СвязьПроект

Здравствуйте, подскажите пожл, есть такая связка IP90 и AS5303, связь по сип, звонок с ип90 на ас5350 проходит, а вот наоборот появилась проблема, сам звонок проходит, но при поднятии трубки ип90 проходит отбой с Remote:CallClear.
конфиг ас5350
dial-peer voice 396877 voip
huntstop
destination-pattern 396877
voice-class codec 2
session protocol sipv2
session target ipv4:ХХ.ХХ.ХХ.ХХ
dtmf-relay rtp-nte sip-notify sip-kpml
fax-relay ecm disable
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
no vad

конфиг ИП90:
Current configuration:
!
version 8.51.001
!
hostname Anton
clock timezone URALS 5
!
username root password router administrator
!
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address ХХ.ХХ.ХХ.ХХ
ip nat outside
speed auto
no qos-control
!
interface FastEthernet0/1
ip address 192.168.10.1 255.255.255.0
ip nat inside
speed auto
no qos-control
!
ip route 0.0.0.0 0.0.0.0 ХХХХХХХ
!
access-list 100 permit ip 192.168.10.0 0.0.0.255 any
!
ip nat inside source list 100 interface FastEthernet0/0 overload
!
snmp name IP90_G2
!
ip tcp keep-alive count 5
ip tcp keep-alive idle 60
ip tcp keep-alive interval 5
!
ftp server
http server
!

!
! IP PHONE OSD configuration.
!
osd
language russian
network signaling sip
network sscp disable
phone save-mode always
phone lcd-type graphic
phone ring-type 6
phone volume ring 5
phone volume input 5
phone volume output 5
phone volume micbooster disable
phone auto-hook-on disable
phone display-name TRS
phone dnd-mode silence
phone pbx-mode general
phone hook-mode digitwithdirect
phone auto-answer disable
phone conference-status disable
phone password 2337
phone password-status disable
phone admin-lock factory status disable
phone admin-lock internet status disable
phone admin-lock voip status disable
phone admin-lock service status disable
phone admin-lock auto-upgrade status disable
phone admin-lock sscp status disable
phone privacy-password 0000
phone privacy-status disable
phone privacy-lock menu status disable
phone privacy-lock incoming status disable
phone privacy-lock outgoing status disable
phone noanswer-sound off
phone noanswer-sound notify interval 30
phone emergency-number 1 112
phone emergency-number 2 119
phone emergency-number 3 911
phone emergency-number 4 999
phone contact-profile id 0 version 0
phone recording disable
!
! SSCP configuration.!
!
!
! SSCP Static CM List
sscp
!
! SSCP Dynamic CM List
sscp
!
!
sscp
call-manager broadcast port 8855
logger disable
logger level info
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate disable
timeout tmohdt 300
call-barring unconfigured-ip-address
voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! SPEECH
voice-port 0/0
fax-early-detect
!
!
! N/A
voice-port 0/1
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 396877
port 0/0
huntstop
recording all
!
dial-peer voice 1 pots
destination-pattern #8
port 0/1
huntstop
recording all
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
destination-pattern T
session target ip УУ.УУ.УУ.УУ
codec g711ulaw
voice-class codec 0
no vad
dtmf-relay rtp-2833
translate-outgoing called-number 1
fax protocol t38 redundancy 0
fax rate 9600
recording all
!
dial-peer voice 101 voip
destination-pattern 8T
session target ip ZZ.ZZ.ZZ.ZZ
codec g711ulaw
voice-class codec 0
no vad
dtmf-relay rtp-2833
fax protocol t38 redundancy 0
fax rate 9600
recording all
!
dial-peer voice 1001 voip
destination-pattern T
session target sip-server
voice-class codec 0
no vad
dtmf-relay rtp-2833
huntstop
shutdown
recording all
!
dial-peer voice 1003 voip
voice-class codec 0
no vad
dtmf-relay rtp-2833
shutdown
recording all
!
dial-peer voice 1004 voip
voice-class codec 0
no vad
dtmf-relay rtp-2833
shutdown
recording all
!
!
!
dial-peer ipaddr-prefix n
dial-peer call-hold n
dial-peer call-transfer n
!
!
!
! Gateway configuration.
!
gateway
!
!
!
! Recording configuration.
!
recording
direction all
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729
codec preference 4 g7231r63
codec preference 5 g726r32
!
!
!
! Translation Rule configuration.
!
translation-rule 1
rule 1 1 9991
rule 2 2 9992
rule 3 3 9993
rule 4 4 9994
rule 5 5 9995
rule 6 6 9996
rule 7 7 9997
rule 8 0 9990
rule 9 9 9999
!
!
!
! SIP UA configuration.
!
sip-ua
no fault-tolerance
no rport
call-transfer-mode attended
recording-info-notify
!
!
! SMS UA configuration.
!
sms-ua
check-to-header enable
!
!
! Tones
!
!
!
!
line console
!
line vty
!
!
sms
quota 30
!
!
! PS Client configuration.
!
psclient
service disable
retry_timer 10000
alive_timer 15000
!
!
end

Дебаг можете показать?

Anton#
Received SIP PDU from ( (ИП ас5350)ХХХХ:53702 )
INVITE sip:396877@(ИП ип90)УУУУ:5060 SIP/2.0
Via: SIP/2.0/UDPХХХХ:5060;branch=z9hG4bK7B014C1
Remote-Party-ID: <sip:83512396801@УУУУ>;party=calling;screen=yes;privacy=off
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>
Date: Tue, 01 Mar 2011 02:55:18 GMT
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 754866311-1122374112-2323972105-3893905184
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 65
Timestamp: 1298948118
Contact: <sip:83512396801@ХХХХ:5060>
Call-Info: <sip:ХХХХ:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309

v=0
o=CiscoSystemsSIP-GW-UserAgent 7700 6269 IN IP4 ХХХХ
s=SIP Call
c=IN IP4 ХХХХ
t=0 0
m=audio 17544 RTP/AVP 0 8 18 101
c=IN IP4 ХХХХ
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


Sending SIP PDU to ( ХХХХ:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B014C1
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
CSeq: 101 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


537 <Call 29> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1262387150) ****
538 <SIP 29> : Receive INVITE Request
539 <NetCon 29> : Found inbound voip peer(100) result(2) peer->fixedPatternSize(0) mostMatchingSize(-1)
540 <NetCon 29> : Found inbound voip peer by dest-pattern id(100)
541 <NetCon 29> : Found inbound voip peer(101) result(2) peer->fixedPatternSize(1) mostMatchingSize(0)
542 <NetCon 29> : Found inbound voip peer by dest-pattern id(101)
543 <NetCon 29> : Found inbound voip peer(1001) result(2) peer->fixedPatternSize(0) mostMatchingSize(1)
544 <Call 29> : From Net - calledParty(396877) callingParty(83512396801)
545 <Call 29> : MatchedPerfect
546 <Call 29> : MatchAllProcess After Sorted
<0> id(0) dest(396877) prefer(0) selected(17)
547 <Call 29> : Initiate callee with dial-peer(396877) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
548 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(83512396801), callerPort(ffffffff) type(SPEECH)
549 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(29)
CallState(Ringing:4) Softkey(Ringing:7) callId(29) proto(0) peer(0x65) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE)
550 <CCA 0> : Call Received from Net : Address(ХХХХ) Number(83512396801) Name(83512396801)
551 <SIP29> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
552 <SIP 29> : SetLocalAudioFormats : myVoipPeer(101) is not NULL, codec(1)
553 <PhonePlay 29> : Audio Count(1)
554 <PhonePlay 29> : rtpSessionId(1) Second Audio Port(-1)
555 <SIP 29> : SetAlerting
556 <SIP 29> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE)
557 <SIP 29> : SetLocalAudioFormats : myVoipPeer(101) is not NULL, codec(1)
558 <SIP 29> : Add Local Audio MediaFormat : 0

Sending SIP PDU to ( ХХХХ:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B014C1
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
CSeq: 101 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:396877@УУУУ
RSeq: 169898
Require: 100rel
Content-Type: application/sdp
Content-Length: 235

v=0
o=396877 1262387150 1262387150 IN IP4 УУУУ
s=AddPac Gateway SDP
c=IN IP4 УУУУ
t=1262387150 0
m=audio 23044 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Received SIP PDU from ( ХХХХ:53702 )
PRACK sip:396877@УУУУ:5060 SIP/2.0
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B11691
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Date: Tue, 01 Mar 2011 02:55:18 GMT
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
CSeq: 102 PRACK
RAck: 169898 101 INVITE
Max-Forwards: 65
Content-Length: 0


559 <SIP 29> : Receive PRACK Request

Sending SIP PDU to ( ХХХХ:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B11691
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
CSeq: 102 PRACK
User-Agent: AddPac SIP Gateway
Content-Length: 0


[ohx] [OSD->CCC] REQ (CallConnect) (voip_commands.cxx/24041)
[ohx] [OSD->CCC] REQ(1) (voip_commands.cxx/24042)
560 <CCA 0> : Call Connect Request from HANDSET
561 <Call 29> : Connected from(0)
562 <CEP 000000> : StopSignal
563 <SIP 29> : SetConnected

Sending SIP PDU to ( ХХХХ:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B014C1
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
CSeq: 101 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:396877@УУУУ
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Length: 0


564 <CCA 0> : Call Connected from Device (cn=29)
CallState(Connected:5) Softkey(Connected:5) callId(29) proto(0) peer(0x65) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE)

Received SIP PDU from ( ХХХХ:53702 )
ACK sip:396877@УУУУ:5060 SIP/2.0
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B2A6B
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Date: Tue, 01 Mar 2011 02:55:18 GMT
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
Max-Forwards: 65
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0


565 <SIP 29> : ACK received
566 <SIP 29> : Receive ACK Request
567 <SIP 29> : Set Terminated Success for 101 INVITE

Received SIP PDU from ( ХХХХ:53702 )
BYE sip:396877@УУУУ:5060 SIP/2.0
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B31130
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Date: Tue, 01 Mar 2011 02:55:18 GMT
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 65
Timestamp: 1298948121
CSeq: 103 BYE
Reason: Q.850;cause=96
Content-Length: 0


568 <SIP 29> : Receive BYE Request

Sending SIP PDU to ( ХХХХ:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B31130
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
CSeq: 103 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0


569 <SIP 29> : ReleaseWithNothing
570 <Call 29> : Terminated from(fffffffe) this(Remote:CallClear) before((null)) forced(0) time(1262387153)
[ohx] CCC(id:0x0) --> SetASStatus(OnHooKStatus -> OnHooKStatus) : (voip_call.cxx,13276)
571 <CEP 000000> : SetAdditional Service Status fr(OnHooKStatus) to(OnHooKStatus)
CallState(OnHook:2) Softkey(OnHook:1) callId(29) proto(0) peer(0x65) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE)
572 <CEP 000000> : DisconnectCall at Busy
573 <CEP 000000> : StopSignal
CallState(OnHook:2) Softkey(OnHook:1) callId(29) proto(0) peer(0xffffffff) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE)
574 <CEP 000000> : Disconnect (0)
575 <CEP 000000> : Disconnect (0)
576 <CCA 0> : Call Disconnected Error : Current call is not assigned callNumber(29), [0][0][0][0]:[0][0][0][0]
CallState(OnHook:2) Softkey(OnHook:1) callId(29) proto(0) peer(0xffffffff) secur(0) reason(0) isPTT(FALSE) isAutoReceived(FALSE)
577 <NetEP 29> : Call FROM <83512396801> terminated reason(Remote:CallClear)
578 <CEP 000000> : Disconnected(16) at Disconnecting
[ohx] [OSD->CCC] REQ (CallDisconn) (voip_commands.cxx/24041)
[ohx] [OSD->CCC] REQ(2) (voip_commands.cxx/24042)
579 <CCA 0> : Call Disconnect Request from HANDSET callNumber(-1)
580 <CCA 0> : Call Disconnect Error : Current call is not assigned callNumber(-1) [0][0][0][0]

Почему-то bye c AS5350 приходит c кодом 96:

Received SIP PDU from ( ХХХХ:53702 )
BYE sip:396877@УУУУ:5060 SIP/2.0
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B31130
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Date: Tue, 01 Mar 2011 02:55:18 GMT
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 65
Timestamp: 1298948121
CSeq: 103 BYE
Reason: Q.850;cause=96
Content-Length: 0

спасибо, но стал смотреть дальше и увидел вот что
Received SIP PDU from ( ХХХХ:53702 )
ACK sip:396877@УУУУ:5060 SIP/2.0
Via: SIP/2.0/UDP ХХХХ:5060;branch=z9hG4bK7B2A6B
From: <sip:83512396801@ХХХХ>;tag=37B38AE8-FCE
To: <sip:396877@УУУУ>;tag=ce4b382ba4
Date: Tue, 01 Mar 2011 02:55:18 GMT
Call-ID: 2D002A04-42E611E0-9A41ED8C-8076660B@ХХХХ
Max-Forwards: 65
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0

как я понимаю это сообщение об авторизации, и как-то кто-то должен авторизироваться, хотя на циске все разрешения прописаны только для ип адресов.

Пакеты ACK используются для подтверждений приема других пакетов. У Вас в схеме авторизация не используется.

нет авторизации нет, есть ацл листы на воипный трафик и все.