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не ходят звнки в одну сторону 401 Unauthorized [Решено]

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На freePBX настроен обычный пользователь с номером 4567.

На Астере с Реалтаймом подключен транков номер 4567 в FreePBX, регистрация проходит. Входящие звонки с FreePBX на Realtime идут нормально.

Исходящеи не идет с Realtime на freePBX, в дебаге высвечивает 401 Unauthorized, уже мачаюсь пол дня чето никак не добпру почему.

Настройки все правильные раньше всегда работало.

На астере с реалтямом настроен 1 peer 200, он регестрируется и на него входящие звонки приходят.

настройки Aster-Realtime

sip.conf

[general]
rtcachefriends=yes
rtcache=yes
type=friend
disallow=all
allow=alaw
context=test

register => 4567:PASS@192.168.0.100/4567

[4567]
username=4567
type=friend
secter=PASS
nat=no
fromuser=4567
fromdomain=192.168.0.100
insecure=port,invite
host=192.168.0.100
dtmfmode=rfc2833
context=incoming
canreinvite=yes

Дебаг astrealtime

  == Using SIP RTP CoS mark 5
Audio is at 17078
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.100:5060:
INVITE sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK031a9d01
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as11aca1a8
To: <sip:101@192.168.0.100>
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 047d3ae356f68c10504c6d5114aad5aa@192.168.0.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.0
Date: Wed, 22 Jan 2014 05:22:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2064116826 2064116826 IN IP4 192.168.0.250
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 17078 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

---
    -- Called SIP/4567/101

<--- SIP read from UDP:192.168.0.100:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK031a9d01;received=192.168.0.250
From: <sip:4567@192.168.0.100>;tag=as11aca1a8
To: <sip:101@192.168.0.100>;tag=as62288ca8
Call-ID: 047d3ae356f68c10504c6d5114aad5aa@192.168.0.100
CSeq: 102 INVITE
Server: FPBX-2.8.1(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f3e8eab"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.100:5060:
ACK sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK031a9d01
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as11aca1a8
To: <sip:101@192.168.0.100>;tag=as62288ca8
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 047d3ae356f68c10504c6d5114aad5aa@192.168.0.100
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.5.0
Content-Length: 0


---
Audio is at 17078
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.100:5060:
INVITE sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK210a42be
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as11aca1a8
To: <sip:101@192.168.0.100>
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 047d3ae356f68c10504c6d5114aad5aa@192.168.0.100
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.5.0
Authorization: Digest username="4567", realm="asterisk", algorithm=MD5, uri="sip:101@192.168.0.100", nonce="0f3e8eab", response="788403e49432a29926405a30bceb0ca8"
Date: Wed, 22 Jan 2014 05:22:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2064116826 2064116827 IN IP4 192.168.0.250
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 17078 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

---

<--- SIP read from UDP:192.168.0.100:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK210a42be;received=192.168.0.250
From: <sip:4567@192.168.0.100>;tag=as11aca1a8
To: <sip:101@192.168.0.100>;tag=as62288ca8
Call-ID: 047d3ae356f68c10504c6d5114aad5aa@192.168.0.100
CSeq: 103 INVITE
Server: FPBX-2.8.1(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.100:5060:
ACK sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK210a42be
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as11aca1a8
To: <sip:101@192.168.0.100>;tag=as62288ca8
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 047d3ae356f68c10504c6d5114aad5aa@192.168.0.100
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.5.0
Content-Length: 0


---
[Jan 22 09:22:58] WARNING[5588][C-00000009]: chan_sip.c:22927 handle_response_invite: Received response: "Forbidden" from '<sip:4567@192.168.0.100>;tag=as11aca1a8'
Scheduling destruction of SIP dialog '047d3ae356f68c10504c6d5114aad5aa@192.168.0.100' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/200-00000012' status is 'CHANUNAVAIL'

Дебаг Астер FreeePBX

<--- SIP read from UDP:192.168.0.250:5060 --->
INVITE sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK1d47f67e
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as357caf30
To: <sip:101@192.168.0.100>
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.0
Date: Wed, 22 Jan 2014 05:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1245284008 1245284008 IN IP4 192.168.0.250
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 11230 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 192.168.0.250:5060 (no NAT)
Sending to 192.168.0.250:5060 (no NAT)
Using INVITE request as basis request - 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
Found peer '4567' for '4567' from 192.168.0.250:5060

<--- Reliably Transmitting (no NAT) to 192.168.0.250:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK1d47f67e;received=192.168.0.250
From: <sip:4567@192.168.0.100>;tag=as357caf30
To: <sip:101@192.168.0.100>;tag=as355a7343
Call-ID: 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
CSeq: 102 INVITE
Server: FPBX-2.8.1(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39647031"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.250:5060 --->
ACK sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK1d47f67e
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as357caf30
To: <sip:101@192.168.0.100>;tag=as355a7343
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.5.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.250:5060 --->
INVITE sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK243025a4
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as357caf30
To: <sip:101@192.168.0.100>
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.5.0
Authorization: Digest username="4567", realm="asterisk", algorithm=MD5, uri="sip:101@192.168.0.100", nonce="39647031", response="2035ca0d1b9e482c601b483c3baefd79"
Date: Wed, 22 Jan 2014 05:24:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1245284008 1245284009 IN IP4 192.168.0.250
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 11230 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
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.0.250:5060 (no NAT)
Using INVITE request as basis request - 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
Found peer '4567' for '4567' from 192.168.0.250:5060

<--- Reliably Transmitting (no NAT) to 192.168.0.250:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK243025a4;received=192.168.0.250
From: <sip:4567@192.168.0.100>;tag=as357caf30
To: <sip:101@192.168.0.100>;tag=as355a7343
Call-ID: 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
CSeq: 103 INVITE
Server: FPBX-2.8.1(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.250:5060 --->
ACK sip:101@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK243025a4
Max-Forwards: 70
From: <sip:4567@192.168.0.100>;tag=as357caf30
To: <sip:101@192.168.0.100>;tag=as355a7343
Contact: <sip:4567@192.168.0.250:5060>
Call-ID: 3eb3681b2dfd0fe472500a66126d0e5b@192.168.0.100
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.5.0
Content-Length: 0

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

РЕШЕНО

МЛЯ.... разобрался косяк был в правописание

secter=PASS - не правильно
secret=PASS - правильно

Все пошло.

удалить закрыть спам изменить тег редактировать

спросил 2014-01-22 12:47:15 +0400

romariosar Gravatar romariosar flag of Russian Federation
588 ● 93 ● 11 ● 44
http://www.webunix.ru/

обновил 2014-01-22 14:07:24 +0400

1 Ответ

Сорт. по » старые новые больше голосов
0

Попробуйте поставить canreinvite=no в настройках пира 4567

ссылка удалить спам редактировать

ответил 2014-01-22 13:39:53 +0400

kolyanius Gravatar kolyanius
11 ● 3 ● 3

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Задан: 2014-01-22 12:47:15 +0400

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Обновлен: Jan 22 '14

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Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
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