similar to: conference room possible bug

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

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

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

any idea on how we can setup threeway calling in * thanks moe smadi

hi; does any body know what are the physical dimension of a digium care 400pm for example? thanks m.smadi

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]

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

hi; Does asterisk handle the "refer method"? If we use asterisk as both a sip proxy and a GW what would happen to POTS leg (SIP --> * --> POTS) after that send client send a "refer" request which refers the POTS destination to let's say another SIP phone? thanks m. smadi

i want the to find out the delay between two events: 1) the instance a call is recieved on an FXO port and the 2) the instance a SIP INVITE is sent to the SIP destination. i need to attach timestamps to the events before logging them. How can i: 1) log `ALL' events. 2) Attach timestamps to them? thanks m.smadi

https://bugzilla.samba.org/show_bug.cgi?id=4132 Summary: Does not delete partial files upon completion of transfer Product: rsync Version: 2.6.8 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

If i leave my asterisk server running for a long time then try to dial outbound on the zaptel channel i get this high pitch static noise and won't dial out. This behavior is happening over two different servers i am using. Rebooting asterisk does not sovle the problem. I rmmod the zaptel driver then reload and that solved the problem. But i cannot continue to do that. Also sip to sip

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change

Hello, I'm trying to setup an Asterix PBX solution in our office. We plan to have 5 active lines open available at any point in time. We'd like to use VoIP Phones, and possibly Software Based phone (*NIX/Windows enviroment). I was researching the various cards and I think I'd want to go with the Digium TDM40B - 4-port. However, I can't figure the differences between FXS &

Hello, I have an Asterisk with one local Cisco ATA and one remote Cisco ATA connected to the Asterisk, the remore connection is a satellite link with an 900ms delay. I can make calls from the remote site to the local site, but when try to call from local to remote it doesn't work. The Asterisk timesout, it sais no one answered and can?t establish the connection. Can anybody help me with

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing

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

Hello All, I have installed Asterisk 1.6 with openVPN in the same machine. I have set up a VPN connection between 2 SIP clients and Asterisk using x-lite. The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn tunnel. When attempting to make a call between the clients, the siganling part of the call goes well. But, when the call is set up, some RTP packets are exchanged at

How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan

Hi all, I didn't find yet any info about this. Is there any way for a Conference Room Owner to change his own password? A kind of Menu like calling his conference room: example:8200 And an IVR option to change password. That seems to me interesting, because i may not want the same users entering two diferent days on my conference room... Also I don't think it is a good choice to