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

[Deka, Rajib IN MAA SL]

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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