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muon

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About muon

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  1. Задумавшись об оплаченном 2,5 месяца назад заказе, не пришедшем до сих пор, прочитал тему как "компания "Наг" переезжает в новый офис (и больше никогда работать не будет)"
  2. NAT нет, LAN, абонент в одной подсети, астериск в другой. IP у телефона с DHCP, не резервированный, может разные получать. RTP ходит: Sent RTP packet to 192.168.20.116:10036 (type 00, seq 036509, ts 082400, len 000160) Got RTP packet from 192.168.20.116:10036 (type 00, seq 022940, ts 1225231604, len 000160) Sent RTP packet to 192.168.20.116:10036 (type 00, seq 036510, ts 082560, len 000160) Got RTP packet from 192.168.20.116:10036 (type 00, seq 022941, ts 1225231764, len 000160) Sent RTP packet to 192.168.20.116:10036 (type 00, seq 036511, ts 082720, len 000160) Got RTP packet from 192.168.20.116:10036 (type 00, seq 022942, ts 1225231924, len 000160) Sent RTP packet to 192.168.20.116:10036 (type 00, seq 036512, ts 082880, len 000160) Got RTP packet from 192.168.20.116:10036 (type 00, seq 022943, ts 1225232084, len 000160) sip debug на проблемный номер, в простое: <--- SIP read from UDP:192.168.20.116:5060 ---> REGISTER sip:asterisk0.msun.int SIP/2.0 Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK19300236068938577;rport From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317 To: Namen <sip:7396@asterisk0.msun.int:5060> Call-ID: 1992342718936-119531866030069@192.168.20.116 CSeq: 61 REGISTER Contact: <sip:7396@192.168.20.116:5060> Max-Forwards: 70 Expires: 87 Supported: path User-Agent: Greenlite ATOM V2.0 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.20.116:5060 (NAT) <--- Transmitting (no NAT) to 192.168.20.116:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK19300236068938577;received=192.168.20.116;rport=5060 From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317 To: Namen <sip:7396@asterisk0.msun.int:5060>;tag=as3fbfbcd5 Call-ID: 1992342718936-119531866030069@192.168.20.116 CSeq: 61 REGISTER Server: FPBX-2.10.0(1.8.16.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b1bd3bd" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1992342718936-119531866030069@192.168.20.116' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.20.116:5060 ---> REGISTER sip:asterisk0.msun.int SIP/2.0 Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK456630781636429961;rport From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317 To: Namen <sip:7396@asterisk0.msun.int:5060> Call-ID: 1992342718936-119531866030069@192.168.20.116 CSeq: 62 REGISTER Contact: <sip:7396@192.168.20.116:5060> Authorization: Digest username="7396", realm="asterisk", nonce="1b1bd3bd", uri="sip:asterisk0.msun.int", response="3fefc3f0163a787543ed5478e5dc8b04", algorithm=MD5 Max-Forwards: 70 Expires: 87 Supported: path User-Agent: Greenlite ATOM V2.0 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.20.116:5060 (no NAT) Reliably Transmitting (no NAT) to 192.168.20.116:5060: OPTIONS sip:7396@192.168.20.116:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK01a3e1fe Max-Forwards: 70 From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as754d0017 To: <sip:7396@192.168.20.116:5060> Contact: <sip:Unknown@192.168.10.79:5060> Call-ID: 74754af1257dfffc4711040a6e753e2a@192.168.10.79:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.10.0(1.8.16.0) Date: Tue, 16 Oct 2012 04:07:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- Transmitting (no NAT) to 192.168.20.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.116:5060;branch=z9hG4bK456630781636429961;received=192.168.20.116;rport=5060 From: Namen <sip:7396@asterisk0.msun.int:5060>;tag=300359317 To: Namen <sip:7396@asterisk0.msun.int:5060>;tag=as3fbfbcd5 Call-ID: 1992342718936-119531866030069@192.168.20.116 CSeq: 62 REGISTER Server: FPBX-2.10.0(1.8.16.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 87 Contact: <sip:7396@192.168.20.116:5060>;expires=87 Date: Tue, 16 Oct 2012 04:07:07 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060' in 6400 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 192.168.20.116:5060: NOTIFY sip:7396@192.168.20.116:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK61403ec8 Max-Forwards: 70 From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as37819252 To: <sip:7396@192.168.20.116:5060> Contact: <sip:Unknown@192.168.10.79:5060> Call-ID: 45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060 CSeq: 102 NOTIFY User-Agent: FPBX-2.10.0(1.8.16.0) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 88 Messages-Waiting: no Message-Account: sip:*97@192.168.10.79 Voice-Message: 0/0 (0/0) --- Scheduling destruction of SIP dialog '1992342718936-119531866030069@192.168.20.116' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:192.168.20.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK01a3e1fe From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as754d0017 To: <sip:7396@192.168.20.116:5060> Call-ID: 74754af1257dfffc4711040a6e753e2a@192.168.10.79:5060 CSeq: 102 OPTIONS Contact: <sip:7396@192.168.20.116:5060> Supported: 100rel, replaces, timer Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Accept: application/sdp, message/sipfrag, application/dtmf-relay Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '74754af1257dfffc4711040a6e753e2a@192.168.10.79:5060' Method: OPTIONS <--- SIP read from UDP:192.168.20.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.79:5060;branch=z9hG4bK61403ec8 From: "Unknown" <sip:Unknown@192.168.10.79>;tag=as37819252 To: <sip:7396@192.168.20.116:5060> Call-ID: 45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060 CSeq: 102 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '45f79b775055328d2ee0b01a0bd74816@192.168.10.79:5060' Method: NOTIFY Really destroying SIP dialog '1992342718936-119531866030069@192.168.20.116' Method: REGISTER Мне это не нравится, слишком много всего, ещё и ответ 401. Blink (софтфон), например, постоянно посылает только <--- SIP read from UDP:192.168.20.12:5060 ---> jaK <-------------> и изредка OPTIONS, на который получает ответ 200. Посмотрю ещё во время звонка.
  3. Здравствуйте, есть IP-телефон SNR-7030, подключён к Asterisk 1.8.16, астериск по Е1 связан с аналоговой станцией. После перезагрузки IP-телефона можно поговорить с аналоговым абонентом, но во время второго и последующих звонков в трубке IP-телефона тишина, звук с аналогового телефона не приходит. Аналоговый абонент при этом всё слышит. Если вместо IP-телефона использовать софтфон, то всё хорошо работает. В чём дело? Что делать дальше, настраивать астериск или обращаться в техподдержку SNR? Пытался сравнивать выхлоп астериска: SNR ... -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g0/5396 -- DAHDI/i1/5396-4c is proceeding passing it to SIP/7396-0000005a -- DAHDI/i1/5396-4c is ringing -- DAHDI/i1/5396-4c is making progress passing it to SIP/7396-0000005a -- DAHDI/i1/5396-4c answered SIP/7396-0000005a -- fixed jitterbuffer created on channel DAHDI/i1/5396-4c -- Span 1: Channel 0/1 got hangup request, cause 16 -- Executing [h@macro-dialout-trunk:1] Macro("SIP/7396-0000005a", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/7396-0000005a", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] Hangup("SIP/7396-0000005a", "") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/7396-0000005a' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/7396-0000005a' -- Hungup 'DAHDI/i1/5396-4c' -- fixed jitterbuffer destroyed on channel DAHDI/i1/5396-4c == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/7396-0000005a' in macro 'dialout-trunk' == Spawn extension (from-internal, 5396, 7) exited non-zero on 'SIP/7396-0000005a' софтфон (Blink): ... -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g0/5396 -- DAHDI/i1/5396-4d is proceeding passing it to SIP/7397-0000005b -- DAHDI/i1/5396-4d is ringing -- DAHDI/i1/5396-4d is making progress passing it to SIP/7397-0000005b -- DAHDI/i1/5396-4d answered SIP/7397-0000005b -- fixed jitterbuffer created on channel DAHDI/i1/5396-4d -- Span 1: Channel 0/1 got hangup request, cause 16 -- Executing [h@macro-dialout-trunk:1] Macro("SIP/7397-0000005b", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/7397-0000005b", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] Hangup("SIP/7397-0000005b", "") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/7397-0000005b' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/7397-0000005b' -- Hungup 'DAHDI/i1/5396-4d' -- fixed jitterbuffer destroyed on channel DAHDI/i1/5396-4d == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/7397-0000005b' in macro 'dialout-trunk' == Spawn extension (from-internal, 5396, 7) exited non-zero on 'SIP/7397-0000005b' фатальных различий не увидел. :(