<div>HI Guys,</div>
<div> </div>
<div>I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? My goal is to do the paging using a multicast IP address. </div>
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<div>I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast.</div>
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<div><a href="https://issues.asterisk.org/view.php?id=11797">https://issues.asterisk.org/view.php?id=11797</a></div>
<div><a href="http://svnview.digium.com/svn/asterisk?revision=101218&view=revision">http://svnview.digium.com/svn/asterisk?revision=101218&view=revision</a></div>
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<div>My dialplan command is as below.</div>
<div> </div>
<div>exten => 1234,1,RTPPage(basic|224.1.1.1:7000|ulaw|ef)</div>
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<div>i have the same IP/Port to be listened on for multicast traffic on the Snom 320's. But when i make a call to 1234, the snom 320 does not get answered at all.</div>
<div> </div>
<div>If i use the same command and the IP of the Snom instead of the multicase IP, i was able to have the snom auto answer the call on Speaker.</div>
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<div>I would like get assistance from the community in this issue.</div>
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<div>Thanks as always</div>
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<div>Regards</div>
<div>Krishna<br></div>
<div class="gmail_quote">On Wed, May 13, 2009 at 9:21 AM, Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hello everyone,<br><br>A month ago I took on an issue on the Asterisk issue tracker (<a href="https://issues.asterisk.org/view.php?id=11797" target="_blank">https://issues.asterisk.org/view.php?id=11797</a>) dealing with multicast RTP paging.<br>
<br>This is the ability to send audio to phones (the phone must support it) and have it played out the speakerphone. Using multicast RTP is great for<br>this because it does not incur the cost and weight of setting up a potentially short call. Depending on the setup this can actually get to be quite<br>
a big problem because when you involve phones subscribed to the state of another they get told that the phone is in use. The amount of SIP traffic can<br>just spiral out of control.<br><br>Originally this issue was filed with a new application that performed the paging. I took this application and turned it into a channel driver. This means<br>
that instead of having a dedicated paging application for it you can just use Dial(). This also means that in mixed environments you can use the Page()<br>application along with other phones that do not support the multicast RTP paging.<br>
<br>So far I have gotten very little response on the issue so I am asking anyone on this mailing list who is interested and has the time to test to please test<br>and provide some feedback.<br><br>A branch based off of trunk (as that is where the channel driver will go) is available at <a href="http://svn.asterisk.org/svn/asterisk/team/file/issue11797" target="_blank">http://svn.asterisk.org/svn/asterisk/team/file/issue11797</a><br>
<br>The dial string for the channel driver is in the form of MulticastRTP/<type>/<destination>/<control address> where type is either basic or linksys. The<br>control address is only needed for the linksys type.<br>
<br>Any feedback is welcome as a note on <a href="https://issues.asterisk.org/view.php?id=11797" target="_blank">https://issues.asterisk.org/view.php?id=11797</a> and will help to getting this into the tree.<br><br>Thanks!<br>
<br>--<br>Joshua Colp<br>Digium, Inc. | Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>Check us out at: <a href="http://www.digium.com/" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org/" target="_blank">www.asterisk.org</a><br>
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