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При входящем вызове на порт FXS на AddPAC отсутствует КПВ, но телефон звонит.
Почему AddPac не выдает КПВ?

конфиг

Вот конфиг

Current configuration:
!
version 8.41.084
!
hostname AP700
!
username root password router administrator
!
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 10.12.32.111 255.255.255.0
speed auto
no qos-control
!
interface FastEthernet0/1
no ip address
speed auto
no qos-control
!
ip route 0.0.0.0 0.0.0.0 10.12.32.1
!
!
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
protocol sip
dtmf-relay out-of-band
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
timeout tttl 300
local-ringback-tone early
inband-ringback-tone
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 4636009
port 0/0
user-password jh5kjd11
!
dial-peer voice 2 pots
destination-pattern 2600101
port 0/1
user-password jh5kjd11
!
!
!
! Voip peer configuration.
!
dial-peer voice 11 voip
destination-pattern T
session target sip-server
session protocol sip
answer-address 4636009
voice-class codec 1
no vad
dtmf-relay rtp-2833
!
dial-peer voice 12 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 1
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.12.32.111
no ignore-msg-from-other-gk
register
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
!
!
!
! SIP UA configuration.
!
sip-ua
no fault-tolerance
sip-server 10.12.33.4
register e164
!
!
! MGCP configuration.
!
mgcp
vad
!
!
! Tones
voice class ring-back-tone 425 425 1000 4000 0 0 -12
!
!
!
!
!
line console
!
line vty
!
end

en
conf t
voice serv voip
no inband
local-ring

en
conf t
gatew
no reg

Не помогает.
Тут дело в том, что в качестве сервера используется Huawei SoftX3000.
C этим же SoftX работает AP1100 и все нормально.

conf t
sip
media early

дебаг
deb voip call
deb rta ipc
deb voip sip

не помогло

Вот дебаг
VoIP in service.
1 <Time 0> : SIP_TREGISTER timer timeout.
2 <SIP 0> : Adding authentication information
3 <SIP 7> : Send REGISTER Request
4 <SIP 7> : Receive 200 OK
5 <SIP 7> : Transaction (7 REGISTER) completed
6 <SIP 0> : Adding authentication information
7 <SIP 8> : Send REGISTER Request
8 <SIP 8> : Receive 200 OK
9 <SIP 8> : Transaction (8 REGISTER) completed
10 <SIP 7> : Set Terminated Success for 7 REGISTER
11 <SIP 8> : Set Terminated Success for 8 REGISTER

Received SIP PDU from ( 10.12.33.4:5061 )
INVITE sip:4636009@10.12.32.111:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>
CSeq: 1 INVITE
Contact: <sip:Anonymous@10.12.33.4:5060>
Min-SE: 90
Session-Expires: 1800
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE
,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R601
Supported: 100rel,timer
Max-Forwards: 70
Content-Length: 246
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 7970 7970 IN IP4 10.12.33.4
s=Sip Call
c=IN IP4 10.12.34.15
t=0 0
m=audio 6216 RTP/AVP 8 0 18 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=fmtp:18 annexb=no


Sending SIP PDU to ( 10.12.33.4:5061 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


12 <Call 3> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(1274802480) ****
13 <SIP 3> : Receive INVITE Request
14 <NetCon 3> : Found inbound voip peer by dest-pattern id(11)
15 <Call 3> : From Net - calledParty(4636009) callingParty(Anonymous
)
16 <Call 3> : MatchedPerfect
17 <Call 3> : MatchAllProcess After Sorted
<0> id(1) dest(4636009) prefer(0) selected(2)
18 <Call 3> : Initiate callee with dial-peer(4636009) status(CalleeD
eterminedAll) id(00000000-0000-0000-0000-000000000000)
19 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(), callerPo
rt(ffffffff) type(FXS)
[142.870] RTA(0/0/0) Rx CC_RING_REQ [80 18 01 08 30 35 32 35 31 35 34 38 04 01 4
f 07 09 41 6e 6f 6e 79 6d 6f 75 73 ] peerId(-1)
[142.870] VM(0/0/0) DaTime [L=8] 30 35 32 35 31 35 34 38
[142.870] VM(0/0/0) CgNoNu [L=1] 4f
[142.870] VM(0/0/0) CgName [L=9] 41 6e 6f 6e 79 6d 6f 75 73
[142.870] VM(0/0/0) Line Reverse
[142.870] VM(0/0/0) Start ring actv
[142.870] VM(0/0/0) SW to -72V
20 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(3)
[142.870] VM(0/0/0) set T38 enable by CCC
[142.870] VM(0/0/0) set T38 mode STD
[142.870] VM(0/0/0) Fax rate 9600
21 <SIP 3> : SetAlerting
22 <SIP 3> : Add Local Audio MediaFormat : 8

Sending SIP PDU to ( 10.12.33.4:5061 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>;tag=304b2d04a4
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:4636009@10.12.32.111
RSeq: 864122
Require: 100rel
Content-Type: application/sdp
Content-Length: 180

v=0
o=4636009 1274802480 1274802480 IN IP4 10.12.32.111
s=AddPac Gateway SDP
c=IN IP4 10.12.32.111
t=1274802480 0
m=audio 23010 RTP/AVP 8
a=ptime:20
a=rtpmap:8 PCMA/8000/1


Received SIP PDU from ( 10.12.33.4:5061 )
PRACK sip:4636009@10.12.32.111 SIP/2.0
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK24b93c5ac75089873b192600a
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>;tag=304b2d04a4
CSeq: 2 PRACK
Max-Forwards: 70
RAck: 864122 1 INVITE
Content-Length: 0


23 <SIP 3> : Receive PRACK Request

Sending SIP PDU to ( 10.12.33.4:5061 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK24b93c5ac75089873b192600a
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>;tag=304b2d04a4
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
CSeq: 2 PRACK
User-Agent: AddPac SIP Gateway
Content-Length: 0


[143.870] VM(0/0/0) Gen ring idle
[144.370] VM(0/0/0) Tx CID enable
[144.370] VP(0/0/0) CallerId enable, std/gain 0/6
[144.370] VP(0/0/0) open channel
[144.370] VM(0/0/0) play mute
[144.370] VP(0/0/0) Tx IBS signal 2/0
[144.370] VP(0/0/0) Tx IBS dir 0
[144.430] VM(0/0/0) Tx CID DATA [L=26] 80 18 01 08 30 35 32 35 31 35 34 38 04 01
4f 07 09 41 6e 6f 6e 79 6d 6f 75 73
[144.430] VP(0/0/0) play CallerId
[144.445] VP(0/0/0) GeneralEvent IBS gen end
[145.430] VM(0/0/0) Tx CID fin
[145.430] VP(0/0/0) stop CallerId
[145.430] VM(0/0/0) vopp idle
[145.430] VP(0/0/0) close channel
[145.870] VM(0/0/0) Gen ring actv
[146.870] VM(0/0/0) Gen ring idle
[148.870] VM(0/0/0) Gen ring actv
[149.870] VM(0/0/0) Gen ring idle
[151.870] VM(0/0/0) Gen ring actv
[152.870] VM(0/0/0) Gen ring idle
[154.870] VM(0/0/0) Gen ring actv
[155.870] VM(0/0/0) Gen ring idle
24 <Time 3> : Call Forwarding No Answer timer timeout.
[157.870] VM(0/0/0) Gen ring actv
[158.870] VM(0/0/0) Gen ring idle

Received SIP PDU from ( 10.12.33.4:5061 )
CANCEL sip:4636009@10.12.32.111:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>
CSeq: 1 CANCEL
Reason: Q.850;cause=19;text="no answer from the user"
Max-Forwards: 70
Content-Length: 0


25 <SIP 3> : Receive CANCEL Request
26 <SIP 3> : Cancel received

Sending SIP PDU to ( 10.12.33.4:5061 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
CSeq: 1 CANCEL
User-Agent: AddPac SIP Gateway
Content-Length: 0



Sending SIP PDU to ( 10.12.33.4:5061 ) from 5060
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>;tag=304b2d04a4
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


27 <Call 3> : Terminated from(fffffffe) this(Remote:CallClear) befor
e(NULL) forced(0) time(1274802497)
28 <CEP 000000> : DisconnectCall at Busy
29 <CEP 000000> : StopSignal
[160.135] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
30 <CEP 000000> : Disconnect (0)
[160.135] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[160.135] VM(0/0/0) SW to -48V
[160.135] VM(0/0/0) Line Forward
[160.135] VM(0/0/0) Tx DISCONN_CNF
31 <NetEP 3> : Call FROM <Anonymous> terminated reason(Remote:CallCle
ar)
32 <CEP 000000> : Disconnected(16) at Disconnecting

Received SIP PDU from ( 10.12.33.4:5061 )
ACK sip:4636009@10.12.32.111:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.12.33.4:5061;branch=z9hG4bK39fd6a952d2ff06cf7fbc0204
Call-ID: d8e2fa1462a84f990da9686722249103@SoftX3000
From: Anonymous <sip:Anonymous@10.12.33.4>;tag=e68a1d9c-CC-22-TRC-633
To: <sip:4636009@10.12.32.111;user=phone>;tag=304b2d04a4
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


33 <SIP 3> : Receive ACK Request
34 <Time 0> : SIP_TREGISTER timer timeout.
35 <SIP 0> : Adding authentication information
36 <SIP 9> : Send REGISTER Request

Sending SIP PDU to ( 10.12.33.4:5060 ) from 5060
REGISTER sip:10.12.33.4 SIP/2.0
Via: SIP/2.0/UDP 10.12.32.111:5060;branch=z9hG4bKb64bf100a49
From: <sip:4636009@10.12.33.4>;tag=b64bf100a4
To: sip:4636009@10.12.33.4
Call-ID: b6f0fb4b-e91e-f167-8000-0002a4072a1e@10.12.32.111
CSeq: 9 REGISTER
Date: Tue, 25 May 2010 15:48:22 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="4636009", realm="Huawei", nonce="1344c59834389e1
d3e19c38498c86ea4", uri="sip:10.12.33.4", response="43b1a950e8c523945e6a31da8f1e
74cc", algorithm=MD5
Contact: <sip:4636009@10.12.32.111>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70


37 <SIP 3> : Set Terminated Success for 1 INVITE

Received SIP PDU from ( 10.12.33.4:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.12.32.111:5060;branch=z9hG4bKb64bf100a49
Call-ID: b6f0fb4b-e91e-f167-8000-0002a4072a1e@10.12.32.111
From: <sip:4636009@10.12.33.4>;tag=b64bf100a4
To: <sip:4636009@10.12.33.4>;tag=7b7dab82
CSeq: 9 REGISTER
Expires: 60
Contact: <sip:4636009@10.12.32.111;user=phone>;expires=60
Server: Huawei SoftX3000 V300R601
Content-Length: 0


38 <SIP 9> : Receive 200 OK
39 <SIP 9> : Transaction (9 REGISTER) completed
40 <SIP 0> : Adding authentication information
41 <SIP 10> : Send REGISTER Request

Sending SIP PDU to ( 10.12.33.4:5060 ) from 5060
REGISTER sip:10.12.33.4 SIP/2.0
Via: SIP/2.0/UDP 10.12.32.111:5060;branch=z9hG4bKb64b6e02a410
From: <sip:2600101@10.12.33.4>;tag=b64b6e02a4
To: sip:2600101@10.12.33.4
Call-ID: b6f0fb4b-03a1-6eec-8002-0002a4072a1e@10.12.32.111
CSeq: 10 REGISTER
Date: Tue, 25 May 2010 15:48:22 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2600101", realm="Huawei", nonce="f83dd33a1272824
61b957cb63c43401c", uri="sip:10.12.33.4", response="3d6a4c4ba8b4419f8c022184b515
b74a", algorithm=MD5
Contact: <sip:2600101@10.12.32.111>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 10.12.33.4:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.12.32.111:5060;branch=z9hG4bKb64b6e02a410
Call-ID: b6f0fb4b-03a1-6eec-8002-0002a4072a1e@10.12.32.111
From: <sip:2600101@10.12.33.4>;tag=b64b6e02a4
To: <sip:2600101@10.12.33.4>;tag=f41544bb
CSeq: 10 REGISTER
Expires: 60
Contact: <sip:2600101@10.12.32.111;user=phone>;expires=60
Server: Huawei SoftX3000 V300R601
Content-Length: 0


42 <SIP 10> : Receive 200 OK
43 <SIP 10> : Transaction (10 REGISTER) completed
44 <SIP 9> : Set Terminated Success for 9 REGISTER
45 <SIP 10> : Set Terminated Success for 10 REGISTER
46 <SIP 3> : Set Terminated Success for 2 PRACK
47 <SIP 3> : Set Terminated Success for 1 CANCE

Можете прислать дамп wireshark перед AP700
И вместе с дампом ещё раз
deb voip call
deb rta ipc
deb voip sip

Похоже что проблема в прошивке, нужно исправлять.

Меня настораживает запись в дебаге

[144.370] VM(0/0/0) play mute

Я так понимаю, это оно и есть?

А может есть какая либо старая прошивка на шлюз, просто на офсайте эта единственная?

переделал дамп wiresharka. Там нет исходящих пакетов.