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GS 1002 помогите настроить звонок с сипа на gsm
http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=2368
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Автор:  Repear [ 04 мар 2011, 04:58 ]
Заголовок сообщения:  GS 1002 помогите настроить звонок с сипа на gsm

Специалисты, помогите пожалуйста
Пытаюсь прозвониться с сипа на gsm звонок приходит на шлюз там выбирается gsm а дальше не идет, в итоге потом возвращается на сип на тот же номер
(192.168.0.185 Voip сервер, 192.168.0.200 Addpac

debug voip call и sip

Код:
Received SIP PDU from ( 192.168.0.185:5060 )
INVITE sip:89182010597@192.168.0.200 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>
Call-ID: 5c9973664db548a7b8242318818135bb
User-Agent: Infra Call Center Server 4.0.636.187
Contact: <sip:192.168.0.185:5060;transport=udp>
CSeq: 2 INVITE
Allow: INVITE,CANCEL,ACK,BYE,OPTIONS
Content-Type: application/sdp
Content-Length: 192

v=0
o=- 22190 1843 IN IP4 192.168.0.185
s=-
c=IN IP4 192.168.0.185
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


        [1964.301] Sending SIP PDU to ( 192.168.0.185:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


38      <Call   2>      : ******  Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(1299114905) ****
39      <SIP    2>      : Receive INVITE Request
40      <NetCon 2>      : Found inbound voip peer by IP address id(100)
41      <Call   2>      : From Net - calledParty(89182010597) callingParty(102)
42      <Call   2>      : MatchedAll
43      <Call   2>      : MatchAllProcess After Sorted
                          <0>  id(4584) dest([78]9[18]T) prefer(0) selected(0)
                          <1>  id(3561) dest(T) prefer(0) selected(0)
                          <2>  id(3560) dest(T) prefer(0) selected(1)
44      <Call   2>      : Initiate callee with dial-peer([78]9[18]T) status(Call
eeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
45      <CEP    000000> : InitiateOutCall :  calledNum(89182010597), callingNum(
102), callerPort(ffffffff) type(GSM)
46      <CEP    000000> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(2)
47      <SIP    2>      : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE
)
48      <SIP    2>      : SetLocalAudioFormats : myVoipPeer(100) is not NULL, co
dec(0)
49      <PhonePlay      2>      : Audio Count(1)
50      <PhonePlay      2>      : rtpSessionId(1) Second Audio Port(-1)
51      <SIP    2>      : SetAlerting
52      <Call   2>      : PreConnected from(0)
53      <SIP    2>      : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE
)
54      <SIP    2>      : SetLocalAudioFormats : myVoipPeer(100) is not NULL, co
dec(0)
55      <SIP    2>      : Add Local Audio MediaFormat : 8

        [1964.325] Sending SIP PDU to ( 192.168.0.185:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>;tag=994d3001a4
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:89182010597@192.168.0.200
Content-Type: application/sdp
Content-Length: 196

v=0
o=89182010597 1299114905 1299114905 IN IP4 192.168.0.200
s=AddPac Gateway SDP
c=IN IP4 192.168.0.200
t=1299114905 0
m=audio 23004 RTP/AVP 8
a=ptime:20
a=rtpmap:8 PCMA/8000
a=sendrecv

56      <Time   2>      : Call Forwarding No Answer timer timeout.
57      <CEP    000000> : Disconnected(16) at Busy
58      <Call   2>      : Terminated from(0) this(Local:CallClear) before((null)
) forced(0) time(1299114922)

        [1981.518] Sending SIP PDU to ( 192.168.0.185:5060 ) from 5060
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>;tag=994d3001a4
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


59      <NetEP  2>      : Call FROM <Oper2> terminated reason(Local:CallClear)
60      <CEP    000000> : DisconnectCall at Idle

        Received SIP PDU from ( 192.168.0.185:5060 )
ACK sip:89182010597@192.168.0.200 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
User-Agent: Infra Call Center Server 4.0.636.187
To: <sip:89182010597@192.168.0.200>;tag=994d3001a4
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 ACK
Content-Length: 0


61      <SIP    2>      : Receive ACK Request
62      <SIP    2>      : Set Terminated Success for 2 INVITE


config
Код:
Welcome, APOS(tm) Kernel Version 8.51.002.
Copyright (c) 1999-2010 AddPac Technology Co., Ltd.

Login:
Login: root
Password:
GS1002> enable
GS1002# show running-config
Building configuration...

Current configuration:
!
version 8.51.002
!
hostname GS1002
!
username root password router administrator
username guest password guest user
!
!
interface Loopback0
 ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
 ip address 192.168.0.200 255.255.255.0
 speed auto
 no qos-control
!
interface FastEthernet0/1
 ip address 192.168.10.1 255.255.255.0
 speed auto
 no qos-control
!
ip route 0.0.0.0 0.0.0.0 192.168.0.1 10
!
!
!
!
http server
!
logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
 protocol sip
 dtmf-relay out-of-band
 fax protocol t38 redundancy 0
 fax rate disable
 h323 call start fast
 h323 call tunnel enable
 no call-barring unconfigured-ip-address
 no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
 connection plar 79881413242
 translate-incoming called-number 900
 caller-id enable
!
!
! GSM
voice-port 0/1
 connection plar 79284452546
 translate-incoming called-number 1
 caller-id enable
!
!
! FXO
voice-port 0/2
 connection plar 78622649060
 ring detect-timeout 50
 ring detect-timer 500
 caller-id enable
 caller-id type etsi
 caller-id name disable
 shutdown
!
!
! FXO
voice-port 0/3
 connection plar 78622644314
 ring detect-timeout 50
 ring detect-timer 500
 caller-id enable
 caller-id type etsi
 caller-id name disable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
 shutdown
!
dial-peer voice 3560 pots
 destination-pattern T
 port 0/2
!
dial-peer voice 3561 pots
 destination-pattern T
 port 0/3
!
dial-peer voice 4584 pots
 destination-pattern [78]9[18]T
 port 0/0
!
dial-peer voice 4585 pots
 destination-pattern 79[2345679]T
 port 0/1
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
 destination-pattern 7T
 session target ip 192.168.0.185
 session protocol sip
 codec g711alaw
 no vad
 dtmf-relay info
!
dial-peer voice 101 voip
 destination-pattern 79881413242
 session target ip 192.168.0.185
 session protocol sip
 codec g711alaw
 no vad
 dtmf-relay info
!
dial-peer voice 102 voip
 destination-pattern 79284452546
 session target ip 192.168.0.185
 session protocol sip
 codec g711alaw
 no vad
 dtmf-relay info
!
dial-peer voice 103 voip
 destination-pattern 78622649060
 session target ip 192.168.0.185
 session protocol sip
 codec g711alaw
 no vad
 dtmf-relay info
!
dial-peer voice 104 voip
 destination-pattern 78622644314
 session target ip 192.168.0.185
 session protocol sip
 codec g711alaw
 no vad
 dtmf-relay info
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.0.200
 no ignore-msg-from-other-gk
!
!
! Translation Rule configuration.
!
translation-rule 1
 rule 1      T                        111T
!
translation-rule 2
 rule 1      111T                     T
!
translation-rule 3
 rule 0      7T                       8T
!
translation-rule 900
 rule 0      T                        79881413242
!
!
!
! SIP UA configuration.
!
sip-ua
!
!
! Tones
voice class clear-down-tone 0 432 0 352 346
!
!
!
!
!
line console
!
line vty
!
gsm dev-restart-by-unreg 300
!
gsm 0/0
 sms-language utf8
!
gsm 0/1
 sms-language utf8
!
end

Автор:  Yuri [ 04 мар 2011, 07:07 ]
Заголовок сообщения:  Re: GS 1002 помогите настроить звонок с сипа на gsm

У Вас sim-карты в сети зарегистрированы? На вход из GSM работает? Посмотрите отладку GSM-части debug gsm all-ports call.

Автор:  Repear [ 04 мар 2011, 13:06 ]
Заголовок сообщения:  Re: GS 1002 помогите настроить звонок с сипа на gsm

Спасибо разобрался, посмотрел что все идет поменял симку в мтс на новую все заработало, какой то глюк с мобильного телефона звонки через нее шли, и с fxo тоже шайтан однако :)
Код:
[162605.130] GSM-0/0-DEV: DIALING
[162606.130] GSM-0/0-DEV: REMOTE RINGING
906     <Time   24>     : Call Forwarding No Answer timer timeout.
[162621.330] GSM-0/0-DEV: HOOK ON
[162621.430] GSM-0/0: CALL REQUEST FAILED, status(-1), err(0), phone failure

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