sip:+19001@asterisk-IP>;tag=as2071f796
Call-ID: 0fd24ab73500c75851d4c9125799a92f@asterisk-IP:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 4.9.5.1 stamp 81564
Allow-Events: talk, hold
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>
Reliably Transmitting (NAT) to 10.0.16.51:5060:
OPTIONS sip:2@10.0.16.51:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk-IP:5060;branch=z9hG4bK5462f1ed;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@asterisk-IP>;tag=as3d56082a
To: <sip:2@10.0.16.51:5060>
Contact: <sip:Unknown@asterisk-IP:5060>
Call-ID: 326f7aea13cc4d7a59d483bf09341019@asterisk-IP:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Date: Thu, 26 Jan 2017 05:11:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:exten-IP:57519 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP asterisk-IP:5060;branch=z9hG4bK1b847fc6
Contact: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>
To: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>;tag=109f8b31
From: "Пользователь" <sip:+19001@asterisk-IP>;tag=as2071f796
Call-ID: 0fd24ab73500c75851d4c9125799a92f@asterisk-IP:5060
CSeq: 102 INVITE
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.9.5.1 stamp 81564
Content-Length: 228

v=0
o=- 2756743031 3 IN IP4 exten-IP
s=X-Lite release 4.9.5.1 stamp 81564
c=IN IP4 exten-IP
t=0 0
m=audio 54830 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 0 RTP/AVP 99 104
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h264|mpeg4), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port exten-IP:54830
Peer doesn't provide video
sip_route_dump: route/path hop: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>
set_destination: Parsing <sip:203@exten-IP:57519;rinstance=326fead00d49a771> for address/port to send to
set_destination: set destination to exten-IP:57519
Transmitting (no NAT) to exten-IP:57519:
ACK sip:203@exten-IP:57519;rinstance=326fead00d49a771 SIP/2.0
Via: SIP/2.0/UDP asterisk-IP:5060;branch=z9hG4bK13099748
Max-Forwards: 70
From: "Пользователь" <sip:+19001@asterisk-IP>;tag=as2071f796
To: <sip:203@exten-IP:57519;rinstance=326fead00d49a771>;tag=109f8b31
Contact: <sip:+19001@asterisk-IP:5060>
Call-ID: 0fd24ab73500c75851d4c9125799a92f@asterisk-IP:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Content-Length: 0


---
Audio is at 13712
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to lync-IP:55920 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP lync-IP:55920;branch=z9hG4bK58afb89e;received=lync-IP
From: "Пользователь"<sip:+19001@domain.com;user=phone>;epid=A81DB9A5BE;tag=3c8343e62e
To: <sip:203@asterisk-IP;user=phone>;tag=as5a78464a
Call-ID: 565e42a6-81d0-4502-b739-b2429190ed9b
CSeq: 29386 INVITE
Server: FPBX-AsteriskNOW-12.0.76.2(13.0.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:203@asterisk-IP:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 927905744 927905744 IN IP4 asterisk-IP
s=Asterisk PBX 13.0.1
c=IN IP4 asterisk-IP
t=0 0
m=audio 13712 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=maxptime:150
a=sendrecv

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

<--- SIP read from TCP:lync-IP:55920 --->
ACK sip:203@asterisk-IP:5060;transport=TCP SIP/2.0
FROM: <sip:+19001@domain.com;user=phone>;epid=A81DB9A5BE;tag=3c8343e62e
TO: <sip:203@asterisk-IP;user=phone>;tag=as5a78464a
CSEQ: 29386 ACK
CALL-ID: 565e42a6-81d0-4502-b739-b2429190ed9b
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP lync-IP:55920;branch=z9hG4bKd913fb8
CONTENT-LENGTH: 0
USER-AGENT: RTCC/5.0.0.0 MediationServer

Как узанть причину падения астериска? Почему ему не нравится входящий звонок с Линка?