[asterisk-users] Trouble registering Cisco 7942

[Warren Selby]

I think your featureLabel definition is wrong.

On the login issue, ssh to the ip of the phone and login first with  
the user/pass you defined in the file (admin/123), then at the second  
login prompt use log/log. That should get you the log files which will  
show you your error.



Thanks,
--Warren Selby

On Nov 7, 2009, at 9:45 AM, Stephen Reese <rsreese at gmail.com> wrote:

> On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby  
> <wcselby at selbytech.com> wrote:
>> That typically means you've got an error in your phone specific  
>> config file,
>> the SEP[MAC].cnf.xml.
>>
>> You need to login to the phone via ssh and use the log/log login.   
>> Once
>> you've done that, look at the logs and see what line of the config  
>> is giving
>> it grief.  Once you know that, you'll know what's causing the  
>> Unprovisioned
>> message.
>
> I set the username and password but am unable to log into the phone. I
> provided an updated config below. I am prompted for the username and
> password though.
>
> Secondly should I be using IP or hostnames for the <proxy> and
> <processNodeName> or does it not matter? Thanks
>
> <device>
> <deviceProtocol>SIP</deviceProtocol>
> <sshUserId>admin</sshUserId>
> <sshPassword>123</sshPassword>
> <devicePool>
> <callManagerGroup>
>   <members>
>      <member priority="0">
>         <callManager>
>            <ports>
>               <ethernetPhonePort>2000</ethernetPhonePort>
>               <sipPort>5060</sipPort>
>               <securedSipPort>5061</securedSipPort>
>            </ports>
>            <processNodeName>SIPSERVER</processNodeName>
>         </callManager>
>      </member>
>   </members>
> </callManagerGroup>
> </devicePool>
> <sipCallFeatures>
>   <cnfJoinEnabled>true</cnfJoinEnabled>
>   <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
>   <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
>   <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
>   <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
>   <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
>   <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
>   <rfc2543Hold>false</rfc2543Hold>
>   <callHoldRingback>2</callHoldRingback>
>   <localCfwdEnable>true</localCfwdEnable>
>   <semiAttendedTransfer>true</semiAttendedTransfer>
>   <anonymousCallBlock>2</anonymousCallBlock>
>   <callerIdBlocking>2</callerIdBlocking>
>   <dndControl>0</dndControl>
>   <remoteCcEnable>true</remoteCcEnable>
> </sipCallFeatures>
>     <natEnabled>true</natEnabled>
>     <natAddress>172.16.2.1</natAddress>
>     <phoneLabel>102</phoneLabel>
> <sipLines>
>  <line button="1">
>  <featureID>9</featureID>
>  <featureLabel>102</featureLabel>
>  <contact>102</contact>
>  <proxy>SIPSERVER</proxy>
>  <port>5060</port>
>  <name>102</name>
>  <displayName>Atlas</displayName>
>  <authName>102</authName>
>  <authPassword>PASS</authPassword>
>    <sharedLine>false</sharedLine>
>    </line>
> </sipLines>
> </device>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users