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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.

...YM66VWp=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...

Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran