similar to: Crappy sound on Console (chan_oss)

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Hi, i'm new in asterisk programming. Maybe my question was posted thousand times but i found nothing using google. I'm looking for a method to limit the total simultaneous calls (inbound and outbound) that pass from internal phones to 2 SIP providers. I found the calllimit option but it works only on a per-channel basis. Instead i want limit the total amount of calls, abstracting from

Hi all, i'm trying from several days to setup a SLA on my machine with some THOMSON 2030. My goal is to bind every F key to an extension (NOT a trunk). So, F1 = 201, F2 = 202, F3 = 203, and so on... I'm googled thousand of pages and many more confusing concepts are in my mind. My server uses extensions with numbering 2XX placed in context 'phones'. I set yet in sip.conf:

-------------------------------------------------------------------------------------------------------- dev*CLI> zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 dev*CLI> zap show channels Chan Extension Context Language MOH Interpret pseudo

Hi list! Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian 9). Well, I copied the configuration and changed what needed, so basically, it works, at least with my tests. But when Asterisk will be started, in the message log I get this error: [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel

Zitat von Tzafrir Cohen <tzafrir.cohen at xorcom.com>: Hi, > Off-topic: any reason you don't use chan_alsa? This was the "Armbian installation", I didn't configured it extra... > Are you sure you quote the error message right? Copy+Paste... ;) But I searched a little bit and I really don't think, I need this module... As I undestand, I just need it, if I

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I've got a boom box style radio plugged in with a patch cable (Our radio shack only had a mono cable). I'm getting a volume off of the stream, but it is a very low volume. I've got the radio turned all the way up. I'm using setmixer to adjust volumes, but it hasn't changed anything. I know it's probably just a simple setting, I just don't know what. Marion

That was part of it, the other thing I believe was the mono cable, I put a stereo cable on it and now it's working great. Your howto has been a life savior in getting this set up. Thanks, Marion Hall -----Original Message----- From: Kerry Cox [mailto:kerryjcox@gmail.com] Sent: Friday, October 01, 2004 2:30 PM To: hallm@1satcom.com Subject: Re: [Icecast] Low volume coming through live

Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace

Trying all the stupid methods to solve the issue of Win2k hosts from different subnets that can't be seen in the NetworkNeighbourhood when server is Samba on Linux (including reading the F***ing manuals and not finding anything useful or browsing through the source of samba until my eyes fall in my mouth) I have decided that they were, well... ,stupid and chose to try crappy solutions! One of

While burning in a new desktop Epox KT-880 workstation, I'm running on my old K6-III at 360MHz. Before new X updates (put everything in today, kernel, elfutils, ntpd - Selinux issues? - xorg, ...) with my PCI ATI 7000 (Radeon) on a 19" Samsung 950b @ 1024x768 (one of 5 different resolutions I commonly use), I could scroll smoothly at reasonable rates. With the new stuff I get severe

Hi, I am trying to setup an overhead paging system with asterisk. I have followed some of the advice from the list and have oss.conf set for autoanswer. The sound card and speakers work because they can play mp3s just fine. When I call the extension, the asterisk console looks like everything is working, but I get no sound. Here is what I get on the console: -- Executing

hi i'm a newbie in asterisk...i installed asterisk but when i tried to dial 1000 for the first time i got the following error messages and i don't hear anything... May 29 20:46:03 WARNING[262160]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: Device or resource busy May 29 20:46:03 WARNING[262160]: chan_oss.c:572 oss_write: Unable to set device to input mode May 29

How do I disable chan_skinny and chan_oss? I think chan_skinny is associated with Cisco hardware, since I don't have any I don't need this channel. I just want to get rid of those warning messages at start up. -- #Joseph

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

>> 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 >

Hi Does anyone used asterisk as a operator phone,with multiple lines and features like transfer forward and etc.I used chan_alsa driver to make asterisk as SIP Phone,but it has limitation,we cant make or receive multiple calls,and will not able to do any features like transfer forward etc. Is any other application available in asterisk to do this . Thanks Nikhil

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20