similar to: SOLVED: 100% cpu usage causes big problems

I have been using Xen on a daily basis on a production (but not critical) machine for a number of months now. It''s looking really good. One thing that I have not yet seen anyone mention as a feature that I would really like to see is the ability to overcommit memory. I have 2G of RAM in my machine. I would like to give a developer his own virtual domain to sandbox his application

When I try to flash my 7960 with SIP I get messages like this in the tftp server logfile: Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin and the phone says something similar on the display for a brief moment and puts a funny char where the space in the filename above is. Seems like around 1 in 4 of the 7960's I have flashed with SIP have this problem. Anyone

This is a call coming in through the ISDN to 7940's. Answering with non-codec capability 1 - Is that the problem? SIP Debugging Enabled We're at 10.1.0.11 port 18406 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 <<<<<<------------- 12

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

We are in urgent need of some help getting Asterisk to gateway between an incoming H323 connection and SIP to a Lucent TNT. We have the incoming H323 already set up and the SIP going to the TNT but the media stream is getting lost somewhere as no audio is heard. We are willing to pay $$$ for an extra set of eyes to get this resolved fast. It's probably something quick and easy and we are just

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

I am so far unable to get the busy lamps on a Snom 220 to work either with current cvs or asterisk 1.0. I am using the hint extension and the Snom 220 just as described in the "mini-howto" on: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html There are also a couple of wiki pages referencing this: http://www.voip-info.org/wiki-Asterisk+standard+extensions

Compiled Asterisk from FreeBSD port (0.9.0_2) When I start asterisk it uses 100% cpu. Searches on Google say to comment the noload => chan_oss.so in modules.conf But this is already commented. Make.conf contains some optimizations. modules.conf: ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; If you want, load the GTK console right away. ;

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 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\"

== 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

Some of you may recall that I have been working on building a box to convert H323 to SIP. After a significant amount of outside help and slicing and dicing of the ohh323 code to get it to compile on AMD64 we finally got it working. Now we are working on improving the performance. This box takes H323 from one device and converts to SIP and spits it back out to another device. The codec is g729 but

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing

Hello, I need to send a tiff via fax with my asterisk 1.6.1.0. I tried in the dialplan [default] exten => _X.,1,SendFax(/root/test.tiff) but I have: salledeconf1*CLI> console dial 111 at default [Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [111 at default:1]

I did a usual "svn update", "./configure" and "make" and got [CC] chan_oss.c -> chan_oss.o gcc: @SDL_INCLUDE@: No such file or directory I don't see any changes to chan_oss recently, so don't understand this. What could be going on?

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

Hello Im trying to install an old version of Asterisk. But it isnt working: when I run "make install": gcc -o gentone gentone.c -lm ./gentone busy 480 620 Wavelength 1 (in samples): 16.66667 Minimum samples (1): 50 (3.000000.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.000000.3 wavelengths) Need 400 samples Wrote busy.h ./gentone ringtone 440 480

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

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

Hi, am building the latest version of Asterisk (1.4.13) on a self-build Linux host (based on LFS-6.3). Version 1.4.12 built installed and worked fine. Last night I upgraded the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1 against it. That seemed to build and install O.K. too. I dl'd the 1.4.13 source tarball and tried to build that: ./configure ran O.k. 'make