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"asterisk за натом и перенаправление звонка на внешний SIP"
Отправлено PetrovR, 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] ---

Исходное сообщение
"asterisk за натом и перенаправление звонка на внешний SIP" Отправлено PetrovR, 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] ---

Ваше сообщение
Имя*:
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Заголовок*:
Сообщение*:	>> 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] ---
	Введите код, изображенный на картинке:

При общении не допускается: неуважительное отношение к собеседнику, хамство, унизительное обращение, ненормативная лексика, переход на личности, агрессивное поведение, обесценивание собеседника, провоцирование флейма голословными и заведомо ложными заявлениями. Не отвечайте на сообщения, явно нарушающие правила - удаляются не только сами нарушения, но и все ответы на них. Лог модерирования.

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