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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
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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
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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 ...

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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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