hey try with app_rpt in asterisk regards dhaval
On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield <[email protected]>wrote: > In article < > 2658e54b540d284981ea57e6a549ea70abd1fdf...@inblrk77m1msx.in002.siemens.net > >, > Deka, Rajib IN MAA SL <[email protected]> wrote: > > > > The requirement is little complicated as it is H/W specific. > > Basically we are integrating a radio gateway (SIP) with asterisk. The > gateway will be > > connected to a meetme room, so that any operator (with IP phone > registered as SIP user to > > asterisk) can login to the room and listen to radio communications and > talk. > > > > Using a PTT button someone can talk on a radio channel. Once someone > presses the PTT button > > a SIP MESSAGE is sent to the gateway with a string as payload to enable > half duplex > > communication. So, we were planning to run an AGI script with meetme > (AGI_BACKGROUND) to > > receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and > to generate a > > VarSet AMI event. > > > > Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> SIP:MESSAGE > -> radio gateway > > And vise versa. > > > > Any suggestions on the above scenario. > > I don't think it can be done without making modifications to Asterisk. > > The first thing I would do, if you haven't done so already, would be to > try it without MeetMe: > > Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE -> > radio gateway > > If that works, then it would suggest that the SIP MESSAGE is > successfully getting translated into an ast_frame, which is then getting > translated back into a SIP MESSAGE. If that is not happening, you might > need to add some code to chan_sip.c to do those steps. > > Once Asterisk is converting the message to and from an ast_frame, the > next step would be to add some code to app_meetme.c in the conf_run() > function, to pass those frames through, in the same way as DTMF frames > get passed through when the F option is enabled. > > Presumably the messages represent PTT PRESS and PTT RELEASE. You will > need to decide what to do if you have two operators connected and they > both press the PTT. > > You might also need to automatically unmute or mute the operator > channel when their PTT is pressed or released. That could also be done > within the MeetMe code. > > There may be other approaches too... > > Hope this helps! > Tony > -- > Tony Mountifield > Work: [email protected] - http://www.softins.co.uk > Play: [email protected] - http://tony.mountifield.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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