similar to: Too many open files

Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this?

Hi, my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14

I don't understand the reasoning here so could somebody please help me out? chan_h323 is causing a segmentation fault when trying to connect a call. I tracked the problem back to chan_h323.c in the oh323_new() function. the code is: tmp = ast_channel_alloc( 1 ); After this point, tmp->pvt is not allocated (null pointer). HOWEVER, in the ast_channel_alloc() function right before the

Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited)

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You will need to add the following line before running > asterisk. > > ulimit -n 32768 > >

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19

I've got a brand new Eicon Diva Server BRI card and I want to configure it with Asterisk. I managed to get asterisk and zaptel to compile and install, I've compiled and installed the drivers for the Diva card and now I need to compile and install the chan_driver for chan_capi. Unfortunately this fails miserably. I get the following messages: I'm using: Kernel 2.6.16.37.4,

After many many hours of frustration and failures I''m almost to the point that I don''t think this is even currently possible with Linux. With out going in to too much detail, I am effectively wanting to do the following. I want to be able to take traffic in from a local LAN on eth0 and route it out eth1 to a default gateway with a static IP. I want said default gateway

Hi, I'm developing a new channel driver for a device. I'm calling ast_channel_alloc from one of my functions, and as I step through the execution of ast_channel_alloc with gdb, everything is being initialized properly. However, once ast_channel_alloc returns, the pvt field of the data structure created is reset to null. I've verified that the structure created in the function and

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Thank you Tzafrir. I had been using different users in earlier attempts to make this work. Decided to try everything where root is the only user, simply to verify it's working. For problem 2, where asterisk is writing to the log but doesn't seem to receive the SIP packets even though tcpdump indicates they are making it to the box on 5060, I am starting asterisk while logged in as root.

Some background information. I have used Debian with Asterisk for several years. Have encountered zero problems. I am now trying to setup an Asterisk on a CentOS7 box using VMWare Workstation. I am brand new to CentOS and RHEL so I may be missing something obvious. I am installing CentOS Minimal Install and Asterisk from Source. (I have installed Asterisk from Source on Debian hundreds of

Hello! In my agi-debug i get the following error-message: AGI Rx << Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set normal priority AGI Tx >> 510 Invalid or unknown command AGI Rx << SET VARIABLE MODCLI 00434345452 the agi i call is a very simple shellscript that simply removes wrong charakters: #!/bin/bash modcli=`echo $1 | sed -e

Hi, I can't read a environment variable in a asterisk dialplan. When logged in as user root on the system an 'echo $HOSTNAME' gives the hostame of the machine. Asterisk (1.4) is started from the same console. I try to read it like this: exten => s,n,NoOp(host=${ENV(HOSTNAME)}) Does anyone know what i am missing? Ipv een saaie e-mail een leuk videobericht? Ga naar

Hi all I have been looking at my asterisk CDR in the mysql database and some channel names are set to "**Unknown**" string. When I look at the code, everybody when calling ast_channel_alloc set a channel format like SIP/%s or Zap/%s Only app_voicemail.c doesn't when sending emails and I don't use voicemail. Why app_voicemail needs to allocate a channel to send emails ? And in

Both updated from CVS within the same script but:- channel.c:44:2: #error "You need newer zaptel! Please cvs update zaptel" channel.c: In function `ast_channel_alloc': channel.c:298: error: `ZT_TIMERPONG' undeclared (first use in this function) channel.c:298: error: (Each undeclared identifier is reported only once channel.c:298: error: for each function it appears in.)

It seems like asterisk-addons in SVN has been broken for the last few weeks: gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 - DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/ chan_h323.lo src/chan_h323.c: In function 'ooh323_new': src/chan_h323.c:250: error: too few arguments