similar to: Is there a way to remove launching shell command from Asterisk CLI

Marcelo Terres wrote: > I noticed another different behaviour. > > In older versions, when I call rasterisk, I receive some informations > about it. Fox example: > > [root at pbx2 ~]# rasterisk > Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others. > Created by Mark Spencer<markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type

Hi, everyone I'm investigating if I could give asterisk CLI access to one of our clients. If I add that user to asterisk group and set his shell to /usr/sbin/rasterisk, is there a possibility for a user to brake our of asterisk CLI to normal shell? Thanks in advance

Hello, I'm remotely managing an asterisk setup using an OpenVPN client on this Asterisk box, connecting to an OpenVPN server of mine). This box is mainly connected to PSTN. It is also connected to the Internet, only for remote management. The former ADSL link has recently been replaced by a new 4G link (UMTS). I'm connecting to this box from a Debian Jessie/Gnome Terminal combo. With

2016-03-03 18:08 GMT+01:00 Duncan Turnbull <duncan at e-simple.co.nz>: > > > On 4/03/2016, at 5:31 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Hello, > > > > I'm remotely managing an asterisk setup using an OpenVPN client on this > Asterisk box, connecting to an OpenVPN server of mine). > > > > This box is mainly connected to

Hello! After upgrading my production machine to 1.2.1(used to be 1.2.0) on friday i experienced strange behaviour yesterday, i received deadlock-avoided-messages and channels refusing to hangup on span1(used for inbound calls), both messages in all cases paired: Jan 9 17:40:01 WARNING[30003] chan_zap.c: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. Jan 9 17:40:01

Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried getting a coredump, but it doesn't coredump. I know there are a lot of errors in the log below, but most of those just say it'll not load a module, and no big deal. When launching from commandline (not service script) here is what happens. http://pastebin.com/3GFe6fG9 Travis Ryan Director of

Hi, I need to run /bin/mount and /sbin/mount.cifs commands as nobody user (it has (bin/bash shell). So, I've edited /etc/sudoers and added: Cmnd_Alias CMD_MOUNT = /bin/mount Cmnd_Alias CMD_CIFS ) = /sbin/mount.cifs nobody ALL = NOPASSWD: CMD_MOUNT nobody ALL = NOPASSWD: CMD_CIFS But when I run the command as nobody (in the shell), I get the error: "mount

Hi guys I just did a ps -Af | grep asterisk on the machine and got several screens full of this: root 6970 6946 0 13:10 ? 00:00:00 rasterisk rxcore show channels verbose root 6987 6948 0 13:10 ? 00:00:00 rasterisk rxcore show channels verbose root 7005 6985 0 13:10 ? 00:00:00 rasterisk rxcore show channels verbose root 7021 7003 0 13:10 ?

Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten => TestTrakt,1,Dial(ZAP/1-2/517255333) exten => TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [TestTrakt at

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

fact loads: I use facter to search the rsync path and then i write this sudoers line: Cmnd_Alias BACKUPCOMMANDS = /usr/bin/rsync I call puppet like this: puppetd --factsync --server=xxx.yyy.61.131 --waitforcert 60 --test the recipe is: sudoensure_line { sudobackupcommands: ligne => "Cmnd_Alias BACKUPCOMMANDS = $rsyncpath",

One more thing about my last email: I think that you forgot to update the configs/samples/res_odbc.conf.sample file, because it still contains idlecheck and limit parameters. Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Mon, Sep 19,

Hi, Is there a way to show callerID of calls waiting in queue? queue show shows only channel not callerID Cheers, Marcin

Hello, I am trying to enable call forwarding feature on asterisk 1.6.0.9 with asterisk-gui. Sure there is no menu for that on gui but, when i try to write some example scripts to extensions.conf to make it work. I totally failed. I dont wanna install smthing like freepbx etc on the system so, i need help to add unconditional etc call forwarding feature for 1.6 Thanks

Hello Everyone, I am trying to change our /etc/sudoers (using visudo) to allow 2 commands to be run as root without a password, but it isn't working. Here is the part of the sudoers file that is in question. # User alias specification User_Alias FULLACCESS = doug, scott # members of the FULLACCESS User_Alias may run chown and chmod without a password FULLACCESS ALL = (root) NOPASSWD:

I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it

Hello my friends We are having seriously problems with our asterisk server, our versions are as follows: WANPIPE Release: 3.4.7 Asterisk 1.4.21.2 Zaptel Version: 1.4.11 libpri version: 1.4.10.2 The symptoms are very weird, the CLI stop working suddenly, a core show channels shows MANY channels FREEZED, the incoming and outgoing calls stop working, the internal calls stop working, in resume we

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

All, Did a little research and found this little trick that works well. First create this batch file and modify to your system. winword ------------------------------------ FNAME=`wine -- winepath -l "$1"` wine -- "/wine/c/Program Files/Microsoft Office/Office/WINWORD.exe" "$FNAME" ------------------------------------ chmod a+x winword and put it some where in the