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503 Service Unavailable / busy / Решено

0

клиент звонит из сети sip 485485 на ружу 390009, номер 390009 занят, а почему получаем 503 Service Unavailable ????


--- Reliably Transmitting (NAT) to 172.16.0.6:5060 ---
503 Service Unavailable
Via: SIP/2.0/UDP
172.16.0.6:5060;branch=z9hG4bK187f10c6183ee57f;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2&gt;;tag=f15afbc4-44180 <br="">To: <sip:390009@192.168.3.2;user=phone>;tag=as29aefb6f
Call-ID: BD21-1117-470441808BB61EFDBA9E-009@SipHost
CSeq: 1281 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0

удалить закрыть спам изменить тег редактировать

спросил 2012-05-21 18:12:12 +0400

voznyaa Gravatar voznyaa
1 ● 6 ● 4 ● 5

обновил 2012-05-22 13:40:08 +0400

4 Ответа

Сорт. по » старые новые больше голосов
0

по этому куску скзать ничего нельзя даже невидно куда звонит

может callerid выставить? или у провайдера спросить

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ответил 2012-05-21 20:04:59 +0400

meral Gravatar meral flag of Ukraine
23347 ● 24 ● 20 ● 177
http://pro-sip.net/

Comments

callerid выставлен, это логи моего астериска и моего подключенного шлюза , шлюз может звонить и принимать звонки со своего номера 485485

я обратил внимание на данный лог из за то что , при звонке с абонентского устройства (к примеру dvg-2102s), в трубке слышим отбой (когда действительно занято)

а если у клиента стоит Asterisk+freepbx (dahdi), то по его логам получается что "канал" не доступен, и в трубке сообщение что канал перегружен(клиент настроен как friend, и Call-limit = 3 )

вотс думаю куда копать....

voznyaa ( 2012-05-21 22:57:37 +0400 )редактировать

этот кусок ни о чем вообще.

meral ( 2012-05-22 01:52:47 +0400 )редактировать
0

город - cisco - asterisk 

GENERIC:
SetupTime=14400477400 ms
Index=298670
PeerAddress=390009
PeerSubAddress=
PeerId=200
PeerIfIndex=135
LogicalIfIndex=8
DisconnectCause=11  
DisconnectText=user busy (17)
ConnectTime=0 ms
DisconnectTime=14400477760 ms
CallDuration=00:00:00 sec
CallOrigin=1
ReleaseSource=3
ChargedUnits=0
InfoType=speech
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=0
TELE:
ConnectionId=[0xE3D29F57 0xA30711E1 0x9A0E8E4A 0x86303D0A]
IncomingConnectionId=[0xE3D29F57 0xA30711E1 0x9A0E8E4A 0x86303D0A]
CallID=537091
Port=0/0/0:15 (537091)
BearerChannel=0/0/0.2
TxDuration=0 ms
VoiceTxDuration=0 ms
FaxTxDuration=0 ms
CoderTypeRate=None
NoiseLevel=0
ACOMLevel=0
SessionTarget=
ImgPages=0
CallerName=485485
CallerIDBlocked=False
Target tg label=MTA
LongDurationCallDetected=no
LongDurCallTimeStamp=
LongDurCallDuration=
OriginalCallingNumber=485485
OriginalCallingOctet=0x0
OriginalCalledNumber=390009
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x80
TranslatedCallingNumber=485485
TranslatedCallingOctet=0x0
TranslatedCalledNumber=390009
TranslatedCalledOctet=0x0
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0x80
GwReceivedCalledNumber=390009
GwReceivedCalledOctet3=0x0
GwOutpulsedCalledNumber=390009
GwOutpulsedCalledOctet3=0x0
GwReceivedCallingNumber=485485
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x80
GwOutpulsedCallingNumber=485485
GwOutpulsedCallingOctet3=0x0
GwOutpulsedCallingOctet3a=0x80
DSPIdentifier=0/3:1
MlppServiceDomain=0 (NO_MLPP_DOMAIN)
PrecedenceLevel=-1 (PRECEDENCE_LEVEL_NONE)

GENERIC:
SetupTime=14400477320 ms
Index=298671
PeerAddress=485485
PeerSubAddress=
PeerId=101
PeerIfIndex=134
LogicalIfIndex=0
DisconnectCause=11  
DisconnectText=user busy (17)
ConnectTime=0 ms
DisconnectTime=14400477770 ms
CallDuration=00:00:00 sec
CallOrigin=2
ReleaseSource=3
ChargedUnits=0
InfoType=speech
TransmitPackets=0
TransmitBytes=0
ReceivePackets=0
ReceiveBytes=0
VOIP:
ConnectionId[0xE3D29F57 0xA30711E1 0x9A0E8E4A 0x86303D0A]
IncomingConnectionId[0xE3D29F57 0xA30711E1 0x9A0E8E4A 0x86303D0A]
CallID=537090
RemoteIPAddress=192.168.4.2
RemoteUDPPort=13066
RemoteSignallingIPAddress=192.168.4.2
RemoteSignallingPort=5060
RemoteMediaIPAddress=192.168.4.2
RemoteMediaPort=13066
SRTP = off
TextRelay = off
Fallback Icpif=0
Fallback Loss=0
Fallback Delay=0
RoundTripDelay=0 ms
SelectedQoS=best-effort
tx_DtmfRelay=rtp-nte
FastConnect=FALSE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=FALSE

SessionProtocol=sipv2
ProtocolCallId=1e885fd23c351ab4713efacd647c2292@192.168.4.2:5060
SessionTarget=192.168.4.2
OnTimeRvPlayout=0
GapFillWithSilence=0 ms
GapFillWithPrediction=0 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=0 ms
LoWaterPlayoutDelay=0 ms
ReceiveDelay=0 ms
LostPackets=0
EarlyPackets=0
LatePackets=0
VAD = disabled
CoderTypeRate=g711alaw
CodecBytes=160
cvVoIPCallHistoryIcpif=0
MediaSetting=flow-through
CallerName=485485
CallerIDBlocked=False
OriginalCallingNumber=485485
OriginalCallingOctet=0x0
OriginalCalledNumber=390009
OriginalCalledOctet=0x0
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0x80
TranslatedCallingNumber=485485
TranslatedCallingOctet=0x0
TranslatedCalledNumber=390009
TranslatedCalledOctet=0x0
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0x80
GwReceivedCalledNumber=390009
GwReceivedCalledOctet3=0x0
GwReceivedCallingNumber=485485
GwReceivedCallingOctet3=0x0
GwReceivedCallingOctet3a=0x80
MediaInactiveDetected=no
MediaInactiveTimestamp=
MediaControlReceived=
LongDurationCallDetected=no
LongDurationCallTimerStamp=
LongDurationCallDuration=
Username=485485
MlppServiceDomain=0 (NO_MLPP_DOMAIN)
PrecedenceLevel=0 (PRECEDENCE_LEVEL_NONE)

CPA Call History Parameters
  CPA Event Status: DISABLE
ссылка удалить спам редактировать

ответил 2012-05-22 09:40:40 +0400

voznyaa Gravatar voznyaa
1 ● 6 ● 4 ● 5

обновил 2012-05-22 09:46:41 +0400

0
<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK1dc51679dd2f51ca
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1482 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229

v=0
o=485485 2207805700 2207805700 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-471006102A80D015A4BF-012@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK1dc51679dd2f51ca;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as1b333b47
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1482 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f62044a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'BD21-1117-471006102A80D015A4BF-012@SipHost' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK1dc51679dd2f51ca
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as1b333b47
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1482 ACK
Max-Forwards:70
Content-Length: 0

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

<--- SIP read from UDP:172.16.0.6:5060 --->
INVITE sip:390009@192.168.3.2;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 INVITE
Contact: <sip:485485@172.16.0.6:5060>
Expires:90
Max-Forwards:70
Authorization:Digest username="485485",realm="asterisk",nonce="7f62044a",uri="sip:390009@192.168.3.2;user=phone",response="46ad1c02712007c4f7f31706c4443e5b",algorithm=MD5
Supported: replaces
User-Agent: dlink 12-3868-1709-0.10.56.1-DSA
Content-Type: application/sdp
Content-Length: 229

v=0
o=485485 2207805700 2207805700 IN IP4 172.16.0.6
s=Session SDP
c=IN IP4 172.16.0.6
t=0 0
m=audio 9000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 172.16.0.6:5060 (NAT)
Using INVITE request as basis request - BD21-1117-471006102A80D015A4BF-012@SipHost
Found peer '485485' for '485485' from 172.16.0.6:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.0.6:9000
Looking for 390009 in local-phones (domain 192.168.3.2)
list_route: hop: <sip:485485@172.16.0.6:5060>

<--- Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:390009@192.168.3.2:5060>
Content-Length: 0


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

<--- Reliably Transmitting (NAT) to 172.16.0.6:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df;received=172.16.0.6;rport=5060
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as356451b2
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0


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

<--- SIP read from UDP:172.16.0.6:5060 --->
ACK sip:390009@192.168.3.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.6:5060;branch=z9hG4bK3c5e3f90d78251df
From: "485485" <sip:485485@192.168.3.2>;tag=5634894f-100610
To: <sip:390009@192.168.3.2;user=phone>;tag=as356451b2
Call-ID: BD21-1117-471006102A80D015A4BF-012@SipHost
CSeq: 1483 ACK
Max-Forwards:70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'BD21-1117-471006102A80D015A4BF-012@SipHost' Method: ACK

    enter code here
ссылка удалить спам редактировать

ответил 2012-05-22 09:23:45 +0400

voznyaa Gravatar voznyaa
1 ● 6 ● 4 ● 5

обновил 2012-05-22 09:49:58 +0400

Comments

-- Executing [390009@local-phones:1] Dial("SIP/485485-00004f4a", "SIP/390009@192.168.4.1,120") in new stack

== Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

-- Called <a href="mailto:SIP/390009@192.168.4.1">SIP/390009@192.168.4.1</a>

-- Got SIP response 486 "Busy here" back from 192.168.4.1:5060

-- SIP/192.168.4.1-00004f4b is busy

== Everyone is busy/congested at this time (1:1/0/0)

-- Executing [390009@local-phones:2] Congestion("SIP/485485-00004f4a", "") in new stack
voznyaa ( 2012-05-22 10:04:20 +0400 )редактировать
-1

все спасибо , починил ...

ссылка удалить спам редактировать

ответил 2012-05-22 12:47:51 +0400

voznyaa Gravatar voznyaa
1 ● 6 ● 4 ● 5

Comments

Как починил то?

Matvey ( 2012-11-12 16:35:39 +0400 )редактировать

по какой-то странной причине прописал Congestion, переписал на hangup

voznyaa ( 2013-03-18 09:29:56 +0400 )редактировать

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Задан: 2012-05-21 18:12:12 +0400

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