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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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