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I've been doing a little playing around with conference calls and I've encountered some odd behaviour that perhaps someone can account for.
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<div>I have a system with FreeSWITCH version 1.7 along with the FSClient (Client Id 5006). I have another system running FSClient (ClientId 5007).</div>
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<div>I have the following extension added in my public dialplan:</div>
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<div><span style="font-size: 12pt;">&nbsp; &lt;extension name=&quot;group_dial_surveillance&quot;&gt;</span></div>
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<div>&nbsp; &nbsp; &lt;condition field=&quot;destination_number&quot; expression=&quot;^1000$&quot;&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;hangup_after_bridge=true&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;continue_on_fail=true&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;originate_continue_on_timeout=true&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;call_timeout=0&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;ignore_early_media=true&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;bridge&quot; data=&quot;${group_call(surveillance@${domain_name}&#43;A)}&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;transfer&quot; data=&quot;1000 XML default&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;hangup&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &lt;/condition&gt;</div>
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<div>&nbsp; &lt;/extension&gt;</div>
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<div>If I dial the number 1000 from a third softphone, then both FSClient phones ring and when one answers the call the other stops ringing.</div>
<div>No problem there - that's exactly what I would expect as the bridge application is being passed the output for&nbsp;a&nbsp;group_call on the surveillance group (consisting of users 5006 and 5007).</div>
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<div>I also have the following Mad Boss extension in my dialplan:</div>
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<div>&nbsp; &lt;extension name=&quot;Mad Boss&quot;&gt;</div>
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<div>&nbsp; &nbsp; &lt;condition field=&quot;destination_number&quot; expression=&quot;^0911$&quot;&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;conference_auto_outcall_caller_id_name=Mad Boss1&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;conference_auto_outcall_caller_id_number=0911&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;conference_auto_outcall_timeout=60&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;conference_auto_outcall_flags=mute&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;end-conf-grace-time=1&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;conference_auto_outcall_prefix={sip_auto_answer=true}&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;set&quot; data=&quot;sip_exclude_contact=${network_addr}&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;conference_set_auto_outcall&quot; data=&quot;${group_call(surveillance@${domain_name}&#43;A)}&quot;/&gt;</div>
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<div>&lt;!-- &nbsp; &nbsp; &nbsp;&lt;action application=&quot;conference_set_auto_outcall&quot; data=&quot;user/5006@$${domain}&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;conference_set_auto_outcall&quot; data=&quot;user/5007@$${domain}&quot;/&gt; --&gt;</div>
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<div>&nbsp; &nbsp; &nbsp; &lt;action application=&quot;conference&quot; data=&quot;madboss_intercom1@default&#43;flags{endconf|deaf}&quot;/&gt;</div>
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<div>&nbsp; &nbsp; &lt;/condition&gt;</div>
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<div>&nbsp; &lt;/extension&gt;</div>
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<div>The problem is that when the third softphone is used to dial 0911, only one of the two phones answers.</div>
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<div>Looking through the log files I've found that when 1000 is dialled and the group_dial_surveillance extension is processed the following is logged:</div>
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<div><span>EXECUTE sofia/internal/4008@10.51.54.185 bridge([^^:sip_invite_domain=10.51.54.76:presence_id=5006@10.51.54.76]sofia/internal/sip:gw&#43;1@10.51.54.76:12346;transport=udp;gw=1,[^^:sip_invite_domain=10.51.54.76:presence_id=5007@10.51.54.76]<span style="background-color: rgb(255, 255, 0);">sofia/internal/sip:gw&#43;3@10.51.54.185:12346;transport=udp;gw=3</span>)</span><br>
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<div><span>but when 0911 is dialled and the Mad&nbsp;Boss extension is executed the following is logged:</span></div>
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<div><span><span>EXECUTE sofia/internal/4008@10.51.54.185 conference_set_auto_outcall([^^:sip_invite_domain=10.51.54.76:presence_id=5006@10.51.54.76]sofia/internal/sip:gw&#43;1@10.51.54.76:12346;transport=udp;gw=1,[^^:sip_invite_domain=10.51.54.76:presence_id=5007@10.51.54.76]<span style="background-color: rgb(255, 255, 0);">error/user_not_registered</span>)</span>&nbsp;</span></div>
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<div><span style="font-size: 12pt;">As far as I can see</span><span style="font-size: 12pt;"> the data passed to the
</span><b style="font-size: 12pt;">bridge </b><span style="font-size: 12pt;">and </span>
<b style="font-size: 12pt;">conference_set_auto_outcall</b><span style="font-size: 12pt;"> should be the same - the
</span><span style="font-size: 12pt;"><b>data</b></span><span style="font-size: 12pt;"><b>
</b>entries for both&nbsp;</span><span style="font-size: 12pt;">are identical, but in the case of the conference call for some reason the group_call is not locating the second user. If I switch to explicitly listing the users instead of using the group_call both
 phones will ring, if I switch back to </span><span style="font-size: 12pt;">using</span><span style="font-size: 12pt;"> group</span><span style="font-size: 12pt;">_</span><span style="font-size: 12pt;">call then the second phone doesn</span><span style="font-size: 12pt;">'</span><span style="font-size: 12pt;">t
 ring. and the same user_not_registered message is logged.</span><br>
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<div><span style="font-size: 12pt;">It's completely reproducible as every call to 1000 works every time and every call to 0911 fails every time. As mentioned, I can work around the issue, but I'm kind of curios as to what would be causing it.</span></div>
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<div><span style="font-size: 12pt;">Paul</span></div>
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<div><span style="font-size: 12pt;">PS. On a slightly different note, is there a way to terminate the conference when the person who initiates it (the moderator?) hangs up?</span></div>
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