Такая ситуация, пытаюсь сделать ip телефонию в компании, после недели мучений у меня получилось подключится к провайдеру(правда к серверу который не требует авторизации) но вот незадача, при наборе номера происходит отсилы 2-3 гудка и сбрасывается, при этом мобильный только начинает реагировать, и при этом он(мобильный) продолжает звонить, теряюсь куда смотреть, Ringtime Default: поставил на 300 но что то мне кажется проблема кроется в нате...
Вот лог вызова
| Код: |
| == Using SIP RTP CoS mark 5 -- Executing [9Мобильный@from-internal:1] Macro("SIP/300-00000017", "user-callerid,SKIPTTL,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/300-00000017", "AMPUSER=300") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/300-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/300-00000017", "1?Set(REALCALLERIDNUM=300)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/300-00000017", "AMPUSER=300") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/300-00000017", "AMPUSERCIDNAME=Igor") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/300-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/300-00000017", "AMPUSERCID=300") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/300-00000017", "CALLERID(all)="Igor" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/300-00000017", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/300-00000017", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/300-00000017", "CALLERID(number)=300") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/300-00000017", "CALLERID(name)=Igor") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/300-00000017", "Using CallerID "Igor" ") in new stack -- Executing [9Мобильный@from-internal:2] NoOp("SIP/300-00000017", "Calling Out Route: 9_outside") in new stack -- Executing [9Мобильный@from-internal:3] Set("SIP/300-00000017", "MOHCLASS=default") in new stack -- Executing [9Мобильный@from-internal:4] Set("SIP/300-00000017", "_NODEST=") in new stack -- Executing [9Мобильный@from-internal:5] Macro("SIP/300-00000017", "record-enable,300,OUT,") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/300-00000017", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/300-00000017", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/300-00000017", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/300-00000017", "0?IN") in new stack -- Executing [s@macro-record-enable:16] ExecIf("SIP/300-00000017", "1?MacroExit()") in new stack -- Executing [9Мобильный@from-internal:6] Macro("SIP/300-00000017", "dialout-trunk,3,Мобильный,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/300-00000017", "DIAL_TRUNK=3") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/300-00000017", "0?sub-pincheck,s,1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/300-00000017", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/300-00000017", "DIAL_NUMBER=Мобильный") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/300-00000017", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/300-00000017", "OUTBOUND_GROUP=OUT_3") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/300-00000017", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/300-00000017", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/300-00000017", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/300-00000017", "outbound-callerid,3") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/300-00000017", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/300-00000017", "0?Set(REALCALLERIDNUM=300)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/300-00000017", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/300-00000017", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/300-00000017", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/300-00000017", "TRUNKOUTCID=Номер_выданый_провайдером") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/300-00000017", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/300-00000017", "1?Set(CALLERID(all)=Номер_выданый_провайдером)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/300-00000017", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/300-00000017", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/300-00000017", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/300-00000017", "0?sub-flp-3,s,1") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/300-00000017", "OUTNUM=Мобильный") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/300-00000017", "custom=SIP/Номер_выданый_провайдером") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/300-00000017", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/300-00000017", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/300-00000017", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/300-00000017", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/300-00000017", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("SIP/300-00000017", "SIP/Номер_выданый_провайдером/Мобильный,300,") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 19910 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/Номер_выданый_провайдером/Мобильный Retransmitting #1 (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #4 (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #5 (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- Retransmitting #6 (NAT) to ИП_ПРОВАЙДЕРА:5060: INVITE sip:Мобильный@ИП_ПРОВАЙДЕРА:5060 SIP/2.0 Via: SIP/2.0/UDP МОЙ_ИП:5060;branch=z9hG4bK44084c51;rport Max-Forwards: 70 From: "Номер_выданый_провайдером" ;tag=as70a68aee To: Contact: Call-ID: 23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.20.0) Date: Mon, 14 Oct 2013 13:28:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 978971700 978971700 IN IP4 МОЙ_ИП s=Asterisk PBX 1.8.20.0 c=IN IP4 МОЙ_ИП t=0 0 m=audio 19910 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/300-00000017", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18") in new stack -- Executing [s@macro-dialout-trunk:21] Goto("SIP/300-00000017", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/300-00000017", "RC=18") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/300-00000017", "18,1") in new stack -- Goto (macro-dialout-trunk,18,1) -- Executing [18@macro-dialout-trunk:1] Goto("SIP/300-00000017", "s-NOANSWER,1") in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp("SIP/300-00000017", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack -- Executing [s-NOANSWER@macro-dialout-trunk:2] Progress("SIP/300-00000017", "") in new stack -- Executing [s-NOANSWER@macro-dialout-trunk:3] Playback("SIP/300-00000017", "number-not-answering,noanswer") in new stack -- Playing 'number-not-answering.gsm' (language 'en') Really destroying SIP dialog '23b029500f65180e1f2bd0ff0fd50798@ИП_ПРОВАЙДЕРА' Method: INVITE -- Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion("SIP/300-00000017", "20") in new stack == Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on 'SIP/300-00000017' in macro 'dialout-trunk' == Spawn extension (from-internal, 9Мобильный, 6) exited non-zero on 'SIP/300-00000017' -- Executing [h@from-internal:1] Macro("SIP/300-00000017", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/300-00000017", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/300-00000017", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/300-00000017", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("SIP/300-00000017", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("SIP/300-00000017", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("SIP/300-00000017", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("SIP/300-00000017", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("SIP/300-00000017", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("SIP/300-00000017", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("SIP/300-00000017", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("SIP/300-00000017", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("SIP/300-00000017", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("SIP/300-00000017", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/300-00000017' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/300-00000017' |
Еще пара вопросов:
1. На сколько безопасно использовать канал провайдера без авторизации?
2. При входящем в CLI я ничего не вижу но при этом tcpdump мне показывает какое то обращение от провайдера к моему серверу
3. У меня команда sip show registry показывает что ничего нету, это связано с нем что у меня канал без регистрации, или просто какой то косяк?
Спасибо.
З.Ы. Бегает через микротик, может он рубит, пробовал софтфоном напрямую к провайдеру, то все ок было.
Added after 1 hours 41 minutes:
Думаю микротик можно исключить, подключился через софтфон пробросил порты и все пошло правда, с мобильного на софт фон голос идет долго, но в обратном направлении без проблем
Все таки проблема в микротике, подключил напрямую и все работает, завтра сброшу фаервол и буду смотреть что к чему
