similar to: ConfBridge / internal_sample_rate=auto / warning

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 All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single video source of the conf. The video mode stays follow_talker. I

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

Hi Raid ! This patch fixes some issues with llvm-top: a) it always makes a debug build of llvm, even when OPTIMIZED="1" is in .options b) it now handles ASSERTIONS, CHECKING and DOXYGEN from .options, it handles DEBUG a bit (not much) better c) it adds support for THREAD After this patch there still is support missing for DEBUG. PROFILING, OPT_FOR_SIZE, STRIPPED missing. I

Hi, On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START and BRIDGE_END events zone-s*CLI> module reload cel The module 'cel' reported a reload failure     -- Reloading module 'cel' (CEL Engine) [2020-07-10 17:57:01] ERROR[16163]: cel.c:428 ast_cel_str_to_event_type: Unknown event name 'BRIDGE_START' [2020-07-10 17:57:01] ERROR[16163]:

On Wed, Jul 22, 2020 at 12:44 PM Administrator <admin at tootai.net> wrote: > No one on this ? > > Le 10/07/2020 à 18:06, Administrator a écrit : > > Hi, > > > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > > and BRIDGE_END events > > > > zone-s*CLI> module reload cel > > The module 'cel' reported a

No one on this ? Le 10/07/2020 à 18:06, Administrator a écrit : > Hi, > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > and BRIDGE_END events > > zone-s*CLI> module reload cel > The module 'cel' reported a reload failure >     -- Reloading module 'cel' (CEL Engine) > [2020-07-10 17:57:01] ERROR[16163]: cel.c:428 >

Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a "Dial(DAHDI/g1/voicenumber,r)". The call is bridged and everything is fine. "dahdi show

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! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option suitable for category '3567' named 'inband_progress' at line 867 of [May 12 15:33:04] ERROR[21137]: res_sorcery_config.c:317 sorcery_config_internal_load: Could not create an object of type

Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI> dahdi <tab tab> No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI> root at campbx1:/etc/wanpipe# wanrouter hwprobe ------------------------------- | Wanpipe Hardware

We have been distributing asterisk servers for several years now, and early on decided that hardware echo can was the way to go. Our first few boxes without it had horrid echo problems, and attempts at tuning in 2006 didn't make any difference. We installed a new server yesterday at a client's location with a Rhino 4 port FXO card (HW EC included), and when an inbound call was

On 15-10-04 09:54 AM, Bryant Zimmerman wrote: > I have a pjsip install that is not pulling it's realtime registrations > from an ODBC database. > When I have them directly pull from a MySQL database everything seems to > work. > I am having trouble finding a requirements document for the pjsip > realtime registrations. > Is there some kind of special config that has to be

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hey Guys! We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make

Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts

the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 & 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (<10%) success rates. we're currently using asterisk

Hi all, I was using dahdi 1.6.2.0.9 version for a long time. We decided to upgrade to 1.6.2.22 a few days ago. After that we started to have some problems with dahdi channels. PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2 We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for outside calls. At begining everything works fine but in a few hours, calls from asterisk to ericsson