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AddPack AP-IP100 - а в ответ тишина...
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Автор:  Valerson [ 10 сен 2008, 03:42 ]
Заголовок сообщения:  AddPack AP-IP100 - а в ответ тишина...
Помогите разобраться с проблемой.
телефон AddPac AP-IP100, 100B (пробовал 4 шт.)

Звонок идёт на телефон, поднимаю трубку, тишина, через 6 сек, отбой. Хотя на другом конце идёт вызов, как будто трубу не сняли и не отбились. На программный телефон звоню, всё нормально.

Телефон не за NAT-ом, в той же сети, что шлюз Dialogic
DMG 2030
SIP-сервер поднят на CommuniGate Pro 5

Версия прошивки:
Welcome, APOS(tm) Kernel Version 8.41.047.
Copyright (c) 1999-2006 AddPac Technology Co., Ltd.

вот что пишет debug время этого звонка:
IP100# debug voip call
IP100# 1 <Call> : ****** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(340485) ****
2 <SIP> : Receive INVITE Request
3 <NetCon> : Found inbound voip peer by dest-pattern id(1001)
4 <Call> : From Net - calledParty(303) callingParty(04101)
5 <Call> : MatchedPerfect
6 <Call> : MatchAllProcess After Sorted <0> id(0) dest(303) prefer(0) selected(25)
7 <Call> : Initiate callee with dial-peer(303) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
8 <CEP> : InitiateOutCall : calledNum(), callingNum(04101), callerPort(ffffffff) type(SPEECH)
9 <CEP> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(33)
10 <SIP> : SetAlerting
11 <SIP> : Add Local Audio MediaFormat : 8
12 <SIP> : Receive PRACK Request
13 <Time> : SIP_TREGISTER timer timeout.
14 <SIP> : Adding authentication information
15 <SIP> : Send REGISTER Request
16 <SIP> : Receive 200 OK
17 <SIP> : Transaction (11772 REGISTER) completed
18 <CCA> : Call Connect Request from HANDSET
19 <Call> : Connected from(0)
20 <CEP> : StopSignal
21 <SIP> : SetConnected
22 <SIP> : Transaction Server (1 INVITE) Timeout (retry #1)
23 <SIP> : Send 200 Response
24 <SIP> : Set Terminated Success for 11772 REGISTER
25 <SIP> : Transaction Server (1 INVITE) Timeout (retry #2)
26 <SIP> : Send 200 Response
27 <SIP> : Transaction Server (1 INVITE) Timeout (retry #3)
28 <SIP> : Send 200 Response
29 <SIP> : Set Terminated Retries Exceeded for 1 INVITE
30 <Call> : Terminated from(fffffffe) this(Local:NoConnectFromDestination) before(NULL) forced(0) time(340503)
31 <CEP> : DisconnectCall at Busy
32 <CEP> : StopSignal
33 <CEP> : Disconnect (0)
34 <CCA> : Call Disconnected from fffffffe (42)
35 <NetEP> : Call FROM <04101> terminated reason(Local:NoConnectFromDestination)
36 <CCA> : Call Disconnect Request from HANDSET
37 <CEP> : Disconnected(16) at Disconnecting

SIP-сервер поднят на CommuniGate Pro.
Автор:  Denis [ 10 сен 2008, 09:08 ]
Заголовок сообщения:  Re: AddPack AP-IP100 - а в ответ тишина...
Пришлите конфиг телефона. Что происходит при исходящем звонке?
Автор:  Valerson [ 10 сен 2008, 23:29 ]
Заголовок сообщения:  Re: AddPack AP-IP100 - а в ответ тишина...
Denis писал(а):
Пришлите конфиг телефона. Что происходит при исходящем звонке?




Исходящие звонки проходят нормально.


Welcome, APOS(tm) Kernel Version 8.41.047.
Copyright (c) 1999-2006 AddPac Technology Co., Ltd.

User Access Verification

IP100# sh ru
Building configuration...

Current configuration:
!
version 8.41.047
!
hostname IP100
!
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 10.14.22.2 255.255.255.0
bridge-group 1
speed auto
no qos-control
!
interface FastEthernet0/1
no ip address
bridge-group 1
speed auto
no qos-control
!
no ip routing
ip route 0.0.0.0 0.0.0.0 10.14.22.1
!
!
!
snmp name IP100_G2
!
!
ftp server
http server
!
!
!
!
! IP PHONE OSD configuration.
!
osd
language english
network signaling sip
network sscp disable
network lan-dhcp dhcp-bridge
phone lcd-type graphic
phone ring-type 1
phone volume ring 4
phone volume input 5
phone volume output 5
phone volume micbooster disable
phone auto-hook-on disable
phone display-name AP-IP100
phone voice-codec 0
phone dnd-mode silence
phone pbx-mode general
phone auto-answer disable
phone save-mode always
phone forward-status disable
phone conference-status enable
phone factory-default-password NONE
phone factory-default-password-status 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
h323 call start fast
h323 call channel early
h323 call tunnel enable
translate-voip-incoming called-number 0
translate-voip-incoming calling-number 0
h323 call response alert
voice-confirmed-connect 25
timeout tttl 20
timeout tmohdt 300
local-ringback-tone early
inband-ringback-tone
static-jitter-buffer 200
!
!
! Voice port configuration.
!
! SPEECH
voice-port 0/0
!
!
! FXS
voice-port 0/1
caller-id enable
!

! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 303
port 0/0
!
! Voip peer configuration.
!
dial-peer voice 1001 voip
destination-pattern T
session target sip-server
clid network-number 04303
session protocol sip
voice-class codec 0
vad
dtmf-relay rtp-2833
huntstop
!
dial-peer voice 1002 voip
destination-pattern T
session target ras
voice-class codec 0
vad
dtmf-relay rtp-2833
preference 1
huntstop
!
dial-peer voice 3000 voip
destination-pattern T
session protocol sip
no vad
dtmf-relay rtp-2833
description localconference
preference 1
!
dial-peer call-hold h
dial-peer call-transfer h
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.14.22.2
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
codec preference 4 g7231r63
!
! SIP UA configuration.
!
sip-ua
user-register
sip-username jit
sip-password jit
sip-server 192.168.1.111
retry-counter 3
rport enable
call-transfer-mode attended
register e164
3way-conference local
fault-tolerance 3 500
!
!
! MGCP configuration.
!
mgcp
codec g711alaw
vad
!
! Tones
!
line console
!
line vty
!
!
sms
quota 30
!
end


вот что происходит при входящем звонке, с проблемой описанной выше, кодеки пробовал менять различные и на шлюзе и на телефоне - не помогло:

IP100# deb rta ipc

[71397.920] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 RING_REQ 128
[71397.920] VP(0/0/0) use speaker
[71397.920] VP(0/1/0) add speaker
[71397.920] VP(0/0/0) enable AEC
[71397.920] VP(0/0/0) open channel
[71397.920] VM(0/0/0) ring play start PhoneBell
[71397.920] VM(0/0/0) set T38 mode STD
[71397.920] VM(0/0/0) Fax rate disab
[71413.255] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
OFF_HOOK 0(handset)
[71413.255] VM(0/0/0) ring play stop
[71413.255] VP(0/0/0) close channel
[71413.255] VP(0/0/0) use handset
[71413.255] VP(0/1/0) add handset
[71413.255] VP(0/0/0) enable AEC
[71413.255] VP(0/0/0) open channel
[71413.255] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[71413.255] VM(0/0/0) play mute
[71413.255] VP(0/0/0) Tx IBS signal 2/0
[71413.255] VP(0/0/0) Tx IBS dir 0
[71413.255] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 1
[71413.255] VM(0/0/0) VAD enable
[71413.255] VM(0/0/0) SID enable by CCC
[71413.260] RTA(0/0/0) Rx RS_LISTEN_REQ callId=46 ssId=1 G711U
peer=10.14.22.77 mp=23088/23089 hp=49028/49029
[71413.260] VM(0/0/0) codec same G711U
[71413.260] RTA(0/0/0) Rx RS_OPEN_REQ callId=46 ssId=1 G711U
peer=10.14.22.77 mp=23088/23089 hp=49028/49029
[71413.260] VM(0/0/0) codec same G711U
[71413.265] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[71413.265] VM(0/0/0) DTMF enable
[71413.265] VM(0/0/0) DTMF_RTP_RFC2833 enable
[71413.265] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65
[71413.320] VP(0/0/0) GeneralEvent IBS gen end
[71420.710] RTA(0/0/0) Rx RS_CLOSE_REQ callId=46 ssId=1 dir=reve
[71420.710] RTA(0/0/0) Rx RS_CLOSE_REQ callId=46 ssId=1 dir=forw
[71420.710] RTA(0/0/0) close Media socket
[71420.710] RTA(0/0/0) close RTCP socket
[71420.710] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[71420.710] VM(0/0/0) play mute
[71420.710] VP(0/0/0) Tx IBS signal 2/0
[71420.710] VP(0/0/0) Tx IBS dir 0
[71420.710] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[71420.710] VM(0/0/0) play Reorder tone
[71420.710] VP(0/0/0) Tx IBS signal 6/3
[71420.710] VP(0/0/0) Tx IBS dir 0
[71443.450] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
ON_HOOK
[71443.450] VP(0/0/0) use none
[71443.450] VP(0/0/0) close channel
[71443.455] VM(0/0/0) Tx DISCONN_CNF53 <CEP 000000> : Disconnected(16) at Disconnecting
[71443.455] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[71443.455] VM(0/0/0) Tx DISCONN_CNF
Автор:  Denis [ 12 сен 2008, 10:29 ]
Заголовок сообщения:  Re: AddPack AP-IP100 - а в ответ тишина...
Ну первым делом вам следует удалить всё лишнее. Зачем 3 voip dial-peera, если вы используете только один? Оставьте только 1001, через который идут звонки. Удалите команду user-reg, она используется если логин и пароль прописаны в конкретном dial-p pots. То есть уберите из конфигурации всё, что не используете. После этого попробуйте принять звонок. Если не поможет пришлите одновременно дебаги: deb voip call, deb rta ipc и deb voip sip.
Автор:  Valerson [ 14 сен 2008, 21:28 ]
Заголовок сообщения: 
Все сделал - не помогло, вот debag

Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


241 <Call 7> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(84369) ****
242 <SIP 7> : Receive INVITE Request
243 <SIP 6> : Transaction (1 INVITE) aborted
244 <NetCon 7> : Found inbound voip peer by dest-pattern id(1001)
245 <Call 7> : From Net - calledParty(303) callingParty(04101)
246 <Call 7> : MatchedPerfect
247 <Call 7> : MatchAllProcess After Sorted
<0> id(0) dest(303) prefer(0) selected(2)
248 <Call 7> : Initiate callee with dial-peer(303) status(CalleeDeter
minedAll) id(00000000-0000-0000-0000-000000000000)
249 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(04101), cal
lerPort(ffffffff) type(SPEECH)
[84377.515] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
RING_REQ 128
[84377.515] VP(0/0/0) use speaker
[84377.515] VP(0/1/0) add speaker
[84377.515] VP(0/0/0) enable AEC
[84377.515] VP(0/0/0) open channel
[84377.515] VM(0/0/0) ring play start PhoneBell
250 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(7)
[84377.515] VM(0/0/0) set T38 mode STD
[84377.515] VM(0/0/0) Fax rate disab
251 <SIP 7> : SetAlerting
252 <SIP 7> : Add Local Audio MediaFormat : 0

Sending SIP PDU to ( 192.168.1.111:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK20;rport=5060
Via: SIP/2.0/TCP 10.14.22.77:5060;branch=z9hG4bK2D40C97DF7D2B7F9F818D0FACF1D3425
From: <sip:04101@192.168.1.111:5060>;tag=077832463135364100328F3C;vnd.pimg.port=
30
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 2 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:303@10.14.22.2
RSeq: 548783
Require: 100rel
Content-Type: application/sdp
Content-Length: 211

v=0
o=303 84369 84369 IN IP4 10.14.22.2
s=AddPac Gateway SDP
c=IN IP4 10.14.22.2
t=0 0
m=audio 23012 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15


Received SIP PDU from ( 10.14.22.77:1048 )
PRACK sip:303@10.14.22.2 SIP/2.0
RAck:548783 2 INVITE
To:<sip:303@192.168.1.111>;tag=91000808a4
From:<sip:04101@192.168.1.111:5060>;vnd.pimg.port=30;tag=077832463135364100328F3
C
Call-ID:01B22C1ADC8140000000280E@192.168.1.111
CSeq:3 PRACK
Max-Forwards:70
User-Agent:PBX-IP Media Gateway
Via:SIP/2.0/UDP 10.14.22.77:5060;branch=z9hG4bKC2A23DAF3D188FCE48092B66778BD85A
Content-Length:0


253 <SIP 7> : Receive PRACK Request

Sending SIP PDU to ( 10.14.22.77:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.14.22.77:5060;branch=z9hG4bKC2A23DAF3D188FCE48092B66778BD85A
From: <sip:04101@192.168.1.111:5060>;vnd.pimg.port=30;tag=077832463135364100328F
3C
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 3 PRACK
User-Agent: AddPac SIP Gateway
Content-Length: 0


254 <CCA 0> : Call Connect Request from HANDSET
[84378.265] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
OFF_HOOK 0(handset)
[84378.265] VM(0/0/0) ring play stop
[84378.265] VP(0/0/0) close channel
[84378.265] VP(0/0/0) use handset
[84378.265] VP(0/1/0) add handset
[84378.265] VP(0/0/0) enable AEC
[84378.265] VP(0/0/0) open channel
255 <Call 7> : Connected from(0)
256 <CEP 000000> : StopSignal
[84378.265] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[84378.265] VM(0/0/0) play mute
[84378.265] VP(0/0/0) Tx IBS signal 2/0
[84378.265] VP(0/0/0) Tx IBS dir 0
[84378.265] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 1
[84378.265] VM(0/0/0) VAD enable
[84378.265] VM(0/0/0) SID enable by CCC
257 <SIP 7> : SetConnected

Sending SIP PDU to ( 192.168.1.111:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK20;rport=5060
Via: SIP/2.0/TCP 10.14.22.77:5060;branch=z9hG4bK2D40C97DF7D2B7F9F818D0FACF1D3425
From: <sip:04101@192.168.1.111:5060>;tag=077832463135364100328F3C;vnd.pimg.port=
30
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 2 INVITE
Supported: timer, replaces, early-session
Session-Expires: 1800;refresher=uac
User-Agent: AddPac SIP Gateway
Contact: sip:303@10.14.22.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Require: timer
Content-Length: 0


[84378.270] RTA(0/0/0) Rx RS_LISTEN_REQ callId=7 ssId=1 G711U
peer=10.14.22.77 mp=23012/23013 hp=49046/49047
[84378.270] VM(0/0/0) codec same G711U
[84378.275] RTA(0/0/0) Rx RS_OPEN_REQ callId=7 ssId=1 G711U
peer=10.14.22.77 mp=23012/23013 hp=49046/49047
[84378.275] VM(0/0/0) codec same G711U
[84378.275] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[84378.275] VM(0/0/0) DTMF enable
[84378.275] VM(0/0/0) DTMF_RTP_RFC2833 enable
[84378.275] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65
[84378.340] VP(0/0/0) GeneralEvent IBS gen end
258 <SIP 7> : Transaction Server (2 INVITE) Timeout (retry #1)
259 <SIP 7> : Send 200 Response

Sending SIP PDU to ( 192.168.1.111:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK20;rport=5060
Via: SIP/2.0/TCP 10.14.22.77:5060;branch=z9hG4bK2D40C97DF7D2B7F9F818D0FACF1D3425
From: <sip:04101@192.168.1.111:5060>;tag=077832463135364100328F3C;vnd.pimg.port=
30
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 2 INVITE
Supported: timer, replaces, early-session
Session-Expires: 1800;refresher=uac
User-Agent: AddPac SIP Gateway
Contact: sip:303@10.14.22.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Require: timer
Content-Length: 0


260 <SIP 7> : Transaction Server (2 INVITE) Timeout (retry #2)
261 <SIP 7> : Send 200 Response

Sending SIP PDU to ( 192.168.1.111:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK20;rport=5060
Via: SIP/2.0/TCP 10.14.22.77:5060;branch=z9hG4bK2D40C97DF7D2B7F9F818D0FACF1D3425
From: <sip:04101@192.168.1.111:5060>;tag=077832463135364100328F3C;vnd.pimg.port=
30
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 2 INVITE
Supported: timer, replaces, early-session
Session-Expires: 1800;refresher=uac
User-Agent: AddPac SIP Gateway
Contact: sip:303@10.14.22.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Require: timer
Content-Length: 0


262 <SIP 7> : Transaction Server (2 INVITE) Timeout (retry #3)
263 <SIP 7> : Send 200 Response

Sending SIP PDU to ( 192.168.1.111:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK20;rport=5060
Via: SIP/2.0/TCP 10.14.22.77:5060;branch=z9hG4bK2D40C97DF7D2B7F9F818D0FACF1D3425
From: <sip:04101@192.168.1.111:5060>;tag=077832463135364100328F3C;vnd.pimg.port=
30
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 2 INVITE
Supported: timer, replaces, early-session
Session-Expires: 1800;refresher=uac
User-Agent: AddPac SIP Gateway
Contact: sip:303@10.14.22.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Require: timer
Content-Length: 0


264 <Time 0> : SIP_TREGISTER timer timeout.
265 <SIP 0> : Adding authentication information
266 <SIP 3628> : Send REGISTER Request

Sending SIP PDU to ( 192.168.1.111:5060 ) from 5060
REGISTER sip:192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 10.14.22.2:5060;branch=z9hG4bK0900f100a43628
From: <sip:303@192.168.1.111>;tag=0900f100a4
To: sip:303@192.168.1.111
Call-ID: 09000000-e91e-f167-8000-0002a404f68a@10.14.22.2
CSeq: 3628 REGISTER
Date: Thu, 01 Jan 1970 23:26:17 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="jit", realm="arka.ru", nonce="D848601ED32BA
1B7698F", opaque="opaqueData", uri="sip:192.168.1.111", qop=auth, nc=00000001, c
nonce="5d67a195", response="40222104ce19f959414039ae4b8e812f", algorithm=MD5
Contact: <sip:303@10.14.22.2>;expires=30
Expires: 30
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 192.168.1.111:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.14.22.2:5060;branch=z9hG4bK0900f100a43628
From: <sip:303@192.168.1.111>;tag=0900f100a4
To: <sip:303@192.168.1.111>;tag=DF91AD67
Call-ID: 09000000-e91e-f167-8000-0002a404f68a@10.14.22.2
CSeq: 3628 REGISTER
Expires: 30
Contact: <sip:303@10.14.22.2>;expires=30
Event: registration
Date: Sun, 14 Sep 2008 21:11:15 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path
Allow-Events: presence,message-summary,reg,dialog,keep-alive,refer
Server: CommuniGatePro/5.1.6
Content-Length: 0


267 <SIP 3628> : Receive 200 OK
268 <SIP 3628> : Transaction (3628 REGISTER) completed
269 <SIP 7> : Set Terminated Retries Exceeded for 2 INVITE
[84385.710] RTA(0/0/0) Rx RS_CLOSE_REQ callId=7 ssId=1 dir=reve
[84385.710] RTA(0/0/0) Rx RS_CLOSE_REQ callId=7 ssId=1 dir=forw
[84385.710] RTA(0/0/0) close Media socket
[84385.710] RTA(0/0/0) close RTCP socket
270 <Call 7> : Terminated from(fffffffe) this(Local:NoConnectFromDest
ination) before(NULL) forced(0) time(84378)
271 <CEP 000000> : DisconnectCall at Busy
272 <CEP 000000> : StopSignal
[84385.710] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[84385.710] VM(0/0/0) play mute
[84385.710] VP(0/0/0) Tx IBS signal 2/0
[84385.710] VP(0/0/0) Tx IBS dir 0
273 <CEP 000000> : Disconnect (0)
[84385.710] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[84385.710] VM(0/0/0) play Reorder tone
[84385.710] VP(0/0/0) Tx IBS signal 6/3
[84385.710] VP(0/0/0) Tx IBS dir 0
274 <CCA 0> : Call Disconnected from fffffffe (42)
275 <NetEP 7> : Call FROM <04101> terminated reason(Local:NoConnectFro
mDestination)

Received SIP PDU from ( 10.14.22.77:1048 )
BYE sip:303@10.14.22.2 SIP/2.0
Reason:E.182;text="Normal"
To:<sip:303@192.168.1.111>;tag=91000808a4
From:<sip:04101@192.168.1.111:5060>;vnd.pimg.port=30;tag=077832463135364100328F3
C
Call-ID:01B22C1ADC8140000000280E@192.168.1.111
CSeq:4 BYE
Max-Forwards:70
User-Agent:PBX-IP Media Gateway
Via:SIP/2.0/UDP 10.14.22.77:5060;branch=z9hG4bKE7A9A9CACCEA7E19FAA862D4B1998E10
Content-Length:0



Sending SIP PDU to ( 10.14.22.77:5060 ) from 5060
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.14.22.77:5060;branch=z9hG4bKE7A9A9CACCEA7E19FAA862D4B1998E10
From: <sip:04101@192.168.1.111:5060>;vnd.pimg.port=30;tag=077832463135364100328F
3C
To: <sip:303@192.168.1.111>;tag=91000808a4
Call-ID: 01B22C1ADC8140000000280E@192.168.1.111
CSeq: 4 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0
Автор:  Mi1ovidoff [ 15 сен 2008, 11:18 ]
Заголовок сообщения: 
Valerson писал(а):
Все сделал - не помогло


Valerson, будьте добры, покажите вашу конечную конфигурацию на APешке, после всех удалений-исправлений.
Автор:  Valerson [ 16 сен 2008, 04:01 ]
Заголовок сообщения: 
Welcome, APOS(tm) Kernel Version 8.41.047.
Copyright (c) 1999-2006 AddPac Technology Co., Ltd.

IP100# sh ru
Building configuration...

Current configuration:
!
version 8.41.047
!
hostname IP100
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 10.14.22.2 255.255.255.0
bridge-group 1
speed auto
no qos-control
!
interface FastEthernet0/1
no ip address
bridge-group 1
speed auto
no qos-control
!
no ip routing
ip route 0.0.0.0 0.0.0.0 10.14.22.1
!
!
!
snmp name IP100_G2
!
!
ftp server
http server
!
!
!
!
! IP PHONE OSD configuration.
!
osd
language english
network signaling sip
network sscp disable
network lan-dhcp dhcp-bridge
phone lcd-type graphic
phone ring-type 1
phone volume ring 1
phone volume input 8
phone volume output 6
phone volume micbooster disable
phone auto-hook-on disable
phone display-name AP-IP100
phone voice-codec 0
phone dnd-mode silence
phone pbx-mode general
phone auto-answer disable
phone save-mode always
phone forward-status disable
phone conference-status enable
phone factory-default-password NONE
phone factory-default-password-status 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
h323 call start fast
h323 call channel early
h323 call tunnel enable
translate-voip-incoming called-number 0
translate-voip-incoming calling-number 0
h323 call response alert
voice-confirmed-connect 25
timeout tttl 20
timeout tmohdt 300
local-ringback-tone early
inband-ringback-tone
static-jitter-buffer 200
rtp-nat-pat
!
!
! Voice port configuration.
!
! SPEECH
voice-port 0/0
no comfort-noise
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 303
port 0/0
!
!
!
! Voip peer configuration.
!
dial-peer voice 1001 voip
destination-pattern T
session target sip-server
clid network-number 04303
session protocol sip
voice-class codec 0
vad
dtmf-relay rtp-2833
huntstop
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.14.22.2
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729
codec preference 4 g7231r63
!
!
!
! SIP UA configuration.
!
sip-ua
sip-username jit
sip-password jit
sip-server 192.168.1.111
retry-counter 3
rport enable
call-transfer-mode attended
register e164
3way-conference local
fault-tolerance 3 500
!
!
! MGCP configuration.
!
mgcp
codec g711alaw
vad
!
!
! Tones
!
!
!
!
line console
!
line vty
!
!
sms
quota 30
!
end
IP100#
Автор:  Geniu$$ [ 16 сен 2008, 13:26 ]
Заголовок сообщения: 
1. У Вас SIP сервер в другой сети, включите маршрутизацию на шлюзе.
conf t
ip routing

2. conf t
sip
no rel
Автор:  Valerson [ 16 сен 2008, 21:41 ]
Заголовок сообщения: 
Спасибо всё заработало, маршрутизацию включил на телефоне!
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