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Перезагрузка AP1000 при входящем звонке
http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=1257
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Автор:  OlegZeml [ 04 мар 2010, 14:36 ]
Заголовок сообщения:  Перезагрузка AP1000 при входящем звонке

Привет Всем.

Настроил AP1000 подключение по SIP к провайдеру (ЮТК).
Исходящий звонок - без проблем. Входящий звонок снаружи - проходит 1 из 10 или 15. В остальных случаях при снятии трубки на addpac тишина, у вызывающего абонента продолжаются гудки вызова. После того как трубка опущена, еще секунд 15 продолжает мигать лампочка порта FXS будто идет разговор, а потом addpac перезагружается.

Если подключить ЮТК к Asterisk , то все нормально.
Раньше сам аппарат тоже раньше работал с Asterisk.

Что может быть? Куда копать?

!
version 8.30U
!
hostname AP1000
!
!
no bridge spanning-tree
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128
!
!
no ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 10.9.4.11 255.255.255.0
!
interface ether1.0
ip address 192.168.1.123 255.255.255.0
!
snmp name AP1000
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
route 10.10.10.0 255.255.255.0 10.9.4.9
!
!
!
!
!
! 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 10 pots
destination-pattern 2687023
port 0/0
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 1
no vad
codec-variant g7231 standard
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.9.4.11
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g729
codec preference 2 g711alaw
codec preference 3 g711ulaw
codec preference 4 g726r32
!
!
!
! SIP UA configuration.
!
sip-ua
sip-username ********
sip-password *********
sip-server 10.10.10.100
register e164
!
!
! MGCP configuration.
!
mgcp
no codec
vad
!
!
! Tones
!

Дебаг успешного звонка.

AP1000#
Received SIP PDU from ( 10.10.10.100:5070 )
INVITE sip:2687069@10.9.4.11;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
To: <sip:2687069@10.9.4.11;user=phone>
CSeq: 1 INVITE
Contact: <sip:Anonymous@10.10.10.100:5070>
Supported: 100rel
User-Agent: Huawei SoftX3000 V300R006B06D061
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Content-Length: 248
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 3293963 3293963 IN IP4 10.10.10.100
s=Sip Call
c=IN IP4 10.10.10.10
t=0 0
m=audio 29908 RTP/AVP 18 8 97
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no

Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
To: <sip:2687069@10.9.4.11;user=phone>
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


1 <Call 3> : ****************** Call Created status(InitiatedByNet) *******************
2 <SIP 3> : Receive INVITE Request
3 <NetCon 3> : Found inbound voip peer by dest-pattern id(1000)
4 <Call 3> : From Net - calledParty(2687069) callingParty(Anonymous)
5 <Call 3> : MatchedPerfect
6 <Call 3> : MatchAllProcess After Sorted
<0> id(10) dest(2687069) prefer(0) selected(0)
7 <Call 3> : Initiate callee with dial-peer(2687069) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
8 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS)
[1451.575] RTA(0/0/0) Rx CC_RING_REQ [80 18 01 08 30 33 30 34 31 37 30 34 04 01 4f 07 09 41 6e 6f 6e 79 6d 6f 75 73 ] peerId(-1)
[1451.575] VM(0/0/0) DaTime [L=8] 30 33 30 34 31 37 30 34
[1451.575] VM(0/0/0) CgNoNu [L=1] 4f
[1451.575] VM(0/0/0) CgName [L=9] 41 6e 6f 6e 79 6d 6f 75 73
[1451.575] VM(0/0/0) Line Reverse
[1451.575] VM(0/0/0) Start ring actv
9 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(3)
[1451.580] VM(0/0/0) set T38 mode STD
[1451.580] VM(0/0/0) Fax rate 9600
10 <SIP 3> : SetAlerting

Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
To: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:2687069@10.9.4.11
Content-Length: 0


[1452.575] VM(0/0/0) Gen ring idle
[1453.075] VM(0/0/0) Tx CID enable
[1453.075] VM(0/0/0) vopp enable
[1453.075] VM(0/0/0) play mute
[1453.135] VM(0/0/0) Tx CID ON
[1453.180] VM(0/0/0) Rx CID_ACK
[1453.180] VM(0/0/0) Tx CID DATA [L=54] 80 01 18 02 01 05 08 06 30 08 33 08 30 08 34 08 31 07 37 07 30 07 34 07 04 05 01 06 4f 0c 07 05 09 06 41 0b 6e 0b 6f 0b 6e 0b 79 0b 6d 0b 6f 0b 75 0b 73 0b 00 0f
[1454.180] VM(0/0/0) Tx CID fin
[1454.180] VM(0/0/0) vopp idle
[1454.180] VM(0/0/0) VoPP ready
[1454.365] VM(0/0/0) vmOffHook
[1454.425] VM(0/0/0) vmTmoOffHook
[1454.425] VM(0/0/0) Line Forward
[1454.485] VM(0/0/0) vmTmoOffHook
[1454.485] VM(0/0/0) Rx OffHook
[1454.485] VM(0/0/0) vopp enable
[1454.485] VM(0/0/0) Fax enable
[1454.485] VM(0/0/0) play mute
[1454.485] VM(0/0/0) Tx CONNECT_CNF
11 <Call 3> : Connected from(0)
[1454.485] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[1454.485] VM(0/0/0) VAD disable
[1454.485] VM(0/0/0) SID enable by CCC
12 <SIP 3> : SetConnected
13 <SIP 3> : Add Local Audio MediaFormat : 18
[1454.490] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[1454.490] VM(0/0/0) VAD disable

Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK2693cd7f3
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
To: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
CSeq: 1 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:2687069@10.9.4.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 250

v=0
o=2687069 1267722296 1267722296 IN IP4 10.9.4.11
s=AddPac Gateway SDP
c=IN IP4 10.9.4.11
t=1267722296 0
m=audio 23002 RTP/AVP 18 97
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000/1
a=fmtp:97 0-15
a=ptime:20

[1454.525] RTA(0/0/0) Rx RS_LISTEN_REQ callId=3 ssId=1 G729A
peer=10.10.10.10 mp=23002/23003 hp=29908/29909
[1454.525] VM(0/0/0) codec same G729A
[1454.525] RTA(0/0/0) Rx RS_OPEN_REQ callId=3 ssId=1 G729A
peer=10.10.10.10 mp=23002/23003 hp=29908/29909
[1454.525] VM(0/0/0) codec same G729A
[1454.525] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[1454.525] VM(0/0/0) DTMF_RTP_RFC2833 enable
[1454.525] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x61, RxPT=0x61

Received SIP PDU from ( 10.10.10.100:5070 )
ACK sip:2687069@10.9.4.11 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bKfbf01e6fc
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
To: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

14 <SIP 3> : ACK received
15 <SIP 3> : Receive ACK Request
16 <SIP 3> : Set Terminated Success for 1 INVITE
[1457.760] RTA(0/0/0) RTP play loss, xCnt=1
[1458.020] RTA(0/0/0) RTP play loss, xCnt=1
[1458.030] RTA(0/0/0) RTP play loss, xCnt=1
[1458.780] RTA(0/0/0) RTP play loss, xCnt=1
[1458.790] RTA(0/0/0) RTP play loss, xCnt=1
[1459.960] RTA(0/0/0) RTP play loss, xCnt=1
[1460.740] RTA(0/0/0) RTP play loss, xCnt=1
[1460.750] RTA(0/0/0) RTP play loss, xCnt=1
[1463.420] RTA(0/0/0) RTP play loss, xCnt=1
[1464.580] RTA(0/0/0) RTP play loss, xCnt=1
[1466.120] RTA(0/0/0) RTP play loss, xCnt=1
[1466.660] RTA(0/0/0) RTP play loss, xCnt=1
[1466.670] RTA(0/0/0) RTP play loss, xCnt=1
[1468.240] RTA(0/0/0) RTP play loss, xCnt=1
[1468.380] RTA(0/0/0) RTP play loss, xCnt=1
[1469.820] RTA(0/0/0) RTP play loss, xCnt=1
[1469.830] RTA(0/0/0) RTP play loss, xCnt=1
[1470.875] VM(0/0/0) vmOnHook
[1470.925] VM(0/0/0) vmTmoOnHook
[1470.975] VM(0/0/0) vmTmoOnHook
[1471.020] RTA(0/0/0) RTP play loss, xCnt=1
[1471.025] VM(0/0/0) vmTmoOnHook
[1471.030] RTA(0/0/0) RTP play loss, xCnt=1
[1471.075] VM(0/0/0) vmTmoOnHook
[1471.125] VM(0/0/0) vmTmoOnHook
[1471.175] VM(0/0/0) vmTmoOnHook
[1471.225] VM(0/0/0) vmTmoOnHook
[1471.275] VM(0/0/0) vmTmoOnHook
[1471.300] RTA(0/0/0) RTP play loss, xCnt=1
[1471.310] RTA(0/0/0) RTP play loss, xCnt=1
[1471.325] VM(0/0/0) vmTmoOnHook
[1471.375] VM(0/0/0) vmTmoOnHook
[1471.425] VM(0/0/0) vmTmoOnHook
[1471.460] RTA(0/0/0) RTP play loss, xCnt=1
[1471.470] RTA(0/0/0) RTP play loss, xCnt=1
[1471.475] VM(0/0/0) vmTmoOnHook
[1471.525] VM(0/0/0) vmTmoOnHook
[1471.575] VM(0/0/0) vmTmoOnHook
[1471.575] VM(0/0/0) Rx OnHook
[1471.575] VM(0/0/0) vopp idle
[1471.575] VM(0/0/0) VoPP ready
[1471.575] VM(0/0/0) Tx DISCONN_CNF
17 <CEP 000000> : Disconnected(16) at Busy
18 <Call 3> : Terminated from(0) this(Local:CallClear) before(NULL) forced(0)
19 <SIP 3> : ReleaseWithBYE
20 <SIP 3> : Send BYE Request

Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060
BYE sip:Anonymous@10.10.10.100:5070 SIP/2.0
Via: SIP/2.0/UDP 10.9.4.11:5060;branch=z9hG4bK354be505a41
From: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4
To: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
CSeq: 1 BYE
Date: Thu, 04 Mar 2010 17:05:13 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:2687069@10.9.4.11>
Content-Length: 0
Max-Forwards: 70


[1471.595] RTA(0/0/0) Rx RS_CLOSE_REQ callId=3 ssId=1 dir=reve
[1471.595] RTA(0/0/0) Rx RS_CLOSE_REQ callId=3 ssId=1 dir=forw
[1471.595] RTA(0/0/0) close Media socket
[1471.595] RTA(0/0/0) close RTCP socket
21 <NetEP 3> : Call TO <Anonymous> terminated reason(Local:CallClear)
22 <CEP 000000> : DisconnectCall at Idle

Received SIP PDU from ( 10.10.10.100:5070 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.4.11:5060;branch=z9hG4bK354be505a41
Call-ID: 9cb24487b416906cee80efc66f84944d@10.10.10.100
From: <sip:2687069@10.9.4.11;user=phone>;tag=354be505a4
To: Anonymous <sip:Anonymous@10.10.10.100>;tag=3ec41bff
CSeq: 1 BYE
Content-Length: 0

23 <SIP 3> : Receive 200 OK
24 <SIP 3> : Transaction (1 BYE) completed
25 <SIP 3> : Set Terminated Success for 1 BYE

Дебаг до перезагрузки:

Received SIP PDU from ( 10.10.10.100:5070 )
INVITE sip:2687069@10.9.4.11;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK8f4f32f60
Call-ID: cc502cb22e7a31344ad15ac96e586928@10.10.10.100
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=581d6a1c
To: <sip:2687069@10.9.4.11;user=phone>
CSeq: 1 INVITE
Contact: <sip:Anonymous@10.10.10.100:5070>
Supported: 100rel
User-Agent: Huawei SoftX3000 V300R006B06D061
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Content-Length: 248
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 3293976 3293976 IN IP4 10.10.10.100
s=Sip Call
c=IN IP4 10.10.10.10
t=0 0
m=audio 29950 RTP/AVP 18 8 97
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=no

Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK8f4f32f60
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=581d6a1c
To: <sip:2687069@10.9.4.11;user=phone>
Call-ID: cc502cb22e7a31344ad15ac96e586928@10.10.10.100
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


26 <Call 4> : ****************** Call Created status(InitiatedByNet) *******************
27 <SIP 4> : Receive INVITE Request
28 <NetCon 4> : Found inbound voip peer by dest-pattern id(1000)
29 <Call 4> : From Net - calledParty(2687069) callingParty(Anonymous)
30 <Call 4> : MatchedPerfect
31 <Call 4> : MatchAllProcess After Sorted
<0> id(10) dest(2687069) prefer(0) selected(1)
32 <Call 4> : Initiate callee with dial-peer(2687069) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
33 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS)
[1488.775] RTA(0/0/0) Rx CC_RING_REQ [80 18 01 08 30 33 30 34 31 37 30 35 04 01 4f 07 09 41 6e 6f 6e 79 6d 6f 75 73 ] peerId(-1)
[1488.775] VM(0/0/0) DaTime [L=8] 30 33 30 34 31 37 30 35
[1488.775] VM(0/0/0) CgNoNu [L=1] 4f
[1488.775] VM(0/0/0) CgName [L=9] 41 6e 6f 6e 79 6d 6f 75 73
[1488.775] VM(0/0/0) Line Reverse
[1488.775] VM(0/0/0) Start ring actv
34 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(4)
[1488.780] VM(0/0/0) set T38 mode STD
[1488.780] VM(0/0/0) Fax rate 9600
35 <SIP 4> : SetAlerting

Sending SIP PDU to ( 10.10.10.100:5070 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.100:5070;branch=z9hG4bK8f4f32f60
From: Anonymous <sip:Anonymous@10.10.10.100>;tag=581d6a1c
To: <sip:2687069@10.9.4.11;user=phone>;tag=5a4b2706a4
Call-ID: cc502cb22e7a31344ad15ac96e586928@10.10.10.100
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:2687069@10.9.4.11
Content-Length: 0

Автор:  Geniu$$ [ 05 мар 2010, 08:53 ]
Заголовок сообщения: 

Попробуйте перепрошить
С питанием все нормально?

Автор:  OlegZeml [ 05 мар 2010, 09:12 ]
Заголовок сообщения: 

С питание все нормально :(
Прошивка 8_30U

Автор:  OlegZeml [ 23 мар 2010, 09:02 ]
Заголовок сообщения: 

В общем проблема решилась заменой блока питания....

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