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List:       asterisk-dev
Subject:    Re: [asterisk-dev] detect called channel hang-up even with dial application g argument
From:       Yves <yves030 () gmx ! de>
Date:       2017-01-26 8:02:03
Message-ID: 8ee2fe08-48c2-e263-4c9d-4c2a1a18f351 () gmx ! de
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[Attachment #2 (multipart/alternative)]


I see.

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
to server 1.
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
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
the servers operate on different sites.

yves

Am 25.01.2017 um 22:55 schrieb Fred Muteesa:
>
> Thanks Yves,
>
> 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.
>
> Regards,
>
> Fred
>
> Sent from Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for 
> Windows 10
>
> *From: *Yves <mailto:yves030@gmx.de>
> *Sent: *Wednesday, January 25, 2017 8:37 PM
> *To: *Asterisk Developers Mailing List 
> <mailto:asterisk-dev@lists.digium.com>
> *Subject: *Re: [asterisk-dev] detect called channel hang-up even with 
> dial application g argument
>
> Hi,
>
> how about evaluating the DIALSTATUS Variable in Server2 right after 
> Dial and Hangup the call accordingly instead of waiting (wait(15))...
>
> yves
>
>
> Am 24.01.2017 um 01:38 schrieb Fred Muteesa:
>>
>> Hello Dev team,
>>
>> I have been playing with asterisk dial function and I have the 
>> senarial below.
>>
>> 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.
>>
>> Though I placed server 2 in the middle which is able to modify my 
>> parameters of the dial function and control call duration.
>>
>> 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.
>>
>> This is of extreme importance to me all advise and help will be 
>> appreciated.
>>
>> *On Server 1*
>>
>> [to_server2]
>>
>> exten => 1234,1,Dial(SIP/server2/1234,3,S(3))
>>
>> exten =>1234,2,Hangup()
>>
>> *on Server 2*
>>
>> [from_server1]
>>
>> exten => 1234,1,Dial(SIP/server3/1234,,gS(15))
>>
>> exten =>1234,2,wait(15)
>>
>> *on Server 3*
>>
>> [from_server2]
>>
>> exten =>1234,1,answer()
>>
>> exten =>1234,2,wait(3)
>>
>> Best regards,
>>
>> Fred
>>
>> VoIP Engineer
>>
>>
>>
>


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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>
    </div>
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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>
        </div>
        <p class="MsoNormal"><o:p> </o:p></p>
      </div>
      <div>
        <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>
        </div>
        <blockquote
cite="mid:AM4PR07MB15873E86E2606031C1521E2BA4750@AM4PR07MB1587.eurprd07.prod.outlook.com"
          type="cite">
          <div class="WordSection1">
            <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 =&gt;
              1234,1,Dial(SIP/server2/1234,3,S(3))</p>
            <p class="MsoNormal">exten =&gt;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 =&gt;
              1234,1,Dial(SIP/server3/1234,,gS(15))</p>
            <p class="MsoNormal">exten =&gt;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 =&gt;1234,1,answer()</p>
            <p class="MsoNormal">exten =&gt;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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