similar to: chan_alsa and confbridge

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230908/dee530c8/attachment.html>

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav > Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit > Little Endian, Rate: 48000 Hz, mono > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >

Hello, I can't get the sound over alsa to work with Asterisk. My current version is 1.4.21.2~dfsg-3 running on debian stable. All settings are the default ones with exception of: /etc/asterisk/modules.conf: load => chan_alsa.so noload => chan_oss.so /etc/asterisk/alsa.conf: input_device=default output_device=default asterisk is started up and doesn't complain about alsa in

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? -- Chris

Hi some problem with chan_alsa. Depending on the configuration I don't get any sound output (output_device not set in alsa.conf - same as output_device=default) or very strange output (output_device=hw:0,0) when dialing into something like exten => 10,1,Answer exten => 10,n,Playback(soundfile) exten => 10,n,Hangup Other alsa applictions do work without problems and for example this

We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-00000003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does

Hi, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

I'm trying to migrate from MeetMe to ConfBridge: [conferences] exten=>_8[1-9],1,Answer() ;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234) exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms) exten=>_8[1-9],n,Hangup And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81 exten => 81,hint,ConfBridge:81 ;;exten => 82,hint,MeetMe:82 exten => 82,hint,ConfBridge:82 ;;exten

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ?

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show profile bridge default_bridge" I see: Video Mode: no video I can change it

I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme and I'd like to switch to confbridge to service more callers. Can anyone reply with their experience along the lines of 'using meetme I was only getting x callers per server but with confbridge I now get y callers per server?' -- Thanks in advance,

I have a very old server that is used only for conferences on Meetme. To manage the conference rooms we use Web Meetme. Now it is time to upgrade everything but since Meetme is no longer available I need to find a replacement GUI to manage the conference rooms. Anyone know a solution that works with Confbridge? I found "Asterisk Confbridge Manager" from a russian company but it

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

We have a system where two calls are in a ConfBridge with recording. This is Asterisk 16.3.0 Channel A seems to work perfectly. Wireshark is showing the RTP to/from working fine and having no jitter/lag issues. This call hears everything from channel B. Channel B we have more issues capturing a wireshark trace because their channel can be in the system for hours. When the two calls are in the

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference