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NAT и провайдер TEKMI

0

Всем привет. Давненько я тут не появлялся :) Возникли проблемы тут с у меня с одним провайдером под названием tekmi. Стандартная схема asterisk за NATом. FreePBX 13.0.192.16 и Asterisk 11.23.1. Настроен транк:

host=login.tekmi.ru
fromdomain=login.tekmi.ru
dtmfmode=auto
username=user
defaultuser=user
fromuser=user
secret=1234
type=friend
nat=yes
disallow=all
allow=alaw&ulaw
qualify=400
insecure=invite,port
directmedia=no

Типичная настройка транка и звонки входящие/исходящие работают нооо звука нет в обе стороны. В астериске настроена локальная сеть и прописан внешний ip т.е. необходимые настройки для успешного прохождения трафика через NAT(в других офисах с такой же схемой по оборудованию и сети все работает без проблем но с другими sip провайдерами). Провайдер зачем-то просит делать проброс tcp 5060 и udp 10000-65535 до астериска, зачем мне не понятно ну до ладно сделал но ожидаемо это не дало никакого результата. Сделал sip дамп нужного пира с исходящим звонком но как не умел читать дамп, так и не научился) может кто поможет. Заранее благодарен за ответ. ip провайдера 109.69.176.249 ip офиса 78.24.178.6 с внутренного номера 190, делаю звонок себе на 89779359537

SIP Debugging Enabled for IP: 109.69.176.249
[2017-09-07 20:14:30] NOTICE[2298]: chan_sip.c:15176 sip_reregister:    -- Re-registration for  user@login.tekmi.ru
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 109.69.176.249:5060:
REGISTER sip:login.tekmi.ru SIP/2.0
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK235b7585;rport
Max-Forwards: 70
From: <sip:user@login.tekmi.ru>;tag=as563e3a93
To: <sip:user@login.tekmi.ru>
Call-ID: 082286d862b136c33d03370f64057b50@[::1]
CSeq: 106 REGISTER
User-Agent: FPBX-13.0.192.16(11.23.1)
Authorization: Digest username="user", realm="itoolabs.net", algorithm=MD5, uri="sip:login.tekmi.ru", nonce="694397F83DE82F70A8AE", response="babbf154f5d9e3ffb0321f4c8f06fb1f", opaque="opaq", qop=auth, cnonce="12456378", nc=00000004
Expires: 120
Contact: <sip:user@78.24.178.6:5060>
Content-Length: 0


---

<--- SIP read from UDP:109.69.176.249:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK235b7585;rport=5060
From: <sip:user@login.tekmi.ru>;tag=as563e3a93
To: <sip:user@login.tekmi.ru>;tag=D95AADF0
Call-ID: 082286d862b136c33d03370f64057b50@[::1]
CSeq: 106 REGISTER
Expires: 120
Contact: <sip:user@78.24.178.6:5060>;expires=120
Event: registration
Date: Thu, 07 Sep 2017 17:14:31 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path,gruu
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/5.4.12d
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
[2017-09-07 20:14:31] NOTICE[2298]: chan_sip.c:23739 handle_response_register: Outbound Registration: Expiry for login.tekmi.ru is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '082286d862b136c33d03370f64057b50@[::1]' Method: REGISTER
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [89779359537@from-internal:1] Macro("SIP/190-00000004", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/190-00000004", "TOUCH_MONITOR=1504804482.4") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/190-00000004", "AMPUSER=190") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/190-00000004", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/190-00000004", "1?Set(REALCALLERIDNUM=190)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/190-00000004", "AMPUSER=190") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/190-00000004", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/190-00000004", "AMPUSERCIDNAME=190") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/190-00000004", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/190-00000004", "AMPUSERCID=190") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/190-00000004", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/190-00000004", "CALLERID(all)="190" <190>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/190-00000004", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/190-00000004", "1?Set(GROUP(concurrency_limit)=190)") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("SIP/190-00000004", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:15] GotoIf("SIP/190-00000004", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s@macro-user-callerid:29] Set("SIP/190-00000004", "CALLERID(number)=190") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/190-00000004", "CALLERID(name)=190") in new stack
    -- Executing [s@macro-user-callerid:31] GotoIf("SIP/190-00000004", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/190-00000004", "CDR(cnam)=190") in new stack
    -- Executing [s@macro-user-callerid:33] Set("SIP/190-00000004", "CDR(cnum)=190") in new stack
    -- Executing [s@macro-user-callerid:34] Set("SIP/190-00000004", "CHANNEL(language)=en") in new stack
    -- Executing [89779359537@from-internal:2] Gosub("SIP/190-00000004", "sub-record-check,s,1(out,89779359537,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/190-00000004", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("SIP/190-00000004", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("SIP/190-00000004", "NOW=1504804482") in new stack
    -- Executing [s@sub-record-check:4] Set("SIP/190-00000004", "__DAY=07") in new stack
    -- Executing [s@sub-record-check:5] Set("SIP/190-00000004", "__MONTH=09") in new stack
    -- Executing [s@sub-record-check:6] Set("SIP/190-00000004", "__YEAR=2017") in new stack
    -- Executing [s@sub-record-check:7] Set("SIP/190-00000004", "__TIMESTR=20170907-201442") in new stack
    -- Executing [s@sub-record-check:8] Set("SIP/190-00000004", "__FROMEXTEN=190") in new stack
    -- Executing [s@sub-record-check:9] Set("SIP/190-00000004", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("SIP/190-00000004", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/190-00000004", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/190-00000004", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/190-00000004", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("SIP/190-00000004", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("SIP/190-00000004", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] NoOp("SIP/190-00000004", "Outbound Recording Check from 190 to 89779359537") in new stack
    -- Executing [out@sub-record-check:2] Set("SIP/190-00000004", "RECMODE=dontcare") in new stack
    -- Executing [out@sub-record-check:3] ExecIf("SIP/190-00000004", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [out@sub-record-check:7] Gosub("SIP/190-00000004", "recordcheck,1(dontcare,out,89779359537)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/190-00000004", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("SIP/190-00000004", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("SIP/190-00000004", "") in new stack
    -- Executing [out@sub-record-check:8] Return("SIP/190-00000004", "") in new stack
    -- Executing [89779359537@from-internal:3] ExecIf("SIP/190-00000004", "0 ?Set(CDR(accountcode)=)") in new stack
    -- Executing [89779359537@from-internal:4] Set("SIP/190-00000004", "MOHCLASS=default") in new stack
    -- Executing [89779359537@from-internal:5] Set("SIP/190-00000004", "_NODEST=") in new stack
    -- Executing [89779359537@from-internal:6] Macro("SIP/190-00000004", "dialout-trunk,2,89779359537,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/190-00000004", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/190-00000004", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/190-00000004", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/190-00000004", "DIAL_NUMBER=89779359537") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/190-00000004", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/190-00000004", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/190-00000004", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/190-00000004", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/190-00000004", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/190-00000004", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/190-00000004", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/190-00000004", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/190-00000004", "0?Set(REALCALLERIDNUM=190)") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/190-00000004", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,7)
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/190-00000004", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/190-00000004", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/190-00000004", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/190-00000004", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/190-00000004", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/190-00000004", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/190-00000004", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/190-00000004", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/190-00000004", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:20] Set("SIP/190-00000004", "CDR(outbound_cnum)=190") in new stack
    -- Executing [s@macro-outbound-callerid:21] Set("SIP/190-00000004", "CDR(outbound_cnam)=190") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/190-00000004", "0?sub-flp-2,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/190-00000004", "OUTNUM=89779359537") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/190-00000004", "custom=SIP/tekmi") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/190-00000004", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/190-00000004", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/190-00000004", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/190-00000004", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/190-00000004", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/190-00000004", "1?Set(CONNECTEDLINE(num,i)=89779359537)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/190-00000004", "1?Set(CONNECTEDLINE(name,i)=CID:190)") in new stack
    -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/190-00000004", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)190)") in new stack
    -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/190-00000004", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:23] Dial("SIP/190-00000004", "SIP/tekmi/89779359537,300,T") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 18492
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 109.69.176.249:5060:
INVITE sip:89779359537@login.tekmi.ru SIP/2.0
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK467e360b;rport
Max-Forwards: 70
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>
Contact: <sip:user@78.24.178.6:5060>
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 102 INVITE
User-Agent: FPBX-13.0.192.16(11.23.1)
Date: Thu, 07 Sep 2017 17:14:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 320733777 320733777 IN IP4 78.24.178.6
s=Asterisk PBX 11.23.1
c=IN IP4 78.24.178.6
t=0 0
m=audio 18492 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=ptime:20
a=sendrecv

---
    -- Called SIP/tekmi/89779359537

<--- SIP read from UDP:109.69.176.249:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK467e360b;rport=5060
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 102 INVITE
Server: CommuniGatePro/5.4.12d
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:109.69.176.249:5060 --->
SIP/2.0 401 Authentication required
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK467e360b;rport=5060
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>;tag=1AAC3EE9
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="itoolabs.net",nonce="3D98289F371E6DB5F086",opaque="opaq",qop="auth",algorithm=MD5
Server: CommuniGatePro/5.4.12d
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 109.69.176.249:5060:
ACK sip:89779359537@login.tekmi.ru SIP/2.0
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK467e360b;rport
Max-Forwards: 70
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>;tag=1AAC3EE9
Contact: <sip:user@78.24.178.6:5060>
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 102 ACK
User-Agent: FPBX-13.0.192.16(11.23.1)
Content-Length: 0


---
Audio is at 18492
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 109.69.176.249:5060:
INVITE sip:89779359537@login.tekmi.ru SIP/2.0
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK5fc4f62e;rport
Max-Forwards: 70
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>
Contact: <sip:user@78.24.178.6:5060>
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 103 INVITE
User-Agent: FPBX-13.0.192.16(11.23.1)
Authorization: Digest username="user", realm="itoolabs.net", algorithm=MD5, uri="sip:89779359537@login.tekmi.ru", nonce="3D98289F371E6DB5F086", response="44aeb12ffa9b78aed3227044c40417bc", opaque="opaq", qop=auth, cnonce="1a3b7873", nc=00000001
Date: Thu, 07 Sep 2017 17:14:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 320733777 320733778 IN IP4 78.24.178.6
s=Asterisk PBX 11.23.1
c=IN IP4 78.24.178.6
t=0 0
m=audio 18492 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=ptime:20
a=sendrecv

---

<--- SIP read from UDP:109.69.176.249:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK5fc4f62e;rport=5060
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 103 INVITE
Server: CommuniGatePro/5.4.12d
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:109.69.176.249:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK5fc4f62e;rport=5060
Record-Route: <sip:13823764-10.69.96.16.dialog.cgatepro;lr>
Record-Route: <sip:10.69.96.16:5060;lr>
Record-Route: <sip:109.69.176.249:5060;lr>
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>;tag=EC3F4E35-421582-F9591009_aoncfjs-52B5
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 103 INVITE
Contact: <sip:signode-421582-F9591009_aoncfjs-52B5@109.69.176.249>
Require: timer
Supported: 100rel,timer,replaces,histinfo,precondition
Session-Expires: 7200;refresher=uas
User-Agent: CommuniGatePro-callLeg/5.4.12d
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
Content-Type: application/sdp
Content-Length: 312

v=0
o=CGPLeg421582 4235710519 2117855260 IN IP4 109.69.176.144
s=-
c=IN IP4 109.69.176.144
t=0 0
m=audio 14880 RTP/AVP 0 8 101
c=IN IP4 109.69.176.144
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-49,64-89
a=sendrecv
a=rtcpping:M1:3332200:1604178221
<------------->
--- (17 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 109.69.176.144:14880
list_route: hop: <sip:109.69.176.249:5060;lr>
list_route: hop: <sip:10.69.96.16:5060;lr>
list_route: hop: <sip:13823764-10.69.96.16.dialog.cgatepro;lr>
set_destination: Parsing <sip:109.69.176.249:5060;lr> for address/port to send to
set_destination: set destination to 109.69.176.249:5060
Transmitting (NAT) to 109.69.176.249:5060:
ACK sip:signode-421582-F9591009_aoncfjs-52B5@109.69.176.249 SIP/2.0
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK6c849a57;rport
Route: <sip:109.69.176.249:5060;lr>,<sip:10.69.96.16:5060;lr>,<sip:13823764-10.69.96.16.dialog.cgatepro;lr>
Max-Forwards: 70
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>;tag=EC3F4E35-421582-F9591009_aoncfjs-52B5
Contact: <sip:user@78.24.178.6:5060>
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 103 ACK
User-Agent: FPBX-13.0.192.16(11.23.1)
Content-Length: 0


---
    -- SIP/tekmi-00000005 answered SIP/190-00000004
[2017-09-07 20:14:42] NOTICE[2805][C-00000003]: res_rtp_asterisk.c:4476 ast_rtp_read: Unknown RTP codec 95 received from '192.168.1.33:8000'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [5555972597634123@from-sip-external:1] NoOp("SIP/78.24.178.6-00000006", "Received incoming SIP connection from unknown peer to 5555972597634123") in new stack
    -- Executing [5555972597634123@from-sip-external:2] Set("SIP/78.24.178.6-00000006", "DID=5555972597634123") in new stack
    -- Executing [5555972597634123@from-sip-external:3] Goto("SIP/78.24.178.6-00000006", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/78.24.178.6-00000006", "1?setlanguage:checkanon") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [s@from-sip-external:2] Set("SIP/78.24.178.6-00000006", "CHANNEL(language)=en") in new stack
    -- Executing [s@from-sip-external:3] GotoIf("SIP/78.24.178.6-00000006", "1?noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/78.24.178.6-00000006", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2017-09-07 20:15:01.672 MSK.
    -- Executing [s@from-sip-external:6] Log("SIP/78.24.178.6-00000006", "WARNING,"Rejecting unknown SIP connection from 37.187.75.53"") in new stack
[2017-09-07 20:14:46] WARNING[2806][C-00000004]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 37.187.75.53"
    -- Executing [s@from-sip-external:7] Answer("SIP/78.24.178.6-00000006", "") in new stack
    -- Executing [s@from-sip-external:8] Wait("SIP/78.24.178.6-00000006", "2") in new stack
    -- Executing [s@from-sip-external:9] Playback("SIP/78.24.178.6-00000006", "ss-noservice") in new stack
    -- <SIP/78.24.178.6-00000006> Playing 'ss-noservice.ulaw' (language 'en')
    -- Executing [s@from-sip-external:10] PlayTones("SIP/78.24.178.6-00000006", "congestion") in new stack
    -- Executing [s@from-sip-external:11] Congestion("SIP/78.24.178.6-00000006", "5") in new stack
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/190-00000004", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] ExecIf("SIP/190-00000004", "0?Set(CDR(recordingfile)=.wav)") in new stack
    -- Executing [s@macro-hangupcall:2] GotoIf("SIP/190-00000004", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] ExecIf("SIP/190-00000004", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:5] Hangup("SIP/190-00000004", "") in new stack
  == Spawn extension (macro-hangupcall, s, 5) exited non-zero on 'SIP/190-00000004' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/190-00000004'
Scheduling destruction of SIP dialog '60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:109.69.176.249:5060;lr> for address/port to send to
set_destination: set destination to 109.69.176.249:5060
Reliably Transmitting (NAT) to 109.69.176.249:5060:
BYE sip:signode-421582-F9591009_aoncfjs-52B5@109.69.176.249 SIP/2.0
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK4b5bbc11;rport
Route: <sip:109.69.176.249:5060;lr>,<sip:10.69.96.16:5060;lr>,<sip:13823764-10.69.96.16.dialog.cgatepro;lr>
Max-Forwards: 70
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>;tag=EC3F4E35-421582-F9591009_aoncfjs-52B5
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 104 BYE
User-Agent: FPBX-13.0.192.16(11.23.1)
Authorization: Digest username="user", realm="itoolabs.net", algorithm=MD5, uri="sip:signode-421582-F9591009_aoncfjs-52B5@109.69.176.249", nonce="3D98289F371E6DB5F086", response="be07038e6f192578db2a7eb8a1606575", opaque="opaq", qop=auth, cnonce="65398e44", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/190-00000004' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 89779359537, 6) exited non-zero on 'SIP/190-00000004'

<--- SIP read from UDP:109.69.176.249:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.24.178.6:5060;branch=z9hG4bK4b5bbc11;rport=5060
From: "190" <sip:user@login.tekmi.ru>;tag=as067876a0
To: <sip:89779359537@login.tekmi.ru>;tag=EC3F4E35-421582-F9591009_aoncfjs-52B5
Call-ID: 60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru
CSeq: 104 BYE
Server: CommuniGatePro/5.4.12d
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '60bdb4e04cfa22a67772d8211161787a@login.tekmi.ru' Method: INVITE
  == Spawn extension (from-sip-external, s, 11) exited non-zero on 'SIP/78.24.178.6-00000006'
    -- Executing [h@from-sip-external:1] Hangup("SIP/78.24.178.6-00000006", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/78.24.178.6-00000006'
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спросил 2017-09-08 11:19:55 +0400

DJs3000 Gravatar DJs3000 flag of Russian Federation
559 42 13 33
http://retrogamesbattle.c...

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Задан: 2017-09-08 11:19:55 +0400

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Обновлен: Sep 08

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.