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?