similar to: asterisk freeze due to "too many open file" error

hello, im having trouble with asterisk with medium load, it seems im running out of files, here is a chunk of the logs with grep "\(file\|pipe\)": Apr 18 15:40:46 WARNING[11644] res_agi.c: Unable to create toast pipe: Too many open files Apr 18 15:40:46 WARNING[11574] channel.c: Channel allocation failed: Can't create alert pipe! Apr 18 15:40:46 WARNING[11574] channel.c: Channel

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

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 For whatever reason we've getting 2 or 3 CDR lines logged for each call, often in different formats: as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

On 4/20/2023 7:16 AM, Rhys Hanrahan wrote: > > Hi All, > > I’m having issues figuring out how to set no DTMF timeout on the AGI > GET DATA command as “0” still has a multi-second timeout if no input > is given. I am trying to find the source code of the AGI command to > figure it out, but I can’t for the life of me find the underlying > source code. The closest I’ve

Hello everyone, I have a problem with a FastAGI connection, could you help me fix this problem please? Here is my log: [2015-12-15 16:17:09] WARNING[23936][C-00000015]: res_agi.c:1658 handle_connection: Connecting to '10.171.54.149:9110' failed for url

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

The AGI debug command worked well, and I found the offending command: SetCallerPres(allowed) That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -----Original

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

Hi, I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've noticed that the CDR logging in MySQL (on a different computer) has stopped. I thought it wasn't logging anything at all, but I realized after a bit of searching that there were log files in /var/log/asterisk/cdr_custom and /var/log/asterisk/cdr_csv with up to date logs. My cdr_mysql.conf is set up

Yes I did. Fair question. I think it`s working, but is there anyway to know for sure? Show modules show app_cdr.so as existing... Mike On Thursday 26 Jan 2006 16:50, Micha?l Gaudette wrote: > Hi, > > I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've > noticed that the CDR logging in MySQL (on a different computer) has > stopped. I thought it

Hi i am upgrading an asterisk server from 1.2.4 to 1.4. i've installed libpri 1.4 i've installed zaptel 1.4 I've installed the new version of misdn with the script of beronet. i use this configure script: ./configure --with-postgres=/usr/local/pgsql then: make menuselect [*] 1. cdr_csv [*] 2. cdr_custom [*] 3. cdr_manager XXX 4. cdr_odbc [*] 5. cdr_pgsql XXX 6. cdr_radius

Hello, On a production system, I'm seeing this: [Feb 13 16:47:00] WARNING[14742] res_agi.c: Unable to create toast pipe: Too many open files [Feb 13 16:47:00] WARNING[9283] acl.c: Cannot create socket [Feb 13 16:47:00] WARNING[9283] rtp.c: Unable to allocate RTCP socket: Too many open files [Feb 13 16:47:00] WARNING[14732] acl.c: Cannot create socket [Feb 13 16:47:00] WARNING[14732]

Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

Thank you for your time. I've put my replies to your questions in-line, below. On Wednesday, June 07, 2017 10:19:41 AM Antony Stone wrote: > On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > > > Hi all, > > > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > > discovered that my server crashes as soon as I leave a