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"asterisk за натом и перенаправление звонка на внешний SIP"	+/–
Сообщение от petrovr on 09-Фев-18, 08:25
Добрый день товарищи Спецы! Помогите советом, как решить вопрос мой. У меня значит входящий вызов на выданный провайдером номер должен переводиться на внешний SIP сервер, которым не я заведую. мой asterisk (10.1.1.5) за натом. Порты 5060 и 10000-20000 проброшены. Вот организация проброса: ----------------- ${FW} nat 1 config ip ${IP_EXT} reset same_ports \         redirect_port   udp     10.1.1.5:5060 5060 \         redirect_port   udp     10.1.1.5:10000-20000 10000-20000 sip.conf ----------------- register=aaa:bbb@voip-prov.ru externaddr=a.b.c.d externip=a.b.c.d localnet=192.168.0.0/255.255.255.0 language=ru context=default allowexternaldomains=yes allowoverlap=no udpbindaddr=10.1.1.5 tcpenable=no transport=udp srvlookup=no allowguest=no alwaysauthreject=yes limitonpeers=yes nat=force_rport,comedia .... клиентские линии и пир провайдера .... extensions.conf ------------------- [prov-in] ;;входящий звонок на номер мне надо перевести на внешний SIP сервер. exten   => 1234567,1,Dial(SIP/${EXTEN}@external_ip_server) exten   => 1234567,n,HangUp() Вот лог происходящего в астериске: ----------------------------------- [Feb  9 09:15:40]        > 0x83c0ed000 -- Strict RTP learning after remote address set to: 37.228.85.6:29732 [Feb  9 09:15:40]     -- Executing [1234567@prov-in:1] Dial("SIP/mcn-0000004b", "SIP/1234567@external_ip_server") in new stack [Feb  9 09:15:40]     -- Called SIP/1234567@external_ip_server [Feb  9 09:16:05]   == Spawn extension (prov-in, 1234567, 1) exited non-zero on 'SIP/mcn-0000004b' [2018-02-09 09:16:12.647] WARNING[101424]: chan_sip.c:4065 int retrans_pkt(const void *): Retransmission timeout reached on transmission 0bd4150e75986daa525775a83e0244c7@my_external_ip:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 31999ms with no response Вот tcpdump происходящего: ------------------------------- 09:22:37.812206 IP my_external_ip.5060 > external_ip_server.5060: SIP: INVITE sip:1234567@external_ip_server SIP/2.0 09:22:37.823816 IP external_ip_server.5060 > my_external_ip.5060: SIP: SIP/2.0 100 Trying 09:22:37.823860 IP my_external_ip > external_ip_server: ICMP my_external_ip udp port 5060 unreachable, length 36 09:22:37.824054 IP external_ip_server.5060 > my_external_ip.5060: SIP: SIP/2.0 200 OK 09:22:37.824079 IP my_external_ip > external_ip_server: ICMP my_external_ip udp port 5060 unreachable, length 36 09:22:37.882674 IP external_ip_server.15578 > my_external_ip.18158: UDP, length 172 09:22:37.903137 IP external_ip_server.15578 > my_external_ip.18158: UDP, length 172 Подскажите что не так? Повторюсь, внешний udp 5060 порт проброшен на 10.1.1.5:5060 а всё указывает на то, что пакет с внешнего SIP сервера на 5060 не достигает астериска.
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1. "asterisk за натом и перенаправление звонка на внешний SIP"	+/–
Сообщение от _ (??) on 09-Фев-18, 09:25
>[оверквотинг удален] > 09:22:37.823860 IP my_external_ip > external_ip_server: ICMP my_external_ip udp port > 5060 unreachable, length 36 > 09:22:37.824054 IP external_ip_server.5060 > my_external_ip.5060: SIP: SIP/2.0 200 OK > 09:22:37.824079 IP my_external_ip > external_ip_server: ICMP my_external_ip udp port > 5060 unreachable, length 36 > 09:22:37.882674 IP external_ip_server.15578 > my_external_ip.18158: UDP, length 172 > 09:22:37.903137 IP external_ip_server.15578 > my_external_ip.18158: UDP, length 172 > Подскажите что не так? Повторюсь, внешний udp 5060 порт проброшен на 10.1.1.5:5060 > а всё указывает на то, что пакет с внешнего SIP сервера > на 5060 не достигает астериска. Это tcpdump с маршрутизатора или с сервера Asterisk? В консоли астериска дайте команду sip set debug ip <external_ip_server>. Что показывает астериск если включить этот дебаг?
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	2. "asterisk за натом и перенаправление звонка на внешний SIP"	+/–
	Сообщение от PetrovR (ok) on 09-Фев-18, 09:43
	> Это tcpdump с маршрутизатора или с сервера Asterisk? > В консоли астериска дайте команду sip set debug ip <external_ip_server>. Что показывает > астериск если включить этот дебаг? Астериск в клетке на шлюзе работает. Этот tcpdump со шлюза на котором получается фактически и астериск. Вот лог вызова с телефона 9221122345 [Feb  9 11:36:56]     -- Executing [1234567@prov-in:1] Set("SIP/mcn-0000018d", "Var_TO=1234567") in new stack [Feb  9 11:36:56]     -- Executing [1234567@prov-in:2] Dial("SIP/mcn-0000018d", "SIP/1234567@external_ip_server") in new stack [Feb  9 11:36:56] Audio is at 12860 [Feb  9 11:36:56] Adding codec alaw to SDP [Feb  9 11:36:56] Adding codec ulaw to SDP [Feb  9 11:36:56] Adding codec gsm to SDP [Feb  9 11:36:56] Adding non-codec 0x1 (telephone-event) to SDP [Feb  9 11:36:56] Reliably Transmitting (NAT) to external_ip_server:5060: [Feb  9 11:36:56] INVITE sip:1234567@external_ip_server SIP/2.0 [Feb  9 11:36:56] Via: SIP/2.0/UDP my_external_ip:5060;branch=z9hG4bK739fdfbf;rport [Feb  9 11:36:56] Max-Forwards: 70 [Feb  9 11:36:56] From: "79221122356" <sip:79221122356@my_external_ip>;tag=as0fa9b5df [Feb  9 11:36:56] To: <sip:1234567@external_ip_server> [Feb  9 11:36:56] Contact: <sip:79221122356@my_external_ip:5060> [Feb  9 11:36:56] Call-ID: 2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060 [Feb  9 11:36:56] CSeq: 102 INVITE [Feb  9 11:36:56] User-Agent: Asterisk PBX 13.19.0 [Feb  9 11:36:56] Date: Fri, 09 Feb 2018 06:36:56 GMT [Feb  9 11:36:56] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Feb  9 11:36:56] Supported: replaces, timer [Feb  9 11:36:56] Content-Type: application/sdp [Feb  9 11:36:56] Content-Length: 289 [Feb  9 11:36:56] [Feb  9 11:36:56] v=0 [Feb  9 11:36:56] o=root 1124322562 1124322562 IN IP4 my_external_ip [Feb  9 11:36:56] s=Asterisk PBX 13.19.0 [Feb  9 11:36:56] c=IN IP4 my_external_ip [Feb  9 11:36:56] t=0 0 [Feb  9 11:36:56] m=audio 12860 RTP/AVP 8 0 3 101 [Feb  9 11:36:56] a=rtpmap:8 PCMA/8000 [Feb  9 11:36:56] a=rtpmap:0 PCMU/8000 [Feb  9 11:36:56] a=rtpmap:3 GSM/8000 [Feb  9 11:36:56] a=rtpmap:101 telephone-event/8000 [Feb  9 11:36:56] a=fmtp:101 0-16 [Feb  9 11:36:56] a=maxptime:150 [Feb  9 11:36:56] a=sendrecv [Feb  9 11:36:56] [Feb  9 11:36:56] --- [Feb  9 11:36:56]     -- Called SIP/1234567@external_ip_server [Feb  9 11:36:56] Retransmitting #1 (NAT) to external_ip_server:5060: [Feb  9 11:36:56] INVITE sip:1234567@external_ip_server SIP/2.0 [Feb  9 11:36:56] Via: SIP/2.0/UDP my_external_ip:5060;branch=z9hG4bK739fdfbf;rport [Feb  9 11:36:56] Max-Forwards: 70 [Feb  9 11:36:56] From: "79221122356" <sip:79221122356@my_external_ip>;tag=as0fa9b5df [Feb  9 11:36:56] To: <sip:1234567@external_ip_server> [Feb  9 11:36:56] Contact: <sip:79221122356@my_external_ip:5060> [Feb  9 11:36:56] Call-ID: 2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060 [Feb  9 11:36:56] CSeq: 102 INVITE [Feb  9 11:36:56] User-Agent: Asterisk PBX 13.19.0 [Feb  9 11:36:56] Date: Fri, 09 Feb 2018 06:36:56 GMT [Feb  9 11:36:56] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Feb  9 11:36:56] Supported: replaces, timer [Feb  9 11:36:56] Content-Type: application/sdp [Feb  9 11:36:56] Content-Length: 289 [Feb  9 11:36:56] [Feb  9 11:36:56] v=0 [Feb  9 11:36:56] o=root 1124322562 1124322562 IN IP4 my_external_ip [Feb  9 11:36:56] s=Asterisk PBX 13.19.0 [Feb  9 11:36:56] c=IN IP4 my_external_ip [Feb  9 11:36:56] t=0 0 [Feb  9 11:36:56] m=audio 12860 RTP/AVP 8 0 3 101 [Feb  9 11:36:56] a=rtpmap:8 PCMA/8000 [Feb  9 11:36:56] a=rtpmap:0 PCMU/8000 [Feb  9 11:36:56] a=rtpmap:3 GSM/8000 [Feb  9 11:36:56] a=rtpmap:101 telephone-event/8000 [Feb  9 11:36:56] a=fmtp:101 0-16 [Feb  9 11:36:56] a=maxptime:150 [Feb  9 11:36:56] a=sendrecv [Feb  9 11:36:56] [Feb  9 11:36:56] --- [Feb  9 11:36:57] Retransmitting #2 (NAT) to external_ip_server:5060: [Feb  9 11:36:57] INVITE sip:1234567@external_ip_server SIP/2.0 [Feb  9 11:36:57] Via: SIP/2.0/UDP my_external_ip:5060;branch=z9hG4bK739fdfbf;rport [Feb  9 11:36:57] Max-Forwards: 70 [Feb  9 11:36:57] From: "79221122356" <sip:79221122356@my_external_ip>;tag=as0fa9b5df [Feb  9 11:36:57] To: <sip:1234567@external_ip_server> [Feb  9 11:36:57] Contact: <sip:79221122356@my_external_ip:5060> [Feb  9 11:36:57] Call-ID: 2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060 [Feb  9 11:36:57] CSeq: 102 INVITE [Feb  9 11:36:57] User-Agent: Asterisk PBX 13.19.0 [Feb  9 11:36:57] Date: Fri, 09 Feb 2018 06:36:56 GMT [Feb  9 11:36:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Feb  9 11:36:57] Supported: replaces, timer [Feb  9 11:36:57] Content-Type: application/sdp [Feb  9 11:36:57] Content-Length: 289 [Feb  9 11:36:57] [Feb  9 11:36:57] v=0 [Feb  9 11:36:57] o=root 1124322562 1124322562 IN IP4 my_external_ip [Feb  9 11:36:57] s=Asterisk PBX 13.19.0 [Feb  9 11:36:57] c=IN IP4 my_external_ip [Feb  9 11:36:57] t=0 0 [Feb  9 11:36:57] m=audio 12860 RTP/AVP 8 0 3 101 [Feb  9 11:36:57] a=rtpmap:8 PCMA/8000 [Feb  9 11:36:57] a=rtpmap:0 PCMU/8000 [Feb  9 11:36:57] a=rtpmap:3 GSM/8000 [Feb  9 11:36:57] a=rtpmap:101 telephone-event/8000 [Feb  9 11:36:57] a=fmtp:101 0-16 [Feb  9 11:36:57] a=maxptime:150 [Feb  9 11:36:57] a=sendrecv [Feb  9 11:36:57] [Feb  9 11:36:57] --- [Feb  9 11:36:59] Retransmitting #3 (NAT) to external_ip_server:5060: [Feb  9 11:36:59] INVITE sip:1234567@external_ip_server SIP/2.0 [Feb  9 11:36:59] Via: SIP/2.0/UDP my_external_ip:5060;branch=z9hG4bK739fdfbf;rport [Feb  9 11:36:59] Max-Forwards: 70 [Feb  9 11:36:59] From: "79221122356" <sip:79221122356@my_external_ip>;tag=as0fa9b5df [Feb  9 11:36:59] To: <sip:1234567@external_ip_server> [Feb  9 11:36:59] Contact: <sip:79221122356@my_external_ip:5060> [Feb  9 11:36:59] Call-ID: 2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060 [Feb  9 11:36:59] CSeq: 102 INVITE [Feb  9 11:36:59] User-Agent: Asterisk PBX 13.19.0 [Feb  9 11:36:59] Date: Fri, 09 Feb 2018 06:36:56 GMT [Feb  9 11:36:59] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Feb  9 11:36:59] Supported: replaces, timer [Feb  9 11:36:59] Content-Type: application/sdp [Feb  9 11:36:59] Content-Length: 289 [Feb  9 11:36:59] [Feb  9 11:36:59] v=0 [Feb  9 11:36:59] o=root 1124322562 1124322562 IN IP4 my_external_ip [Feb  9 11:36:59] s=Asterisk PBX 13.19.0 [Feb  9 11:36:59] c=IN IP4 my_external_ip [Feb  9 11:36:59] t=0 0 [Feb  9 11:36:59] m=audio 12860 RTP/AVP 8 0 3 101 [Feb  9 11:36:59] a=rtpmap:8 PCMA/8000 [Feb  9 11:36:59] a=rtpmap:0 PCMU/8000 [Feb  9 11:36:59] a=rtpmap:3 GSM/8000 [Feb  9 11:36:59] a=rtpmap:101 telephone-event/8000 [Feb  9 11:36:59] a=fmtp:101 0-16 [Feb  9 11:36:59] a=maxptime:150 [Feb  9 11:36:59] a=sendrecv [Feb  9 11:36:59] [Feb  9 11:36:59] --- [Feb  9 11:37:03] Retransmitting #4 (NAT) to external_ip_server:5060: [Feb  9 11:37:03] INVITE sip:1234567@external_ip_server SIP/2.0 [Feb  9 11:37:03] Via: SIP/2.0/UDP my_external_ip:5060;branch=z9hG4bK739fdfbf;rport [Feb  9 11:37:03] Max-Forwards: 70 [Feb  9 11:37:03] From: "79221122356" <sip:79221122356@my_external_ip>;tag=as0fa9b5df [Feb  9 11:37:03] To: <sip:1234567@external_ip_server> [Feb  9 11:37:03] Contact: <sip:79221122356@my_external_ip:5060> [Feb  9 11:37:03] Call-ID: 2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060 [Feb  9 11:37:03] CSeq: 102 INVITE [Feb  9 11:37:03] User-Agent: Asterisk PBX 13.19.0 [Feb  9 11:37:03] Date: Fri, 09 Feb 2018 06:36:56 GMT [Feb  9 11:37:03] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Feb  9 11:37:03] Supported: replaces, timer [Feb  9 11:37:03] Content-Type: application/sdp [Feb  9 11:37:03] Content-Length: 289 [Feb  9 11:37:03] [Feb  9 11:37:03] v=0 [Feb  9 11:37:03] o=root 1124322562 1124322562 IN IP4 my_external_ip [Feb  9 11:37:03] s=Asterisk PBX 13.19.0 [Feb  9 11:37:03] c=IN IP4 my_external_ip [Feb  9 11:37:03] t=0 0 [Feb  9 11:37:03] m=audio 12860 RTP/AVP 8 0 3 101 [Feb  9 11:37:03] a=rtpmap:8 PCMA/8000 [Feb  9 11:37:03] a=rtpmap:0 PCMU/8000 [Feb  9 11:37:03] a=rtpmap:3 GSM/8000 [Feb  9 11:37:03] a=rtpmap:101 telephone-event/8000 [Feb  9 11:37:03] a=fmtp:101 0-16 [Feb  9 11:37:03] a=maxptime:150 [Feb  9 11:37:03] a=sendrecv [Feb  9 11:37:03] [Feb  9 11:37:03] --- [Feb  9 11:37:09] Scheduling destruction of SIP dialog '2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060' in 32000 ms (Method: INVITE) [Feb  9 11:37:09]   == Spawn extension (prov-in, 1234567, 2) exited non-zero on 'SIP/mcn-0000018d' [Feb  9 11:37:11] Retransmitting #5 (NAT) to external_ip_server:5060: [Feb  9 11:37:11] INVITE sip:1234567@external_ip_server SIP/2.0 [Feb  9 11:37:11] Via: SIP/2.0/UDP my_external_ip:5060;branch=z9hG4bK739fdfbf;rport [Feb  9 11:37:11] Max-Forwards: 70 [Feb  9 11:37:11] From: "79221122356" <sip:79221122356@my_external_ip>;tag=as0fa9b5df [Feb  9 11:37:11] To: <sip:1234567@external_ip_server> [Feb  9 11:37:11] Contact: <sip:79221122356@my_external_ip:5060> [Feb  9 11:37:11] Call-ID: 2792b45d3fc43ee4698069bd326ef6f3@my_external_ip:5060 [Feb  9 11:37:11] CSeq: 102 INVITE [Feb  9 11:37:11] User-Agent: Asterisk PBX 13.19.0 [Feb  9 11:37:11] Date: Fri, 09 Feb 2018 06:36:56 GMT [Feb  9 11:37:11] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Feb  9 11:37:11] Supported: replaces, timer [Feb  9 11:37:11] Content-Type: application/sdp [Feb  9 11:37:11] Content-Length: 289 [Feb  9 11:37:11] [Feb  9 11:37:11] v=0 [Feb  9 11:37:11] o=root 1124322562 1124322562 IN IP4 my_external_ip [Feb  9 11:37:11] s=Asterisk PBX 13.19.0 [Feb  9 11:37:11] c=IN IP4 my_external_ip [Feb  9 11:37:11] t=0 0 [Feb  9 11:37:11] m=audio 12860 RTP/AVP 8 0 3 101 [Feb  9 11:37:11] a=rtpmap:8 PCMA/8000 [Feb  9 11:37:11] a=rtpmap:0 PCMU/8000 [Feb  9 11:37:11] a=rtpmap:3 GSM/8000 [Feb  9 11:37:11] a=rtpmap:101 telephone-event/8000 [Feb  9 11:37:11] a=fmtp:101 0-16 [Feb  9 11:37:11] a=maxptime:150 [Feb  9 11:37:11] a=sendrecv [Feb  9 11:37:11] [Feb  9 11:37:11] ---
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	3. "asterisk за натом и перенаправление звонка на внешний SIP"	+/–
	Сообщение от Andrey (??) on 10-Фев-18, 13:24
	>[оверквотинг удален] > [Feb  9 11:37:11] m=audio 12860 RTP/AVP 8 0 3 101 > [Feb  9 11:37:11] a=rtpmap:8 PCMA/8000 > [Feb  9 11:37:11] a=rtpmap:0 PCMU/8000 > [Feb  9 11:37:11] a=rtpmap:3 GSM/8000 > [Feb  9 11:37:11] a=rtpmap:101 telephone-event/8000 > [Feb  9 11:37:11] a=fmtp:101 0-16 > [Feb  9 11:37:11] a=maxptime:150 > [Feb  9 11:37:11] a=sendrecv > [Feb  9 11:37:11] > [Feb  9 11:37:11] --- А у вас вообще регистрация на внешнем SIP проходит?
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