Cannot register Cisco 7942 to Asterisk

Discussion in 'General' started by leungs, Mar 21, 2011.

  1. fasilkaks

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    Hi leungs,
    I have successfully configured the phones. As I told earlier, I am using 7911G and 7945G phones. Is it possible to store directory in these phones. I tried creating directory.xml file as explained in voipinfo.org and put it in my webserver. In the SEP<mac>.cnf.xml file I have added the tag <directoryURL>url</directoryURL>. But the phones are not taking that value. I have checked this in the phones. If I try to register the phone with other extension number, that values are getting updated correctly. Is there anything that I miss in my configuration file with which I can proceed successfully? Please advice.
     
  2. leungs

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    Here is the format of my directory.xml file. Hope it helps.

    <CiscoIPPhoneMenu>
    <Title>IP Telephony Directory</Title>
    <Prompt>People reachable via VoIP</Prompt>

    <MenuItem>
    <Name>Beijing</Name>
    <URL>Dial:900608610xxxxxxx</URL>
    </MenuItem>

    </CiscoIPPhoneMenu>
     
  3. Clintg

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    Guys,
    I am facing the same problem, do you mind emailing the files over as well

    ccgaddes@gmail.com


    Cheers
     
  4. RicMarques

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    I've run into this problem before, and what I discovered is that Cisco dropped SIP UDP support in v9... only SIP TCP support...

    see here for more discussion, and how to make v9 SIP TCP work with Asterisk 1.6.2.13+

    -Ric
     
  5. arshadahmad

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    Hi

    Please help me as well, i have 7942 phones, i have tried to modify the following Files

    OS79XX.TXT ( i put the following text into it SCCP42.8-3-3S-----SIP42.8-3-3S and 03-07-5-00)
    XMLDefault.cnf.xml (i put the same above mentioned files name in this file one by one)

    But it gives following error.

    Connection received from 192.168.0.113 on port 49152 [16/11 09:37:45.036]
    Read request for file <CTLSEP0026CBBD93B6.tlv>. Mode octet [16/11 09:37:45.038]
    File <CTLSEP0026CBBD93B6.tlv> : error 2 in system call CreateFile The system cannot find the file specified. [16/11 09:37:45.038]
    Connection received from 192.168.0.113 on port 49153 [16/11 09:37:45.114]
    Read request for file <SEP0026CBBD93B6.cnf.xml>. Mode octet [16/11 09:37:45.115]
    Using local port 49682 [16/11 09:37:45.115]
    <SEP0026CBBD93B6.cnf.xml>: sent 14 blks, 6961 bytes in 0 s. 0 blk resent [16/11 09:37:45.143]


    Please help me on this.
     
  6. leungs

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    Leave down your email and I would send you some files and instruction.
     
  7. jordan.turner1974

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  8. trincity

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  9. AlanDaniel

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    You need to use TCP on Asterisk on each extension when you use firmware 9.x, (79XX) and change the xml like this:


    ASTERISK SIP GENERAL CUSTOM
    tcpenable=yes
    tcpbindaddr=0.0.0.0

    sip_custom_post.conf
    ASTERISK
    [300](+)
    transport=tcp


    XML
    <line button="1">
    <featureID>9</featureID>
    <featureLabel>300</featureLabel>
    <name>300</name>
    <displayName>300</displayName>
    <contact>300</contact>
    <proxy>USECALLMANAGER</proxy> ----CHECK THIS.....+++++++
    <port>5060</port>
    <autoAnswer>
    <autoAnswerEnabled>2</autoAnswerEnabled>
    </autoAnswer>
    <callWaiting>3</callWaiting>

    <authName>300</authName>
    <authPassword>mypassword</authPassword>
     
  10. skatal

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    please send me the config file too, i tried 8.5 and it didnt work i still get provisioning

    my email is brett@tgco.com
     
  11. AlanDaniel

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    You can Download the XML example from here:
    http://www.minded.ca/default/2009-12-16 ... -asterisk/

    On the XML Change the proxy for "USECALLMANAGER"
    XML
    <line button="1">
    <featureID>9</featureID>
    <featureLabel>300</featureLabel>
    <name>300</name>
    <displayName>300</displayName>
    <contact>300</contact>
    <proxy>USECALLMANAGER</proxy> ----CHECK THIS.....+++++++

    Use the DHCP option 150 and put the ip of Asterisk, if you can´t you must change the ip tftp server on the phone.

    Best Regards
     
  12. KernelPanic

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    This is not a problematic error message. It simply means that the phone cant download the security certificates for encryption. If you dont have security set up, you dont have certificates, and hence the phone cant get them.
    Anything in the logs about *.tlv not being available can be safely ignored.
    The file its looking for, the .cnf.xml file has been sent to the phone successfully.
    The phone must be barfing on something in the config file. Is it formatted correctly? Can you log into the phone via http or ssh and view the logs to get some clues as to what is wrong?

    Id personally start by simplifying the config file as much as possible to trim it down to where the phone will boot, and then add back in the lines and buttons.
     
  13. mmdigital

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  14. ocecom

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    Hi I have the same issue, the IPPhone not pass from registering step, and the logs on the phone says thath can not parse in some point that I do not understand. I re-check the .cnf.xml file and everything looks good for me, any one who have faced this same issue.


    |=== Syslogd === Fri Nov 2 18:57:19 2007
    ====================================================

    NOT 18:57:19.025265 TFTP: [8]:Requesting SEPB41489A2A32C.cnf.xml from 10.11.10.10

    NOT 18:57:19.037861 TFTP: [8]:Finished --> rcvd 6780 bytes

    WRN 18:57:19.353772 JVM: Startup Module Loader|cip.xml.as: - XML Parser Warning: Unknown element 'uid' in element '/device/networkLocaleInfo' (line=64)

    ERR 18:57:19.355205 JVM: Startup Module Loader|cip.xml.as: - XML Parser Exception: Element '/device/sipProfile/natEnabled' Invalid EnumValue: 0 (line=152)

    ERR 18:57:19.356739 JVM: Startup Module Loader|cip.cfg.h:? - ERROR PARSING CONFIG file:scratchpad/SEPB41489A2A32C.cnf.xml

    NOT 18:57:19.359922 SECD: setSecMode: sec mode set to NONE (was NONE)

    ERR 18:57:19.391754 JVM: Parse error: Unknown key name: g711ulaw

    ERR 18:57:19.392510 JVM: config_set_string: Parse function failed. ID: 5

    NOT 18:57:19.476729 JVM: Startup Module Loader|cip.cfg.h:? - DELETE ConfigFile:(scratchpad/SEPB41489A2A32C.cnf.xml)WAS SUCCESSFUL

    WRN 18:57:19.509560 SECD: WARN:lookupCTL: ** no CTL, assume CCM 10.11.10.10 NONSECURE

    NOT 18:57:19.517506 JVM: Startup Module Loader|cip.sipcc.SipCcAdapter: - cmname=10.11.10.10 cmIp=10.11.10.10 port=5060 isValid=true

    WRN 18:57:19.518894 JVM: Startup Module Loader|cip.sipcc.SipCcAdapter: - No Valid CCMs. Defaulting to TFTP Server: 10.11.10.10

    WRN 18:57:19.521455 SECD: WARN:lookupSRST: no CTL, treat SRST as non-secure

    NOT 18:57:19.663511 JVM: Startup Module Loader|cip.sipcc.SipCcAdapter: - addSrstToCAgentList: cmname=10.11.10.1 cmIp=10.11.10.1 port=5060 isValid=true

    NOT 18:57:19.664889 JVM: Startup Module Loader|cip.sipcc.d: - initializeLinePlane(): Mgmt Interface is in Service now..

    ERR 18:57:19.666520 JVM: Startup Module Loader|cip.sipcc.d: - sipMinCfgCheck():Line 1 not configured..

    WRN 18:57:49.786731 JVM: Startup Module Loader|cip.sipcc.SipCcAdapter: - unprovisioned: Trying to get config again..

    NOT 18:57:49.788081 JVM: Startup Module Loader|cip.sipcc.d: - retryCfgUnprovisioned:Mgmt Interface Going out of Service..

    NOT 18:57:49.823705 SECD: updateCTL: starting CTL update


    Any
     
  15. keler

    Joined:
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    Hello,

    I have the same problem, please can you send me the files at keler2000@hotmail.com

    here's a details when I set the debug on.



    <--- SIP read from UDP:192.168.1.252:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK00a752c2;rport
    From: "Unknown" <sip:Unknown@192.168.0.117>;tag=as70122b0d
    To: "Samuel" <sip:102@192.168.1.252:5060>;tag=763F157D-63E07A08
    CSeq: 102 OPTIONS
    Call-ID: 25fe3efe19202dab0bbda96c3a545c6e@192.168.0.117:5060
    Contact: <sip:102@192.168.1.252:5060>
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
    Supported: 100rel,replaces,100rel,timer,replaces,norefersub
    User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.6.0314
    Accept-Language: en
    Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
    Accept-Encoding: identity
    Content-Length: 0

    <------------->
    --- (14 headers 0 lines) ---
    Really destroying SIP dialog '25fe3efe19202dab0bbda96c3a545c6e@192.168.0.117:5060' Method: OPTIONS

    <--- SIP read from UDP:192.168.1.161:56030 --->
    REGISTER sip:192.168.0.117:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-fb17256e98560a5a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    To: "104"<sip:104@192.168.0.117:5060>
    From: "104"<sip:104@192.168.0.117:5060>;tag=b23d0641
    Call-ID: ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.
    CSeq: 83 REGISTER
    Expires: 120
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Supported: replaces
    User-Agent: 3CXPhone 5.0.14900.0
    Authorization: Digest username="104",realm="asterisk",nonce="05881b9f",uri="sip:192.168.0.117:5060",response="410564ab80d405fe78dbb288b1cd2c84",algorithm=MD5
    Content-Length: 0

    <------------->
    --- (14 headers 0 lines) ---
    Sending to 192.168.1.161:56030 (no NAT)

    <--- Transmitting (NAT) to 192.168.1.161:56030 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-fb17256e98560a5a-1---d8754z-;received=192.168.1.161;rport=56030
    From: "104"<sip:104@192.168.0.117:5060>;tag=b23d0641
    To: "104"<sip:104@192.168.0.117:5060>
    Call-ID: ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.
    CSeq: 83 REGISTER
    Server: FPBX-2.8.1(1.8.5.0)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0


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

    <--- Transmitting (NAT) to 192.168.1.161:56030 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-fb17256e98560a5a-1---d8754z-;received=192.168.1.161;rport=56030
    From: "104"<sip:104@192.168.0.117:5060>;tag=b23d0641
    To: "104"<sip:104@192.168.0.117:5060>;tag=as4a8fc0d8
    Call-ID: ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.
    CSeq: 83 REGISTER
    Server: FPBX-2.8.1(1.8.5.0)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f3c138f"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.' in 32000 ms (Method: REGISTER)

    <--- SIP read from UDP:192.168.1.161:56030 --->
    REGISTER sip:192.168.0.117:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-600d6e54624e5c64-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    To: "104"<sip:104@192.168.0.117:5060>
    From: "104"<sip:104@192.168.0.117:5060>;tag=b23d0641
    Call-ID: ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.
    CSeq: 84 REGISTER
    Expires: 120
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Supported: replaces
    User-Agent: 3CXPhone 5.0.14900.0
    Authorization: Digest username="104",realm="asterisk",nonce="1f3c138f",uri="sip:192.168.0.117:5060",response="7b84292eb479a43c2d56781c9c83eff0",algorithm=MD5
    Content-Length: 0

    <------------->
    --- (14 headers 0 lines) ---
    Sending to 192.168.1.161:56030 (NAT)

    <--- Transmitting (NAT) to 192.168.1.161:56030 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-600d6e54624e5c64-1---d8754z-;received=192.168.1.161;rport=56030
    From: "104"<sip:104@192.168.0.117:5060>;tag=b23d0641
    To: "104"<sip:104@192.168.0.117:5060>
    Call-ID: ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.
    CSeq: 84 REGISTER
    Server: FPBX-2.8.1(1.8.5.0)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0


    <------------>
    Reliably Transmitting (NAT) to 192.168.1.161:56030:
    OPTIONS sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK5f26f805;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@192.168.0.117>;tag=as20027259
    To: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    Contact: <sip:Unknown@192.168.0.117:5060>
    Call-ID: 1c3a381f0ac4409a2be685795f37ffad@192.168.0.117:5060
    CSeq: 102 OPTIONS
    User-Agent: FPBX-2.8.1(1.8.5.0)
    Date: Thu, 24 May 2012 20:51:06 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Length: 0


    ---

    <--- Transmitting (NAT) to 192.168.1.161:56030 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-600d6e54624e5c64-1---d8754z-;received=192.168.1.161;rport=56030
    From: "104"<sip:104@192.168.0.117:5060>;tag=b23d0641
    To: "104"<sip:104@192.168.0.117:5060>;tag=as4a8fc0d8
    Call-ID: ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.
    CSeq: 84 REGISTER
    Server: FPBX-2.8.1(1.8.5.0)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Expires: 120
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>;expires=120
    Date: Thu, 24 May 2012 20:51:06 GMT
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'ZmU1ZWY0YTkwYzVkZDJiZDU3YWZhYmQ2NjllNTYzMWU.' in 32000 ms (Method: REGISTER)

    <--- SIP read from UDP:192.168.1.161:56030 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK5f26f805;rport=5060
    Contact: <sip:192.168.1.161:56030>
    To: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>;tag=f606c83a
    From: "Unknown"<sip:Unknown@192.168.0.117>;tag=as20027259
    Call-ID: 1c3a381f0ac4409a2be685795f37ffad@192.168.0.117:5060
    CSeq: 102 OPTIONS
    Accept: application/sdp
    Accept-Language: en
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Supported: replaces
    Allow-Events: presence, message-summary, tunnel-info
    Content-Length: 0

    <------------->
    --- (13 headers 0 lines) ---
    Really destroying SIP dialog '1c3a381f0ac4409a2be685795f37ffad@192.168.0.117:5060' Method: OPTIONS

    <--- SIP read from UDP:192.168.1.161:56030 --->
    SUBSCRIBE sip:Unknown@192.168.0.117:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-a350373bc63fd667-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    To: "104"<sip:104@192.168.0.117:5060>;tag=as354fd767
    From: "104"<sip:104@192.168.0.117:5060>;tag=02084937
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 44 SUBSCRIBE
    Expires: 120
    User-Agent: 3CXPhone 5.0.14900.0
    Authorization: Digest username="104",realm="asterisk",nonce="218cae61",uri="sip:Unknown@192.168.0.117:5060",response="fb8654aa64bdbfe99974aafbd2f2a192",algorithm=MD5
    Event: message-summary
    Content-Length: 0

    <------------->
    --- (13 headers 0 lines) ---
    Found peer '104' for '104' from 192.168.1.161:56030

    <--- Transmitting (NAT) to 192.168.1.161:56030 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-a350373bc63fd667-1---d8754z-;received=192.168.1.161;rport=56030
    From: "104"<sip:104@192.168.0.117:5060>;tag=02084937
    To: "104"<sip:104@192.168.0.117:5060>;tag=as354fd767
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 44 SUBSCRIBE
    Server: FPBX-2.8.1(1.8.5.0)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="574c4107", stale=true
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.' in 6592 ms (Method: SUBSCRIBE)

    <--- SIP read from UDP:192.168.1.161:56030 --->
    SUBSCRIBE sip:Unknown@192.168.0.117:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-d7288f338134c94c-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    To: "104"<sip:104@192.168.0.117:5060>;tag=as354fd767
    From: "104"<sip:104@192.168.0.117:5060>;tag=02084937
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 45 SUBSCRIBE
    Expires: 120
    User-Agent: 3CXPhone 5.0.14900.0
    Authorization: Digest username="104",realm="asterisk",nonce="574c4107",uri="sip:Unknown@192.168.0.117:5060",response="3d926c57015e1153055a70570b5e6a66",algorithm=MD5
    Event: message-summary
    Content-Length: 0

    <------------->
    --- (13 headers 0 lines) ---
    Found peer '104' for '104' from 192.168.1.161:56030
    Scheduling destruction of SIP dialog 'Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.' in 130000 ms (Method: SUBSCRIBE)

    <--- Transmitting (NAT) to 192.168.1.161:56030 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.161:56030;branch=z9hG4bK-d8754z-d7288f338134c94c-1---d8754z-;received=192.168.1.161;rport=56030
    From: "104"<sip:104@192.168.0.117:5060>;tag=02084937
    To: "104"<sip:104@192.168.0.117:5060>;tag=as354fd767
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 45 SUBSCRIBE
    Server: FPBX-2.8.1(1.8.5.0)
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Expires: 120
    Contact: <sip:Unknown@192.168.0.117:5060>;expires=120
    Content-Length: 0


    <------------>
    Reliably Transmitting (NAT) to 192.168.1.161:56030:
    NOTIFY sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK3d49885f
    Max-Forwards: 70
    Route: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    From: "Unknown" <sip:Unknown@192.168.0.117>;tag=as354fd767
    To: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>;tag=02084937
    Contact: <sip:Unknown@192.168.0.117:5060>
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 124 NOTIFY
    User-Agent: FPBX-2.8.1(1.8.5.0)
    Event: message-summary
    Content-Type: application/simple-message-summary
    Subscription-State: active
    Content-Length: 93

    Messages-Waiting: no
    Message-Account: sip:*97@192.168.0.117:5060
    Voice-Message: 0/0 (0/0)

    ---
    Retransmitting #1 (NAT) to 192.168.1.161:56030:
    NOTIFY sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK3d49885f
    Max-Forwards: 70
    Route: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    From: "Unknown" <sip:Unknown@192.168.0.117>;tag=as354fd767
    To: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>;tag=02084937
    Contact: <sip:Unknown@192.168.0.117:5060>
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 124 NOTIFY
    User-Agent: FPBX-2.8.1(1.8.5.0)
    Event: message-summary
    Content-Type: application/simple-message-summary
    Subscription-State: active
    Content-Length: 93

    Messages-Waiting: no
    Message-Account: sip:*97@192.168.0.117:5060
    Voice-Message: 0/0 (0/0)

    ---

    <--- SIP read from UDP:192.168.1.161:56030 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK3d49885f
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    To: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>;tag=02084937
    From: "Unknown"<sip:Unknown@192.168.0.117>;tag=as354fd767
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 124 NOTIFY
    User-Agent: 3CXPhone 5.0.14900.0
    Content-Length: 0

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

    <--- SIP read from UDP:192.168.1.161:56030 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK3d49885f
    Contact: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>
    To: <sip:104@192.168.1.161:56030;rinstance=29066b5d7f1496a7>;tag=02084937
    From: "Unknown"<sip:Unknown@192.168.0.117>;tag=as354fd767
    Call-ID: Y2UwOGE2N2U4ZWQxZjY2YjFlNjQ3ZWNiZWFkNTQzMDE.
    CSeq: 124 NOTIFY
    User-Agent: 3CXPhone 5.0.14900.0
    Content-Length: 0

    <------------->
    --- (9 headers 0 lines) ---
    wv-srvoip*CLI>
    Disconnected from Asterisk server
    Executing last minute cleanups
    [root@wv-srvoip ~]#
    [root@wv-srvoip ~]#
     
  16. ahmedjalal

    Joined:
    May 28, 2012
    Messages:
    1
    Likes Received:
    0
    Cisco 7942 stuck during firmware upgrade

    Hi everyone I need help with my 7942. I was trying to upgrade the firmware from SCCP to SIP which I successfully accomplished at version 8.2.2 (2). Soon after I tried to further upgrade the SIP firmware to 9.2 and now the phone is stuck with nothing showing on the screen and only the mic button light is red. I waited for 24 hours before I tried to use the 3491672850*# key strokes to factory restore the phone and that does not work either. I have attached a picture of the phone for your reference. Please tell me what I did wrong and if I can fix the problem.
     
  17. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Re:Cisco 7942 stuck during firmware upgrade

    Hi and welcome to our Elastix Forum.

    Perhaps try to downgrade your firmware!?
    Maybe using tftpd32 (on Windows).
    With this way, you could see all tftp request and all message log.

    Regards
     
  18. franklin

    Joined:
    Oct 22, 2010
    Messages:
    254
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    0
    Hi leungs -- can you email me the firmware and other files to program the 7942? I have them for the 7940/7960, but i understand the 7942 is different. Thanks. themarshal@live.com
     
  19. sharox

    Joined:
    Aug 6, 2012
    Messages:
    1
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    0
    Hi everyone i need help configuring Cisco 7942G Phone with asterisk i have been trying with different XML files and end up with nothing, i used two firmware virsions SIP42.9-3-1-1S and SIP42.8-5-2SR1S

    With the version SIP42.9-3-1-1S giving the error massage Unprovisioned
    With the Version SIP42.8-5-2SR1S managed to provisioned the phone, although Phone still not registering with the asterisk PBX, displaying the message “your current option” not able to make any calls.

    I really appreciate If anyone can give me correct XML file or the Firmware to use with this Phone
     
  20. oneway

    Joined:
    Apr 29, 2011
    Messages:
    2
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    0
    Hi Leungs, could you send me a instruction to ww1970@gmail.com

    Thanks a lot
     

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