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<p>Hi folks.</p>
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<p>I have a FreeSWITCH server with a couple of FSClient softphones attached. I've been experimenting with conference calls using the following <span style="font-size: 12pt;">extension configuration (based upon the Mad Boss example):</span></p>
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<div><span style="font-size: 11pt;"> <extension name="<span style="font-family: Calibri, Arial, Helvetica, sans-serif, EmojiFont, "Apple Color Emoji", "Segoe UI Emoji", NotoColorEmoji, "Segoe UI Symbol", "Android Emoji", EmojiSymbols; font-size: 14.6667px;">Broadcast
Announcement</span>"></span></div>
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<div><span style="font-size: 11pt;"> <condition field="destination_number" expression="^0911$"></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="conference_auto_outcall_caller_id_name=Broadcast"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="conference_auto_outcall_timeout=60"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="conference_utils_auto_outcall_flags=mute"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="end-conf-grace-time=0"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="set" data="sip_exclude_contact=${network_addr}"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="conference_set_auto_outcall" data="user/5006@$${domain}"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="conference_set_auto_outcall" data="user/5007@$${domain}"/></span></div>
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<div><span style="font-size: 11pt;"> <action application="conference" data="SystemBroadcast@default++flags{deaf|moderator|endconf}"/></span></div>
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<div><span style="font-size: 11pt;"> </condition></span></div>
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<div><span style="font-size: 11pt;"> </extension></span></div>
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<p>I've noticed a couple of issues which seem to be tied to the sip_auto_answer setting.</p>
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<li>The phone making the call is providing a live radio feed, so the callees should receive audio, but they do not</li><li>When the caller disconnects from the conference, the callees remain connected despite the fact that the moderator and endconf flags are set.</li></ol>
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<div>If I change the sip_auto_answer setting from true to false and then place the call, each phone rings until answered but</div>
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<li>Once a phone is answered you hear the radio feed.<br>
</li><li>When the caller disconnects, so do all the callees.</li></ol>
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<div>Can anyone explain why I am seeing the difference in behaviour between the true and false settings for sip_auto_answer?</div>
<div>I was expecting the same behaviour in both cases and that setting the sip_auto_answer flag would simply mean that I didn't have to explicitly answer each phone added to the conference.</div>
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<div>Paul</div>
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