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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) --- listroute: hop: <sip:103@83.83.216.181:5061> setdestination: 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

setdestination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to setdestination: 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) setdestination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to setdestination: 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

Отбой входящего звонка при поднятии трубки

Добрый день. Есть Asterisk + FreePBX

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) --- ~~listroute:~~ [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: 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

setdestination: 0 --- set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to ~~set~~destination: 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

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) ~~setdestination:~~ set_destination: Parsing <sip:103@83.83.216.181:5061> for address/port to send to ~~set~~destination: 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

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

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

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

INVITE