is your problem solved or not<br><br><div class="gmail_quote">On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL <span dir="ltr"><<a href="mailto:rajib.deka@siemens.com">rajib.deka@siemens.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Thanks a lot Tony and Dhaval for your much appreciable suggestions.<br>
<br>
Regards,<br>
Rajib<br>
<br>
Rajib Deka<br>
SIEMENS Ltd.<br>
Robert V Chandran Tower, First Floor, West Wing,<br>
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.<br>
<a href="http://www.siemens.com" target="_blank">www.siemens.com</a><br>
<br>
Mob: +91-9176780669 | E-Mail: <a href="mailto:rajib.deka@siemens.com">rajib.deka@siemens.com</a><br>
<br>
Date: Wed, 20 Apr 2011 13:55:25 +0530<br>
From: DHAVAL INDRODIYA <<a href="mailto:dhaval.it01034@gmail.com">dhaval.it01034@gmail.com</a>><br>
Subject: Re: [asterisk-users] No voice in MeetMe for SIP with<br>
AGI_BACKGROUND<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <BANLkTikgRHjCVJhBC097S8n9YM66VWp=<a href="mailto:QA@mail.gmail.com">QA@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
hey try with app_rpt in asterisk<br>
<br>
regards<br>
dhaval<br>
<br>
On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield <<a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a>>wrote:<br>
<br>
> In article <<br>
> <a href="mailto:2658E54B540D284981EA57E6A549EA70ABD1FDF921@INBLRK77M1MSX.in002.siemens.net">2658E54B540D284981EA57E6A549EA70ABD1FDF921@INBLRK77M1MSX.in002.siemens.net</a><br>
> >,<br>
> Deka, Rajib IN MAA SL <<a href="mailto:rajib.deka@siemens.com">rajib.deka@siemens.com</a>> wrote:<br>
> ><br>
> > The requirement is little complicated as it is H/W specific.<br>
> > Basically we are integrating a radio gateway (SIP) with asterisk. The<br>
> gateway will be<br>
> > connected to a meetme room, so that any operator (with IP phone<br>
> registered as SIP user to<br>
> > asterisk) can login to the room and listen to radio communications and<br>
> talk.<br>
> ><br>
> > Using a PTT button someone can talk on a radio channel. Once someone<br>
> presses the PTT button<br>
> > a SIP MESSAGE is sent to the gateway with a string as payload to enable<br>
> half duplex<br>
> > communication. So, we were planning to run an AGI script with meetme<br>
> (AGI_BACKGROUND) to<br>
> > receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and<br>
> to generate a<br>
> > VarSet AMI event.<br>
> ><br>
> > Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> SIP:MESSAGE<br>
> -> radio gateway<br>
> > And vise versa.<br>
> ><br>
> > Any suggestions on the above scenario.<br>
><br>
> I don't think it can be done without making modifications to Asterisk.<br>
><br>
> The first thing I would do, if you haven't done so already, would be to<br>
> try it without MeetMe:<br>
><br>
> Operator (wants to talk) -> SIP:MESSAGE ->Dial(asterisk) SIP:MESSAGE -><br>
> radio gateway<br>
><br>
> If that works, then it would suggest that the SIP MESSAGE is<br>
> successfully getting translated into an ast_frame, which is then getting<br>
> translated back into a SIP MESSAGE. If that is not happening, you might<br>
> need to add some code to chan_sip.c to do those steps.<br>
><br>
> Once Asterisk is converting the message to and from an ast_frame, the<br>
> next step would be to add some code to app_meetme.c in the conf_run()<br>
> function, to pass those frames through, in the same way as DTMF frames<br>
> get passed through when the F option is enabled.<br>
><br>
> Presumably the messages represent PTT PRESS and PTT RELEASE. You will<br>
> need to decide what to do if you have two operators connected and they<br>
> both press the PTT.<br>
><br>
> You might also need to automatically unmute or mute the operator<br>
> channel when their PTT is pressed or released. That could also be done<br>
> within the MeetMe code.<br>
><br>
> There may be other approaches too...<br>
><br>
> Hope this helps!<br>
> Tony<br>
> --<br>
> Tony Mountifield<br>
> Work: <a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a> - <a href="http://www.softins.co.uk" target="_blank">http://www.softins.co.uk</a><br>
> Play: <a href="mailto:tony@mountifield.org">tony@mountifield.org</a> - <a href="http://tony.mountifield.org" target="_blank">http://tony.mountifield.org</a><br>
><br>
> --<br>
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</blockquote></div><br>