Чат  ::   FAQ  ::   Поиск  ::   Регистрация  ::   Вход

SIP BEELINE Forbidden

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

Модератор: april22

Re: SIP BEELINE Forbidden

Сообщение skynetyar » 21 авг 2017, 15:29

Да кстате о кривом пароле To: <sip:eE@96XXXXXXXXX1> то что жирным это пароль почему то тут засвечивается, перед eE в пароле знак ?

Строка регистрации так и не побеждена там так же Registration for 'eE@mpbx.sip.beeline.ru' timed out, trying again (Attempt #16)
Всё знают и всё понимают только дураки да шарлатаны.(с)А.П Чехов.
skynetyar
 
Сообщений: 402
Зарегистрирован: 18 авг 2016, 14:25

Re: SIP BEELINE Forbidden

Сообщение skynetyar » 21 авг 2017, 17:50

Прогресс однозначно есть , спасибо за наводку, именно знак "?" вопроса в пароле был проблемой со строкой регистрации!
В правильном варианте для облачной АТС Билайна она будет такой
Код: выделить все
96ХХХХХХХ@mpbx.sip.beeline.ru:PASS:96XXXXXXXX@mpbx.sip.beeline.ru@mpbx.sip.beeline.ru/96XXXXXXX

Ну а вот в трубке по прежнему "все линии заняты"

sip set debug on
SIP Debugging enabled

[Показать] Спойлер:
<--- SIP read from UDP:192.168.0.103:49154 --->
INVITE sip:89066357777@192.168.0.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:50607;branch=z9hG4bK207bddc1
From: "703" <sip:703@192.168.0.7>;tag=0016473ee629142af04673d7-f593d15d
To: <sip:89066357777@192.168.0.7>
Call-ID: 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
Max-Forwards: 70
Date: Mon, 21 Aug 2017 13:34:35 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.5.3
Contact: <sip:703@192.168.0.103:50607;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 278
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 14849 0 IN IP4 192.168.0.103
s=SIP Call
t=0 0
m=audio 26428 RTP/AVP 8 0 18 101
c=IN IP4 192.168.0.103
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 13 lines) ---
Sending to 192.168.0.103:50607 (no NAT)
Sending to 192.168.0.103:50607 (no NAT)
Using INVITE request as basis request - 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
Found peer '703' for '703' from 192.168.0.103:49154

<--- Reliably Transmitting (no NAT) to 192.168.0.103:50607 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.103:50607;branch=z9hG4bK207bddc1;received=192.168.0.103
From: "703" <sip:703@192.168.0.7>;tag=0016473ee629142af04673d7-f593d15d
To: <sip:89066357777@192.168.0.7>;tag=as10580ee8
Call-ID: 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
CSeq: 101 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e2c53d7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.103:51012 --->
ACK sip:89066357070@192.168.0.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:50607;branch=z9hG4bK207bddc1
From: "703" <sip:703@192.168.0.7>;tag=0016473ee629142af04673d7-f593d15d
To: <sip:89066357070@192.168.0.7>;tag=as10580ee8
Call-ID: 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
Date: Mon, 21 Aug 2017 13:34:35 GMT
CSeq: 101 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.103:49154 --->
INVITE sip:89066357070@192.168.0.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:50607;branch=z9hG4bKc27ab063
From: "703" <sip:703@192.168.0.7>;tag=0016473ee629142af04673d7-f593d15d
To: <sip:89066357070@192.168.0.7>
Call-ID: 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
Max-Forwards: 70
Date: Mon, 21 Aug 2017 13:34:35 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7970G/8.5.3
Contact: <sip:703@192.168.0.103:50607;transport=udp>
Authorization: Digest username="703",realm="asterisk",uri="sip:89066357777@192.168.0.7",response="0c6dcfacaa789a204105d493b5da5744",nonce="2e2c53d7",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 278
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 14849 0 IN IP4 192.168.0.103
s=SIP Call
t=0 0
m=audio 26428 RTP/AVP 8 0 18 101
c=IN IP4 192.168.0.103
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 13 lines) ---
Sending to 192.168.0.103:50607 (no NAT)
Using INVITE request as basis request - 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
Found peer '703' for '703' from 192.168.0.103:49154
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g729)/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 192.168.0.103:26428
set_destination: Parsing <sip:712@192.168.0.135:5060;transport=UDP> for address/port to send to
set_destination: set destination to 192.168.0.135:5060
Reliably Transmitting (no NAT) to 192.168.0.135:5060:
NOTIFY sip:712@192.168.0.135:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7:50607;branch=z9hG4bK18f2da3a
Max-Forwards: 70
From: <sip:703@192.168.0.7>;tag=as18a68658
To: <sip:712@192.168.0.135>;tag=00230432ea9a000a92a17a87-92b2de80
Contact: <sip:703@192.168.0.7:50607>
Call-ID: 00230432-ea9a0007-151c13c5-b3d2eb8a@192.168.0.135
CSeq: 1009 NOTIFY
User-Agent: FPBX-13.0.192.16(13.17.0)
Subscription-State: active
Event: presence
Content-Type: application/cpim-pidf+xml
Content-Length: 346

<?xml version="1.0"?>
<!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
<presentity uri="sip:712@192.168.0.135;method=SUBSCRIBE" />
<atom id="703">
<address uri="sip:703@192.168.0.7;user=ip" priority="0.800000">
<status status="inuse" />
<msnsubstatus substatus="onthephone" />
</address>
</atom>
</presence>

---
== Extension Changed 703[ext-local] new state InUse for Notify User 712
Looking for 89066357777 in from-internal (domain 192.168.0.7)
sip_route_dump: route/path hop: <sip:703@192.168.0.103:50607;transport=udp>

<--- Transmitting (no NAT) to 192.168.0.103:50607 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.103:50607;branch=z9hG4bKc27ab063;received=192.168.0.103
From: "703" <sip:703@192.168.0.7>;tag=0016473ee629142af04673d7-f593d15d
To: <sip:89066357070@192.168.0.7>
Call-ID: 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
CSeq: 102 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:89066357070@192.168.0.7:50607>
Content-Length: 0

<--- Reliably Transmitting (no NAT) to 192.168.0.103:50607 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.103:50607;branch=z9hG4bKc27ab063;received=192.168.0.103
From: "703" <sip:703@192.168.0.7>;tag=0016473ee629142af04673d7-f593d15d
To: <sip:89066357777@192.168.0.7>;tag=as7a09f1cd
Call-ID: 0016473e-e629000f-d10cb93b-4dee88f9@192.168.0.103
CSeq: 102 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0
Всё знают и всё понимают только дураки да шарлатаны.(с)А.П Чехов.
skynetyar
 
Сообщений: 402
Зарегистрирован: 18 авг 2016, 14:25

Re: SIP BEELINE Forbidden

Сообщение april22 » 21 авг 2017, 19:25

Странно, вы звоните сами себе?
Своими вопросами , вы загоняете меня в ГУГЛЬ.
april22
 
Сообщений: 2122
Зарегистрирован: 09 июл 2012, 09:47

Re: SIP BEELINE Forbidden

Сообщение virus_net » 22 авг 2017, 08:10

вы привели дамп правого плеча вызова (телефон - ваш астериск), а смотреть вам нужно левое плечо ( ваш астериск - Билайн).
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru - Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)

ENUMER - звони бесплатно и напрямую.
virus_net
 
Сообщений: 2319
Зарегистрирован: 05 июн 2013, 08:12
Откуда: Москва

Re: SIP BEELINE Forbidden

Сообщение skynetyar » 22 авг 2017, 11:06

Простите туплю, как Вы это поняли что я сам себе звоню?
Как не странно но я не вижу пакетов относительно моего звонка при дебаге,что то затупил жестко :cry:

[Показать] Спойлер:
-- Executing [89066357777@from-internal:1] Macro("SIP/703-00000ab0", "user-callerid,LIMIT") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/703-00000ab0", "TOUCH_MONITOR=1503385136.2931") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/703-00000ab0", "AMPUSER=703") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/703-00000ab0", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/703-00000ab0", "1?Set(REALCALLERIDNUM=703)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/703-00000ab0", "AMPUSER=703") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/703-00000ab0", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/703-00000ab0", "AMPUSERCIDNAME=Evstif P.S. 703") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/703-00000ab0", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/703-00000ab0", "AMPUSERCID=703") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/703-00000ab0", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/703-00000ab0", "CALLERID(all)="Evsti P.S. 703" <703>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/703-00000ab0", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/703-00000ab0", "1?Set(GROUP(concurrency_limit)=703)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/703-00000ab0", "1?Set(CHANNEL(language)=ru)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/703-00000ab0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("SIP/703-00000ab0", "CALLERID(number)=703") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/703-00000ab0", "CALLERID(name)=Evstif P.S. 703") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("SIP/703-00000ab0", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/703-00000ab0", "CDR(cnam)=Evstif P.S. 703") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/703-00000ab0", "CDR(cnum)=703") in new stack
-- Executing [s@macro-user-callerid:34] Set("SIP/703-00000ab0", "CHANNEL(language)=ru") in new stack
-- Executing [89066357070@from-internal:2] Set("SIP/703-00000ab0", "ROUTEUSER=703") in new stack
-- Executing [89066357070@from-internal:3] Set("SIP/703-00000ab0", "ROUTEUSER=703") in new stack
-- Executing [89066357070@from-internal:4] GotoIf("SIP/703-00000ab0", "1?notblind") in new stack
-- Goto (from-internal,89066357777,7)
-- Executing [89066357070@from-internal:7] GotoIf("SIP/703-00000ab0", "1?restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc,89066357777,2:outbound-allroutes,89066357777,2") in new stack
-- Goto (restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc,89066357070,2)
-- Executing [89066357777@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:2] Gosub("SIP/703-00000ab0", "sub-record-check,s,1(out,89066357777,yes)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/703-00000ab0", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/703-00000ab0", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/703-00000ab0", "NOW=1503385136") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/703-00000ab0", "__DAY=22") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/703-00000ab0", "__MONTH=08") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/703-00000ab0", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/703-00000ab0", "__TIMESTR=20170822-095856") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/703-00000ab0", "__FROMEXTEN=703") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/703-00000ab0", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/703-00000ab0", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/703-00000ab0", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/703-00000ab0", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/703-00000ab0", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/703-00000ab0", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/703-00000ab0", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/703-00000ab0", "Outbound Recording Check from 703 to 89066357777") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/703-00000ab0", "RECMODE=yes") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/703-00000ab0", "0?Goto(routewins)") in new stack
-- Executing [out@sub-record-check:4] ExecIf("SIP/703-00000ab0", "0?Goto(routewins)") in new stack
-- Executing [out@sub-record-check:5] Gosub("SIP/703-00000ab0", "recordcheck,1(yes,out,89066357777)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/703-00000ab0", "Starting recording check against yes") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/703-00000ab0", "yes") in new stack
-- Goto (sub-record-check,recordcheck,9)
-- Executing [recordcheck@sub-record-check:9] ExecIf("SIP/703-00000ab0", "0?Return()") in new stack
-- Executing [recordcheck@sub-record-check:10] Set("SIP/703-00000ab0", "__REC_POLICY_MODE=YES") in new stack
-- Executing [recordcheck@sub-record-check:11] Goto("SIP/703-00000ab0", "startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [recordcheck@sub-record-check:16] NoOp("SIP/703-00000ab0", "Starting recording: out, 89066357777") in new stack
-- Executing [recordcheck@sub-record-check:17] Set("SIP/703-00000ab0", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [recordcheck@sub-record-check:18] Set("SIP/703-00000ab0", "__CALLFILENAME=out-89066357777-703-20170822-095856-1503385136.2931") in new stack
-- Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/703-00000ab0", "/mnt/backup/monitor/2017/08/22/out-89066357070-703-20170822-095856-1503385136.2931.wav,abi(LOCAL_MIXMON_ID),/etc/asterisk/scripts/mixmon-mp3-2.sh ^{YEAR} ^{MONTH} ^{DAY} ^{CALLFILENAME} ^{MIXMON_FORMAT} ^{MIXMON_DIR}") in new stack
-- Executing [recordcheck@sub-record-check:20] Set("SIP/703-00000ab0", "__MIXMON_ID=0xaca19c8") in new stack
== Begin MixMonitor Recording SIP/703-00000ab0
-- Executing [recordcheck@sub-record-check:21] Set("SIP/703-00000ab0", "__RECORD_ID=SIP/703-00000ab0") in new stack
-- Executing [recordcheck@sub-record-check:22] Set("SIP/703-00000ab0", "__REC_STATUS=RECORDING") in new stack
-- Executing [recordcheck@sub-record-check:23] Set("SIP/703-00000ab0", "CDR(recordingfile)=out-89066357070-703-20170822-095856-1503385136.2931.wav") in new stack
-- Executing [recordcheck@sub-record-check:24] Return("SIP/703-00000ab0", "") in new stack
-- Executing [out@sub-record-check:6] Return("SIP/703-00000ab0", "") in new stack
-- Executing [89066357070@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:3] ExecIf("SIP/703-00000ab0", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [89066357070@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:4] Set("SIP/703-00000ab0", "MOHCLASS=default") in new stack
-- Executing [89066357070@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:5] Set("SIP/703-00000ab0", "_NODEST=") in new stack
-- Executing [89066357070@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:6] Macro("SIP/703-00000ab0", "dialout-trunk,65,89066357070,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/703-00000ab0", "DIAL_TRUNK=65") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/703-00000ab0", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/703-00000ab0", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/703-00000ab0", "DIAL_NUMBER=89066357777") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/703-00000ab0", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/703-00000ab0", "OUTBOUND_GROUP=OUT_65") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/703-00000ab0", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/703-00000ab0", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/703-00000ab0", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/703-00000ab0", "DIAL_TRUNK_OPTIONS=T") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/703-00000ab0", "outbound-callerid,65") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/703-00000ab0", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/703-00000ab0", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/703-00000ab0", "0?Set(REALCALLERIDNUM=703)") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/703-00000ab0", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,7)
-- Executing [s@macro-outbound-callerid:7] Set("SIP/703-00000ab0", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/703-00000ab0", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] Set("SIP/703-00000ab0", "TRUNKOUTCID=96XXXXXXXXX") in new stack
-- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/703-00000ab0", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/703-00000ab0", "1?Set(CALLERID(all)=96XXXXXXXX)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/703-00000ab0", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/703-00000ab0", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/703-00000ab0", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/703-00000ab0", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:20] Set("SIP/703-00000ab0", "CDR(outbound_cnum)=96XXXXXXX") in new stack
-- Executing [s@macro-outbound-callerid:21] Set("SIP/703-00000ab0", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/703-00000ab0", "1?sub-flp-65,s,1()") in new stack
-- Executing [s@sub-flp-65:1] ExecIf("SIP/703-00000ab0", "1?Return()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/703-00000ab0", "OUTNUM=89066357777") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/703-00000ab0", "custom=SIP/79XXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/703-00000ab0", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/703-00000ab0", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/703-00000ab0", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/703-00000ab0", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/703-00000ab0", "0?skipcrm") in new stack
-- Executing [s@macro-dialout-trunk:19] Set("SIP/703-00000ab0", "__CRM_DIRECTION=OUTBOUND") in new stack
-- Executing [s@macro-dialout-trunk:20] Set("SIP/703-00000ab0", "__CRM_DESTINATION=89066357777") in new stack
-- Executing [s@macro-dialout-trunk:21] Set("SIP/703-00000ab0", "__CRM_SOURCE=703") in new stack
-- Executing [s@macro-dialout-trunk:22] AGI("SIP/703-00000ab0", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/703-00000ab0>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@macro-dialout-trunk:23] Set("SIP/703-00000ab0", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/703-00000ab0", "CRM Finished") in new stack
-- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/703-00000ab0", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:26] ExecIf("SIP/703-00000ab0", "1?Set(CONNECTEDLINE(num,i)=89066357777)") in new stack
-- Executing [s@macro-dialout-trunk:27] ExecIf("SIP/703-00000ab0", "1?Set(CONNECTEDLINE(name,i)=CID:96XXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/703-00000ab0", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)96XXXXXX)") in new stack
-- Executing [s@macro-dialout-trunk:29] GotoIf("SIP/703-00000ab0", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:30] Dial("SIP/703-00000ab0", "SIP/79XXXXXXX/89066357777,300,T") in new stack
[2017-08-22 09:58:57] WARNING[28274][C-000001b6]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:31] NoOp("SIP/703-00000ab0", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
-- Executing [s@macro-dialout-trunk:32] GotoIf("SIP/703-00000ab0", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/703-00000ab0", "RC=20") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/703-00000ab0", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [20@macro-dialout-trunk:1] Goto("SIP/703-00000ab0", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/703-00000ab0", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/703-00000ab0", "1?Set(CALLERID(number)=703)") in new stack
-- Executing [89066357070@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:7] Macro("SIP/703-00000ab0", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/703-00000ab0", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/703-00000ab0", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/703-00000ab0", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/703-00000ab0", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
-- <SIP/703-00000ab0> Playing 'all-circuits-busy-now.ulaw' (language 'ru')
> 0xb744d490 -- Probation passed - setting RTP source address to 192.168.0.103:20682
[2017-08-22 09:58:59] WARNING[28274][C-000001b6]: file.c:774 ast_openstream_full: File please-try-call-later does not exist in any format
[2017-08-22 09:58:59] WARNING[28274][C-000001b6]: file.c:1247 ast_streamfile: Unable to open please-try-call-later (format (alaw)): No such file or directory
-- Executing [h@restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc:1] Hangup("SIP/703-00000ab0", "") in new stack
== Spawn extension (restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc, h, 1) exited non-zero on 'SIP/703-00000ab0'
-- SIP/703-00000ab0 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/703-00000ab0", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/703-00000ab0", "HANGUP CAUSE: 20") in new stack
== Extension Changed 703[ext-local] new state Idle for Notify User 712
-- Executing [s@crm-hangup:3] ExecIf("SIP/703-00000ab0", "0?Set(__CRM_VOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/703-00000ab0", "MASTER CHANNEL: 1503385136.2931 = 1503385136.2931") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/703-00000ab0", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("SIP/703-00000ab0", "__CRM_HANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("SIP/703-00000ab0", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/703-00000ab0>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("SIP/703-00000ab0", "") in new stack
== Spawn extension (restrictedroute-1679091c5a880faf6fb5e6087eb1b2dc, h, 1) exited non-zero on 'SIP/703-00000ab0'
-- SIP/703-00000ab0 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
== MixMonitor close filestream (mixed)
== Executing [/etc/asterisk/scripts/mixmon-mp3-2.sh 2017 08 22 out-89066357777-703-20170822-095856-1503385136.2931 wav /mnt/backup/monitor/]
== End MixMonitor Recording SIP/703-00000ab0

atc*CLI> sip set debug off
Всё знают и всё понимают только дураки да шарлатаны.(с)А.П Чехов.
skynetyar
 
Сообщений: 402
Зарегистрирован: 18 авг 2016, 14:25

Re: SIP BEELINE Forbidden

Сообщение ded » 22 авг 2017, 11:23

Информативная часть портянки:
-- Executing [s@macro-dialout-trunk:30] Dial("SIP/703-00000ab0", "SIP/79XXXXXXX/89066357777,300,T") in new stack
[2017-08-22 09:58:57] WARNING[28274][C-000001b6]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
Что-то надо ещё комментировать?
ded
 
Сообщений: 14391
Зарегистрирован: 26 авг 2010, 19:00

Re: SIP BEELINE Forbidden

Сообщение skynetyar » 22 авг 2017, 13:01

Да,пир не зарегистрировался , и я блин не знаю тоже как его зарегистрировать...
Код: выделить все
79ХХХХХХХ/96ХХХХХ    (Unspecified)                               No         No             0        UNKNOWN


Ну настолько мало информации про эту облачную АТС от билайна, сами они в ТП говорят что не консультируют "про срм астерипск" дословно цитата тп.
outgoing у меня такой...
Код: выделить все
host=mpbx.sip.beeline.ru
fromdomain=mpbx.sip.beeline.ru
type=peer
secret=ЧЧЧЧЧЧЧ
qualify=yes
insecure=invite,port
fromuser=96ХХХХХХХ
context=from-pstn
directmedia=no
nat=no
dtmfmode=rfc2833
port=5060
Всё знают и всё понимают только дураки да шарлатаны.(с)А.П Чехов.
skynetyar
 
Сообщений: 402
Зарегистрирован: 18 авг 2016, 14:25

Re: SIP BEELINE Forbidden

Сообщение ded » 22 авг 2017, 13:48

Платный, жосский суппорт с БДСМ.
ded
 
Сообщений: 14391
Зарегистрирован: 26 авг 2010, 19:00

Re: SIP BEELINE Forbidden

Сообщение awsswa » 22 авг 2017, 14:12

поиск по слову - mpbx.sip.beeline.ru
выводит примерно более 1000 упоминаний
платный суппорт по мере возможностей
awsswa
 
Сообщений: 2387
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: SIP BEELINE Forbidden

Сообщение skynetyar » 22 авг 2017, 16:26

Только не БДСМ :cry:

Смущает
Код: выделить все
atc*CLI> [2017-08-22 14:37:42] WARNING[3221]: netsock2.c:216 ast_sockaddr_split_hostport: Port disallowed in mpbx.sip.beeline.ru:5060
atc*CLI> [2017-08-22 14:37:42] WARNING[3221]: acl.c:800 resolve_first: Unable to lookup 'mpbx.sip.beeline.ru:5060'


Пока не понял что это такое..
А настройки пира я уже все 1000 вариантов попробовал...
Все одно...
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Всё знают и всё понимают только дураки да шарлатаны.(с)А.П Чехов.
skynetyar
 
Сообщений: 402
Зарегистрирован: 18 авг 2016, 14:25

Пред.След.

Вернуться в Вопросы новичков

Кто сейчас на форуме

Сейчас этот форум просматривают: нет зарегистрированных пользователей и гости: 6

cron
© 2008 — 2020 Asterisk.ru
Digium, Asterisk and AsteriskNOW are registered trademarks of Digium, Inc.
Design and development by PostMet-Netzwerk GmbH