Asterisk 12.4.0 команда channel redirect разрывает оба соединения, звонок не переводится

Добрый день! Пытаюсь командой из CLI сделать blind transfer звонка. Звонок поступает в очередь и падает на агента, соединение установлено, разговор идёт:

asterisk*CLI> core show channels Channel Location State Application(Data) SIP/2001-0000000a (None) Up AppQueue((Outgoing Line)) SIP/2003-00000009 (None) Up Queue(CRM,tc) 2 active channels 1 active call 5 calls processed После этого пытаюсь сделать: asterisk*CLI> channel redirect SIP/2003-00000009 2002 Channel 'SIP/2003-00000009' successfully redirected to 2002 -- Channel SIP/2003-00000009 left 'simple_bridge' basic-bridge <8dcb78a3-f502-46c2-abfe-254a33bcc71e> -- Channel SIP/2001-0000000a left 'simple_bridge' basic-bridge <8dcb78a3-f502-46c2-abfe-254a33bcc71e> -- Auto fallthrough, channel 'SIP/2003-00000009' status is 'UNKNOWN' 

SIP конфиг:

[2001] type=friend regexten=2001 secret=2001 callerid="2001" <2001> host=dynamic directmedia=no insecure=invite disallow=all allow=ulaw registertrying=yes context=internal qualify=100 qualifyfreq=10 call-limit=10 canreinvite=yes 

SIP debug:

asterisk*CLI> channel redirect SIP/2003-0000000b 2002 Channel 'SIP/2003-0000000b' successfully redirected to 2002 -- Channel SIP/2003-0000000b left 'simple_bridge' basic-bridge <0b988562-9eed-4df3-9fe6-34bdfd5a5a7e> -- Auto fallthrough, channel 'SIP/2003-0000000b' status is 'UNKNOWN' Scheduling destruction of SIP dialog '4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain' in 6400 ms (Method: ACK) set_destination: Parsing <sip:2003@10.0.8.171:5060> for address/port to send to set_destination: set destination to 10.0.8.171:5060 -- Channel SIP/2001-0000000c left 'simple_bridge' basic-bridge <0b988562-9eed-4df3-9fe6-34bdfd5a5a7e> Reliably Transmitting (no NAT) to 10.0.8.171:5060: BYE sip:2003@10.0.8.171:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK486ae89d;rport Max-Forwards: 70 From: <sip:9150@10.0.15.40>;tag=as228803e1 To: "Andrey Lyarskiy" <sip:2003@10.0.15.40>;tag=4a32f4f9-6215-e411-9e6d-f0def11a319a Call-ID: 4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain CSeq: 102 BYE User-Agent: Asterisk PBX 12.4.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060' in 6400 ms (Method: INVITE) set_destination: Parsing <sip:2001@10.0.17.128:5060> for address/port to send to set_destination: set destination to 10.0.17.128:5060 Reliably Transmitting (no NAT) to 10.0.17.128:5060: BYE sip:2001@10.0.17.128:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK50d6b5af Max-Forwards: 70 From: "2003" <sip:2003@10.0.15.40>;tag=as49681c37 To: <sip:2001@10.0.17.128:5060>;tag=2784575879 Call-ID: 71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060 CSeq: 103 BYE User-Agent: Asterisk PBX 12.4.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:10.0.8.171:5060 ---> SIP/2.0 200 OK CSeq: 102 BYE Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK486ae89d;rport From: <sip:9150@10.0.15.40>;tag=as228803e1 Call-ID: 4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain To: "Andrey Lyarskiy" <sip:2003@10.0.15.40>;tag=4a32f4f9-6215-e411-9e6d-f0def11a319a Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '4c35f4f9-6215-e411-9e6d-f0def11a319a@localhost.localdomain' Method: ACK <--- SIP read from UDP:10.0.17.128:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK50d6b5af From: "2003" <sip:2003@10.0.15.40>;tag=as49681c37 To: <sip:2001@10.0.17.128:5060>;tag=2784575879 Call-ID: 71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060 CSeq: 103 BYE Contact: <sip:2001@10.0.17.128:5060> Supported: replaces User-Agent: C610A IP/42.076.00.000.000 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '71bd9e5b2ab9af95239b66da72131b61@10.0.15.40:5060' Method: INVITE Reliably Transmitting (no NAT) to 10.0.17.129:5060: OPTIONS sip:2002@10.0.17.129:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK0c66b3f1 Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.0.15.40>;tag=as6a00bc56 To: <sip:2002@10.0.17.129:5060> Contact: <sip:asterisk@10.0.15.40:5060> Call-ID: 087025ab1af750f240e0e5247106336e@10.0.15.40:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 12.4.0 Date: Tue, 29 Jul 2014 07:52:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:10.0.17.129:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.15.40:5060;branch=z9hG4bK0c66b3f1 From: "asterisk" <sip:asterisk@10.0.15.40>;tag=as6a00bc56 To: <sip:2002@10.0.17.129:5060>;tag=ar7a11cb47 Call-ID: 087025ab1af750f240e0e5247106336e@10.0.15.40:5060 CSeq: 102 OPTIONS Supported: replaces User-Agent: C610A IP/42.076.00.000.000 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag Accept-Encoding: identity Accept-Language: en Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '087025ab1af750f240e0e5247106336e@10.0.15.40:5060' Method: OPTIONS asterisk*CLI> sip set debug off SIP Debugging Disabled 

Подскажите, что я делаю не так?