[asterisk-users] No voice in MeetMe for SIP with

[DHAVAL INDRODIYA]

is your problem solved or not

On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL <
rajib.deka at siemens.com> 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: rajib.deka at siemens.com
>
> Date: Wed, 20 Apr 2011 13:55:25 +0530
> From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>
> Subject: Re: [asterisk-users] No voice in MeetMe for SIP with
>        AGI_BACKGROUND
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>        <asterisk-users at lists.digium.com>
> Message-ID: <BANLkTikgRHjCVJhBC097S8n9YM66VWp=QA at mail.gmail.com>
> 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 <tony at softins.co.uk
> >wrote:
>
> > In article <
> >
> 2658E54B540D284981EA57E6A549EA70ABD1FDF921 at INBLRK77M1MSX.in002.siemens.net
> > >,
> > Deka, Rajib IN MAA SL <rajib.deka at siemens.com> 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: tony at softins.co.uk - http://www.softins.co.uk
> > Play: tony at mountifield.org - http://tony.mountifield.org
> >
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