<div dir="ltr"><br><div class="gmail_extra"><div class="gmail_quote">On Wed, Jun 12, 2013 at 8:29 PM, Meadows Hoa <span dir="ltr"><<a href="mailto:meadows_sub@yahoo.com" target="_blank">meadows_sub@yahoo.com</a>></span> wrote:<br>
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<div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif">QUESTION #1<br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif">
If Asterisk's SIP channel driver (version 1.8) receives an SIP:INVITE from an endpoint (calling party), is it possible to play an announcement back to that originating endpoint immediately after the final SIP:ACK is received by the SIP channel driver?</div>
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Would something in the extensions.conf file like the following work?<br></div><div style="font-style:normal;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif">
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<div class=""><i>Exten => 7000,1,Dial(SIP/7000,10)</i></div>
<div class=""><i>Same => n,GotoIf($[“${DIALSTATUS}” =
“ANSWER”]?answer:busy)</i></div>
<div class=""><i>Same => n(answer),Playback(tt-monkeys)</i></div>
<div class=""><i>Same => n(busy),Playback(tt-somethingwrong,noanswer)</i></div><div style="font-style:italic;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif" class="">
<br></div></div></div></blockquote><div><br></div><div style>No. Dial immediately bridges the channels. With no options specified, the Dial application won't return until the bridge is broken, which typically occurs when channels hangup. With no options specified, Asterisk will typically hang up the non-hungup channel when the bridge is broken.</div>
<div style><br></div><div style>If you want to playback a file to the caller, you should do so before the Dial application is called. If you want to playback a file to the called party before bridging, you should use the A option. See documentation for the Dial application on the Asterisk wiki for more information:</div>
<div style><br></div><div style><a href="https://wiki.asterisk.org/wiki/display/AST/Application_Dial">https://wiki.asterisk.org/wiki/display/AST/Application_Dial</a><br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
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QUESTION #2:<br></div><div style="font-style:italic;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif" class=""><span style="font-weight:bold">Also, would there be a way to send an announcement through the terminating channel driver if the originating channel driver hung up,. so that an ANNOUNCEMENT is played to the called party before the terminating channel driver sends a SIP:BYE to the terminating endpoint (called party)?</span></div>
<div style="font-style:italic;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif" class=""><br></div></div></div></blockquote><div><br></div><div style>Asterisk is a B2BUA, so the calling party does not transmit a BYE request to the called party, Asterisk does. As such, yes, you can generally manipulate what is presented and when it is presented.</div>
<div style><br></div><div style>In this particular case, you could use the F option to transfer the called party to a new context, extension, priority and playback a file from that location. You can then hangup the called party.</div>
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<br></div><div style="font-style:italic;font-size:13.3333px;background-color:transparent;font-family:'times new roman','new york',times,serif" class=""><i>QUESTION #3:
<br>Is there a way to have a Channel Driver 'trigger' an annoucement to play at different times in the call flow through that channel driver to its endpoint? Some kind of function to inv </i></div></div></div><br>
</blockquote><div><br></div><div style>There are certain interception routines that can be fired at particular times in the dialplan. These include pre-dial handlers, pre-bridge handlers, and hangup handlers. Several other options allow you to periodically playback announcements to channels, although these are typically used in the context of warning announcements before calls are terminated (L option in Dial/Bridge). Documentation for all of these items can be found on the Asterisk wiki.</div>
<div style><br></div><div style>Breaking out of a bridge between two channels to playback audio can be disruptive to the participants in a call. There are other applications, such as ChanSpy, that can whisper into channels. If you needed this to trigger at particular times, you would most likely need to use AMI to monitor the progress of the call and initiate the action.</div>
<div style><br></div><div style>All of that being said, in general, configuration questions such as these are more appropriate for the asterisk-users mailing list. The asterisk-dev list is used for discussing implementation of features within Asterisk - you will most likely get more and better answers on the users mailing list than here.<br>
</div></div><div class="gmail_extra"><br></div>Matt<br clear="all"><div><br></div>-- <br><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div>
<div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div></div>
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