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Asterisk -> Lync ->Asterisk проблема при переадресации вызова

Newbies/FAQ Forum 3 сообщений -
#1

Настроил связку Астр - Линц. Звонить можно на линц и с линца.
Но возникла такая проблема когда звонок идет на Линц, а в клиенте стоит переадресация на номер который на астре, звонок не проходит.
Астериск принимает переадресация но ничего не делает. В итоге звонок через какое-то время сбрасывается
Вот лог входящего звонка от линца при переадресации

--- (13 headers 0 lines) ---
-- SIP/Lync_Trunk-0000276d is ringing


INVITE sip:1321@10.0.XX.AST;user=phone SIP/2.0
FROM: "Sergey Yakovlev";epid=402B947F2B;tag=9cf4e7f5a4
TO:
CSEQ: 20283 INVITE
CALL-ID: 17c0863d-fef7-4778-b441-6af3a581d9fa
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.0.XX.LYNC:62481;branch=z9hG4bKd63b7a35
CONTACT:
CONTENT-LENGTH: 337
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE

v=0
o=- 368 1 IN IP4 10.0.XX.LYNC
s=session
c=IN IP4 10.0.XX.LYNC
b=CT:1000
t=0 0
m=audio 56174 RTP/AVP 97 101 13 0 8
c=IN IP4 10.0.XX.LYNC
a=rtcp:56175
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20


--- (14 headers 18 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.0.XX.LYNC : 62481 (no NAT)
Using INVITE request as basis request - 17c0863d-fef7-4778-b441-6af3a581d9fa
Found peer '1399' for '1399' from 10.0.XX.LYNC:62481


SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.0.XX.LYNC:62481;branch=z9hG4bKd63b7a35;received=10.0.XX.LYNC
From: "Sergey Yakovlev";epid=402B947F2B;tag=9cf4e7f5a4
To: ;tag=as381f7726
Call-ID: 17c0863d-fef7-4778-b441-6af3a581d9fa
CSeq: 20283 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5c4c4998"
Content-Length: 0



Scheduling destruction of SIP dialog '17c0863d-fef7-4778-b441-6af3a581d9fa' in 6400 ms (Method: INVITE)


ACK sip:1321@10.0.XX.AST;user=phone SIP/2.0
FROM: "Sergey Yakovlev";tag=9cf4e7f5a4;epid=402B947F2B
TO: ;tag=as381f7726
CSEQ: 20283 ACK
CALL-ID: 17c0863d-fef7-4778-b441-6af3a581d9fa
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.0.XX.LYNC:62481;branch=z9hG4bKd63b7a35
CONTENT-LENGTH: 0


Really destroying SIP dialog '17c0863d-fef7-4778-b441-6af3a581d9fa' Method: INVITE
Reliably Transmitting (NAT) to 10.0.XX.LYNC:5068:
OPTIONS sip:10.0.XX.LYNC SIP/2.0
Via: SIP/2.0/TCP 10.0.XX.AST:5060;branch=z9hG4bK103e9627;rport
Max-Forwards: 70
From: "asterisk" ;tag=as78981b17
To:
Contact:
Call-ID: 26873e190c0b722f21b55257177255ef@10.0.XX.AST
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Mon, 01 Oct 2012 13:45:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---


SIP/2.0 200 OK
FROM: "asterisk";tag=as78981b17
TO: ;tag=63a33a8e7
CSEQ: 102 OPTIONS
CALL-ID: 26873e190c0b722f21b55257177255ef@10.0.XX.AST
VIA: SIP/2.0/TCP 10.0.XX.AST:5060;branch=z9hG4bK103e9627;rport
ACCEPT: application/sdp
CONTENT-LENGTH: 0
ACCEPT-ENCODING: gzip
ACCEPT-LANGUAGE: en
ALLOW: NOTIFY
ALLOW: BENOTIFY
SERVER: RTCC/4.0.0.0 MediationServer



--- (13 headers 0 lines) ---


SIP/2.0 405 Method Not Allowed
FROM: "Sergey Yakovlev";tag=as55899e1a
TO: ;tag=d883e37e11;epid=402B947F2B
CSEQ: 102 INVITE
CALL-ID: 35143e334d4412b001aabc991c3fff38@10.0.XX.AST
VIA: SIP/2.0/TCP 10.0.XX.AST:5060;branch=z9hG4bK4b9c1217;rport
CONTENT-LENGTH: 0
ALLOW: None
SERVER: RTCC/4.0.0.0 MediationServer



--- (9 headers 0 lines) ---
-- Got SIP response 405 "Method Not Allowed" back from 10.0.XX.LYNC
Transmitting (NAT) to 10.0.XX.LYNC:5068:
ACK sip:+1599@10.0.XX.LYNC:5068 SIP/2.0
Via: SIP/2.0/TCP 10.0.XX.AST:5060;branch=z9hG4bK4b9c1217;rport
Max-Forwards: 70
From: "Sergey Yakovlev" ;tag=as55899e1a
To: ;tag=d883e37e11
Contact:
Call-ID: 35143e334d4412b001aabc991c3fff38@10.0.XX.AST
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0


---
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1599@all] Hangup("SIP/1399-0000276c", "") in new stack
== Spawn extension (all, 1599, 2) exited non-zero on 'SIP/1399-0000276c'
Really destroying SIP dialog '35143e334d4412b001aabc991c3fff38@10.0.XX.AST' Method: INVITE

Added after 39 seconds:

В чем может быть загвозка ?
Заранее спасибо.
#2

SIP/2.0 405 Method Not Allowed - это попытка уйти на голосовую почту ?

type=peer ?
hasvoicemail = ?
#3

sip.conf для Линца

[Lync_Trunk] ; Our Lync trunk
type=friend
port=5068 ; This is the default Lync Server TCP listening port
host=10.0.XX.LYNC ; This should be the IP address of your Lync Server
dtmfmode=rfc2833
context=all
qualify=yes
transport=tcp
nat = yes

Added after 1 minutes:

awsswa @ Пн Окт 01, 2012 15:13 писал(а):
SIP/2.0 405 Method Not Allowed - это попытка уйти на голосовую почту ?


Нет. Должен быть звонок на sip phone