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Cisco7906 не регистрируется на Asterisk по SIP

Вопросы по использованию и настройке IP телефонов, шлюзов и всего прочего

Модераторы: april22, Zavr2008

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noize
Сообщения: 117
Зарегистрирован: 01 сен 2010, 11:29

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение noize »

Разбираться с чужими конфигами нет ни времени ни желания. В аттаче конфиг телефона, который прекрасно работает с астериском без регистрации
Вложения
SEP00070E6CD276.cnf.xml.zip
(2.6 КБ) 657 скачиваний
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AppendIX
Сообщения: 16
Зарегистрирован: 08 сен 2011, 13:04

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение AppendIX »

2 noize
за конфиг спасибо, к астериску прицепился, чуть позже разберусь в отличиях и отпишу.

но вот теперь следующая проблема. пытаюсь со свежепрошитой циски позвонить на внутренний номер и в ответ от астериска получаю голосое сообщение, что вызываемый номер not in service, хотя номер живой и пробовал не только на него. а при попытке прозвониться на циску слышу тишину а в дампе SIP/SDP Request: INVITE sip:номер циски@192.168.1.7, with session description. и все. ну кроме bye при окончании вызова. в CDR отчете вижу при звонке на циску что вызов на 560 (это номер на циске) FAILED, а когда смотрю на отчет при звонке с нее вижу что 2011-09-09 15:56:43 560 s SIP/192.168.1.7-00000041 ANSWERED 6s. может я чего не понимаю, но почему вместо номера стоит буква s, хотя в дампе при звонке с циски я вижу что идет вызов на нужный номер -
121139 4727.647115 192.168.1.61 192.168.1.7 SIP/SDP Request: INVITE sip:569@192.168.1.7, with session description
и еще
121142 4727.690690 192.168.1.61 192.168.1.7 SIP Request: ACK sip:569@192.168.1.7:5060

помогите разобраться плизз.
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение ded »

AppendIX писал(а): вижу при звонке на циску что вызов на 560 (это номер на циске) FAILED .
и
AppendIX писал(а): SIP Request: ACK sip:569@192.168.1.7:5060
Нестыковочка!
sip show peers?
sip show peer 560?
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AppendIX
Сообщения: 16
Зарегистрирован: 08 сен 2011, 13:04

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение AppendIX »

Name/username Host Dyn Nat ACL Port Status
560 192.168.1.61 A 5060 Unmonitored
568/568 192.168.1.155 D N A 13842 OK (1 ms)
569/569 192.168.1.79 D N A 8859 OK (2 ms)
MOR/логин ip-провайдера N 5060 Unmonitored
4 sip peers [Monitored: 2 online, 0 offline Unmonitored: 2 online, 0 offline]

Код: Выделить всё

* Name       : 560
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 560@device
  VM Extension : *97
  LastMsgsSent : 32767/65535
  Call limit   : 50
  Dynamic      : No
  Callerid     : "device" <560>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Nat          : RFC3581
  ACL          : Yes
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  Forward Loop : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 192.168.1.61
  Addr->IP     : 192.168.1.61 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 
  SIP Options  : join norefersub replaces replace 
  Codecs       : 0xe (gsm|ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20,gsm:20)
  Auto-Framing :  No 
  100 on REG   : No
  Status       : Unmonitored
  Useragent    : 
  Reg. Contact : 
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   :
оно?
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение ded »

Ещё бОльшая нестыковочка:
Request: INVITE sip:569@192.168.1.7
569/569 192.168.1.79 D N A 8859 OK (2 ms)
1) Непонятно на каком ИП ваш 569;
2) Сложно вам будет без регистрации. Телефон грубо говоря не знает, что он 560-й, потому и s
Попробуйте его в extensions объявить как Custom device, со строкой Dial SIP/92.168.1.61/560
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AppendIX
Сообщения: 16
Зарегистрирован: 08 сен 2011, 13:04

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение AppendIX »

569 на 192.168.1.79
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение ded »

А вот тут другой - sip:569@192.168.1.7:5060
AppendIX писал(а): в CDR отчете вижу при звонке на циску что вызов на 560 (это номер на циске) FAILED, а когда смотрю на отчет при звонке с нее вижу что 2011-09-09 15:56:43 560 s SIP/192.168.1.7-00000041 ANSWERED 6s. может я чего не понимаю, но почему вместо номера стоит буква s, хотя в дампе при звонке с циски я вижу что идет вызов на нужный номер -
121139 4727.647115 192.168.1.61 192.168.1.7 SIP/SDP Request: INVITE sip:569@192.168.1.7, with session description
и еще
121142 4727.690690 192.168.1.61 192.168.1.7 SIP Request: ACK sip:569@192.168.1.7:5060.
ded писал(а):2) Сложно вам будет без регистрации. Телефон грубо говоря не знает, что он 560-й, потому и s
Попробуйте его в extensions объявить как Custom device, со строкой Dial SIP/92.168.1.61/560
снимайте sip set debug peer 560
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AppendIX
Сообщения: 16
Зарегистрирован: 08 сен 2011, 13:04

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение AppendIX »

вначале звоню на 560 (прошитый на сип цискофон) с 568 (сипфон x-lite) а потом в обратную сторону. бред какой-то ((

Код: Выделить всё

Connected to Asterisk 1.6.2.13 currently running on rosa (pid = 8625)
Verbosity is at least 3

звонок с 568 на 560

Reliably Transmitting (no NAT) to 192.168.1.61:5060:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK1e2afa59;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as5bac2ed9
To: <sip:192.168.1.61>
Contact: <sip:Unknown@192.168.1.7>
Call-ID: 74cf208e41df270238f6baba1e023a77@192.168.1.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:26:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '74cf208e41df270238f6baba1e023a77@192.168.1.7' Method: OPTIONS
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [560@from-internal:1] Macro("SIP/568-0000006a", "exten-vm,novm,560") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/568-0000006a", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/568-0000006a", "AMPUSER=568") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/568-0000006a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/568-0000006a", "1?Set(REALCALLERIDNUM=568)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/568-0000006a", "AMPUSER=568") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/568-0000006a", "AMPUSERCIDNAME=568") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/568-0000006a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/568-0000006a", "AMPUSERCID=568") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/568-0000006a", "CALLERID(all)="568" <568>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/568-0000006a", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/568-0000006a", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/568-0000006a", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/568-0000006a", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/568-0000006a", "Using CallerID "568" <568>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/568-0000006a", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/568-0000006a", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/568-0000006a", "EXTTOCALL=560") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/568-0000006a", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/568-0000006a", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/568-0000006a", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/568-0000006a", "record-enable,560,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/568-0000006a", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/568-0000006a", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/568-0000006a", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/568-0000006a", "1?IN") in new stack
    -- Goto (macro-record-enable,s,20)
    -- Executing [s@macro-record-enable:20] ExecIf("SIP/568-0000006a", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/568-0000006a", "dial,,tr,560") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/568-0000006a", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/568-0000006a", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is '568' number is '568'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 560 to extension map
    -- dialparties.agi: Extension 560 cf is disabled
    -- dialparties.agi: Extension 560 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
 dialparties.agi: Extension 560 has ExtensionState: 0
    -- dialparties.agi: Checking CW and CFB status for extension 560
    -- dialparties.agi: dbset CALLTRACE/560 to 568
    -- dialparties.agi: Filtered ARG3: 560
    -- <SIP/568-0000006a>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/568-0000006a", "SIP/560/192.168.1.61,,tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 192.168.1.7 port 19094
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.61:5060:
INVITE sip:192.168.1.61@192.168.1.61:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK389819ce;rport
Max-Forwards: 70
From: "568" <sip:568@192.168.1.7>;tag=as2998138f
To: <sip:192.168.1.61@192.168.1.61:5060>
Contact: <sip:568@192.168.1.7>
Call-ID: 4e18410c0e3e1bf2191b88235326b4e3@192.168.1.7
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:26:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 715844044 715844044 IN IP4 192.168.1.7
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.7
t=0 0
m=audio 19094 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called 560/192.168.1.61
Transmitting (no NAT) to 192.168.1.61:5060:
ACK sip:192.168.1.61@192.168.1.61:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK389819ce;rport
Max-Forwards: 70
From: "568" <sip:568@192.168.1.7>;tag=as2998138f
To: <sip:192.168.1.61@192.168.1.61:5060>;tag=001da2f3fbb6001a46562516-19d660cf
Contact: <sip:568@192.168.1.7>
Call-ID: 4e18410c0e3e1bf2191b88235326b4e3@192.168.1.7
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
    -- SIP/560-0000006b is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dial:8] Set("SIP/568-0000006a", "DIALSTATUS=CONGESTION") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/568-0000006a", "0?CONGESTION,1") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/568-0000006a", "0?exit,return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/568-0000006a", "SV_DIALSTATUS=CONGESTION") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/568-0000006a", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/568-0000006a", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/568-0000006a", "DIALSTATUS=CONGESTION") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/568-0000006a", "Voicemail is 'novm'") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/568-0000006a", "1?s-CONGESTION,1") in new stack
    -- Goto (macro-exten-vm,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-exten-vm:1] NoOp("SIP/568-0000006a", "IVR_RETVM:  IVR_CONTEXT: ") in new stack
    -- Executing [s-CONGESTION@macro-exten-vm:2] GotoIf("SIP/568-0000006a", "0?exit,1") in new stack
    -- Executing [s-CONGESTION@macro-exten-vm:3] PlayTones("SIP/568-0000006a", "congestion") in new stack
    -- Executing [s-CONGESTION@macro-exten-vm:4] Congestion("SIP/568-0000006a", "10") in new stack
  == Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on 'SIP/568-0000006a' in macro 'exten-vm'
  == Spawn extension (from-internal, 560, 1) exited non-zero on 'SIP/568-0000006a'
    -- Executing [h@from-internal:1] Macro("SIP/568-0000006a", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/568-0000006a", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/568-0000006a", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/568-0000006a", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/568-0000006a", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/568-0000006a", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/568-0000006a", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/568-0000006a' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/568-0000006a'
Really destroying SIP dialog '4e18410c0e3e1bf2191b88235326b4e3@192.168.1.7' Method: INVITE




а теперь звонок с 560 на 568

Reliably Transmitting (no NAT) to 192.168.1.61:5060:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK38e9f5e9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as567f95bd
To: <sip:192.168.1.61>
Contact: <sip:Unknown@192.168.1.7>
Call-ID: 2504cbfd1969a9ad350b208a5bc4e045@192.168.1.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:27:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '2504cbfd1969a9ad350b208a5bc4e045@192.168.1.7' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.1.61:5060:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK06b33b98;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as50c0cc77
To: <sip:192.168.1.61>
Contact: <sip:Unknown@192.168.1.7>
Call-ID: 5ff1565b5619a15d799a079e4ec68afa@192.168.1.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:28:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '5ff1565b5619a15d799a079e4ec68afa@192.168.1.7' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.1.61:5060:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK281d5128;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as2023b336
To: <sip:192.168.1.61>
Contact: <sip:Unknown@192.168.1.7>
Call-ID: 32b7eb851996602d1868e6f468413b4e@192.168.1.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:29:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '32b7eb851996602d1868e6f468413b4e@192.168.1.7' Method: OPTIONS
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
Reliably Transmitting (no NAT) to 192.168.1.61:5060:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK24db4b15;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as7599eb03
To: <sip:192.168.1.61>
Contact: <sip:Unknown@192.168.1.7>
Call-ID: 44b798eb553ff0ad12599a630b4b7cf3@192.168.1.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:30:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '44b798eb553ff0ad12599a630b4b7cf3@192.168.1.7' Method: OPTIONS
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 192.168.1.61 : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 116
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 G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x150c (ulaw|alaw|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.61:17658
Looking for 568 in from-sip-external (domain 192.168.1.7)
list_route: hop: <sip:560@192.168.1.61:5060;transport=udp>

<--- Transmitting (no NAT) to 192.168.1.61:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK038ad02b;received=192.168.1.61
From: "560" <sip:560@192.168.1.7>;tag=001da2f3fbb6001fdd78d193-7c8bcb8b
To: <sip:568@192.168.1.7>
Call-ID: 001da2f3-fbb60007-8cb5c8b3-d3479beb@192.168.1.61
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:568@192.168.1.7>
Content-Length: 0


<------------>
    -- Executing [568@from-sip-external:1] NoOp("SIP/192.168.1.7-0000006c", "Received incoming SIP connection from unknown peer to 568") in new stack
    -- Executing [568@from-sip-external:2] Set("SIP/192.168.1.7-0000006c", "DID=568") in new stack
    -- Executing [568@from-sip-external:3] Goto("SIP/192.168.1.7-0000006c", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/192.168.1.7-0000006c", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/192.168.1.7-0000006c", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2011-09-09 18:31:23.426 MSD.
    -- Executing [s@from-sip-external:6] Answer("SIP/192.168.1.7-0000006c", "") in new stack
Audio is at 192.168.1.7 port 17074
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.1.61:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK038ad02b;received=192.168.1.61
From: "560" <sip:560@192.168.1.7>;tag=001da2f3fbb6001fdd78d193-7c8bcb8b
To: <sip:568@192.168.1.7>;tag=as412d2cd0
Call-ID: 001da2f3-fbb60007-8cb5c8b3-d3479beb@192.168.1.61
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:568@192.168.1.7>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 831262544 831262544 IN IP4 192.168.1.7
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.7
t=0 0
m=audio 17074 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

<------------>
    -- Executing [s@from-sip-external:7] Wait("SIP/192.168.1.7-0000006c", "2") in new stack
    -- Executing [s@from-sip-external:8] Playback("SIP/192.168.1.7-0000006c", "ss-noservice") in new stack
    -- <SIP/192.168.1.7-0000006c> Playing 'ss-noservice.gsm' (language 'en')
    -- Executing [s@from-sip-external:9] PlayTones("SIP/192.168.1.7-0000006c", "congestion") in new stack
    -- Executing [s@from-sip-external:10] Congestion("SIP/192.168.1.7-0000006c", "5") in new stack
Reliably Transmitting (no NAT) to 192.168.1.61:5060:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK1ac1a762;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.7>;tag=as08cf218d
To: <sip:192.168.1.61>
Contact: <sip:Unknown@192.168.1.7>
Call-ID: 0ca91b3f330a59927ce51e1f50557d17@192.168.1.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 09 Sep 2011 14:31:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '0ca91b3f330a59927ce51e1f50557d17@192.168.1.7' Method: OPTIONS
  == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/192.168.1.7-0000006c'
    -- Executing [h@from-sip-external:1] Hangup("SIP/192.168.1.7-0000006c", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/192.168.1.7-0000006c'
Scheduling destruction of SIP dialog '001da2f3-fbb60007-8cb5c8b3-d3479beb@192.168.1.61' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:560@192.168.1.61:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.1.61, port 5060
Reliably Transmitting (no NAT) to 192.168.1.61:5060:
BYE sip:560@192.168.1.61:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK4a6b0b1a;rport
Max-Forwards: 70
From: <sip:568@192.168.1.7>;tag=as412d2cd0
To: "560" <sip:560@192.168.1.7>;tag=001da2f3fbb6001fdd78d193-7c8bcb8b
Call-ID: 001da2f3-fbb60007-8cb5c8b3-d3479beb@192.168.1.61
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Really destroying SIP dialog '001da2f3-fbb60007-8cb5c8b3-d3479beb@192.168.1.61' Method: ACK
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AppendIX
Сообщения: 16
Зарегистрирован: 08 сен 2011, 13:04

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение AppendIX »

ППЦ. проблема дозвона на прошитый цискофон с софтфона решилась изменением строки dial у пира 560 c SIP/560 на SIP/560/560
Сделал по наитию, но как-то получилось :)

а вот что теперь делать с исходящими с 560 на остальные - ума не приложу блин :(
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Cisco7906 не регистрируется на Asterisk по SIP

Сообщение ded »

Какое к шутам наитие? SIP/560/560 расшифровывает прописаный пир 560 в 92.168.1.61, так что всё в рамках формулы Технология/узел/экстеншн

Дальше два взаимоисключающих метода
1) username & secret
2) insecure=invite

или добить проблему регистрации
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