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Conference station CP-8831 - how to configure SIP with ASTERISK PBX server

janbarabas
Beginner
Beginner

Dear all,

I took over administration of our VoIP phones since the old admin quit unexpectedly.. We have approx. 150 phones from Cisco - IP PHONE 303, registering on Asterisk phone gateway. I now need to urgently add new phone to our network - the Conference station CP-8831 - however, for the love of god, I can't get to any important admin settings in the phone menu. Neither can I enable the web server to access admin settings via web intefrace..

UPDATE 10.5.2014:

Since admin settings are no longer available via WebUI I managed to get hold of CNF.XML file for the CP-8831 - tried editing various options in the XML to connect to our SIP server (and uploaded to phone using TFTP), but no luck! The darn thing doesen't even try to make a connection (IP config 100% OK)

 

Anyone can help please?

 

Jan

 

 

 

 

10 REPLIES 10

Hi.

With http access, you can only display info for troubleshooting purpose.

If you want to change network settings to pair the conf station to your asterisk, you can access through the panel as specified on the following doc.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8831/9_3_3/english/adminguide/CS38_BK_A8C4AC51_00_adminguide-8831/CS38_BK_A8C4AC51_00_adminguide-8831_chapter_0100.html

 

 

HTH

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Thanks Carlo..

however, I knew I could get to these settings - but I need to change more advanced ones!

I managed to get a hold of a sample CP-8831 CNF.XML file and have a TFTP server running and can upload the XML to the phone. However, I need to find the following options (tags) in the XML file:

 

Specifically:

 

1. IP address of my SIP server

2. Username

3. Password

 

The SIP server is based on Asterisk and works OK with Cisco SPA 303 IP phones (these have an admin WebUI enabled - no idea why CISCO removed this feature from CP-8831!)

 

Can someone PLEAAAAASE help? I already lost 2 weeks of my time and energy on this :(

 

Jan

Were you ever able to get your 8831 to communicate with asterisk successfullt? I am in a similar boat right now. It finds the configuration file but is erroring out trying to verify it. I am in desperate need of an example configuration file to base off of.

My english is not well I have the same problem how to configure cisco Conference station CP-8831 SIP with Asterisk Server.

Need yours helps

DId anyone find a solution to this problem.  Also in the same boat and trying to inquire if a solution was found.  Desperately seeking help.

 

Thank You.

Good morning to everyone ? Me also I faced the same problem of cisco ip conference phone CP-8831G . I tried to change it as SIP but i was unable to get the SIP configurations options . 

The user and password and SIP IP are not there in settings . Did anyone had successful configuration and help me to make it works ??  Your assistance shall be helpful . Same to CP-7937G it was not able to works on SIP IP PBX .

Thanks 

JOERG HOCHWALD
Beginner
Beginner

 

Jan,

you might want to try this:

<?xml version="1.0" ?>
<device>
  <deviceProtocol>SIP</deviceProtocol>
  <sshUserId>cisco</sshUserId>
  <sshPassword>cisco</sshPassword>
  <devicePool>
    <dateTimeSetting>
      <dateTemplate>D.M.Y</dateTemplate>
      <timeZone>W. Europe Standard/Daylight Time</timeZone>
      <ntps>
        <ntp>
          <name>YOURNTPSERVERIPADDRESS</name>
          <ntpMode>Unicast</ntpMode>
        </ntp>
      </ntps>
    </dateTimeSetting>
    <callManagerGroup>
      <members>
        <member priority="0">
          <callManager>
            <processNodeName>YOURASTERISKIPADDRESS</processNodeName>
            <ports>
              <sipPort>5060</sipPort>
            </ports>
          </callManager>
        </member>
      </members>
    </callManagerGroup>
  </devicePool>
  <sipProfile>
    <natEnabled>false</natEnabled>
    <natAddress></natAddress>
    <sipProxies>
      <registerWithProxy>true</registerWithProxy>
      <outboundProxy></outboundProxy>
      <outboundProxyPort></outboundProxyPort>
      <backupProxy>YOURASTERISKIPADDRESS</backupProxy>
      <backupProxyPort>5060</backupProxyPort>
    </sipProxies>
    <preferredCodec>YOUMIGHTNEED_OR_WANT_TOSETTHIS</preferredCodec>
    <phoneLabel>_NAME_</phoneLabel>
    <sipLines>
      <line button="1">
        <featureID>9</featureID>
        <featureLabel>_USER_</featureLabel>
        <proxy>USECALLMANAGER</proxy>
        <port>5060</port>
        <name>_USER_</name>
        <displayName>_USER_</displayName>
        <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
        </autoAnswer>
        <callWaiting>3</callWaiting>
        <authName>_USER_</authName>
        <authPassword>_PASSWORD_</authPassword>
        <sharedLine>false</sharedLine>
        <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
        <messagesNumber>*97</messagesNumber>
        <ringSettingIdle>4</ringSettingIdle>
        <ringSettingActive>5</ringSettingActive>
        <contact>_USER_</contact>
        <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>true</callerNumber>
          <redirectedNumber>true</redirectedNumber>
          <dialedNumber>true</dialedNumber>
        </forwardCallInfoDisplay>
      </line>
    </sipLines>
    <dialTemplate>dialplan.xml</dialTemplate>
  </sipProfile>
  <userLocale>
    <name>English_United_Kingdom</name>
    <langCode>en</langCode>
  </userLocale>
  <networkLocale>Germany</networkLocale>
  <networkLocaleInfo>
    <name>Germany</name>
  </networkLocaleInfo>
</device>

Honestly: I didn't try it. I have a 8831, but it runs on an Cisco UC Server and not attached to an Asterisk. But i have other devices attached to the Asterisk and the work with configurations like this one.

Please note: You have to provide your own configuration (e.g. Language, Timezone and Asterisk Server info) to get it up and running. 

Good luck ;-)

/JH

I know this thread is old, but this is worth mentionning.

We were trying to register our CP-8861 with our FreePBX and were unable to do so until we found this XML.

This CNF.XML worked flawlessly for us and we were able to register our Cisco IP Phone 8861 with our PBX system by pushing the config through tftp.

We also updated the SIP firmware to the latest version, for all that matters.

Thank Joerg for providing this XML.

I was able to get the

I'm using an older version of Asterisk, which did not support TCP, so I had to change the TransportLayer to "2" for UDP only since I'm only using UDP transport at the moment.  There is probably a LOT more broken in this config file, but since I haven't achieved successful registration I haven't cleaned it up.  But it does TRY to register to my Asterisk server, at least.

The problem: the password doesn't seem to work.  I simplified it, made all of the authentication and extension nomenclature the same (3127) and even validated that the Asterisk server was correctly working by testing another SIP client on the same credentials.  But the 8831 doesn't want to register.  I see it trying, but it gets a bad password response (authentication failure.)  I've tried multiple passwords to see if it's just a short/long problem - no go. Latest versions of Asterisk were used as of 2016/01/15.

Has anyone been able to successfully get an 8831 registered or calling to Asterisk?

[asterisk.blah.com tftpboot]# cat SEPAC44F2152195.cnf.xml
<?xml version="1.0" ?>
<device>

<vendorConfig>
<wirelessMicRegion>0</wirelessMicRegion>
<settingsAccess>1</settingsAccess>
<webAccess>1</webAccess>
<moreKeyReversionTimer>5</moreKeyReversionTimer>
<g722CodecSupport>0</g722CodecSupport>
<enableCdpSwPort>0</enableCdpSwPort>
<enableLldpSwPort>0</enableLldpSwPort>
<lldpAssetId>8</lldpAssetId>
<powerPriority>0</powerPriority>
<displayRefreshRate>0</displayRefreshRate>
<useEnblocDialing>1</useEnblocDialing>
<sshAccess>0</sshAccess>
<sshPort>22</sshPort>

</vendorConfig>


<loadinformation>ip8831.10-3-1SR2-2</loadinformation>

<deviceProtocol>SIP</deviceProtocol>

<transportLayerProtocol>2</transportLayerProtocol>

<webAccess>0</webAccess>

<sshUserId>admin</sshUserId>
<sshPassword>nottherealpassword</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D.M.Y</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>128.138.141.172</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<processNodeName>204.210.2.69</processNodeName>
<ports>
<sipPort>5060</sipPort>
</ports>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<natEnabled>false</natEnabled>
<natAddress></natAddress>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<backupProxy>204.210.2.69</backupProxy>
<backupProxyPort>5060</backupProxyPort>
</sipProxies>
<phoneLabel>THE-PHONE</phoneLabel>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>Line1</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>3127</name>
<displayName>Line1</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>3127</authName>
<authPassword>9999</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>3127</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<userLocale>
<name>English_United_Kingdom</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>English_United_Kingdom</networkLocale>
<networkLocaleInfo>
<name>English_United_Kingdom</name>
</networkLocaleInfo>
</device>

[asterisk.blah.com tftpboot]#

I think Im late to the party, but I have infos that may be useful.

I recently had troubles registering a 7945 to my Asterisk PBX.  The phone would get the config file but would sit in the "registering" state.  From the PCAPs I took I saw that the phone was sending registration requests then receiving a 401unauthorized challenge but not responding with UN/PW info.  I tried changing the password multiple times but nothing I did with it seemed to make any difference.  The fix that I found was to change "NAT mode" to "no" on the extensions config in the PBX.  I don't know if your issues are the same, but they sound similar.

Hope this was helpful.

Zach

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