I am having the same problem on the 2.02 x64 stable as reported on the issues number 245 and 255 in your bug tracker. I have cisco 7940 phones with SIP firmware 8.8 and after I create 2 extensions and use endpoint configuration to assign the extensions to the discovered phones, the phones pickup the configuration from tftp and you can see the name appear on the cisco phones but I cannot dial between extensions, cant dial voicemail, *65 does not work. I get the following on the cli> == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 elastix*CLI> nothing else happens. When I do a Show sip registry I get undefined and 2 offlines in the monitor. I am trying to replace a 35 phone druid OSE 2.01 for obvious reasons that druid is dead. Elastix seemed to have the best intrface and maturity from all my research and I need to do a migration quick because of all the silly bugs in druid that have not been addressed over the last year. I already yum updated freepbx to the lastest version and its the same result. Where do the extensions get defined in Elastix. Do they get called from custom conf files because I cant see them in extensions.conf or sip.conf. However dialplan show from cli lists them. If you need me to post my conf files let me know which ones and I will. Thanks.