hey try with app_rpt in asterisk<br><br>regards<br>dhaval<br><br><div class="gmail_quote">On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield <span dir="ltr"><<a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">In article <<a href="mailto:2658E54B540D284981EA57E6A549EA70ABD1FDF921@INBLRK77M1MSX.in002.siemens.net">2658E54B540D284981EA57E6A549EA70ABD1FDF921@INBLRK77M1MSX.in002.siemens.net</a>>,<br>
<div class="im">Deka, Rajib IN MAA SL <<a href="mailto:rajib.deka@siemens.com">rajib.deka@siemens.com</a>> wrote:<br>
><br>
</div><div class="im">> The requirement is little complicated as it is H/W specific.<br>
> Basically we are integrating a radio gateway (SIP) with asterisk. The gateway will be<br>
> connected to a meetme room, so that any operator (with IP phone registered as SIP user to<br>
> asterisk) can login to the room and listen to radio communications and talk.<br>
><br>
> Using a PTT button someone can talk on a radio channel. Once someone presses the PTT button<br>
> a SIP MESSAGE is sent to the gateway with a string as payload to enable half duplex<br>
> communication. So, we were planning to run an AGI script with meetme (AGI_BACKGROUND) to<br>
> receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and to generate a<br>
> VarSet AMI event.<br>
><br>
> Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> SIP:MESSAGE -> radio gateway<br>
> And vise versa.<br>
><br>
> Any suggestions on the above scenario.<br>
<br>
</div>I don't think it can be done without making modifications to Asterisk.<br>
<br>
The first thing I would do, if you haven't done so already, would be to<br>
try it without MeetMe:<br>
<br>
Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE -> radio gateway<br>
<br>
If that works, then it would suggest that the SIP MESSAGE is<br>
successfully getting translated into an ast_frame, which is then getting<br>
translated back into a SIP MESSAGE. If that is not happening, you might<br>
need to add some code to chan_sip.c to do those steps.<br>
<br>
Once Asterisk is converting the message to and from an ast_frame, the<br>
next step would be to add some code to app_meetme.c in the conf_run()<br>
function, to pass those frames through, in the same way as DTMF frames<br>
get passed through when the F option is enabled.<br>
<br>
Presumably the messages represent PTT PRESS and PTT RELEASE. You will<br>
need to decide what to do if you have two operators connected and they<br>
both press the PTT.<br>
<br>
You might also need to automatically unmute or mute the operator<br>
channel when their PTT is pressed or released. That could also be done<br>
within the MeetMe code.<br>
<br>
There may be other approaches too...<br>
<br>
Hope this helps!<br>
<div class="im">Tony<br>
--<br>
Tony Mountifield<br>
Work: <a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a> - <a href="http://www.softins.co.uk" target="_blank">http://www.softins.co.uk</a><br>
Play: <a href="mailto:tony@mountifield.org">tony@mountifield.org</a> - <a href="http://tony.mountifield.org" target="_blank">http://tony.mountifield.org</a><br>
<br>
</div><div><div></div><div class="h5">--<br>
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