Currently i have Aastra and Cisco having this behavior, working on getting a polycom lab setup. FreeSWITCH Version 1.4.7+git~20140724T230506Z~9d1c0f7f3d~64bit (git 9d1c0f7 2014-07-24 23:05:06Z 64bit) I have my extensions 101, 102 and 103 So if i call in, and answer on 101, 102 and 103 will successfully show that 101 is on the phone. If i then transfer the call from 101 to 102, 102 and 103 will show 101 get off the phone, however 101, 103 never show that 102 is on the phone. further transferring the call from 102 to 101 or 103 also never shows that they are on the phone *conf/sip_profile/internal.xml:* http://pastebin.freeswitch.com/23020 *conf/autoload_configs/presence_map.conf.xml:* http://pastebin.freeswitch.com/23021 Dial String is as follows: <param name="dial-string" value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@ ${dialed_domain}):transfer_fallback_extension=${dialed_user}}"/> This behavior is similar on Jitsi, but because i set it to do peer mode, it checks every 30 seconds so after 30 seconds it shows that the person (in this case 102) is on the phone.