Элтекс

Есть чудный скрипт для исправления callerid при переводе звонка для asterisk

exten => s,n,NoOp( == add blind transfer number == )
exten => s,n,NoOp( == in CallerID == ${CALLERID(name)} == )
exten => s,n,ExecIf($[ ${LEN(${BLINDTRANSFER})} > 0 ]?Set(TRANSFERUSER=${BLINDTRANSFER}))
exten => s,n,ExecIf($[ ${LEN(${TRANSFERERNAME})} > 0 ]?Set(TRANSFERUSER=${TRANSFERERNAME}) )
exten => s,n,ExecIf($[ ${LEN(${TRANSFERUSER})} > 0 ]?Set(TRANSFERUSER=${CUT(TRANSFERUSER,-,1)}) )
exten => s,n,ExecIf($[ ${LEN(${TRANSFERUSER})} > 0 ]?Set(TRANSFERUSER=${CUT(TRANSFERUSER,\/,2)}) )
exten => s,n,ExecIf($[ "foo${TRANSFERUSER}" = "foo" ]?Set(__CID=${CALLERID(number)}) )
exten => s,n,ExecIf($[ ${LEN(${TRANSFERUSER})} > 0 ]?Set(CALLERID(number)=${CID}-${TRANSFERUSER}) )
exten => s,n,ExecIf($[ "foo${TRANSFERUSER}" != "foo" ]?Set(CALLERID(name)="ORIGINAL:${CID} TRANSFER:${TRANSFERUSER}") )
exten => s,n,ExecIf($[ ${LEN(${BLINDTRANSFER})} > 0 ]?NoOp( == New CallerID == ${CALLERID(name)} == )
exten => s,n,NoOp( == end blind transfer == )
;

При его использования получаю ошибку Bad From Header
выложу с подробностями
перевод звонка не состоялся

== Using SIP RTP CoS mark 5
Audio is at 15050
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100012 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.68:5060:
INVITE sip:101@192.168.0.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK4cc3e71c
Max-Forwards: 70
From: ""ORIGINAL:109 TRANSFER:102"" <sip:109-102@192.168.0.66>;tag=as0ad1cdc4
To: <sip:101@192.168.0.68:5060>
Contact: <sip:109-102@192.168.0.66:5060>
Call-ID: 5b1d76944763dd51065c57d4425c277e@192.168.0.66:5060
CSeq: 102 INVITE
User-Agent: VoiceOne PBX
Date: Fri, 11 Oct 2013 11:58:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Alert-Info: <Bellcore-dr2>;info=<Bellcore-dr2>;x-line-id=0
P-Asserted-Identity: ""ORIGINAL:109 TRANSFER:102"" <sip:109-102@192.168.0.66>
Content-Type: application/sdp
Content-Length: 323

v=0
o=root 1514940027 1514940027 IN IP4 192.168.0.66
s=VoiceOne PBX
c=IN IP4 192.168.0.66
t=0 0
m=audio 15050 RTP/AVP 8 0 18 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/101
set_destination: Parsing <sip:102@192.168.0.68:5060> for address/port to send to
set_destination: set destination to 192.168.0.68:5060
Audio is at 12870
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.68:5060:
INVITE sip:102@192.168.0.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK24577a97
Max-Forwards: 70
From: "test 109" <sip:109@192.168.0.66>;tag=as243495ad
To: <sip:102@192.168.0.68:5060>;tag=7SH03K69Z3Hrc
Contact: <sip:109@192.168.0.66:5060>
Call-ID: 3dfa2b1b4d65fc2225a10c2f1b1a417a@192.168.0.66:5060
CSeq: 103 INVITE
User-Agent: VoiceOne PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "101 101" <sip:101@192.168.0.66>
Content-Type: application/sdp
Content-Length: 228

v=0
o=root 2051062857 2051062858 IN IP4 192.168.0.66
s=VoiceOne PBX
c=IN IP4 192.168.0.66
t=0 0
m=audio 12870 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
set_destination: Parsing <sip:102@192.168.0.68:5060> for address/port to send to
set_destination: set destination to 192.168.0.68:5060
Reliably Transmitting (no NAT) to 192.168.0.68:5060:
UPDATE sip:102@192.168.0.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK1b8d5de4
Max-Forwards: 70
From: "test 109" <sip:109@192.168.0.66>;tag=as243495ad
To: <sip:102@192.168.0.68:5060>;tag=7SH03K69Z3Hrc
Contact: <sip:109@192.168.0.66:5060>
Call-ID: 3dfa2b1b4d65fc2225a10c2f1b1a417a@192.168.0.66:5060
CSeq: 104 UPDATE
User-Agent: VoiceOne PBX
P-Asserted-Identity: "101 101" <sip:101@192.168.0.66>
X-Asterisk-rpid-update: Yes
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.68:5060 --->
SIP/2.0 400 Bad From Header
Via: SIP/2.0/UDP 192.168.0.66:5060;branch=z9hG4bK4cc3e71c
To: <sip:101@192.168.0.68:5060>;tag=Dg9mepBZeSQ8H
Call-ID: 5b1d76944763dd51065c57d4425c277e@192.168.0.66:5060
CSeq: 102 INVITE
Content-Length: 0

Здравствуйте.

У вас проблема в формировании From:
From: ""ORIGINAL:109 TRANSFER:102"" <sip:109-102@192.168.0.66>;tag=as0ad1cdc4

Лишние двойные кавычки.