similar to: Reducing number of Asterisk processes?

Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with "asterisk -rvvv". I need it in debugging purpose for tracking some bug. Thanks Enrico. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b.

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

Hi, I have just installed the fresh svn version of asterisk and when I run it I get the following errors: [Mar 4 14:19:27] WARNING[24527]: loader.c:728 load_modules: No 'modules.conf' found, no modules will be loaded. [Mar 4 14:19:27] NOTICE[24527]: manager.c:2681 init_manager: Unable to open management configuration manager.conf. Call management disabled. [Mar 4 14:19:27] NOTICE[24527]:

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

Hello, I am trying to get a repeatable build setup for asterisk. Part of doing so involves using the -C option to specify the master config file. The problem is that asterisk reads the config file location that i specify, however it still tries to read two other config files, namely: * /etc/asterisk/extconfig.conf, and * /etc/asterisk/logger.conf I have specified in my config file that the

Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' -

Hi All, I have ported OpenSSH on an embedded uClinux platform. My main concern is that sshd takes 1 MB if RAM while its in running state and for each incoming connection it eats up 1 mb ram again. I am looking for help in reducing RAM requirement for ssh daemon. Any comments ? Thanks in advance, Reema.

Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting

Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110810/365d9d56/attachment.htm>

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Hello I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to call 3. Asterisk puts me on hold through Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the

On 13.9.0 https://github.com/traud/asterisk-opus The README: Alternatively, you can use the Makefile of this repository to create just the shared libraries of the modules. That way, you do not have to (re-) make your whole Asterisk. The Makefile generates: codecs/codec_opus_open_source.so formats/format_ogg_opus_open_source.so formats/format_vp8.so res/res_format_attr_opus.so Without any of

Hi Jean, First, I would like to thank for the prompt reply. Basically, what I have to do is to run a speex coder/decoder on a Blackfin processor, on top of the microC/OS II kernel. I wish I could use uclinux, since I'm familiar with Linux, but this doesn't depend on me. You said that the assembly code is written for gcc, not for VisualDSP++. Furthermore, in the link you sent me, the

Dear all, I'm Arun, new in this group. and newbie in Linux also. I am working on linphone and I want to compile linphone, but before that I want to compile speex-1.1.10 for uClinux using bfin-gcc. I used these three files : uClinux-dist-2005R3.tar.bz2 bfin-gcc-3.4-2005R3.1.tar.gz speex-1.1.10.tar I sucessfully compiled and installed bfin-gcc, and

Hi Jean-Marc I'm trying to integrate your speex codec on our custom Blackfin board. The board is not uCLinux compatible and there is no chance that it will ever be. I am using ADI-supplied VisualDSP++ IDE and corresponding toolchain. As long as I am compiling "C"-only version of the library everything is fine. VisualDSP++ produces working library. There is only one not so minor

Dear, Faxter is an opensource email to fax gateway, please check it, let me know if any bug. best -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110130/0f418a92/attachment.htm>

Dear this note is only for fresh administrators don't think about asterisk security. I found fail2ban very useful for anti asterisk hacking, so I want to share it with fresh admins. some hackers try your sip or iax2 ip with a lot of username/password, may be after 1 million try, one username/password was accepted. so in 2-3 hours, they use all of the credit of the hacked user. fail2ban, runs

Dear,is any way to find silent channels , and disconnect them after 30 secs? best -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090824/7709c910/attachment.htm

Dear, do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ? is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110525/9df2050a/attachment.htm>