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

Deka, Rajib IN MAA SL rajib.deka at siemens.com
Tue Apr 19 23:39:42 CDT 2011


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: tony at softins.co.uk (Tony Mountifield)
Subject: Re: [asterisk-users] No voice in MeetMe for SIP with
        AGI_BACKGROUND
To: asterisk-users at lists.digium.com
Message-ID: <iojoq5$183$1 at softins.clara.co.uk>

In article <2658E54B540D284981EA57E6A549EA70ABD1F7C410 at INBLRK77M1MSX.in002.siemens.net>,
Deka, Rajib IN MAA SL <rajib.deka at siemens.com> 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: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org




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