similar to: Function DENOISE not registered

On 5/26/23 01:15, Fourhundred Thecat wrote: > how do I fix this? > What do I have to do to "register" denoise ? confbridge.conf states: "Requires func_speex to be built and installed." I am guessing you have not fulfilled that requirement. Doug -------------- next part -------------- An HTML attachment was scrubbed... URL:

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

To all: No matter what I try, I cannot get the system to wait for the admin to join. It just dumps users into the bridge directly. I do not have a pin for users, does that matter? What am I missing? Another issue the absolute timeout is not working ? ... have recordings that last for over 24 hours... and this should not happen... All calls should hangup after 4 ? Any ideas ? Any help is much

Hi, I'm trying to use AGC in combination with Asterisk 1.8 and an odd telephone which is very loud when used with a headset and more quiet when used "normal". Regarding to the documentation, AGC should be available since * 1.6 - but every time I want to set it, the CLI tells me: -- Executing [0160xxxxxxx at intern:2] Set("SIP/intern-xxx-000000d2",

In article <94191802-6c9c-bdab-615b-001786a2a0ca at gmx.ch>, Fourhundred Thecat <400thecat at gmx.ch> wrote: > > On 2019-11-16 03:29, Fourhundred Thecat wrote: > > Hello, > > > > I am logging directly into file and also to syslog. > > Here is snippet from my /etc/asterisk/logger.conf: > > > > messages => notice,warning,error,verbose

I'm seeing Asterisk crashes with the following frame at func_speex.c:188: (gdb) p *frame $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640, mallocd = 1, mallocd_hdr_len = 232, offset = 64, src = 0x2ac07413e7f8 "siren14tolin32", data = {ptr = 0x3cab9378, uint32 = 1017877368, pad =

I am having a problem with "record_file" working properly depending on when it is called -- basically if it is called immediately upon a call, it acts like it does not hear anything from the callers phone (yes, my phone is setup properly and functions fine otherwise)... if I do a "background" or "festival" command before calling it, it works fine. Details below:

> On 2020-06-06 10:38, Antony Stone wrote: > On Saturday 06 June 2020 at 09:18:11, Fourhundred Thecat wrote: > >> In a situation when I start asterisk, and the remote database is >> unreachable, asterisk waits for several minutes before it actually >> starts (before it loads sip module, etc). >> >> And when database is unreachable during operation, when call

Hello, I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-00000013] pbx.c: NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE! vs. VERBOSE[7609][C-00000013]: pbx.c:2925 in

Hello, my logs are flooded with: WARNING: The 'stasis/m:cdr:aggregator-00000005' task processor queue reached 5000 scheduled tasks again. and then, when call came, I got this: ERROR: Oh dear... we couldn't allocate a port for RTP instance '0x6e1e680fd670' WARNING: Unable to cancel schedule ID 0. This is probably a bug (res_rtp_asterisk.c: dtls_srtp_stop_timeout_timer,

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps

Hello, I am copying /mnt/foo to /mnt/bar/ rsync --info=name1,del2 -rl /mnt/foo /mnt/bar/ /mnt/foo contains deep directory structure, ie: /mnt/foo/aaa/ /mnt/foo/aaa/somestuff/ /mnt/foo/aaa/somestuff/file1 /mnt/foo/bbb/ /mnt/foo/bbb/someotherstuff/ /mnt/foo/bbb/someotherstuff/file2 I am not interested in details which individual files were copied, just the main directory.

Hi. There is denoise for preprocess during encoding. The nature of my source is unpredictable and sometimes the result is better off if I dont have denoise ON when I save my encoded speex file. So, I would like to implment a real time denoise during decoding/playback time, (instead of having the denoise result saved into speex-encoded file). Is there such denoise-preprocess function that runs

> On 2020-05-31 15:59, Antony Stone wrote: > On Sunday 31 May 2020 at 15:44:46, Fourhundred Thecat wrote: > > "%Cn[;n] - Change terminal foreground (and optional background) color to > specified A full list of colors may be found in include/asterisk/term.h" > > So, try: > > export ASTERISK_PROMPT="%C31[%H]: " > > (I got 31 from reading the

Hello, how can I change the color of the asterisk prompt to red ? I read in the wiki that I can use %Cn[;n] https://wiki.asterisk.org/wiki/display/AST/Asterisk+CLI+Configuration But what does this mean ? There is no example how to actually use it. where do I put it? What syntax is that anyway? How do I specify red ? I currently have this in my environment: export ASTERISK_PROMPT="[%H]:

Hi All I am using speex in one of my WinCE project. I have been trying to use speex to perform denoise on the captured audio packet and echo cancellation. Following behavior I have observed while using various options. I would really appreciate if you could help me with the issues I am facing. 1. Denoise: I have written the following to code to perform denoise. //To initialize speex

On 06/04/2010 07:37 PM, Jean-Marc Valin wrote: > On 10-06-04 05:16 AM, Gurinder Singh wrote: > >> I have been developing an audio application using Speex. To reduce the >> background noise in the captured audio I have enabled the denoise >> feature and set the noise suppression level to 60. >> > There you go, don't do that. There's a reason

Hi I am trying to use the Denoise option of speex but unable to do so successfully. I would really appreciate if some one could help me and identify what exactly wrong i am doing... I am using below code to perform denoise. //Initialize speex preprocess state and set the denoise option int nEnable = 1; SpxPreprocessState = speex_preprocess_state_init(160, 8000);

Hi I have been developing an audio application using Speex. To reduce the background noise in the captured audio I have enabled the denoise feature and set the noise suppression level to 60. Although the constant background noise is reduced but using denoise introduces a weird effect in the audio which can be described as 'Drain Pipe' effect. Has anyone faced a simiar issue with the