Cisco ошибка 503

    Introduction

    This document describes how to troubleshoot Mobile and Remote Access (MRA) when calls fail due to «503 Service Unavailable» error.

    Contributed by Ishan Sambhi, Cisco TAC Engineer.

    Prerequisites

    Requirements

    Cisco recomends that you have knowledge of the next topics:

    • Cisco Video Communication Server (VCS) or Expressway-C and Expressway-E plataform
    • Cisco Unified Communication Manager (CUCM)

    The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, ensure that you understand the potential impact of any command.

    Components Used

    This document is not restricted to specific software and hardware versions.

    The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, ensure that you understand the potential impact of any command.

    Problem

    Calls between two Jabber clients registered to the CUCM via MRA fails with error message «503 Service Unavailable«

    Call flow:

    Jabber client 1 > Expressway-E > Expressway-C > CUCM > Expressway-C > Expressway-E > Jabber client 2

    Note: In the call flow, both Jabber client 1 and Jabber client 2, are registered to the same MRA deployment, that means the Expressway pair is the same before and after the CUCM.

    Solution

    A Session Initiation Protocol (SIP) INVITE goes from CUCM to Expressway-C

    2014-06-06T10:27:20+10:00 kcec tvcs: UTCTime="2014-06-06 00:27:20,782" Module="network.sip" Level="DEBUG":  Action="Received"  Local-ip="10.200.1.220"  Local-port="5060"  Src-ip="10.200.1.210"  Src-port="49071"  Msg-Hash="12136618272786736242"
 SIPMSG:
 |INVITE sip:80d713ba-514d-fc2a-6b7e-af527f9eb37f@10.200.1.220:5060;transport=tcp;orig-hostport=10.100.94.116:58991 SIP/2.0
 Via: SIP/2.0/TCP 10.200.1.210:5060;branch=z9hG4bK4ca27d4986
 Call-ID: 50f2a980-39110ae8-1fb3-d201c80a@10.200.1.210
 CSeq: 101 INVITE
 Remote-Party-ID: "Jabber client 1" <sip:sip%3Ajabber.client1@domain.com;x-cisco-number=168;x-cisco-callback-number=168>;party=calling;screen=yes;privacy=off
 Contact: <sip:sip%3Ajabber.client1i@10.200.1.210:5060;transport=tcp>
 From: "Jabber Client 1" <sip:sip%3Ajabber.client1@domain.com>;tag=132472~2b8aa2ec-85b4-4b2c-b662-3d078784a480-27388704
 To: <sip:jabber.client2@10.200.1.210>
 Max-Forwards: 70
 Allow: INVITE,OPTIONS,INFO,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY
 User-Agent: Cisco-CUCM9.1
 Expires: 180
 Date: Fri, 06 Jun 2014 00:27:20 GMT
 Supported: timer,resource-priority,replaces
 Min-SE: 1800
 Allow-Events: presence
 Send-Info: conference,x-cisco-conference
 Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= 1-7197; call-instance= 2
 Alert-Info: <file://Bellcore-dr1/>
 Content-Length: 0

    The Expressway-C tried to invoke call license.

    Module="network.http" Level="DEBUG":  Message="Request" Method="POST" URL="http://127.0.0.1:4370/status/call/call/uuid/86cc754d-eec2-4202-aa07-ea89c8afc47d"; Ref="0x7fd3780034e0"
Module="network.http" Level="DEBUG":  Message="Response" Src-ip="127.0.0.1" Src-port="4370" Dst-ip="127.0.0.1" Dst-port="32769" Response="200 OK" ResponseTime="0.002091" Ref="0x7fd3780034e0"
    
Module="developer.licensemanager.service.licensepool" Level="INFO" CodeLocation="licensepool(818)" Detail="Call license limit reached"license_type="nontraversal" tokens="2" cluster_token_inuse="0" cluster_token_limit="0"
Module="developer.licensemanager.service.licensepool" Level="INFO" CodeLocation="licensepool(818)" Detail="Call license limit reached"license_type="traversal" tokens="2" cluster_token_inuse="0" cluster_token_limit="0"
Module="developer.licensemanager.service.manager" Level="INFO" CodeLocation="licensemanager(155)" Detail="License not granted" call_id="6355985b-0d26-4a48-8b24-ba1ac5de38c6" lic_type="nontraversal" tokens=2
licensemanager: Level="INFO" Detail="License not granted" call_id="6355985b-0d26-4a48-8b24-ba1ac5de38c6" lic_type="nontraversal" tokens=2 UTCTime="2014-06-06 00:27:20,799"
    
Module="network.http" Level="DEBUG":  Message="Response" Src-ip="127.0.0.1" Src-port="9999" Dst-ip="127.0.0.1" Dst-port="32516" Response="200 OK" ResponseTime="0.012601" Ref="0x7fd366097b00"
Event="Search Completed" Reason="Service Unavailable" Service="SIP" Src-alias-type="SIP" Src-alias="sip%3Ajabber.client1@domain.com" Dst-alias-type="SIP" Dst-alias="sip:80d713ba-514d-fc2a-6b7e-af527f9eb37f@10.100.94.116:58991;transport\=tls" Call-serial-number="86cc754d-eec2-4202-aa07-ea89c8afc47d" Tag="92f548cb-a8a6-4339-8b3b-814cb85ae25a" Detail="found:false, searchtype:INVITE, Info:No License Available" Level="1" UTCTime="2014-06-06 00:27:20,800"
Event="Call Rejected" Service="SIP" Src-ip="10.200.1.210" Src-port="5060" Src-alias-type="SIP" Src-alias="sip:sip%3Ajabber.client1@domain.com" Dst-alias-type="SIP" Dst-alias="sip:80d713ba-514d-fc2a-6b7e-af527f9eb37f@10.100.94.116:58991;transport\=tls" Call-serial-number="86cc754d-eec2-4202-aa07-ea89c8afc47d" Tag="92f548cb-a8a6-4339-8b3b-814cb85ae25a" Detail="Service Unavailable" Protocol="TCP" Response-code="503" Level="1" UTCTime="2014-06-06 00:27:20,801"
Module="network.sip" Level="INFO":  Action="Sent"  Local-ip="10.200.1.220"  Local-port="5060"  Dst-ip="10.200.1.210"  Dst-port="49071"   Detail="Sending Response Code=503, Method=INVITE, CSeq=101, To=sip:168@10.200.1.210, Call-ID=50f2a980-39110ae8-1fb3-d201c80a@10.200.1.210, From-Tag=132472~2b8aa2ec-85b4-4b2c-b662-3d078784a480-27388704, To-Tag=111a6c358c828b39, Msg-Hash=17852441825048296421"

    As there is no license available, the Expressway-C replies with the error «Service unavailable»

    2014-06-06T10:27:20+10:00 kcec tvcs: UTCTime="2014-06-06 00:27:20,801" Module="network.sip" Level="DEBUG":  Action="Sent"  Local-ip="10.200.1.220"  Local-port="5060"  Dst-ip="10.200.1.210"  Dst-port="49071"  Msg-Hash="17852441825048296421"
 SIPMSG:
 |SIP/2.0 503 Service Unavailable
 Via: SIP/2.0/TCP 10.200.1.210:5060;branch=z9hG4bK4ca27d4986;received=10.200.1.210;ingress-zone=CEtcp102001210
 Call-ID: 50f2a980-39110ae8-1fb3-d201c80a@10.200.1.210
 CSeq: 101 INVITE
 From: "Jabber Client 1" <sip:sip%3Ajabber.client1@domain.com>;tag=132472~2b8aa2ec-85b4-4b2c-b662-3d078784a480-27388704
 To: <sip:jabber.client2@10.200.1.210>;tag=111a6c358c828b39
 Server: TANDBERG/4129 (X8.1.1)
 Warning: 399 10.200.1.220:5061 "No License Available"
 Content-Length: 0

    The call fails because it does not find an available license, but a call between two jabber clients registered in the same CUCM via MRA does not require any license in the Expressway-C. This issue happens when the Expressway-C Unified communication traversal zone is pointed to the Expressway-E internal IP address instead to the Expressway-E public IP address.

    In single NIC Expressway-E deployment, the Unified Communication Traversal zone must be pointed to the public IP address of the Expressway-E and the firewall between Expressway-E and the Internet must be configured with Network Address Translation (NAT) reflexion to allow communication between Expressway-C and Expressway-E.

    Источник

    Problem:

    All Endpoints from Branch locations can call HQ, but HQ is unable to call them as shown below.

    Topology

    Call Fails from HQ to BR1

    Call Successful from BR1 to HQ

    Troubleshooting:

    All Clusters are configured with ILS/GPDR for Inter-site Dialing with SME as a Centralized Unit.
    Step 1:
    Understand/Analyze Call Flow.

    Also, verify if IP Phone has visibility to Dialed or Learned Number.
    NOTE: DNA doesn’t work in case of ILS/GDPR

    Steps 2:
    Make a test call to any Branch and collect the SDL logs from HQ-SUB (primary call processing node), SME, and BR1.
    You can either use RTMT or CUCM CLI login.
    In case, if you are using CUCM CLI use below command to capture SDL traces.
    “file tail activelog cm/trace/ccm/SDL recent”

    Step 3:
    SIP Call Flow

    This is how the Basic SIP Call Setup looks like when the calls are working properly. Now Let’s have a look at Call Flow Diagram for our scenario.

    HQ-Sub

    As per the logs, we are receiving SIP Response “503 Service Unavailable” from SME, which is the cause of call failure.
    Let’s analyze it further and have a look at SDL logs on SME.

    HQ-SME

    As per the logs, no SIP Request is being forwarded to BR1-Pub. It indicates that the “503 Service Unavailable” was been generated by SME.

    SIP Responses

    Following is the snippet for SIP Response received from SME.

    In our case, Subscriber is Primary Call Manager on HQ, while the IP address configured on SME to HQ SIP trunk is pointing towards HQ Publisher (Backup Call Processing node).

    This is the reason Call from HQ to all Branch Clusters is failing.

    Resolution:

    Currently, SIP trunk on SME which is pointing towards HQ-Pub.

    We need to make sure that SIP Trunk is pointing towards Primary Call Manager (HQ-Sub) and not to Secondary (HQ-Pub) as below.

    Reset the SIP trunk once the destination address is modified.

    Verification:

    Make a test call from HQ to BR 1.

    Call between HQ to all BR is successful.

    Summary:

    In our scenario depicted above HQ-Sub is a primary call processing node. The primary call processing node has the Cisco Call Manager Service enabled. Devices such as phones, gateways, and media resources can register and make calls only to servers with this service enabled.

    Whereas IP address configured on the SME to HQ SIP Trunk is pointing towards HQ Publisher which is not a primary call processing server, whereas the Source IP address that is sending the SIP INVITE is of HQ Subscriber and it does not exist in the SIP Trunk in CUCM. This is the reason behind the 503 Service Unavailable.

    Источник

     Hello Team,

    i have a trunk CUCM to SFB 2015 where skype can call Cisco extension but Cisco cannot call Skype extensions, i get the error 503 service unavailable on the skype server.

    So far i downloaded and installed the latest SFB update that did not fix the issue.

    here are some logs of Cisco to SFB fetch from Skype:

    TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.3D3C::07/01/2019-11:43:13.941.00002000 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [431143657]
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068<-10.10.60.250:35810
    INVITE sip:820993@10.10.30.23:5068 SIP/2.0
    FROM: «AMBOZOO» <sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979
    TO: <sip:820993@10.10.30.23>
    CSEQ: 101 INVITE
    CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250
    MAX-FORWARDS: 69
    VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828
    ALLOW-EVENTS: presence
    CONTACT: <sip:25006@10.10.60.250:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com=»CSFAMBOZOO»;bfcp
    CONTENT-LENGTH: 202
    DATE: Mon, 01 Jul 2019 11:26:00 GMT
    EXPIRES: 180
    SUPPORTED: timer,resource-priority,replaces
    SUPPORTED: X-cisco-srtp-fallback,X-cisco-original-called
    USER-AGENT: Cisco-CUCM11.0
    CONTENT-TYPE: application/sdp
    ALLOW: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    P-ASSERTED-IDENTITY: «AMBOZOO» <sip:25006@10.10.60.250>
    Min-SE: 1800
    Session-ID: 00001cc400105000a0001002b529e1f4;remote=00000000000000000000000000000000
    Cisco-Guid: 4283652480-0000065536-0000000002-4198238730
    Session-Expires: 1800
    Remote-Party-ID: «AMBOZOO» <sip:25006@10.10.60.250>;party=calling;screen=yes;privacy=off
    v=0
    o=CiscoSystemsCCM-SIP 40478 1 IN IP4 10.10.60.250
    s=SIP Call
    c=IN IP4 10.123.123.10
    t=0 0
    m=audio 16386 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

    ————EndOfIncoming SipMessage

    TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.2FE8::07/01/2019-11:43:13.942.00002001 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [431143657]
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068->10.10.60.250:35810
    SIP/2.0 100 Trying
    FROM: «AMBOZOO»<sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979
    TO: <sip:820993@10.10.30.23>
    CSEQ: 101 INVITE
    CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250
    VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828
    CONTENT-LENGTH: 0

    ————EndOfOutgoing SipMessage

    TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.36B0::07/01/2019-11:43:13.953.00002005 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [431143657]
    >>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068->10.10.60.250:35810
    SIP/2.0 503 Service Unavailable
    FROM: «AMBOZOO»<sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979
    TO: <sip:820993@10.10.30.23>;epid=4A4C146B07;tag=707bec76e1
    CSEQ: 101 INVITE
    CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250
    VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828
    CONTENT-LENGTH: 0
    SERVER: RTCC/6.0.0.0 MediationServer

    ————EndOfOutgoing SipMessage

    TL_INFO(TF_PROTOCOL) [SFB\PR-VM-SFB-01]1594.3D3C::07/01/2019-11:43:14.063.0000200C (S4,SipMessage.DataLoggingHelper:sipmessage.cs(801)) [546807437]
    <<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_5291CE>], 10.10.30.23:5068<-10.10.60.250:35810
    ACK sip:820993@10.10.30.23:5068 SIP/2.0
    FROM: «AMBOZOO» <sip:25006@10.10.60.250>;tag=40478~bb16990c-bb8d-474d-e339-4cb6bd7d93e7-23478979
    TO: <sip:820993@10.10.30.23>;epid=4A4C146B07;tag=707bec76e1
    CSEQ: 101 ACK
    CALL-ID: ff535980-d191edc8-41b-fa3c0a0a@10.10.60.250
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP 10.10.60.250:5060;branch=z9hG4bK41e3a437828
    ALLOW-EVENTS: presence
    CONTENT-LENGTH: 0
    DATE: Mon, 01 Jul 2019 11:26:00 GMT
    USER-AGENT: Cisco-CUCM11.0

    ————EndOfIncoming SipMessage

    Any help will be appreciated.

    thank you

    Источник

    Hello everybody!

    Hope you’re doing allright,

    I’m using a Call Manager Express in an SIP scenario where we want to route calls to the pstn via a voip dialpeer (connectivity is not an issue here), but apparently we’re facin errors regarding to Xcoding resources:

    *Feb 19 21:30:15.835: //38/C70B562D8013/CCAPI/cc_api_call_disconnected:

    Call Entry(Responsed=TRUE, Cause Value=47, Retry Count=0)

    .

    Feb 19 21:42:59.479: //63/8E36C4E88022/SIP/Error/sipSPIUpdateSrcSdpFixedPart:

    resolve_sig_ip_address_to_bind failed

    *Feb 19 21:42:59.483: //63/8E36C4E88022/SIP/Error/sipSPICodecTranscoder:

    Disjoint set & xcoder reservation failed. Disconnect call

    Looking at the sip messages, I see a 503 from the Call Manager that implies that the transcoder is not configured (and it comes from the CME, since the server header says so):

    Warning: 399 172.16.5.250 «Transcoder Not Configured»

    Server: Cisco-SIPGateway/IOS-15.4.3.M1

    I understand that the configuration regarding media handling is not an issue (it worked fine in other scenarios), anyway, I felt free to attach outputs of the following commands:

    debug ccsip messages

    debug ccsip error

    as well as the configuration of the box (sho run output)

    I hope someone could take a look at this and shed some light on the issue (it might be simpler than I think but at this point I’m not sure).

    Please let me know should you need more info about this scenario.

    Your comments are very much appreciated!

    Источник

    Модератор: april22

    Got SIP response 503 «Service Unavailable»

    Добрый день, периодически стали слышать в трубке все линии заняты, посмотрел лог, по этим вызовам вижу
    Got SIP response 503 «Service Unavailable» prov IP:5060
    свои настройки не менял. Другие вызовы через данного прова совершаются нормально. Кол-во линий достаточное.
    С провайдером разговор свелся к тому, что.
    дословно.
    от оборудования вызываемого абонента был получен некорректный ответ (вместо «занято» другой сигнал) этот сигнал интерпретировался с истемой как Service Unavailable.
    Вопрос: настройки моего оборудование могут вызвать такую ошибку? или это пров?
    погуглил, часто упоминают cisco какуюто :) , у меня такой нет

    vladv
     
    Сообщений: 229
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    Re: Got SIP response 503 «Service Unavailable»

    Сообщение virus_net » 11 окт 2013, 09:11

    vladv, телепатов тут нет

    vladv писал(а):Вопрос: настройки моего оборудование могут вызвать такую ошибку? или это пров?

    чтобы на это ответить смотри в debug sip`а или tcpdump обмена пакетами в провом
    там будет четко видно кто, кому и что послал

    мой SIP URI sip:virus_net@asterisk.ru
    bitname.ru — Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
    ENUMER — звони бесплатно и напрямую.

    virus_net
     
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    Откуда: Москва

    Re: Got SIP response 503 «Service Unavailable»

    Сообщение vladv » 11 окт 2013, 10:13

    понял, спасибо. пробую tcpdump

    vladv
     
    Сообщений: 229
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    Re: Got SIP response 503 «Service Unavailable»

    Сообщение ded » 11 окт 2013, 10:21

    Изображение

    ded
     
    Сообщений: 15679
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    Re: Got SIP response 503 «Service Unavailable»

    Сообщение vladv » 11 окт 2013, 11:40

    :) пошел по схеме, научился делать tcpdump, wireshark — установлен, ловит пакеты, жду ошибку для изучения. Ну и соответсвенно изучения выхлопа вайршарка. Вот.
    Спасибо за советы.

    vladv
     
    Сообщений: 229
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    Re: Got SIP response 503 «Service Unavailable»

    Сообщение vladv » 11 окт 2013, 19:24

    Добрый вечер друзья.
    Не взглянете, wireshark. Это один звонок, firstleg от телефона к эластиксу, secondleg от эластикса к провайдеру.
    Эластикс, проговаривает «все линии заняты». С мобильного если звонить на этот номер, то «неправильно набран номер». Набор других номер через этого провайдера идет нормально.
    404 Not found в некоторых случаях меняется на 403 Forbidden или 408 Request Timeout.

    Вложения
    first leg.JPG
    firstleg
    second leg.JPG
    secondleg
    vladv
     
    Сообщений: 229
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