Re: Cisco7906 не регистрируется на Asterisk по SIP
Добавлено: 09 сен 2011, 19:34
сейчас ситуация следующая: работает входящая связь с остальных номеров (оба на софтфонах) на номер 560 (прошитая циска), также софтфоны без проблем общаются между собой. а вот исходящей связи с номера 560 нет никуда (ни внутрь ни наружу). ответ один - набранный номер not in service и все тут. дебаг вот такойded писал(а): Дальше два взаимоисключающих метода
1) username & secret
2) insecure=invite
или добить проблему регистрации
Код: Выделить всё
== 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 192.168.1.61:17110
Looking for 569 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=z9hG4bK6b6bc447;received=192.168.1.61
From: "560" <sip:560@192.168.1.7>;tag=001da2f3fbb6007cb7334ad3-44933e97
To: <sip:569@192.168.1.7>
Call-ID: 001da2f3-fbb6001a-fed10e43-dd3a9667@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:569@192.168.1.7>
Content-Length: 0
<------------>
-- Executing [569@from-sip-external:1] NoOp("SIP/192.168.1.7-0000009d", "Received incoming SIP connection from unknown peer to 569") in new stack
-- Executing [569@from-sip-external:2] Set("SIP/192.168.1.7-0000009d", "DID=569") in new stack
-- Executing [569@from-sip-external:3] Goto("SIP/192.168.1.7-0000009d", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/192.168.1.7-0000009d", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/192.168.1.7-0000009d", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2011-09-09 19:33:53.381 MSD.
-- Executing [s@from-sip-external:6] Answer("SIP/192.168.1.7-0000009d", "") in new stack
Audio is at 192.168.1.7 port 12570
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=z9hG4bK6b6bc447;received=192.168.1.61
From: "560" <sip:560@192.168.1.7>;tag=001da2f3fbb6007cb7334ad3-44933e97
To: <sip:569@192.168.1.7>;tag=as35ee3a23
Call-ID: 001da2f3-fbb6001a-fed10e43-dd3a9667@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:569@192.168.1.7>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 2068856907 2068856907 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 12570 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/192.168.1.7-0000009d", "2") in new stack
-- Executing [s@from-sip-external:8] Playback("SIP/192.168.1.7-0000009d", "ss-noservice") in new stack
-- <SIP/192.168.1.7-0000009d> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-sip-external:9] PlayTones("SIP/192.168.1.7-0000009d", "congestion") in new stack
-- Executing [s@from-sip-external:10] Congestion("SIP/192.168.1.7-0000009d", "5") in new stack
== Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/192.168.1.7-0000009d'
-- Executing [h@from-sip-external:1] Hangup("SIP/192.168.1.7-0000009d", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/192.168.1.7-0000009d'
Scheduling destruction of SIP dialog '001da2f3-fbb6001a-fed10e43-dd3a9667@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=z9hG4bK1aa37c43;rport
Max-Forwards: 70
From: <sip:569@192.168.1.7>;tag=as35ee3a23
To: "560" <sip:560@192.168.1.7>;tag=001da2f3fbb6007cb7334ad3-44933e97
Call-ID: 001da2f3-fbb6001a-fed10e43-dd3a9667@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-fbb6001a-fed10e43-dd3a9667@192.168.1.61' Method: ACK
логин и пароль 100% верны, insecure=invite попробую в выходные, но проблема решилась несколько другим способом, а именно разрешением астериску принимать входящие анонимные звонки