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<div class="moz-cite-prefix">I see.<br>
<br>
so if you donīt have control over server 2 the only way I could
think of is some kind of external hook... server 3 has to indicate
the end of call between S2 and S3 somehow<br>
to server 1.<br>
this could be done by calling an URL with call specific parameters
or, if you have access to server 1 and 3, connect them and close
the "circuit" back from <br>
server 3 to 1 and use this "channel" as a signalling channel. this
should be very easy, if server 1 and 3 are on the same network and
slightly more work, when<br>
the servers operate on different sites.<br>
<br>
yves<br>
<br>
Am 25.01.2017 um 22:55 schrieb Fred Muteesa:<br>
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<p class="MsoNormal">Thanks Yves, </p>
<p class="MsoNormal">That makes sense but I am looking at a
situation where, server2 is a service provider that I have no
control over, This is a big issue I am already facing.
</p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Regards,</p>
<p class="MsoNormal">Fred</p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Sent from <a moz-do-not-send="true"
href="https://go.microsoft.com/fwlink/?LinkId=550986">
Mail</a> for Windows 10</p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal" style="border:none;padding:0in"><b>From:
</b><a moz-do-not-send="true" href="mailto:yves030@gmx.de">Yves</a><br>
<b>Sent: </b>Wednesday, January 25, 2017 8:37 PM<br>
<b>To: </b><a moz-do-not-send="true"
href="mailto:asterisk-dev@lists.digium.com">Asterisk
Developers Mailing List</a><br>
<b>Subject: </b>Re: [asterisk-dev] detect called channel
hang-up even with dial application g argument</p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<div class="moz-cite-prefix">Hi,<br>
<br>
how about evaluating the DIALSTATUS Variable in Server2 right
after Dial and Hangup the call accordingly instead of waiting
(wait(15))...<br>
<br>
yves<br>
<br>
<br>
Am 24.01.2017 um 01:38 schrieb Fred Muteesa:<br>
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<p class="MsoNormal">Hello Dev team, </p>
<p class="MsoNormal">I have been playing with asterisk dial
function and I have the senarial below.
</p>
<p class="MsoNormal">I am generating a call from server 1
and receiving it on server 3, but I want server 1 to
control how long this call should be.
</p>
<p class="MsoNormal">Though I placed server 2 in the middle
which is able to modify my parameters of the dial function
and control call duration.</p>
<p class="MsoNormal">How do I detect on server 1 that server
3 has hangup so that server 2 does not keep the call
connected longer than I require.</p>
<p class="MsoNormal">This is of extreme importance to me all
advise and help will be appreciated.</p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>On Server 1<o:p></o:p></b></p>
<p class="MsoNormal">[to_server2]</p>
<p class="MsoNormal">exten =>
1234,1,Dial(SIP/server2/1234,3,S(3))</p>
<p class="MsoNormal">exten =>1234,2,Hangup() </p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>on Server 2<o:p></o:p></b></p>
<p class="MsoNormal">[from_server1]</p>
<p class="MsoNormal">exten =>
1234,1,Dial(SIP/server3/1234,,gS(15))</p>
<p class="MsoNormal">exten =>1234,2,wait(15)</p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>on Server 3<o:p></o:p></b></p>
<p class="MsoNormal">[from_server2]</p>
<p class="MsoNormal">exten =>1234,1,answer()</p>
<p class="MsoNormal">exten =>1234,2,wait(3)</p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Best regards,</p>
<p class="MsoNormal">Fred</p>
<p class="MsoNormal">VoIP Engineer</p>
<p class="MsoNormal"><o:p> </o:p></p>
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