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Факсы ходят только в одну сторону
http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=2498
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Автор:  Silver-D [ 27 сен 2011, 15:24 ]
Заголовок сообщения:  Факсы ходят только в одну сторону

Всем добрый день.
Столкнулся с проблемой.
Имеем AP200B и SIP.

Проблема в том, что факсы ходят только в одну сторону. Т.е. я могу послать факс со шлюза на городской телефон (FAX=>шлюз=>SIP=>городской тел=>FAX). А обратно (FAX=>городской тел=>SIP=>шлюз=>FAX) факсы не идут.

Происходит это так:
- набираем номер на факсе городского тел - факс (1).
- начинает звонить факс на шлюзе - факс (2)
- факс (2) отвечает на звонок и начинает пищать для соединения
- факс (1) и факс (2) пишет, что -=обнаружен ответ факса=-
- и на этом этапе шлюз, вешает линию. Просто берет и отрубает.

Что пробовал делать:
- менял факсы местами (1) и (2). Без изменений. Т.е. (FAX=>городской тел=>SIP=>шлюз=>FAX) факсы не идут. Шлюз вешает трубку.
- всячески менял настройки шлюза. Ничего не помогает.

Провайдер Т38 не поддерживает. Факсы только по g711.

Конфиг:
Цитата:
version 8.30V
!
hostname AP200
!
!
no bridge spanning-tree
!
dhcp-list 0 type server
dhcp-list 0 address server interface ether0.0
dhcp-list 0 option dhcp-lease-time 7200
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.254 255.255.255.0
dhcp-list 1 option dhcp-lease-time 600
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.1.20 255.255.255.0
!
interface ether1.0
no ip address
!
snmp name AP200B
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.1.1
!
dnshost nameserver 192.168.1.1
!
pnp-sktelink debug on
!
auto-script autorun.inf
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol bypass
fax rate 9600
h323 call start fast
h323 call tunnel enable
announcement language english
busyout monitor gatekeeper
busyout monitor sip-server
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
low-dtmf-gain 0
high-dtmf-gain 0
caller-id enable
caller-id type etsi-dtmf
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 14924
port 0/0
!
!
!
! Voip peer configuration.
!
dial-peer voice 2000 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.192.168.1.20
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
voice class codec 1000
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
! SIP UA configuration.
!
sip-ua
sip-username 14924
sip-password xxxxxxxxx
sip-server 188.134.000.243
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711alaw
vad
!
!
! Tones
!
!
!
!


Debug:
Цитата:
17<Call1>: ****************** Call Created status(InitiatedByNet) *******************
18<SIP1>: Receive INVITE Request
19<NetCon1>: Found inbound voip peer by dest-pattern id(2000)
20<Call1>: From Net - calledParty(14924) callingParty(+781274xxxxx)
21<Call1>: MatchedPerfect
22<Call1>: MatchAllProcess After Sorted
<0> id(0) dest(14924) prefer(0) selected(0)
23<Call1>: Initiate callee with dial-peer(14924) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
24<CEP000000>: InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS)

[814.800] RTA(0/0/0) Rx CC_RING_REQ [80 ] peerId(-1)
[814.800] VM(0/0/0) Rx RingReq CID msg ERR
[814.800] VM(0/0/0) Line Reverse
[814.800] VM(0/0/0) Start ring actv

25<CEP000000>: Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(1)

[814.800] VM(0/0/0) Fax rate 9600

26<SIP1>: SetAlerting


Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4204e690;rport

From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5
To: <sip:14924@192.168.1.20>;tag=004ee201a4

Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:14924@192.168.1.20
Content-Length: 0


[815.800] VM(0/0/0) Gen ring idle
[817.800] VM(0/0/0) Gen ring actv
[818.800] VM(0/0/0) Gen ring idle
[820.800] VM(0/0/0) Gen ring actv
[821.800] VM(0/0/0) Gen ring idle
[823.800] VM(0/0/0) Gen ring actv
[824.800] VM(0/0/0) Gen ring idle
[826.120] VM(0/0/0) vmOffHook
[826.180] VM(0/0/0) vmTmoOffHook
[826.180] VM(0/0/0) Line Forward
[826.205] VM(0/0/0) vmOnHook
[826.210] VM(0/0/0) vmOffHook
[826.270] VM(0/0/0) vmTmoOffHook
[826.270] VM(0/0/0) Rx OffHook
[826.270] VM(0/0/0) Modem attribute disable
[826.270] VM(0/0/0) Modem attribute G711A
[826.270] VM(0/0/0) vopp enable
[826.270] VM(0/0/0) Tx CONNECT_CNF

27<Call1>: Connected from(0)
[826.270] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0

VAD_CTRL 0

[826.270] VM(0/0/0) VAD disable
[826.270] VM(0/0/0) SID enable by CCC

28<SIP1>: SetConnected
29<SIP1>: Add Local Audio MediaFormat : 8

[826.280] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0

VAD_CTRL 0

[826.280] VM(0/0/0) VAD disable

Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4204e690;rport
From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5
To: <sip:14924@192.168.1.20>;tag=004ee201a4
Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru

CSeq: 102 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:14924@192.168.1.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 236

v=0
o=14924 1317135372 1317135372 IN IP4 192.168.1.20
s=AddPac Gateway SDP
c=IN IP4 192.168.1.20
t=1317135372 0
m=audio 23002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20


[826.320] RTA(0/0/0) Rx RS_LISTEN_REQ callId=1 ssId=1 G711A
peer=188.134.0.243 mp=23002/23003 hp=13480/13481
[826.320] VM(0/0/0) vopp idle
[826.320] VM(0/0/0) start codec replace timer to G711A
[826.320] RTA(0/0/0) Rx RS_OPEN_REQ callId=1 ssId=1 G711A
peer=188.134.0.243 mp=23002/23003 hp=13480/13481
[826.320] VM(0/0/0) under codec replace to G711A
[826.320] VM(0/0/0) V152_modem TxPT=0x64, RxPT=0x64
[826.320] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[826.320] VM(0/0/0) DTMF_RTP_RFC2833 enable
[826.320] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65


Received SIP PDU from ( 188.134.0.243:5060 )
ACK sip:14924@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK5e989f40;rport
From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5
To: <sip:14924@192.168.1.20>;tag=004ee201a4
Contact: <sip:+781274xxxxx@188.134.0.243>
Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru

CSeq: 102 ACK
User-Agent: InterZet PBX
Max-Forwards: 70
Content-Length: 0


30<SIP1>: ACK received
31<SIP1>: Receive ACK Request
32<SIP1>: Set Terminated Success for 102 INVITE
[826.380] VM(0/0/0) Modem attribute disable
[826.380] VM(0/0/0) Modem attribute G711A
[826.380] VM(0/0/0) vopp enable
[826.380] VM(0/0/0) codec replaced to G711A
[826.380] VM(0/0/0) play mute


Received SIP PDU from ( 188.134.0.243:5060 )
BYE sip:14924@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4a7ba1f6;rport
From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5
To: <sip:14924@192.168.1.20>;tag=004ee201a4
Contact: <sip:+781274xxxxx@188.134.0.243>
Call-ID: 5af2be469bb330d5de76daa5dbce3f0@voip.interzet.ru
CSeq: 103 BYE
User-Agent: InterZet PBX
Max-Forwards: 70
Content-Length: 0


33<SIP1>: Receive BYE Request

Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK4a7ba1f6;rport
From: "+781274xxxxx" <sip:+781274xxxxx@voip.interzet.ru>;tag=as779431c5
To: <sip:14924@192.168.1.20>;tag=004ee201a4
Call-ID: 5aaf2be469bb330d5de76daa5dbce3f0@voip.interzet.ru

CSeq: 103 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0


34<SIP1>: ReleaseWithNothing
[834.790] RTA(0/0/0) Rx RS_CLOSE_REQ callId=1 ssId=1 dir=reve
[834.790] RTA(0/0/0) Rx RS_CLOSE_REQ callId=1 ssId=1 dir=forw
[834.790] RTA(0/0/0) close Media socket
[834.790] RTA(0/0/0) close RTCP socket
35<Call1>: Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0)
36<CEP000000>: DisconnectCall at Busy
37<CEP000000>: StopSignal
[834.795] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0

DTMF_STOP

[834.795] VM(0/0/0) play mute
38<CEP000000>: Disconnect (0)
[834.795] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[834.795] VM(0/0/0) play Reorder tone
39<NetEP1>: Call TO <+781274xxxxx> terminated reason(Remote:CallClear)
40<Time0>: SIP_TREGISTER timer timeout.
41<SIP0>: Adding authentication information
42<SIP35>: Send REGISTER Request


Sending SIP PDU to ( 188.134.000.243:5060 ) from 5060
REGISTER sip:188.134.000.243 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a435
From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4
To: sip:14924@188.134.000.243
Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20
CSeq: 35 REGISTER
Date: Tue, 27 Sep 2011 14:56:28 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="14924", realm="voip.interzet.ru", nonce="75f7a428", uri="sip:188.134.000.243", response="cf373d878a7140e5ddaedfea85b859f5", algorithm=MD5
Contact: <sip:14924@192.168.1.20>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70


Received SIP PDU from ( 188.134.0.243:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a435;received=188.134.30.161
From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4
To: sip:14924@188.134.000.243
Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20
CSeq: 35 REGISTER
User-Agent: InterZet PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14924@188.134.0.243>
Content-Length: 0


43<SIP35>: Receive 100 Trying
44<SIP35>: Transaction (35 REGISTER) proceeding

Received SIP PDU from ( 188.134.0.243:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a435;received=188.134.30.161
From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4
To: sip:14924@188.134.000.243;tag=as217b6d45
Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20
CSeq: 35 REGISTER
User-Agent: InterZet PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14924@188.134.0.243>
WWW-Authenticate: Digest realm="voip.interzet.ru", nonce="661f5f4c"
Content-Length: 0


45<SIP35>: Receive 401 Unauthorized
46<SIP35>: Transaction (35 REGISTER) completed
47<SIP0>: No opaque in authentication
48<SIP0>: Adding authentication information
49<SIP36>: Send REGISTER Request

Sending SIP PDU to ( 188.134.000.243:5060 ) from 5060
REGISTER sip:188.134.000.243 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a436
From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4
To: sip:14924@188.134.000.243
Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20
CSeq: 36 REGISTER
Date: Tue, 27 Sep 2011 14:56:28 GMT
User-Agent: AddPac SIP Gateway

Authorization: Digest username="14924", realm="voip.interzet.ru", nonce="661f5f4c", uri="sip:188.134.000.243", response="c44ef2db3c4a29b0f58f3153b0201b06", algorithm=MD5
Contact: <sip:14924@192.168.1.20>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70


Received SIP PDU from ( 188.134.0.243:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a436;received=188.134.30.161
From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4
To: sip:14924@188.134.000.243
Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20
CSeq: 36 REGISTER
User-Agent: InterZet PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14924@188.134.0.243>
Content-Length: 0


50<SIP36>: Receive 100 Trying
51<SIP36>: Transaction (36 REGISTER) proceeding

Received SIP PDU from ( 188.134.0.243:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bKea4e7d00a436;received=188.134.30.161

From: <sip:14924@188.134.000.243>;tag=ea4e7d00a4
To: sip:14924@188.134.000.243;tag=as217b6d45
Call-ID: eae0814e-7290-7d09-8000-0002a40167c2@192.168.1.20

CSeq: 36 REGISTER
User-Agent: InterZet PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:14924@192.168.1.20>;expires=60
Date: Tue, 27 Sep 2011 10:56:21 GMT
Content-Length: 0


52<SIP36>: Receive 200 OK
53<SIP36>: Transaction (36 REGISTER) completed
54<SIP35>: Set Terminated Success for 35 REGISTER
55<SIP36>: Set Terminated Success for 36 REGISTER

[848.095] VM(0/0/0) vmOnHook
[848.145] VM(0/0/0) vmTmoOnHook
[848.195] VM(0/0/0) vmTmoOnHook
[848.245] VM(0/0/0) vmTmoOnHook
[848.295] VM(0/0/0) vmTmoOnHook
[848.345] VM(0/0/0) vmTmoOnHook
[848.395] VM(0/0/0) vmTmoOnHook
[848.445] VM(0/0/0) vmTmoOnHook
[848.495] VM(0/0/0) vmTmoOnHook
[848.545] VM(0/0/0) vmTmoOnHook
[848.595] VM(0/0/0) vmTmoOnHook
[848.645] VM(0/0/0) vmTmoOnHook
[848.695] VM(0/0/0) vmTmoOnHook
[848.745] VM(0/0/0) vmTmoOnHook
[848.795] VM(0/0/0) vmTmoOnHook
[848.795] VM(0/0/0) Rx OnHook
[848.795] VM(0/0/0) vopp idle
[848.795] VM(0/0/0) Tx DISCONN_CNF

56<CEP000000>: Disconnected(16) at Disconnecting

Received SIP PDU from ( 188.134.0.243:5060 )
OPTIONS sip:14924@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK3cd655ea;rport
From: "asterisk" <sip:asterisk@voip.interzet.ru>;tag=as654a36f6
To: <sip:14924@192.168.1.20>
Contact: <sip:asterisk@188.134.0.243>
Call-ID: 419ee2010136040e48f1c6c5626551bc@voip.interzet.ru
CSeq: 102 OPTIONS
User-Agent: InterZet PBX
Max-Forwards: 70
Date: Tue, 27 Sep 2011 10:56:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Sending SIP PDU to ( 188.134.0.243:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.134.0.243:5060;branch=z9hG4bK3cd655ea;rport
From: "asterisk" <sip:asterisk@voip.interzet.ru>;tag=as654a36f6
To: <sip:14924@192.168.1.20>
Call-ID: 419ee2010136040e48f1c6c5626551bc@voip.interzet.ru
CSeq: 102 OPTIONS
User-Agent: AddPac SIP Gateway
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Length: 0

Автор:  craft [ 28 сен 2011, 09:32 ]
Заголовок сообщения:  Re: Факсы ходят только в одну сторону

sip-server 188.134.000.243 - это сервер провайдера?
Если да, то мучать его

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