asterisk users - Apr 2011 - No voice in MeetMe for SIP with AGI_BACKGROUND

Hello List,

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.

Kind 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<http://www.siemens.com>

Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com


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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

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




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.

Thanks a lot guys.
We planning to use different approach like maintaining a separate extension for
sending and receiving in dialog SIP:MESSAGE using SendText and RECEIVE TEXT.

Regards,
Rajib

DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>
wrote:
>
> is your problem solved or not
It will take a lot more time than that to try out the suggestions!
> 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
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - 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.