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chan_sip.c:20870 handle_response_invite

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

Модераторы: april22, Zavr2008

Ответить
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: chan_sip.c:20870 handle_response_invite

Сообщение Vlad1983 »

включать дебаг общий и sip
будет видно как происходит выбор пира
ЛС: @rostel
Вернуться к началу
test-sip
Сообщения: 19
Зарегистрирован: 12 мар 2014, 09:28

Re: chan_sip.c:20870 handle_response_invite

Сообщение test-sip »

sip debug на астериске где раньше был этот 202 номер

Код: Выделить всё

<--- SIP read from UDP:192.168.111.111:5060 --->
INVITE sip:333@192.168.100.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK2fa87a26
Max-Forwards: 70
From: "202" <sip:202@192.168.111.111>;tag=as1dbbe900
To: <sip:333@192.168.100.111>
Contact: <sip:202@192.168.111.111:5060>
Call-ID: 379b003c1392bdc56ac5fb446692ce60@192.168.111.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Wed, 12 Mar 2014 12:13:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 598210340 598210340 IN IP4 192.168.111.111
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.111.111
t=0 0
m=audio 13524 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.111.111:5060 (NAT)
Using INVITE request as basis request - 379b003c1392bdc56ac5fb446692ce60@192.168.111.111:5060
Found peer '202' for '202' from 192.168.111.111:5060

<--- Reliably Transmitting (no NAT) to 192.168.111.111:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK2fa87a26;received=192.168.111.111
From: "202" <sip:202@192.168.111.111>;tag=as1dbbe900
To: <sip:333@192.168.100.111>;tag=as1bcd3d5c
Call-ID: 379b003c1392bdc56ac5fb446692ce60@192.168.111.111:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.15.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="15a93fec"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '379b003c1392bdc56ac5fb446692ce60@192.168.111.111:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.111.111:5060 --->
ACK sip:333@192.168.100.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK2fa87a26
Max-Forwards: 70
From: "202" <sip:202@192.168.111.111>;tag=as1dbbe900
To: <sip:333@192.168.100.111>;tag=as1bcd3d5c
Contact: <sip:202@192.168.111.111:5060>
Call-ID: 379b003c1392bdc56ac5fb446692ce60@192.168.111.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0

<------------->
sip debug нового тестового астериска

Код: Выделить всё

<--- Reliably Transmitting (no NAT) to 192.168.0.102:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK1257472033145347;received=192.168.0.102;rport=5060
From: Test202 <sip:202@192.168.111.111:5060>;tag=120161177
To: "333" <sip:333@192.168.111.111:5060>;tag=as269826dd
Call-ID: 25211640630300-93132832711735@192.168.0.102
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '176a9116193919393d012858512a46a4@192.168.111.111:5060' Method: INVITE

<--- SIP read from UDP:192.168.0.102:5060 --->
ACK sip:333@192.168.111.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK1257472033145347;rport
From: Test202 <sip:202@192.168.111.111:5060>;tag=120161177
To: "333" <sip:333@192.168.111.111:5060>;tag=as269826dd
Call-ID: 25211640630300-93132832711735@192.168.0.102
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.100.111:5060:
OPTIONS sip:192.168.100.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK2b941aa0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as36625927
To: <sip:192.168.100.111>
Contact: <sip:asterisk@192.168.111.111:5060>
Call-ID: 60a3a4ad70c653f70b40671b5f4dff5b@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Wed, 12 Mar 2014 12:21:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.100.111:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK2b941aa0;received=192.168.111.111;rport=5060
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as36625927
To: <sip:192.168.100.111>;tag=as049c8c5e
Call-ID: 60a3a4ad70c653f70b40671b5f4dff5b@192.168.111.111:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '60a3a4ad70c653f70b40671b5f4dff5b@192.168.111.111:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.100.111:5060 --->
OPTIONS sip:192.168.111.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.111:5060;branch=z9hG4bK542b2a6b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.100.111>;tag=as3c0d29d6
To: <sip:192.168.111.111>
Contact: <sip:asterisk@192.168.100.111:5060>
Call-ID: 1f5a4a676d2e40132ca12256051d8d76@192.168.100.111:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.1
Date: Wed, 12 Mar 2014 12:21:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 192.168.111.111)

<--- Transmitting (NAT) to 192.168.100.111:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.100.111:5060;branch=z9hG4bK542b2a6b;received=192.168.100.111;rport=5060
From: "asterisk" <sip:asterisk@192.168.100.111>;tag=as3c0d29d6
To: <sip:192.168.111.111>;tag=as62d70710
Call-ID: 1f5a4a676d2e40132ca12256051d8d76@192.168.100.111:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1f5a4a676d2e40132ca12256051d8d76@192.168.100.111:5060' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '25211640630300-93132832711735@192.168.0.102' Method: ACK

<--- SIP read from UDP:192.168.0.102:5060 --->
REGISTER sip:192.168.111.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK1889202402375319992;rport
From: Test202 <sip:202@192.168.111.111:5060>;tag=3014613116
To: Test202 <sip:202@192.168.111.111:5060>
Call-ID: 253071584526290-3333111729886@192.168.0.102
CSeq: 2823 REGISTER
Contact: <sip:202@192.168.0.102:5060>
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.102:5060 (NAT)

<--- Transmitting (no NAT) to 192.168.0.102:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK1889202402375319992;received=192.168.0.102;rport=5060
From: Test202 <sip:202@192.168.111.111:5060>;tag=3014613116
To: Test202 <sip:202@192.168.111.111:5060>;tag=as31b9a581
Call-ID: 253071584526290-3333111729886@192.168.0.102
CSeq: 2823 REGISTER
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d6c16cc"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '253071584526290-3333111729886@192.168.0.102' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.0.102:5060 --->
REGISTER sip:192.168.111.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK2762178091941213003;rport
From: Test202 <sip:202@192.168.111.111:5060>;tag=3014613116
To: Test202 <sip:202@192.168.111.111:5060>
Call-ID: 253071584526290-3333111729886@192.168.0.102
CSeq: 2824 REGISTER
Contact: <sip:202@192.168.0.102:5060>
Authorization: Digest username="202", realm="asterisk", nonce="5d6c16cc", uri="sip:192.168.111.111", response="17fc324282a4e4e91263a2b8849f259a", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.102:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
OPTIONS sip:202@192.168.0.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK51cce668
Max-Forwards: 70
From: "asterisk" <sip:202@192.168.111.111>;tag=as5a0d946b
To: <sip:202@192.168.0.102:5060>
Contact: <sip:202@192.168.111.111:5060>
Call-ID: 166448c11700fa7b57a32c694b8293c7@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Wed, 12 Mar 2014 12:21:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.0.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK2762178091941213003;received=192.168.0.102;rport=5060
From: Test202 <sip:202@192.168.111.111:5060>;tag=3014613116
To: Test202 <sip:202@192.168.111.111:5060>;tag=as31b9a581
Call-ID: 253071584526290-3333111729886@192.168.0.102
CSeq: 2824 REGISTER
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:202@192.168.0.102:5060>;expires=60
Date: Wed, 12 Mar 2014 12:21:50 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '253071584526290-3333111729886@192.168.0.102' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.0.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK51cce668
From: "asterisk" <sip:202@192.168.111.111>;tag=as5a0d946b
To: <sip:202@192.168.0.102:5060>
Call-ID: 166448c11700fa7b57a32c694b8293c7@192.168.111.111:5060
CSeq: 102 OPTIONS
Contact: <sip:202@192.168.0.102:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '166448c11700fa7b57a32c694b8293c7@192.168.111.111:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.0.101:5060 --->
REGISTER sip:192.168.111.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK32329250922916629246;rport
From: Test <sip:201@192.168.111.111:5060>;tag=2974532410
To: Test <sip:201@192.168.111.111:5060>
Call-ID: 249002951313917-24202153814174@192.168.0.101
CSeq: 3265 REGISTER
Contact: <sip:201@192.168.0.101:5060>
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.101:5060 (NAT)

<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK32329250922916629246;received=192.168.0.101;rport=5060
From: Test <sip:201@192.168.111.111:5060>;tag=2974532410
To: Test <sip:201@192.168.111.111:5060>;tag=as3a148f2a
Call-ID: 249002951313917-24202153814174@192.168.0.101
CSeq: 3265 REGISTER
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5edf9256"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '249002951313917-24202153814174@192.168.0.101' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.0.101:5060 --->
REGISTER sip:192.168.111.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK74019297149558988;rport
From: Test <sip:201@192.168.111.111:5060>;tag=2974532410
To: Test <sip:201@192.168.111.111:5060>
Call-ID: 249002951313917-24202153814174@192.168.0.101
CSeq: 3266 REGISTER
Contact: <sip:201@192.168.0.101:5060>
Authorization: Digest username="201", realm="asterisk", nonce="5edf9256", uri="sip:192.168.111.111", response="a36380e4782141d4dea19768a1cf7fb1", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.101:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.101:5060:
OPTIONS sip:201@192.168.0.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK1a21f8ef
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as560c2457
To: <sip:201@192.168.0.101:5060>
Contact: <sip:asterisk@192.168.111.111:5060>
Call-ID: 7293c4aa6ce517a239fc8b2c28a1903a@192.168.111.111:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.20.0
Date: Wed, 12 Mar 2014 12:21:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK74019297149558988;received=192.168.0.101;rport=5060
From: Test <sip:201@192.168.111.111:5060>;tag=2974532410
To: Test <sip:201@192.168.111.111:5060>;tag=as3a148f2a
Call-ID: 249002951313917-24202153814174@192.168.0.101
CSeq: 3266 REGISTER
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:201@192.168.0.101:5060>;expires=60
Date: Wed, 12 Mar 2014 12:21:50 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '249002951313917-24202153814174@192.168.0.101' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK1a21f8ef
From: "asterisk" <sip:asterisk@192.168.111.111>;tag=as560c2457
To: <sip:201@192.168.0.101:5060>
Call-ID: 7293c4aa6ce517a239fc8b2c28a1903a@192.168.111.111:5060
CSeq: 102 OPTIONS
Contact: <sip:201@192.168.0.101:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7293c4aa6ce517a239fc8b2c28a1903a@192.168.111.111:5060' Method: OPTIONS
debug peer 202 на новом тестовом астериск

Код: Выделить всё

<--- Reliably Transmitting (no NAT) to 192.168.0.102:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK3095323891252212730;received=192.168.0.102;rport=5060
From: Test202 <sip:202@192.168.111.111:5060>;tag=845518332
To: "333" <sip:333@192.168.111.111:5060>;tag=as07357a29
Call-ID: 30953520110532-254862135811365@192.168.0.102
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.0.102:5060 --->
ACK sip:333@192.168.111.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK3095323891252212730;rport
From: Test202 <sip:202@192.168.111.111:5060>;tag=845518332
To: "333" <sip:333@192.168.111.111:5060>;tag=as07357a29
Call-ID: 30953520110532-254862135811365@192.168.0.102
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '30953520110532-254862135811365@192.168.0.102' Method: ACK
Вернуться к началу
test-sip
Сообщения: 19
Зарегистрирован: 12 мар 2014, 09:28

Re: chan_sip.c:20870 handle_response_invite

Сообщение test-sip »

Вот это неправильно скорее всего realm="asterisk"
Вернуться к началу
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: chan_sip.c:20870 handle_response_invite

Сообщение ded »

Скорее всего Вы недочитали Будущее телефонии.
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: chan_sip.c:20870 handle_response_invite

Сообщение Vlad1983 »

включать дебаг общий и sip
на сервере назначения ("где раньше был этот 202 номер")
ЛС: @rostel
Вернуться к началу
test-sip
Сообщения: 19
Зарегистрирован: 12 мар 2014, 09:28

Re: chan_sip.c:20870 handle_response_invite

Сообщение test-sip »

Включенный core set debug и sip set debug на asterisk где раньше был 202 номер

Код: Выделить всё

<--- SIP read from UDP:192.168.111.111:5060 --->
INVITE sip:333@192.168.100.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5db79df7
Max-Forwards: 70
From: "202" <sip:202@192.168.111.111>;tag=as069d621e
To: <sip:333@192.168.100.111>
Contact: <sip:202@192.168.111.111:5060>
Call-ID: 6d47d4040d8375ba3fb6a47d767f6aa0@192.168.111.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.20.0
Date: Wed, 12 Mar 2014 13:22:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1607241875 1607241875 IN IP4 192.168.111.111
s=Asterisk PBX 1.8.20.0
c=IN IP4 192.168.111.111
t=0 0
m=audio 19052 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->

[Asterisk*CLI> 
[0K--- (14 headers 12 lines) ---

[Asterisk*CLI> 
[0KSending to 192.168.111.111:5060 (NAT)

[Asterisk*CLI> 
[0KUsing INVITE request as basis request - 6d47d4040d8375ba3fb6a47d767f6aa0@192.168.111.111:5060

[Asterisk*CLI> 
[0KFound peer '202' for '202' from 192.168.111.111:5060

[Asterisk*CLI> 
[0K
<--- Reliably Transmitting (no NAT) to 192.168.111.111:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5db79df7;received=192.168.111.111
From: "202" <sip:202@192.168.111.111>;tag=as069d621e
To: <sip:333@192.168.100.111>;tag=as79b1b474
Call-ID: 6d47d4040d8375ba3fb6a47d767f6aa0@192.168.111.111:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.15.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ed30f9b"
Content-Length: 0


<------------>

[Asterisk*CLI> 
[0KScheduling destruction of SIP dialog '6d47d4040d8375ba3fb6a47d767f6aa0@192.168.111.111:5060' in 6400 ms (Method: INVITE)

[Asterisk*CLI> 
[0K
<--- SIP read from UDP:192.168.111.111:5060 --->
ACK sip:333@192.168.100.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.111:5060;branch=z9hG4bK5db79df7
Max-Forwards: 70
From: "202" <sip:202@192.168.111.111>;tag=as069d621e
To: <sip:333@192.168.100.111>;tag=as79b1b474
Contact: <sip:202@192.168.111.111:5060>
Call-ID: 6d47d4040d8375ba3fb6a47d767f6aa0@192.168.111.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.20.0
Content-Length: 0
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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: chan_sip.c:20870 handle_response_invite

Сообщение Vlad1983 »

Found peer '202' for '202' from 192.168.111.111:5060
ещё есть вопросы?
ЛС: @rostel
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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: chan_sip.c:20870 handle_response_invite

Сообщение ded »

Vlad1983 » 46 минут назад ты оочень высоко просто задрал нос.
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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: chan_sip.c:20870 handle_response_invite

Сообщение Vlad1983 »

ded, в чем?

пир 202 присутствует на 192.168.100.111 и любой INVITE c "From: "202" <sip:202@" будет матчиться на него (соответственно и запрос авторизации в ответ)?

не удивлюсь что
asterisk -rx "sip show peer 202"
тоже даст выхлоп

там по утверждению ТС всё зачищено от 202
ЛС: @rostel
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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: chan_sip.c:20870 handle_response_invite

Сообщение ded »

Это я пытаюсь шутить :)
test-sip писал(а):Ну а по поводу обид, обид нет. просто Вы задрали нос оочень высоко.
Конечно ТС не смотрит на дебаг даже когда публикует его.
пир 202 присутствует на 192.168.100.111 и любой INVITE c "From: "202" <sip:202@" будет пытаться авторизоваться.
Но у него всегда есть решение: купить новый телефон (или просто взять с полки другой).
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