HI Guys, 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.
I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast. https://issues.asterisk.org/view.php?id=11797 http://svnview.digium.com/svn/asterisk?revision=101218&view=revision My dialplan command is as below. exten => 1234,1,RTPPage(basic|224.1.1.1:7000|ulaw|ef) 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. 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. I would like get assistance from the community in this issue. Thanks as always Regards Krishna On Wed, May 13, 2009 at 9:21 AM, Joshua Colp <jc...@digium.com> wrote: > Hello everyone, > > A month ago I took on an issue on the Asterisk issue tracker ( > https://issues.asterisk.org/view.php?id=11797) dealing with multicast RTP > paging. > > 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 > 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 > 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 > just spiral out of control. > > 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 > 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() > application along with other phones that do not support the multicast RTP > paging. > > 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 > and provide some feedback. > > A branch based off of trunk (as that is where the channel driver will go) > is available at http://svn.asterisk.org/svn/asterisk/team/file/issue11797 > > 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 > control address is only needed for the linksys type. > > Any feedback is welcome as a note on > https://issues.asterisk.org/view.php?id=11797 and will help to getting > this into the tree. > > Thanks! > > -- > Joshua Colp > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users