is your problem solved or not On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL < [email protected]> wrote:
> Thanks a lot Tony and Dhaval for your much appreciable suggestions. > > Regards, > Rajib > > Rajib Deka > SIEMENS Ltd. > Robert V Chandran Tower, First Floor, West Wing, > #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. > www.siemens.com > > Mob: +91-9176780669 | E-Mail: [email protected] > > Date: Wed, 20 Apr 2011 13:55:25 +0530 > From: DHAVAL INDRODIYA <[email protected]> > Subject: Re: [asterisk-users] No voice in MeetMe for SIP with > AGI_BACKGROUND > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > Content-Type: text/plain; charset="iso-8859-1" > > 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 > > > > > Important notice: This e-mail and any attachment there to contains > corporate proprietary information. If you have received it by mistake, > please notify us immediately by reply e-mail and delete this e-mail and its > attachments from your system. > Thank You. > > -- > _____________________________________________________________________ > -- 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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