similar to: Strange Meetme disconnects

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) any

Has anyone used the G option with the Dial app? I'm looking for a way to control the called party leg. Specifically, I'd like to pass a few variables to the called side for some call control. Here's a synopsis of what I'm doing: Make outbound call w/ AMI Originate action. Called party answers ("Customer") Customer identifies himself, and now I use Dial w/ the G

Hello everybody, I would like to know whether it is possible to run Dial and MeetMe commands simultaneoously on the same channel. I am using a C AGI as below but it seems to me that only the first command that is called in the agi is executed. ........... // Pr?paration de la commande pour l'appel du client fprintf(stderr,"%s%s",numtocall," is the number to

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAARRRRGGGGGGHHHHH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get:

I'm curious as to what versions of everything you are using. Reason being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing it to SIP/5000-00000000". It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to SIP/801-0000000c" [1-1 being the span and channel

About every 10th call coming into my x1000p is not getting the audio it should. You can see the messages scrolling on the console as they usually would, playing the thankyou, then and menu messages. internal phones ring, but when answered there is no audio. The caller gets a full volume echo with about 1/2 second latency. At first I thought it might be related to using the aggressive

Hi, On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading this: # asterisk -rx "dahdi show status" Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO HA8-0000 RED 109 0 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8-0000 UNCONFI 109 0

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

I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

How are you modeling weak aliases in Atoms? mach-o does not support weak aliases. My mental model of a weak alias is: If foo is a weak alias for bar, then if nothing else defines bar, use foo in place of bar. -Nick On Jan 8, 2013, at 4:50 PM, Michael Spencer wrote: > So I just got lua to link and run and work on x86-64 Linux with musl > and lld. It did require one change to hack

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

So I just got lua to link and run and work on x86-64 Linux with musl and lld. It did require one change to hack around incorrect handling of ELF weak aliases. In musl __stdio_exit.c <http://git.musl-libc.org/cgit/musl/tree/src/stdio/__stdio_exit.c> we have: static FILE *const dummy_file = 0; weak_alias(dummy_file, __stdin_used); weak_alias(dummy_file, __stdout_used); weak_alias(dummy_file,

Hi Trying to get WinXP HVM to install as domU on AMD. Got latest Xen Kernel and Xen package (NOT unstable or debug). When start, a window pops up for few seconds.. then it dies and another one comes up... and it disappears too. (Xen is trying to restart and giving up because the restart happens in 2 seconds..) It seems DomU is not able to read/understand something from Cirrus VGABIOS.

Hi Trying to get WinXP HVM to install as domU on AMD. Got latest Xen Kernel and Xen package (NOT unstable or debug). When start, a window pops up for few seconds.. then it dies and another one comes up... and it disappears too. (Xen is trying to restart and giving up because the restart happens in 2 seconds..) It seems DomU is not able to read/understand something from Cirrus VGABIOS.

Game - Tank Combat - city entertainment Set dll d3dx9_36.dll - still get the following error, as follows.. > err:module:find_forwarded_export function not found for forward 'd3dx8.D3DXGetImageInfoFromFileInMemory' used by L"C:\\windows\\system32\\d3dx9_36.dll". If you are using builtin L"d3dx9_36.dll", try using the native one instead. >

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

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

I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues