<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">Please file a jira for this issue.<div><br><div><div>On Aug 8, 2014, at 3:40 AM, Chris Tunbridge <<a href="mailto:blasterjr@gmail.com">blasterjr@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr"><div>Currently i have Aastra and Cisco having this behavior, working on getting a polycom lab setup.<br><br>FreeSWITCH Version 1.4.7+git~20140724T230506Z~9d1c0f7f3d~64bit (git 9d1c0f7 2014-07-24 23:05:06Z 64bit)<br>
<br>I have my extensions 101, 102 and 103<br><br></div><div>So if i call in, and answer on 101, 102 and 103 will successfully show that 101 is on the phone.<br><br>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.<br>
<br></div><div>further transferring the call from 102 to 101 or 103 also never shows that they are on the phone<br><br></div><div><b>conf/sip_profile/internal.xml:</b> <a href="http://pastebin.freeswitch.com/23020">http://pastebin.freeswitch.com/23020</a><br>
</div><div><b>conf/autoload_configs/presence_map.conf.xml:</b> <a href="http://pastebin.freeswitch.com/23021">http://pastebin.freeswitch.com/23021</a><br></div><div><br></div><div>Dial String is as follows:<br><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}}"/><br>
<br>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.<br></div></div></blockquote></div><br></div></body></html>