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| Не проходят звонки на AddPac1000 http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=1250 |
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| Автор: | ДСВ [ 01 мар 2010, 08:15 ] |
| Заголовок сообщения: | Не проходят звонки на AddPac1000 |
Привет всем.В сети есть IP-PBX Asterisk, настроен на SIP.Настроил шлюз AddPac1000, исходящие звонки проходят, а вот на шлюз дозвониться не могут-идут короткие гудки(занято).Вот конфиг: ! 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 1 pots destination-pattern 6025 port 0/0 user-password xxxxxx ! ! ! ! Voip peer configuration. ! dial-peer voice 100 voip destination-pattern T session target 192.168.100.236 session protocol sip codec g711alaw voice-class codec 1 no dtmf-relay no vad ! ! ! ! ! ! ! Gateway configuration. ! gateway h323-id voip.192.168.50.1 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua sip-username 6025 sip-password xxxxxx sip-server 192.168.100.236 register e164 ! ! ! MGCP configuration. ! mgcp codec g711ulaw vad ! ! ! Tones ! ! ! ! Где может быть ошибка? |
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| Автор: | Geniu$$ [ 01 мар 2010, 08:48 ] |
| Заголовок сообщения: | |
dial-peer voice 100 voip session target sip-server посмотрите дебаг, приходит ли что-то на шлюз? deb voip sip deb rta ipc deb voip call conf t deb |
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| Автор: | ДСВ [ 01 мар 2010, 10:44 ] |
| Заголовок сообщения: | |
Geniu$$ писал(а): dial-peer voice 100 voip
session target sip-server посмотрите дебаг, приходит ли что-то на шлюз? deb voip sip deb rta ipc deb voip call conf t deb Сделал debug Received SIP PDU from ( 192.168.100.236:5060 ) OPTIONS sip:6025@192.168.50.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK718e09c9;rport From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as00166edc To: <sip:6025@192.168.50.233> Contact: <sip:Unknown@192.168.100.236> Call-ID: 555cb1a67c3bea8e1118ff08296fd24f@192.168.100.236 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Mar 2010 10:30:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK718e09c9;rport From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as00166edc To: <sip:6025@192.168.50.233> Call-ID: 555cb1a67c3bea8e1118ff08296fd24f@192.168.100.236 CSeq: 102 OPTIONS User-Agent: AddPac SIP Gateway Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) INVITE sip:6025@192.168.50.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233> Contact: <sip:6030@192.168.100.236> Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Mar 2010 10:30:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 428 v=0 o=root 2683 2683 IN IP4 192.168.100.236 s=session c=IN IP4 192.168.100.236 b=CT:384 t=0 0 m=audio 12244 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11300 RTP/AVP 31 34 103 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233> Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233>;tag=4149d201a4 Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Contact: sip:6025@192.168.50.233 Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) CANCEL sip:6025@192.168.50.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233> Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233> Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 CANCEL User-Agent: AddPac SIP Gateway Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233>;tag=4149d201a4 Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) ACK sip:6025@192.168.50.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK73c46299;rport From: "Scherbakov Aleksandr" <sip:6030@192.168.100.236>;tag=as424ddee0 To: <sip:6025@192.168.50.233>;tag=4149d201a4 Contact: <sip:6030@192.168.100.236> Call-ID: 7b828c7167a693960a1b879b4c692ebb@192.168.100.236 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 REGISTER sip:192.168.100.236 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4461 From: <sip:6025@192.168.100.236>;tag=14495200a4 To: sip:6025@192.168.100.236 Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233 CSeq: 461 REGISTER Date: Tue, 07 Apr 2009 23:44:34 GMT User-Agent: AddPac SIP Gateway Authorization: Digest username="6025", realm="asterisk", nonce="3391eb6c", uri=" sip:192.168.100.236", response="02d3f6de146fd2ca1d1752bc3276e53c", algorithm=MD5 Contact: <sip:6025@192.168.50.233>;expires=60 Expires: 60 Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 192.168.100.236:5060 ) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4461;received=192.16 8.50.233 From: <sip:6025@192.168.100.236>;tag=14495200a4 To: sip:6025@192.168.100.236 Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233 CSeq: 461 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:6025@192.168.100.236> Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4461;received=192.16 8.50.233 From: <sip:6025@192.168.100.236>;tag=14495200a4 To: sip:6025@192.168.100.236;tag=as46bc1f4a Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233 CSeq: 461 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e313af3" Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 REGISTER sip:192.168.100.236 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4462 From: <sip:6025@192.168.100.236>;tag=14495200a4 To: sip:6025@192.168.100.236 Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233 CSeq: 462 REGISTER Date: Tue, 07 Apr 2009 23:44:34 GMT User-Agent: AddPac SIP Gateway Authorization: Digest username="6025", realm="asterisk", nonce="6e313af3", uri=" sip:192.168.100.236", response="65ea45a587839b026cae323d49ba0ca0", algorithm=MD5 Contact: <sip:6025@192.168.50.233>;expires=60 Expires: 60 Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 192.168.100.236:5060 ) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4462;received=192.16 8.50.233 From: <sip:6025@192.168.100.236>;tag=14495200a4 To: sip:6025@192.168.100.236 Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233 CSeq: 462 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:6025@192.168.100.236> Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.233:5060;branch=z9hG4bK14495200a4462;received=192.16 8.50.233 From: <sip:6025@192.168.100.236>;tag=14495200a4 To: sip:6025@192.168.100.236;tag=as46bc1f4a Call-ID: 14badb49-bea2-52e1-8000-0002a404bf00@192.168.50.233 CSeq: 462 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: <sip:6025@192.168.50.233>;expires=60 Date: Mon, 01 Mar 2010 10:30:47 GMT Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) NOTIFY sip:6025@192.168.50.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK6b9ac404;rport From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as58124513 To: <sip:6025@192.168.50.233> Contact: <sip:Unknown@192.168.100.236> Call-ID: 6343656f08250546626c47c071b7fffe@192.168.100.236 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: no Message-Account: sip:*97@192.168.100.236 Voice-Message: 0/0 (0/0) Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK6b9ac404;rport From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as58124513 To: <sip:6025@192.168.50.233> Call-ID: 6343656f08250546626c47c071b7fffe@192.168.100.236 CSeq: 102 NOTIFY User-Agent: AddPac SIP Gateway Content-Length: 0 Received SIP PDU from ( 192.168.100.236:5060 ) OPTIONS sip:6025@192.168.50.233 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK280dfde2;rport From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as7789aeb3 To: <sip:6025@192.168.50.233> Contact: <sip:Unknown@192.168.100.236> Call-ID: 011b6ef11a16b0f96c03e88c3c0fb528@192.168.100.236 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Mar 2010 10:31:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Sending SIP PDU to ( 192.168.100.236:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.236:5060;branch=z9hG4bK280dfde2;rport From: "Unknown" <sip:Unknown@192.168.100.236>;tag=as7789aeb3 To: <sip:6025@192.168.50.233> Call-ID: 011b6ef11a16b0f96c03e88c3c0fb528@192.168.100.236 CSeq: 102 OPTIONS User-Agent: AddPac SIP Gateway Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY Content-Length: 0 Вызов как я понимаю проходит, но в ответ гудки.С софтовым клиентом(X-lite) все работает. |
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| Автор: | Geniu$$ [ 01 мар 2010, 12:45 ] |
| Заголовок сообщения: | |
Звонок проходит, но после Ringing на шлюз приходит Cancel. И все. Смотрите на станции или на удаленной стороне причину этого Cancel. |
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| Автор: | ДСВ [ 01 мар 2010, 13:06 ] |
| Заголовок сообщения: | |
Geniu$$ писал(а): Звонок проходит, но после Ringing на шлюз приходит Cancel. И все. Смотрите на станции или на удаленной стороне причину этого Cancel.
Спасибо, будем разбираться. |
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