C595 IP трансфер между трубками одной базы.

При трансфере без консультации (# по умолчанию во features.conf) между трубками одной базы происходит какая-то чепуха, которая завершается -- Got SIP response 486 "Busy Here" back from ... вот лог этой чепухи :

SIP/1003-00000330
srv-ast00*CLI>     -- Blind transferring SIP/1018-0000032f to '1015' (context e-33) priority 1
    -- Stopped music on hold on SIP/1018-0000032f
srv-ast00*CLI> Scheduling destruction of SIP dialog '752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:1003@10.10.10.115:5060> for address/port to send to
set_destination: set destination to 10.10.10.115:5060
Reliably Transmitting (NAT) to 10.10.10.115:5060:
BYE sip:1003@10.10.10.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK7a289665;rport
Max-Forwards: 70
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as5c426352
To: <sip:1003@10.10.10.115:5060>;tag=3440354342
Call-ID: 752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060
CSeq: 103 BYE
User-Agent: Testing PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
srv-ast00*CLI>     -- Executing [1015@e-33:1] Answer("SIP/1018-0000032f", "") in new stack
    -- Executing [1015@e-33:2] ExecIf("SIP/1018-0000032f", "1?Dial(SIP/1015,,tTrR") in new stack
srv-ast00*CLI>   == Using SIP RTP CoS mark 5
Audio is at 10874
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.10.115:5060:
INVITE sip:1015@10.10.10.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK5915abac;rport
Max-Forwards: 70
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as6544bf65
To: <sip:1015@10.10.10.115:5060>
Contact: <sip:1018@10.10.254.8:5060>
Call-ID: 4c07df101a01e74914a022391447f072@10.10.254.8:5060
CSeq: 102 INVITE
User-Agent: Testing PBX
Date: Fri, 23 Mar 2012 14:20:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 2035726969 2035726969 IN IP4 10.10.254.8
s=Asterisk PBX 1.8.10.0-1digium1~squeeze
c=IN IP4 10.10.254.8
t=0 0
m=audio 10874 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/1015
srv-ast00*CLI> Retransmitting #1 (NAT) to 10.10.10.115:5060:
BYE sip:1003@10.10.10.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK7a289665;rport
Max-Forwards: 70
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as5c426352
To: <sip:1003@10.10.10.115:5060>;tag=3440354342
Call-ID: 752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060
CSeq: 103 BYE
User-Agent: Testing PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
srv-ast00*CLI> Retransmitting #1 (NAT) to 10.10.10.115:5060:
INVITE sip:1015@10.10.10.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK5915abac;rport
Max-Forwards: 70
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as6544bf65
To: <sip:1015@10.10.10.115:5060>
Contact: <sip:1018@10.10.254.8:5060>
Call-ID: 4c07df101a01e74914a022391447f072@10.10.254.8:5060
CSeq: 102 INVITE
User-Agent: Testing PBX
Date: Fri, 23 Mar 2012 14:20:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 2035726969 2035726969 IN IP4 10.10.254.8
s=Asterisk PBX 1.8.10.0-1digium1~squeeze
c=IN IP4 10.10.254.8
t=0 0
m=audio 10874 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
srv-ast00*CLI> Retransmitting #2 (NAT) to 10.10.10.115:5060:
BYE sip:1003@10.10.10.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK7a289665;rport
Max-Forwards: 70
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as5c426352
To: <sip:1003@10.10.10.115:5060>;tag=3440354342
Call-ID: 752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060
CSeq: 103 BYE
User-Agent: Testing PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
srv-ast00*CLI> 
<--- SIP read from UDP:10.10.10.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK5915abac;rport=5060
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as6544bf65
To: <sip:1015@10.10.10.115:5060>;tag=2081908448
Call-ID: 4c07df101a01e74914a022391447f072@10.10.254.8:5060
CSeq: 102 INVITE
Contact: <sip:1015@10.10.10.115:5060>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
srv-ast00*CLI> 
<--- SIP read from UDP:10.10.10.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK7a289665;rport=5060
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as5c426352
To: <sip:1003@10.10.10.115:5060>;tag=3440354342
Call-ID: 752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060
CSeq: 103 BYE
Contact: <sip:1003@10.10.10.115:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060' Method: INVITE
srv-ast00*CLI> 
<--- SIP read from UDP:10.10.10.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK7a289665;rport=5060
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as5c426352
To: <sip:1003@10.10.10.115:5060>;tag=3440354342
Call-ID: 752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060
CSeq: 103 BYE
Contact: <sip:1003@10.10.10.115:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
srv-ast00*CLI> 
<--- SIP read from UDP:10.10.10.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK5915abac;rport=5060
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as6544bf65
To: <sip:1015@10.10.10.115:5060>;tag=2081908448
Call-ID: 4c07df101a01e74914a022391447f072@10.10.254.8:5060
CSeq: 102 INVITE
Contact: <sip:1015@10.10.10.115:5060>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
srv-ast00*CLI> 
<--- SIP read from UDP:10.10.10.115:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK7a289665;rport=5060
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as5c426352
To: <sip:1003@10.10.10.115:5060>;tag=3440354342
Call-ID: 752aa8f121fb33655774fa5c1e338d4a@10.10.254.8:5060
CSeq: 103 BYE
Contact: <sip:1003@10.10.10.115:5060>
Supported: replaces
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
srv-ast00*CLI> 
<--- SIP read from UDP:10.10.10.115:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK5915abac;rport=5060
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as6544bf65
To: <sip:1015@10.10.10.115:5060>;tag=2081908448
Call-ID: 4c07df101a01e74914a022391447f072@10.10.254.8:5060
CSeq: 102 INVITE
Contact: <sip:1015@10.10.10.115:5060>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 486 "Busy Here" back from 10.10.10.115:5060
Transmitting (NAT) to 10.10.10.115:5060:
ACK sip:1015@10.10.10.115:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.254.8:5060;branch=z9hG4bK5915abac;rport
Max-Forwards: 70
From: "FAX Line" <sip:1018@10.10.254.8>;tag=as6544bf65
To: <sip:1015@10.10.10.115:5060>;tag=2081908448
Contact: <sip:1018@10.10.254.8:5060>
Call-ID: 4c07df101a01e74914a022391447f072@10.10.254.8:5060
CSeq: 102 ACK
User-Agent: Testing PBX
Content-Length: 0


---
srv-ast00*CLI>     -- SIP/1015-00000331 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
srv-ast00*CLI>     -- Executing [1015@e-33:3] NoOp("SIP/1018-0000032f", "NOT_INUSE") in new stack
srv-ast00*CLI>     -- Executing [1015@e-33:4] Busy("SIP/1018-0000032f", "5") in new stack
srv-ast00*CLI> Really destroying SIP dialog '4c07df101a01e74914a022391447f072@10.10.254.8:5060' Method: INVITE
srv-ast00*CLI>   == Spawn extension (e-33, 1015, 4) exited non-zero on 'SIP/1018-0000032f'

dialplan

exten = _1XXX,1,Answer()
exten = _1XXX,n,ExecIf($[ ${DEVICE_STATE(SIP/${EXTEN})} != BUSY ]?Dial(SIP/${EXTEN},,tTrR)
exten = _1XXX,n,NoOp(${DEVICE_STATE(SIP/${EXTEN})})
exten = _1XXX,n,Busy(5)
exten = _1XXX,n,Hangup()