search for: confroom

hi; i setup a Meetme conference room and i notice the following behavior: if A calls confroom over PSTN channel 1 B call confroom over PSTN channel 2 C calls confroom over SIP/Ethernet then i have all of them talking and the media stream mixed by asterisk. However, if i hang up A, channel 1 is still ocuppied (i try dialing inbound again on channel and it continues to give a busy siganl)...

Running * version 1.6.1.17. My meetme conferences automagically disconnect users approximately 5-15 seconds after the user is connected. This occurs regardless of whether music on hold is active or not. [Mar 23 11:34:36] -- Executing Macro("SIP/SDN_TMCKEE-000000e9", "confroom,1808") [Mar 23 11:34:36] -- Executing [s at macro-confroom:1] Answer("SIP/SDN_TMCKEE-000000e9", "") in new stack [Mar 23 11:34:36] -- Executing [s at macro-confroom:2] Wait("SIP/SDN_TMCKEE-000000e9", "1") in new stack [Mar 23 11:34:37] -- Exe...

...("Customer") Customer identifies himself, and now I use Dial w/ the G option: Dial(Zap/g9/${agentext}|60|mG(Agent_Xfer^s^1) Customer hears MOH while the Dial app gets the "agent" on the line My destination context looks like this: [Agent_Xfer] exten => s,1(Customer),Meetme({$ConfRoom}|qM) exten => s,2(Agent),Macro(Connect_to_agent,${Customerid},${ConfRoom}) Customerid and ConfRoom are channel variables that are set in the Originate action and at the start of the dialplan processing, respectively. The idea is to put the customer in a conference room, listening to MOH, until...

...la commande et lib?ration du buffer fprintf(stderr,"%s\n",cmd); printf("%s\n",cmd); fflush(stdout); resultcode = checkresult(); // Mise en conf?rence de l'operateur strcpy(cmd1,""); strcpy(cmd1,"EXEC MeetMe "); strcat(cmd1,confroom); //confroom is a variable quote from teh database strcat(cmd1,"|q"); fprintf(stderr,"%s\n",cmd1); printf("%s\n",cmd1); fflush(stdout); .......... Any reason on why only the first command is successfull?? Thanks in adavance. Lamine

We've started to use Asterisk for conferencing and have been getting some complaints. Our configuration is that some people call in from home, but we have a physical conference room with a Polycom. When somebody was giving a presentation in the physical conference room, we were told that the remote people kept hearing him cut in and our. To me, this sounds like the talking optimization was

...xtension Changed 612[a10base] new state InUse for Notify User mlapolla == Extension Changed 612[a10base] new state InUse for Notify User sparea1-a10 == Extension Changed 612[a10base] new state InUse for Notify User clerical-a10 == Extension Changed 612[a10base] new state InUse for Notify User confroom-a10 == Extension Changed 612[a10base] new state InUse for Notify User smcclain == Extension Changed 612[a10base] new state InUse for Notify User vdemarco -- Executing [*8617 at a10f:2] Playback("SIP/tkeeley-acc9aaf8", "im-sorry") in new stack -- <SIP/tkeeley-acc9a...

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so