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
