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 <[email protected]> wrote: > On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby > <[email protected]> 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 _______________________________________________ -- 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
