search for: rasterisk

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 ? 00:00:00 rasterisk rxcore show channels verbose root...

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 'core show wa...

...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 The former ADSL link has recently been replaced by a new 4G link (UMTS).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 this new link, whenever I launch a vim, a nano or a rasterisk session, my terminal freezes (rasterisk) or remains empty (nano, vim). When a session is frozon, I can open a new one at the same so it excludes a basic connectivity loss. What would you suggest ? Best regards PS: I was about to determine best MTU value but I always thought a punishment for a...

...ternet, 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 The former ADSL link has recently been replaced by a new 4G link (UMTS).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 this new link, whenever I launch a vim, a nano or a rasterisk > session, my terminal freezes (rasterisk) or remains empty (nano, vim). > > > > When a session is frozon, I can open a new one at the same so it > excludes a basic connectivity loss. > > > Usually incorrect MTU gives you this effect. Use ping with MTU size set to > te...

Hello, Is there a way to let someone access to Asterisk CLI and type whatever command (s)he likes but the shell command (the ones started by !) ? Ideally, it could be an argument to rasterisk: rasterisk --no-shell When done, a session could be like this: > pjsip show endpoints ... > core reload ... > !rm /etc/foobar Forbidden Suggestions ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/aste...

Hi, I'm running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a TE205P card. Then I enter asterisk console, and once I do a "pri show channels", the console no longer works correctly. There's no output for any command after that. If I type "pri show" and hit tab for completion, console freezes there, I need to kill session. Any hints? new bug? known issue?

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,

...om: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Karl J. Vesterling Sent: Wednesday, January 26, 2005 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCardApplication forAsterisk You and me both Brian... I've found MySQL to be fast, reliable, and once you have the replication set up, talk about reliable!!! I haven't used Postgress since 1998, and ummmmm... I'm not going back... There's an old phrase of wisdom that describes his opinion... "The fa...

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

...0.5 31012 12272 ? S 12:51 0:00 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p asterisk 2394 0.0 0.7 35048 15144 ? S 15:53 0:00 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p root 21099 0.0 0.0 4104 1592 pts/0 S+ 13:55 0:01 | \_ rasterisk r root 24101 0.0 0.0 4104 1592 pts/1 S+ 14:20 0:01 | \_ rasterisk r root 6009 0.0 0.0 1500 476 pts/2 S+ 16:18 0:00 | \_ grep asterisk root 29100 0.0 0.0 4104 1576 pts/3 S+ 14:58 0:00 \_ rasterisk r asthost log # a...

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/2...

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/3GF...

...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 na-miasto:1] Dial("SIP/sempron-b2ae1918", "ZAP/1-2/517255333") in new stack [Jun 19 21:22:20] WARNING[23814]: chan_zap.c:7966 zt_request: Unknown option '-' in '1-2/517255333' -- Requested transfer capability: 0x00 - SP...

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

...lid 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 have 1 cdr record with only inbound call leg callid. I can see all the call legs with rasterisk -x "sip show channels", but I would like to get the data in cdrs as well. Does anyone have any idea? I have asterisk 1.6.2 and I don't want to fork cdrs, just another field for the came call. Regards, Kate -------------- next part -------------- An HTML attachment was scrubbed... URL...

...work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it should be starting in the correct order. I've previously done this so Asterisk is on one server and IAXmodem and Hylafax on another, I am stumped. [root at pbxserver ~]# rasterisk Asterisk 1.6.1.20, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public Licens...

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

...asterisk [root at localhost ~]# ps -aux | grep asterisk Warning: bad syntax, perhaps a bogus '-'? See /usr/share/doc/procps-3.2.3/FAQ root 16105 0.4 0.2 28700 11476 ? Sl 14:59 0:09 /usr/sbin/asterisk -f -vvvg -c root 16255 0.0 0.0 3364 1208 pts/0 S+ 14:59 0:00 rasterisk r root 16465 0.0 0.0 4572 640 pts/3 S+ 15:32 0:00 grep asterisk Please HELP! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100317/25f9a38f/attachment.htm

On the asterisk console I see: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected This is happening about once every 5 seconds or so. How do I figure out