9608 is showing [Feature Status Subscription failed. Error code= 404]

Collapse
X
 
  • Filter
  • Time
  • Show
Clear All
new posts
  • katarj
    Aspiring Member
    • Aug 2013
    • 2

    9608 is showing [Feature Status Subscription failed. Error code= 404]

    I have been trying to start the push agent on Avaya IP phone 9608. The crash log shows:
    165>Aug 22 19:08:56 172.17.0.61 MSM: -04:00 2013 000 1 .TEL | 0 Registration process for SM[0] has been started.
    <165>Aug 22 19:08:56 172.17.0.61 AST: -04:00 2013 000 1 .TEL | 0 Feature Status Subscription failed. Error code= 404.
    <165>Aug 22 19:08:56 172.17.0.61 CONTACT: -04:00 2013 000 1 .TEL | 0 UpdateContacts-No data source -OR- Avaya Environment is [0]
    <165>Aug 22 19:08:57 172.17.0.61 PUSHADAP: -04:00 2013 000 1 .TEL | 0 CVxPushAdaptor::StartWebServer, Cannot start Goahead webserver. port (-1) is an invalid port. PUSHCAP will be set to 00000.
    <165>Aug 22 19:08:56 172.17.0.61 NETADAP: -04:00 2013 000 1 .TEL | 0 IP address of Adaptor eth0 = 172.17.0.61.
    <165>Aug 22 19:08:57 172.17.0.61 PPMDATA: -04:00 2013 000 1 .TEL | 0 getContactList: Encountered SOAP Fault.



    Environment:
    Call manager: Asterisk 1.8
    Firmware version on Avaya :6.2.2

    According to my understanding of avaya environment, I have already added the server where asterisk is running to the TPSLIST in 46xxsettings.txt. Also, the extension has been added to the user.conf and sip.conf of asterisk configuration.

    For SIP logs at the asterisk side , this is the debug log when phone logins:
    <------------>
    Really destroying SIP dialog '[email protected]' Method: SUBSCRIBE
    Really destroying SIP dialog '[email protected]' Method: REGISTER

    <--- Transmitting (no NAT) to 172.17.0.61:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 172.17.0.61:5060;branch=z9hG4bK1_52166e094306ad16-5bd336e0_R103;received=172.17.0.61
    From: <sip:[email protected]>;tag=-3f291f052166e09-5bd322a8_F103172.17.0.61
    To: <sip:[email protected]>;tag=as4eefb442
    Call-ID: [email protected]
    CSeq: 1 REGISTER
    Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0383559b"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
    Sending to 172.17.0.61:5060 (no NAT)

    <--- Transmitting (no NAT) to 172.17.0.61:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 172.17.0.61:5060;branch=z9hG4bK2_52166e0926522642-5bd327b7_R103;received=172.17.0.61
    From: <sip:[email protected]>;tag=-3f291f052166e09-5bd322a8_F103172.17.0.61
    To: <sip:[email protected]>;tag=as4eefb442
    Call-ID: [email protected]
    CSeq: 2 REGISTER
    Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Expires: 900
    Contact: <sip:[email protected];transport=udp;avaya-sc-enabled>;expires=900
    Date: Fri, 23 Aug 2013 00:01:14 GMT
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)

    <--- Transmitting (no NAT) to 172.17.0.61:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 172.17.0.61:5060;branch=z9hG4bK1_52166e09-38f1f4c2-5bd32f66_S103;received=172.17.0.61
    From: <sip:[email protected]>;tag=-f7cc74c52166e09-5bd32024_F103172.17.0.61
    To: <sip:[email protected]>;tag=as3aecb239
    Call-ID: [email protected]
    CSeq: 1 SUBSCRIBE
    Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ac4c477"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: SUBSCRIBE)
    Sending to 172.17.0.61:5060 (no NAT)
    Looking for 103 in demo (domain 172.17.0.10)

    <--- Transmitting (no NAT) to 172.17.0.61:5060 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 172.17.0.61:5060;branch=z9hG4bK2_52166e09-64f4697a-5bd3245b_S103;received=172.17.0.61
    From: <sip:[email protected]>;tag=-f7cc74c52166e09-5bd32024_F103172.17.0.61
    To: <sip:[email protected]>;tag=as3aecb239
    Call-ID: [email protected]
    CSeq: 2 SUBSCRIBE
    Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0
  • sinhar
    Member
    .
    • Mar 2012
    • 8

    #2
    To get more idea and fix, please open a trouble ticket with Avaya (if not done till now). Engineer will have the whole picture and move towards fixing the issue quickly.

    Comment

    Loading