СвязьПроект

Привет всем.В сети есть IP-PBX Asterisk, настроен на SIP.Настроил шлюз AddPac1000, исходящие звонки проходят, а вот на шлюз дозвониться не могут-идут короткие гудки(занято).Вот конфиг:

! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor sip-server
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
! FXS
voice-port 0/2
caller-id enable
!
!
! FXS
voice-port 0/3
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern 6025
port 0/0
user-password xxxxxx
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
destination-pattern T
session target 192.168.100.236
session protocol sip
codec g711alaw
voice-class codec 1
no dtmf-relay
no vad
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.50.1
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
! SIP UA configuration.
!
sip-ua
sip-username 6025
sip-password xxxxxx
sip-server 192.168.100.236
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones
!
!
!
!
Где может быть ошибка?

dial-peer voice 100 voip
session target sip-server


посмотрите дебаг, приходит ли что-то на шлюз?
deb voip sip
deb rta ipc
deb voip call
conf t
deb

Сделал debug

Received SIP PDU from ( 192.168.100.236:5060 )
OPTIONS sip:6025@192.168.50.233 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK718e09c9;rport
From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as00166edc
To: <sip:6025@192.168.50.233>
Contact: <sip:Unknown@192.168.100.236>
Call-ID: 555cb1a67c3bea8e1118ff08296fd24f@192.168.100.236
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Mar 2010 10:30:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK718e09c9;rport
From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as00166edc
To: <sip:6025@192.168.50.233>
Call-ID: 555cb1a67c3bea8e1118ff08296fd24f@192.168.100.236
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0



Received SIP PDU from ( 192.168.100.236:5060 )
INVITE sip:6025@192.168.50.233 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>
Contact: <sip:6030@192.168.100.236>
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Mar 2010 10:30:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 428

v=0
o=root 2683 2683 IN IP4 192.168.100.236
s=session
c=IN IP4 192.168.100.236
b=CT:384
t=0 0
m=audio 12244 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 11300 RTP/AVP 31 34 103 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>;tag=4149d201a4
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:6025@192.168.50.233
Content-Length: 0



Received SIP PDU from ( 192.168.100.236:5060 )
CANCEL sip:6025@192.168.50.233 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 CANCEL
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>;tag=4149d201a4
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



Received SIP PDU from ( 192.168.100.236:5060 )
ACK sip:6025@192.168.50.233 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport
From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0
To: <sip:6025@192.168.50.233>;tag=4149d201a4
Contact: <sip:6030@192.168.100.236>
Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
REGISTER sip:192.168.100.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4461
From: <sip:6025@192.168.100.236>;tag=14495200a4
To: sip:6025@192.168.100.236
Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233
CSeq: 461 REGISTER
Date: Tue, 07 Apr 2009 23:44:34 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="6025", realm="asterisk", nonce="3391eb6c", uri="
sip:192.168.100.236", response="02d3f6de146fd2ca1d1752bc3276e53c", algorithm=MD5
Contact: <sip:6025@192.168.50.233>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.100.236:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4461;received=192.16
8.50.233
From: <sip:6025@192.168.100.236>;tag=14495200a4
To: sip:6025@192.168.100.236
Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233
CSeq: 461 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6025@192.168.100.236>
Content-Length: 0


Received SIP PDU from ( 192.168.100.236:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4461;received=192.16
8.50.233
From: <sip:6025@192.168.100.236>;tag=14495200a4
To: sip:6025@192.168.100.236;tag=as46bc1f4a
Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233
CSeq: 461 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e313af3"
Content-Length: 0


Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
REGISTER sip:192.168.100.236 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4462
From: <sip:6025@192.168.100.236>;tag=14495200a4
To: sip:6025@192.168.100.236
Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233
CSeq: 462 REGISTER
Date: Tue, 07 Apr 2009 23:44:34 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="6025", realm="asterisk", nonce="6e313af3", uri="
sip:192.168.100.236", response="65ea45a587839b026cae323d49ba0ca0", algorithm=MD5
Contact: <sip:6025@192.168.50.233>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.100.236:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4462;received=192.16
8.50.233
From: <sip:6025@192.168.100.236>;tag=14495200a4
To: sip:6025@192.168.100.236
Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233
CSeq: 462 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6025@192.168.100.236>
Content-Length: 0


Received SIP PDU from ( 192.168.100.236:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4462;received=192.16
8.50.233
From: <sip:6025@192.168.100.236>;tag=14495200a4
To: sip:6025@192.168.100.236;tag=as46bc1f4a
Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233
CSeq: 462 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:6025@192.168.50.233>;expires=60
Date: Mon, 01 Mar 2010 10:30:47 GMT
Content-Length: 0


Received SIP PDU from ( 192.168.100.236:5060 )
NOTIFY sip:6025@192.168.50.233 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK6b9ac404;rport
From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as58124513
To: <sip:6025@192.168.50.233>
Contact: <sip:Unknown@192.168.100.236>
Call-ID: 6343656f08250546626c47c071b7fffe@192.168.100.236
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90

Messages-Waiting: no
Message-Account: sip:*97@192.168.100.236
Voice-Message: 0/0 (0/0)

Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK6b9ac404;rport
From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as58124513
To: <sip:6025@192.168.50.233>
Call-ID: 6343656f08250546626c47c071b7fffe@192.168.100.236
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0



Received SIP PDU from ( 192.168.100.236:5060 )
OPTIONS sip:6025@192.168.50.233 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK280dfde2;rport
From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as7789aeb3
To: <sip:6025@192.168.50.233>
Contact: <sip:Unknown@192.168.100.236>
Call-ID: 011b6ef11a16b0f96c03e88c3c0fb528@192.168.100.236
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 01 Mar 2010 10:31:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK280dfde2;rport
From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as7789aeb3
To: <sip:6025@192.168.50.233>
Call-ID: 011b6ef11a16b0f96c03e88c3c0fb528@192.168.100.236
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0

Вызов как я понимаю проходит, но в ответ гудки.С софтовым клиентом(X-lite) все работает.

Звонок проходит, но после Ringing на шлюз приходит Cancel. И все. Смотрите на станции или на удаленной стороне причину этого Cancel.

Спасибо, будем разбираться.