similar to: Duplicate incoming channel into two outgoing channels

Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus

Hi, I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls via an AGI-Script. When parsing the AGI-Variables I can see one that look like that: [agi_channel] => DAHDI/i3/211123456-89c What hat do the values mean in detail, please? DAHDI : this is clear i3 : does it mean, that the call comes in via E1-Port 3? 211123456 : Incoming-Call Caller-ID -89c : ? WANPIPE Release:

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus

    I am having a strange problem.  We have an Asterisk 16.12.0 server (we have upgraded at least two versions since we found the problem) where users complain that confbridge calls end after about 30 minutes or so.  The problem is that according to Asterisk the calls are still active.  All users are cut off at the same time but a "core show channels verbose" still shows channels as

Le 2017-07-05 18:51, Joshua Colp a ?crit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate without success. Tried redirecting the channels to a context, playing audio to the agent's channel and then bridging the 2 channels. The problem with this is as soon as the bridge is created the audio stops. I can provide the dialplan details, if anyone is

Hi Dale, I am in fact doing all the items you suggest. here is a log. For normal commands I am logging off just fine. Its just the heartbeat command I am getting an error on when logging out. Thoughts? Jerry ------ asterisk_command() Action: Login asterisk_command() Username: XXXXX asterisk_command() Secret: XXXXX asterisk_command() Events: off asterisk_execute() event_list=0 ret=36

One thing that concerns me is that this post is from 2009, even though the newest version of the app on Sourceforge is up to date. I have a customer who has been using a conference server that I built for him using app_konference for several years now and he routinely runs conferences with anywhere from 10 ? 125 active users. The ultimate goal is several hundred concurrent users and I can see that

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

On Mon, Feb 24, 2020 at 8:07 PM Dan Cropp <dan at amtelco.com> wrote: > We are looking to migrate from AMI to ARI. > > > > We currently rely heavily on ConfBridges for multiple party support. > > Is it possible to add more than 2 channels? > > If so, is there a limit? > > Or a way to configure the limit? > > Yes, you can add more than 2. The bridge

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

I had a similiar problem, but with calls dropping after 30 sec. It turned out that Android didn't support RP-CID (Reverse Party Caller ID) so when I sent the name of the callee to the caller (as some sort of "centralized phonebook function") it caused calls to be dropped as android refused to reply on the packets or sent rejections back. Check if you have some equipment on the line

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE

I seem to have managed to end up with a pool that is confused abut its children disks. The pool is faulted with corrupt metadata: pool: d state: FAULTED status: The pool metadata is corrupted and the pool cannot be opened. action: Destroy and re-create the pool from a backup source. see: http://illumos.org/msg/ZFS-8000-72 scan: none requested config: NAME STATE

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,