similar to: alsa 1.016 compile error on latest kernel centos 5.1

Subject says it all. How can I get the 1.3 version and 0.9.28 to compile on CentOS 5.8 ??? When I compile the two as modules I get errors. My Makefile is: obj-m += 8139cp.o 8139too.o all: make -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules clean: make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean The errors I get are: Entering directory

CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o LD [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o

anyone compiled alsa 1.0.11 on 4.3? I get errors compileing alsa-drivers. Thanks for any suggestions if you have done it. Anyone can try it download from http://alsa-project.org and doing the extract, ./configure and make. With out the make install nothing should be affected on your system if you dont want to use the new version. Thanks, jerry

Any Virtualbox on CentOS guru's out there? I only have 1 VM on my linux host. in the update process to virtualbox 6.1.16 my config was lost. When I run Virtualbox my VM does not show. So I desire to add it back. My structure is this. ls -l /home/silentm/vmware/CentOS\ 64-bit\ LSI/ -rw------- 1 silentm silentm 8684 Aug 13 08:31 CentOS 64-bit LSI.nvram -rw-r--r-- 1 silentm silentm 0 Feb 27

I am trying to use cut to tell me the suffix of a file. for example: echo /home/silentm/log/file.machine.log | cut -d . -f 1- I was expecting to get .log or log but I get the entire string echoed back. doing the opposite gave me what I expected: echo /home/silentm/log/file.machine.log | cut -d . -f 1 gives me /home/silentm/log/file I am trying to find a way to test if the file ends in .log?

I have read the info on apache 2.4... I added to the bottom of httpd.conf these lines (and restarted httpd) ------------------------- <Directory /var/www/html> # old 2.2 config # Order Allow,Deny # Allow from all AllowOverride None Require all granted </Directory> <Directory /home/silentm/public_html/gifs> AllowOverride None Require all granted </Directory>

I am getting this: make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pritest.o -MF .pritest.o.d -MP -c -o pritest.o pritest.c pritest.c:50:28: fatal error: dahdi/tonezone.h: No such file or directory #include <dahdi/tonezone.h> I am using dahdi-linux-complete-2.11.0+2.11.0 I did "find . | grep timezone.h" from the dahdi root and its not

I am running asterisk 1.4.18 trying to use MeetMe and option b. I am getting permissions denied failed to execute conf-background.agi on the CLI lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> /home/silentm/bin/conf-background.agi my conf background is a symbolic link - then my permissions are : [root at devcentos5x64 src]# ls -l /home/silentm/bin/conf-background.agi

I am trying to get my kickstart "repo" line to function correctly. repo --name=Updates --baseurl=http://192.168.1.14/centos/6.2/updates/x86_64/ When I comment the above line my install works, When I uncomment it the install fails with a message about dbus package error. This is my nightly script that runs: =============== PLACE="mirror.team-cymru.org/CentOS" mkdir -p

I have a USB to serial converter attached to my box. pl2303: Prolific PL2303 USB to serial adaptor driver if I login to the box and send/receive serial commands over this unit it works without error EVERY time. however, if I run the same command set from with-in the extensions.conf with System() I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - failed submitting read urb, error

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis <jerry.geis at gmail.com> wrote: > >An issue[1] was already created by asterisk at phreaknet.org and they > also put > >a fix up for review and inclusion[2]. > > >[1] https://github.com/asterisk/asterisk/issues/308 > >[2] https://github.com/asterisk/asterisk/pull/309 > > > The change "seems" to be

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi! I have a Asterisk installation to manage my phones at home (provider is Deutsche Telekom). It works, but very often the voice is "broken"... Yesterday during a call it was very difficult to understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

>An issue[1] was already created by asterisk at phreaknet.org and they also put >a fix up for review and inclusion[2]. >[1] https://github.com/asterisk/asterisk/issues/308 >[2] https://github.com/asterisk/asterisk/pull/309 The change "seems" to be working. Will test more tomorrow - had to leave. THANKS! Jerry -------------- next part -------------- An HTML attachment was

Am 13.06.2020 um 18:06 schrieb Michael Keuter: > So the call used Alaw as Codec. Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'm really puzzled... Luca Bertoncello (lucabert at lucabert.de)

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>