similar to: [modules.conf] Modules still loaded after "noload"

It's already mounted rw somewhere else. This enables the needs_recovery feature which prevents an ext2 mount. -Leslie -----Original Message----- From: Christopher J. PeBenito [mailto:pebenito@ieee.org] Sent: Tuesday, April 22, 2003 11:30 AM To: kuczynski, leslie Cc: 'ext3-users@redhat.com' Subject: Re: ext3 "noload" option to mount returning error in 2.4.9 & 2.4.18 s

I have an ext3 filesystem that I want to mount without loading the journal. I tried the "noload" option with both 2.4.9 and 2.4.18 series kernels and get the errors listed below. [root@host]# mount -t ext3 -o noload /dev/sdf1 /mnt mount: wrong fs type, bad option, bad superblock on /dev/sdf1, or too many mounted file systems /var/log/messages: Apr 18 13:30:23 host kernel: ext3:

I have an ext3 filesystem that I want to mount without loading the journal. I tried the "noload" option with both 2.4.9 and 2.4.18 series kernels and get the errors listed below. [root@host]# mount -t ext3 -o noload /dev/sdf1 /mnt mount: wrong fs type, bad option, bad superblock on /dev/sdf1, or too many mounted file systems /var/log/messages: Apr 18 13:30:23 host kernel: ext3:

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Hi all, I've been told that part of best practices is to noload all modules that you don't use. Adsi being one of them (are there actually people using adsi & asterisk?) I noloaded it in /etc/asterisk/modules.conf: noload => res_adsi.so noload => app_adsiprog.so When I start * (this is cvs head of today by the way) I get: [app_voicemail.so]Aug 31 17:25:49 WARNING[4961]:

Can someone please describe how to use the "noload" option and what the expected results are? -Leslie

Hello I have a couple of questions about using modules in Asterisk (1.4 or 1.6): 1. I'd like to experiment with extensions.lua: What happens if... - I leave extensions.conf enabled by not using "noload=pbx_config.so" in /etc/asterisk/modules.conf? Will the two dialplans get mixed together, with possibly unpredictable results? - I disable extension.conf by setting

Just working on Configing up Voicemail and now that I have got it working and configed and answering the way it should be I have another challange. on the * CLI> I get this -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg0000 format: wav49, 0x8133390 -- x=1, open writing:

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

Hello Using Asterisk 1.4.20 and Zaptel 1.4.3-1, I notice that Asterisk sends FXO calls to a context named "numberplan-local" that is not mentionned in my configuration file, which prevents incoming calls to be successfull: ======= /etc/asterisk/zapata.conf ================== [trunkgroups] [channels] ;Send incoming calls to this context in extensions.conf context=from_fxo

I'm trying to somewhat reduce the security risk of Asterisk, by loading less modules. In my installation I use SIP and IAX2 for incoming calls, and that's it. No voicemail, no call parking, it just plays back voice clips. I can remove /etc/asterisk/modules.conf modules one by one: ------------------------------------ [modules] autoload=yes noload => pbx_gtkconsole.so noload =>

Hello. I'm having problems with the dynamic library of libspeex under win32. I have used the static library for a while with no problems. When I try to compile my application with the dynamic library I get the following link error: codec_speex.obj : error LNK2001: unresolved external symbol _speex_uwb_mode codec_speex.obj : error LNK2001: unresolved external symbol _speex_wb_mode

Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Is that the only problem to use lld to link cygwin programs? On Thu, Feb 8, 2018 at 8:19 AM, Andrew Kelley <superjoe30 at gmail.com> wrote: > Here are the linker errors: > > lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: > __data_start__ > lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: > __data_end__ > lld: warning:

Here are the linker errors: lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __data_start__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __data_end__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __bss_start__ lld: warning: libcygwin.a(_cygwin_crt0_common.o): undefined symbol: __bss_end__ lld: warning:

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

i'm having this error message when trying to run the agents-feature Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application 'AgentsLogin' for extension (from-internal, 28, 1) while chan_agent.so is beeing loaded i still don't seem to have access to the commands like agentlogin or agentcallbacklogin. my agents.conf and queues.conf are configured correctly

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open