I have had a few Cisco phones connected to my installation of FreePBX. I register then with the Legacy_chan_sip driver. I am unable to make a conference with any of these phones. Is this a limitation of the driver? Or is this Cisco being jerks and removing functionality from their phones on their third party systems. Cisco is lucky that their phones are pretty, otherwise no one would ever bother with them.
server based conference will likely be fine, under resourced Cisco phones in the 7 and 8 series need CUCM for anything above ‘very basic sip’
Is there any work around for these phones?
Some would suggest ‘replacement’ 
, some suggest converting to SCCP. (You get what you pay for so that why you can buy them for $5 on ebay)
How does one start a server conference call from the front end of one of these cisco phones?
You can’t start one, you can however join one set up in FreePBX by dialing it’s number.
Well, IMO they also have excellent build quality, reliability and speakerphone voice quality, as well as being available on the used market for next to nothing.
So, I understand the attraction but agree with you that they are jerks.
For a three-way call, you can use Asterisk’s in-call transfer features; see atxferthreeway in
https://wiki.asterisk.org/wiki/display/AST/Feature+Code+Call+Transfers
This is not exposed by the FreePBX GUI, but you could add it to /etc/asterisk/features_featuremap_custom.conf
For a conference call with more than three parties, you can call each one in turn and transfer them to the conference bridge.
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