Hello List, 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. Regards, Rajib Date: Tue, 19 Apr 2011 10:40:05 +0000 (UTC) From: [email protected] (Tony Mountifield) Subject: Re: [asterisk-users] No voice in MeetMe for SIP with AGI_BACKGROUND To: [email protected] Message-ID: <[email protected]> In article <2658e54b540d284981ea57e6a549ea70abd1f7c...@inblrk77m1msx.in002.siemens.net>, Deka, Rajib IN MAA SL <[email protected]> wrote: > > I have seen from the following link that, for SIP channels there is no audio > communication > possible in MeetMe with AGI_BACKGROUND. > http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe > > Currently we are using asterisk-1.6.2 and the problem still persists. Is > there any solution > available to overcome this problem? According to our requirement, we have to > run an AGI > script in MeetMe. The fact that background AGI in meetme only works with Zap channels is a consequence of the original design of Meetme. See these two old posts: http://lists.digium.com/pipermail/asterisk-users/2004-August/050725.html http://lists.digium.com/pipermail/asterisk-users/2004-August/050776.html You will need to change to a different approach to solve your requirement. Could you explain your original requirement? Then people on this list may be able to suggest an alternative way to do it. Cheers Tony -- Tony Mountifield Work: [email protected] - http://www.softins.co.uk Play: [email protected] - http://tony.mountifield.org 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
