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[проблема] Asterisk и звонки в другую подсеть

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

[проблема] Asterisk и звонки в другую подсеть

Сообщение xShepard199x » 15 фев 2019, 12:24

Здравствуйте ув.форумчане. Столкнулся с такой проблемой, имеется вызывная панель Dahua(VTO3221D) и сервер Asterisk 15, при подключении панели в ту же подсеть где находится Asterisk то всё работает нормально, но стоит вынести её в другую подсеть организации то она вечно сбрасывает входящий вызов (и она так же не может совершать вызовы) хотя она спокойно регистрируется на сервере, что странно ATA в той же подсети ведёт себя нормально.
Код: выделить все
Connected to Asterisk 15.4.0 currently running on freepbx (pid = 2008)
Audio is at 19142
Adding codec g722 to SDP
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.10.146.116:5060:
INVITE sip:3105@192.10.146.116:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK439a33da
Max-Forwards: 70
From: "Max" <sip:78193@172.27.0.38>;tag=as6b30909b
To: <sip:3105@192.10.146.116:5060>
Contact: <sip:78193@172.27.0.38:5060>
Call-ID: 47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.3.1(15.4.0)
Date: Fri, 15 Feb 2019 10:55:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Max" <sip:78193@172.27.0.38>
Content-Type: application/sdp
Content-Length: 384

v=0
o=root 269982279 269982279 IN IP4 172.27.0.38
s=Asterisk PBX 15.4.0
c=IN IP4 172.27.0.38
t=0 0
m=audio 19142 RTP/AVP 9 18 0 8 3 111 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK439a33da
From: "Max" <sip:78193@172.27.0.38>;tag=as6b30909b
To: <sip:3105@192.10.146.116:5060>
Call-ID: 47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060
CSeq: 102 INVITE
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK439a33da
From: "Max" <sip:78193@172.27.0.38>;tag=as6b30909b
To: <sip:3105@192.10.146.116:5060>;tag=2122678646
Call-ID: 47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060
CSeq: 102 INVITE
Contact: <sip:3105@192.10.146.116:5060>
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK439a33da
From: "Max" <sip:78193@172.27.0.38>;tag=as6b30909b
To: <sip:3105@192.10.146.116:5060>;tag=2122678646
Call-ID: 47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060
CSeq: 102 INVITE
Contact: <sip:3105@192.10.146.116:5060>
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
MaxRingingTime: 27
MaxConnectingTime: 120
DependentInfo:
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:3105@192.10.146.116:5060>

<--- SIP read from UDP:192.10.146.116:5060 --->
BYE sip:78193@172.27.0.38:5060 SIP/2.0
Via: SIP/2.0/UDP 192.10.146.116:5060;rport;branch=z9hG4bK1807383711
From: <sip:3105@192.10.146.116:5060>;tag=2122678646
To: "Max" <sip:78193@172.27.0.38>;tag=as6b30909b
Call-ID: 47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060
CSeq: 103 BYE
Contact: <sip:3105@192.10.146.116:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.10.146.116:5060 (no NAT)
Scheduling destruction of SIP dialog '47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060' in 11968 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.10.146.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.10.146.116:5060;branch=z9hG4bK1807383711;received=192.10.146.116;rport=5060
From: <sip:3105@192.10.146.116:5060>;tag=2122678646
To: "Max" <sip:78193@172.27.0.38>;tag=as6b30909b
Call-ID: 47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060
CSeq: 103 BYE
Server: FPBX-14.0.3.1(15.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '611759543@192.10.146.116' Method: REGISTER
Really destroying SIP dialog '47ff729a00ea38e7626c40796ae7e51b@172.27.0.38:5060' Method: BYE


Код: выделить все
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x7ff600009c20 -- Strict RTP learning after remote address set to: 172.27.0.97:4028
    -- Executing [3105@from-internal:1] GotoIf("SIP/78193-00000128", "1?ext-local,3105,1:followme-check,3105,1") in new stack
    -- Goto (ext-local,3105,1)
    -- Executing [3105@ext-local:1] Set("SIP/78193-00000128", "__RINGTIMER=60") in new stack
    -- Executing [3105@ext-local:2] Macro("SIP/78193-00000128", "exten-vm,novm,3105,0,0,0") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/78193-00000128", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/78193-00000128", "TOUCH_MONITOR=1550231013.296") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/78193-00000128", "AMPUSER=78193") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/78193-00000128", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/78193-00000128", "1?Set(REALCALLERIDNUM=78193)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/78193-00000128", "AMPUSER=78193") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/78193-00000128", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/78193-00000128", "AMPUSERCIDNAME=Max") in new stack
    -- Executing [s@macro-user-callerid:8] ExecIf("SIP/78193-00000128", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/78193-00000128", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/78193-00000128", "AMPUSERCID=78193") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/78193-00000128", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/78193-00000128", "CALLERID(all)="Max" <78193>") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/78193-00000128", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("SIP/78193-00000128", "0?Set(GROUP(concurrency_limit)=78193)") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("SIP/78193-00000128", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:16] NoOp("SIP/78193-00000128", "Macro Depth is 2") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("SIP/78193-00000128", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] GotoIf("SIP/78193-00000128", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:19] ExecIf("SIP/78193-00000128", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/78193-00000128", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:21] GotoIf("SIP/78193-00000128", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,37)
    -- Executing [s@macro-user-callerid:37] Set("SIP/78193-00000128", "CALLERID(number)=78193") in new stack
    -- Executing [s@macro-user-callerid:38] Set("SIP/78193-00000128", "CALLERID(name)=Max") in new stack
    -- Executing [s@macro-user-callerid:39] GotoIf("SIP/78193-00000128", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:40] Set("SIP/78193-00000128", "CDR(cnam)=Max") in new stack
    -- Executing [s@macro-user-callerid:41] Set("SIP/78193-00000128", "CDR(cnum)=78193") in new stack
    -- Executing [s@macro-user-callerid:42] Set("SIP/78193-00000128", "CHANNEL(language)=en") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/78193-00000128", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/78193-00000128", "__EXTTOCALL=3105") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/78193-00000128", "__PICKUPMARK=3105") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/78193-00000128", "RT=") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:6] ExecIf("SIP/78193-00000128", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:7] ExecIf("SIP/78193-00000128", "0?MacroExit()") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:8] ExecIf("SIP/78193-00000128", "0?Gosub(ext-intercom,*803105,1())") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:9] ExecIf("SIP/78193-00000128", "0?MacroExit()") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:10] ExecIf("SIP/78193-00000128", "0?ChanSpy(SIP/3105,q)") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:11] ExecIf("SIP/78193-00000128", "0?MacroExit()") in new stack
[2019-02-15 11:43:33] ERROR[16659][C-000000a3]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
    -- Executing [s@macro-exten-vm:12] Gosub("SIP/78193-00000128", "sub-record-check,s,1(exten,3105,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/78193-00000128", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("SIP/78193-00000128", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("SIP/78193-00000128", "NOW=1550231013") in new stack
    -- Executing [s@sub-record-check:4] Set("SIP/78193-00000128", "__DAY=15") in new stack
    -- Executing [s@sub-record-check:5] Set("SIP/78193-00000128", "__MONTH=02") in new stack
    -- Executing [s@sub-record-check:6] Set("SIP/78193-00000128", "__YEAR=2019") in new stack
    -- Executing [s@sub-record-check:7] Set("SIP/78193-00000128", "__TIMESTR=20190215-114333") in new stack
    -- Executing [s@sub-record-check:8] Set("SIP/78193-00000128", "__FROMEXTEN=78193") in new stack
    -- Executing [s@sub-record-check:9] Set("SIP/78193-00000128", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("SIP/78193-00000128", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/78193-00000128", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/78193-00000128", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/78193-00000128", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("SIP/78193-00000128", "5?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("SIP/78193-00000128", "1?sub-record-check,exten,1") in new stack
    -- Goto (sub-record-check,exten,1)
    -- Executing [exten@sub-record-check:1] NoOp("SIP/78193-00000128", "Exten Recording Check between 78193 and 3105") in new stack
    -- Executing [exten@sub-record-check:2] Set("SIP/78193-00000128", "CALLTYPE=internal") in new stack
    -- Executing [exten@sub-record-check:3] ExecIf("SIP/78193-00000128", "0?Set(CALLTYPE=)") in new stack
    -- Executing [exten@sub-record-check:4] Set("SIP/78193-00000128", "CALLEE=dontcare") in new stack
    -- Executing [exten@sub-record-check:5] ExecIf("SIP/78193-00000128", "0?Set(CALLEE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:6] GotoIf("SIP/78193-00000128", "0?callee") in new stack
    -- Executing [exten@sub-record-check:7] GotoIf("SIP/78193-00000128", "1?caller") in new stack
    -- Goto (sub-record-check,exten,13)
    -- Executing [exten@sub-record-check:13] Set("SIP/78193-00000128", "RECMODE=dontcare") in new stack
    -- Executing [exten@sub-record-check:14] ExecIf("SIP/78193-00000128", "0?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:15] ExecIf("SIP/78193-00000128", "1?Set(RECMODE=dontcare)") in new stack
    -- Executing [exten@sub-record-check:16] Gosub("SIP/78193-00000128", "recordcheck,1(dontcare,internal,3105)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/78193-00000128", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("SIP/78193-00000128", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("SIP/78193-00000128", "") in new stack
    -- Executing [exten@sub-record-check:17] Return("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-exten-vm:13] GotoIf("SIP/78193-00000128", "1?macrodial") in new stack
    -- Goto (macro-exten-vm,s,19)
    -- Executing [s@macro-exten-vm:19] GosubIf("SIP/78193-00000128", "0?clrheader,1()") in new stack
    -- Executing [s@macro-exten-vm:20] Macro("SIP/78193-00000128", "dial-one,,HhTtr,3105") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/78193-00000128", "DEXTEN=3105") in new stack
    -- Executing [s@macro-dial-one:2] Set("SIP/78193-00000128", "__CRM_SOURCE=78193") in new stack
    -- Executing [s@macro-dial-one:3] ExecIf("SIP/78193-00000128", "0?Set(__EXTTOCALL=3105)") in new stack
    -- Executing [s@macro-dial-one:4] Set("SIP/78193-00000128", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:5] GosubIf("SIP/78193-00000128", "0?screen,1()") in new stack
    -- Executing [s@macro-dial-one:6] GosubIf("SIP/78193-00000128", "0?cf,1()") in new stack
    -- Executing [s@macro-dial-one:7] GotoIf("SIP/78193-00000128", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,10)
    -- Executing [s@macro-dial-one:10] GotoIf("SIP/78193-00000128", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/78193-00000128", "0?continue") in new stack
    -- Executing [s@macro-dial-one:12] ExecIf("SIP/78193-00000128", "0?Set(D_OPTIONS=g)") in new stack
    -- Executing [s@macro-dial-one:13] Set("SIP/78193-00000128", "EXTHASCW=ENABLED") in new stack
    -- Executing [s@macro-dial-one:14] GotoIf("SIP/78193-00000128", "0?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,26)
    -- Executing [s@macro-dial-one:26] GotoIf("SIP/78193-00000128", "0?next3:continue") in new stack
    -- Goto (macro-dial-one,s,28)
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/78193-00000128", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:29] GosubIf("SIP/78193-00000128", "1?dstring,1():dlocal,1()") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/78193-00000128", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("SIP/78193-00000128", "DEVICES=3105") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/78193-00000128", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/78193-00000128", "0?Set(DEVICES=105)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/78193-00000128", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/78193-00000128", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/78193-00000128", "THISDIAL=SIP/3105") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/78193-00000128", "1?zap2dahdi,1()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/78193-00000128", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/78193-00000128", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/78193-00000128", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/78193-00000128", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/78193-00000128", "THISPART2=SIP/3105") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/78193-00000128", "0?Set(THISPART2=DAHDI/3105)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/78193-00000128", "NEWDIAL=SIP/3105&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/78193-00000128", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/78193-00000128", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/78193-00000128", "THISDIAL=SIP/3105") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/78193-00000128", "") in new stack
    -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/78193-00000128", "1?docheck") in new stack
    -- Goto (macro-dial-one,dstring,14)
    -- Executing [dstring@macro-dial-one:14] GotoIf("SIP/78193-00000128", "0?skipset") in new stack
    -- Executing [dstring@macro-dial-one:15] Set("SIP/78193-00000128", "DSTRING=SIP/3105&") in new stack
    -- Executing [dstring@macro-dial-one:16] Set("SIP/78193-00000128", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:17] GotoIf("SIP/78193-00000128", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:18] ExecIf("SIP/78193-00000128", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:19] Set("SIP/78193-00000128", "DSTRING=SIP/3105") in new stack
    -- Executing [dstring@macro-dial-one:20] Return("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-dial-one:30] GotoIf("SIP/78193-00000128", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:31] GotoIf("SIP/78193-00000128", "0?skiptrace") in new stack
    -- Executing [s@macro-dial-one:32] GosubIf("SIP/78193-00000128", "1?ctset,1():ctclear,1()") in new stack
    -- Executing [ctset@macro-dial-one:1] Set("SIP/78193-00000128", "DB(CALLTRACE/3105)=78193") in new stack
    -- Executing [ctset@macro-dial-one:2] Return("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-dial-one:33] Set("SIP/78193-00000128", "D_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-dial-one:34] NoOp("SIP/78193-00000128", "Blind Transfer: , Attended Transfer: , User: 78193, Alert Info: ") in new stack
    -- Executing [s@macro-dial-one:35] ExecIf("SIP/78193-00000128", "1?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:36] ExecIf("SIP/78193-00000128", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:37] ExecIf("SIP/78193-00000128", "0?Set(ALERT_INFO=)") in new stack
    -- Executing [s@macro-dial-one:38] ExecIf("SIP/78193-00000128", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
    -- Executing [s@macro-dial-one:39] ExecIf("SIP/78193-00000128", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
    -- Executing [s@macro-dial-one:40] GosubIf("SIP/78193-00000128", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
    -- Executing [s@macro-dial-one:41] ExecIf("SIP/78193-00000128", "0?Set(CHANNEL(musicclass)=)") in new stack
    -- Executing [s@macro-dial-one:42] GosubIf("SIP/78193-00000128", "0?qwait,1()") in new stack
    -- Executing [s@macro-dial-one:43] Set("SIP/78193-00000128", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:44] Set("SIP/78193-00000128", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:45] GotoIf("SIP/78193-00000128", "0?usegoto,1") in new stack
    -- Executing [s@macro-dial-one:46] GotoIf("SIP/78193-00000128", "0?godial") in new stack
    -- Executing [s@macro-dial-one:47] Gosub("SIP/78193-00000128", "sub-presencestate-display,s,1(3105)") in new stack
    -- Executing [s@sub-presencestate-display:1] Goto("SIP/78193-00000128", "state-not_set,1") in new stack
    -- Goto (sub-presencestate-display,state-not_set,1)
    -- Executing [state-not_set@sub-presencestate-display:1] Set("SIP/78193-00000128", "PRESENCESTATE_DISPLAY=") in new stack
    -- Executing [state-not_set@sub-presencestate-display:2] Return("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-dial-one:48] Set("SIP/78193-00000128", "CONNECTEDLINE(name,i)=3105") in new stack
    -- Executing [s@macro-dial-one:49] Set("SIP/78193-00000128", "CONNECTEDLINE(num)=3105") in new stack
    -- Executing [s@macro-dial-one:50] Set("SIP/78193-00000128", "D_OPTIONS=HhTtrI") in new stack
    -- Executing [s@macro-dial-one:51] Macro("SIP/78193-00000128", "dialout-one-predial-hook,") in new stack
    -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-dial-one:52] ExecIf("SIP/78193-00000128", "0?Set(D_OPTIONS=HhtrII)") in new stack
    -- Executing [s@macro-dial-one:53] NoOp("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-dial-one:54] Dial("SIP/78193-00000128", "SIP/3105,,HhTtrIb(func-apply-sipheaders^s^1)") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- SIP/3105-00000129 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/3105-00000129", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
    -- Executing [s@func-apply-sipheaders:2] NoOp("SIP/3105-00000129", "Applying SIP Headers to channel") in new stack
    -- Executing [s@func-apply-sipheaders:3] Set("SIP/3105-00000129", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:4] While("SIP/3105-00000129", "0") in new stack
    -- Jumping to priority 8
    -- Executing [s@func-apply-sipheaders:9] Return("SIP/3105-00000129", "") in new stack
  == Spawn extension (from-internal, 3105, 1) exited non-zero on 'SIP/3105-00000129'
    -- SIP/3105-00000129 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
Audio is at 15050
Adding codec g722 to SDP
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.10.146.116:5060:
INVITE sip:3105@192.10.146.116:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK677cbac4
Max-Forwards: 70
From: "Max" <sip:78193@172.27.0.38>;tag=as37580e4c
To: <sip:3105@192.10.146.116:5060>
Contact: <sip:78193@172.27.0.38:5060>
Call-ID: 31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.3.1(15.4.0)
Date: Fri, 15 Feb 2019 11:43:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Max" <sip:78193@172.27.0.38>
Content-Type: application/sdp
Content-Length: 384

v=0
o=root 235540019 235540019 IN IP4 172.27.0.38
s=Asterisk PBX 15.4.0
c=IN IP4 172.27.0.38
t=0 0
m=audio 15050 RTP/AVP 9 18 0 8 3 111 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/3105
    -- Connected line update to SIP/78193-00000128 prevented.

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK677cbac4
From: "Max" <sip:78193@172.27.0.38>;tag=as37580e4c
To: <sip:3105@192.10.146.116:5060>
Call-ID: 31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060
CSeq: 102 INVITE
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK677cbac4
From: "Max" <sip:78193@172.27.0.38>;tag=as37580e4c
To: <sip:3105@192.10.146.116:5060>;tag=2099849453
Call-ID: 31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060
CSeq: 102 INVITE
Contact: <sip:3105@192.10.146.116:5060>
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

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

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK677cbac4
From: "Max" <sip:78193@172.27.0.38>;tag=as37580e4c
To: <sip:3105@192.10.146.116:5060>;tag=2099849453
Call-ID: 31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060
CSeq: 102 INVITE
Contact: <sip:3105@192.10.146.116:5060>
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
MaxRingingTime: 27
MaxConnectingTime: 120
DependentInfo:
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:3105@192.10.146.116:5060>
    -- SIP/3105-00000129 is ringing
Really destroying SIP dialog '84797043@192.10.146.116' Method: REGISTER

<--- SIP read from UDP:192.10.146.116:5060 --->
BYE sip:78193@172.27.0.38:5060 SIP/2.0
Via: SIP/2.0/UDP 192.10.146.116:5060;rport;branch=z9hG4bK1670358657
From: <sip:3105@192.10.146.116:5060>;tag=2099849453
To: "Max" <sip:78193@172.27.0.38>;tag=as37580e4c
Call-ID: 31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060
CSeq: 103 BYE
Contact: <sip:3105@192.10.146.116:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.10.146.116:5060 (no NAT)
Scheduling destruction of SIP dialog '31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060' in 10560 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.10.146.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.10.146.116:5060;branch=z9hG4bK1670358657;received=192.10.146.116;rport=5060
From: <sip:3105@192.10.146.116:5060>;tag=2099849453
To: "Max" <sip:78193@172.27.0.38>;tag=as37580e4c
Call-ID: 31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060
CSeq: 103 BYE
Server: FPBX-14.0.3.1(15.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- No one is available to answer at this time (1:0/0/0)
    -- Executing [s@macro-dial-one:55] ExecIf("SIP/78193-00000128", "0?MacroExit()") in new stack
    -- Executing [s@macro-dial-one:56] ExecIf("SIP/78193-00000128", "0?Set(DIALSTATUS=)") in new stack
    -- Executing [s@macro-dial-one:57] GosubIf("SIP/78193-00000128", "0?s-NOANSWER,1()") in new stack
    -- Executing [s@macro-dial-one:58] MacroExit("SIP/78193-00000128", "") in new stack
    -- Executing [s@macro-exten-vm:21] Set("SIP/78193-00000128", "SV_DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:22] GosubIf("SIP/78193-00000128", "0?docfu,1()") in new stack
    -- Executing [s@macro-exten-vm:23] GosubIf("SIP/78193-00000128", "0?docfb,1()") in new stack
    -- Executing [s@macro-exten-vm:24] Set("SIP/78193-00000128", "DIALSTATUS=NOANSWER") in new stack
    -- Executing [s@macro-exten-vm:25] ExecIf("SIP/78193-00000128", "0?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:26] GotoIf("SIP/78193-00000128", "1?s-NOANSWER,1") in new stack
    -- Goto (macro-exten-vm,s-NOANSWER,1)
    -- Executing [s-NOANSWER@macro-exten-vm:1] GotoIf("SIP/78193-00000128", "0?exit,1") in new stack
    -- Executing [s-NOANSWER@macro-exten-vm:2] PlayTones("SIP/78193-00000128", "congestion") in new stack
    -- Executing [s-NOANSWER@macro-exten-vm:3] Congestion("SIP/78193-00000128", "10") in new stack
  == Spawn extension (macro-exten-vm, s-NOANSWER, 3) exited non-zero on 'SIP/78193-00000128' in macro 'exten-vm'
  == Spawn extension (ext-local, 3105, 2) exited non-zero on 'SIP/78193-00000128'
    -- Executing [h@ext-local:1] Macro("SIP/78193-00000128", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/78193-00000128", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/78193-00000128", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("SIP/78193-00000128", " monior file= ") in new stack
    -- Executing [s@macro-hangupcall:5] AGI("SIP/78193-00000128", "attendedtransfer-rec-restart.php,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
    -- <SIP/78193-00000128>AGI Script attendedtransfer-rec-restart.php completed, returning 0
    -- Executing [s@macro-hangupcall:6] Hangup("SIP/78193-00000128", "") in new stack
  == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/78193-00000128' in macro 'hangupcall'
  == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/78193-00000128'
Really destroying SIP dialog '31ae12304f25b5503ab2e5232ee12ba6@172.27.0.38:5060' Method: BYE

<--- SIP read from UDP:192.10.146.116:5060 --->
REGISTER sip:172.27.0.38 SIP/2.0
Via: SIP/2.0/UDP 192.10.146.116:5060;rport;branch=z9hG4bK1595706800
From: <sip:3105@172.27.0.38:5060>;tag=1779615067
To: <sip:3105@172.27.0.38:5060>
Call-ID: 1330533396@192.10.146.116
CSeq: 1 REGISTER
Contact: <sip:3105@192.10.146.116:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Expires: 60
PhoneState: 0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.10.146.116:5060 (no NAT)
Sending to 192.10.146.116:5060 (no NAT)

<--- Transmitting (no NAT) to 192.10.146.116:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.10.146.116:5060;branch=z9hG4bK1595706800;received=192.10.146.116;rport=5060
From: <sip:3105@172.27.0.38:5060>;tag=1779615067
To: <sip:3105@172.27.0.38:5060>;tag=as7887f751
Call-ID: 1330533396@192.10.146.116
CSeq: 1 REGISTER
Server: FPBX-14.0.3.1(15.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b24c4b8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1330533396@192.10.146.116' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.10.146.116:5060 --->
REGISTER sip:172.27.0.38 SIP/2.0
Via: SIP/2.0/UDP 192.10.146.116:5060;rport;branch=z9hG4bK1729931070
From: <sip:3105@172.27.0.38:5060>;tag=1779615067
To: <sip:3105@172.27.0.38:5060>
Call-ID: 1330533396@192.10.146.116
CSeq: 2 REGISTER
Contact: <sip:3105@192.10.146.116:5060>
Authorization: Digest username="3105", realm="asterisk", nonce="1b24c4b8", uri="sip:172.27.0.38", response="6216662dd0861a32d786760bc1a8a142", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Expires: 60
PhoneState: 0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.10.146.116:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.10.146.116:5060:
OPTIONS sip:3105@192.10.146.116:5060 SIP/2.0
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK1088d7a6
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.27.0.38>;tag=as1ca7e222
To: <sip:3105@192.10.146.116:5060>
Contact: <sip:Unknown@172.27.0.38:5060>
Call-ID: 0706fa8c050b62b56f73c9172aee59bc@172.27.0.38:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.1(15.4.0)
Date: Fri, 15 Feb 2019 11:44:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.10.146.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.10.146.116:5060;branch=z9hG4bK1729931070;received=192.10.146.116;rport=5060
From: <sip:3105@172.27.0.38:5060>;tag=1779615067
To: <sip:3105@172.27.0.38:5060>;tag=as7887f751
Call-ID: 1330533396@192.10.146.116
CSeq: 2 REGISTER
Server: FPBX-14.0.3.1(15.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:3105@192.10.146.116:5060>;expires=60
Date: Fri, 15 Feb 2019 11:44:05 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1330533396@192.10.146.116' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.10.146.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.0.38:5060;branch=z9hG4bK1088d7a6
From: "Unknown" <sip:Unknown@172.27.0.38>;tag=as1ca7e222
To: <sip:3105@192.10.146.116:5060>;tag=232954513
Call-ID: 0706fa8c050b62b56f73c9172aee59bc@172.27.0.38:5060
CSeq: 102 OPTIONS
User-Agent: Dahua UAC/3.0 VTO3221D V3.200.0.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '0706fa8c050b62b56f73c9172aee59bc@172.27.0.38:5060' Method: OPTIONS
freepbx*CLI>


В чём может быть проблема? Заранее благодарен
xShepard199x
 
Сообщений: 2
Зарегистрирован: 15 фев 2019, 12:11

Re: [проблема] Asterisk и звонки в другую подсеть

Сообщение sasa » 15 фев 2019, 16:34

Панель сама завершает звонок BYE
Ищите на ней какие то логи или играйтесь настройками
sasa
 
Сообщений: 119
Зарегистрирован: 22 янв 2019, 15:41

Re: [проблема] Asterisk и звонки в другую подсеть

Сообщение ded » 15 фев 2019, 17:27

Отключите на панели все кодеки, кроме alaw & ulaw.
ded
 
Сообщений: 15801
Зарегистрирован: 26 авг 2010, 19:00

Re: [проблема] Asterisk и звонки в другую подсеть

Сообщение xShepard199x » 15 фев 2019, 19:15

К сожалению не помогло
xShepard199x
 
Сообщений: 2
Зарегистрирован: 15 фев 2019, 12:11

Re: [проблема] Asterisk и звонки в другую подсеть

Сообщение ded » 16 фев 2019, 15:33

Проблема SIP ALG на роутере 192.10.146.116? За ним ведь больше, чем одно устройство SIP?

Соединяйте без НАТа, через VPN, и будет всё ОК.
ded
 
Сообщений: 15801
Зарегистрирован: 26 авг 2010, 19:00


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