Отбой входящего звонка при поднятии трубки

Добрый день. Есть Asterisk + FreePBX

Есть удалённый клиент на SIP-шлюзе Grandstream. При поступлении на этот сип звонка, сам звонок проходит, но при снятии трубки происходит отбой. Исходящие звонки с этого клиента осуществляются нормально.

В дебаге пишет такое:

    <--- SIP read from UDP:83.83.216.181:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK376e33ab;rport=5060
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 102 INVITE
Contact: <sip:103@83.83.216.181:5061>
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.2A 1.0.1.41
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 2450
o=103 8002 8000 IN IP4 83.83.216.181
s=SIP Call
c=IN IP4 83.83.216.181
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
<------------->
--- (23 headers 0 lines) ---
[2015-11-16 22:27:18] WARNING[1800][C-000039c0]: chan_sip.c:23033 handle_response_invite: Received response: "200 OK" from '103' without SDP
list_route: hop: <sip:103@83.83.216.181:5061>
set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to
set_destination: set destination to 83.83.216.181:5061
Transmitting (NAT) to 83.83.216.181:5061:
ACK sip:103@83.83.216.181:5061 SIP/2.0
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK7a852fc1;rport
Max-Forwards: 70
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Contact: <sip:500@109.74.139.75:5060>
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
Content-Length: 0


---
set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to
set_destination: set destination to 83.83.216.181:5061
Reliably Transmitting (NAT) to 83.83.216.181:5061:
BYE sip:103@83.83.216.181:5061 SIP/2.0
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK08fe6c29;rport
Max-Forwards: 70
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 103 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
Scheduling destruction of SIP dialog '0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to
set_destination: set destination to 83.83.216.181:5061
Reliably Transmitting (NAT) to 83.83.216.181:5061:
BYE sip:103@83.83.216.181:5061 SIP/2.0
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK23edd013;rport
Max-Forwards: 70
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 104 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.2(11.20.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:83.83.216.181:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK08fe6c29;rport=5060
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 103 BYE
Contact: <sip:103@83.83.216.181:5061>
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.2A 1.0.1.41
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<--- SIP read from UDP:83.83.216.181:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.74.139.75:5060;branch=z9hG4bK23edd013;rport=5060
From: "500" <sip:500@109.74.139.75>;tag=as3d937f1c
To: <sip:103@83.83.216.181:5061>;tag=1603997517
Call-ID: 0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060
CSeq: 104 BYE
Contact: <sip:103@83.83.216.181:5061>
Supported: replaces, path, timer
User-Agent: Grandstream HT-502 V1.2A 1.0.1.41
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '0d61048a691077cb1472dfe518ce989c@109.74.139.75:5060' Method: INVITE