search for: internal_sample_rate

...ere, I am running Asterisk 11.9.0 WANPIPE Release: 7.0.10 DAHDI Version: 2.9.0 Echo Canceller: HWEC libpri version: 1.4.12 When I start the ConfBridge application I get the following warning: [2014-10-24 14:36:21] WARNING[29177][C-00006934]: config_options.c:790 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) I do not specify a specific user- or bridge-profile so the default profiles are used in confbridge.conf (and there the profiles are empty). But before calling the ConfBridge-App I set the channel var "internal_sample_rate=auto". Am I making...

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

...e, he does not become the single video source of the conf. The video mode stays follow_talker. I have tried changing conf mode dynamically in dialplan but does not work following is my settings and dialplan: confbridge.conf ---------------------- [common_bridge] type=bridge record_conference=yes internal_sample_rate=auto mixing_interval=20 video_mode = none conference participants. release_as_single_video_src [default_user] type=user dsp_talking_threshold=128 dsp_silence_threshold=2000 talk_detection_events=yes extensions.conf ----------------------- exten => 200,1,Noop(Going to ConfBridge now) same =&gt...

...or the first couple seconds channel B talks. After that, everything seems to be perfectly fine. For each channel added to the ConfBridge, the user profile has... jitterbuffer = yes denoise = no dsp_drop_silence = yes dsp_silence_threshold = 2500 dsp_talking_threshold = 160 On the bridge profile. internal_sample_rate = 0 mixing_interval = 20 jitterbuffer is not being set. According to the wiki, this defaults to no binaural_active is not being set. According to the wiki, this defaults to no One other possible coincidence in the samples I have received, channel B seems to always start talking roughly 2500 ms...

...used by Chrome's video sent to asterisk being some format which asterisk can't use in the confbridge? The endpoints for both Firefox and Chrome calls are set to disallow = all allow = opus,ulaw,vp9,vp8,h264 Both calls use the same bridge and user settings. [3] type = bridge language = en internal_sample_rate = 0 mixing_interval = 20 record_file_append = no max_members = 10 video_mode = none [4] type = user admin = no marked = no startmuted = no music_on_hold_when_empty = no quiet = yes wait_marked = no end_marked = no dsp_drop_silence = yes dsp_silence_threshold = 2500 dsp_talking_threshold = 160 deno...