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=101218view=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
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