similar to: queue show name - callerID

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

Hi, I'm expiriencing very annoying issue with Asterisk while using ODBC on Linux box running Debian with following packages: Asterisk-node0:/tmp# dpkg -l | grep -i odbc ii libiodbc2 3.52.6-2 iODBC Driver Manager ii libmyodbc 3.51.15r409-4 the MySQL ODBC driver ii libodbcinstq1c2

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

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,

The Asterisk Development Team has announced the release of libpri 1.4.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.13 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Outgoing

Hi, I need help with modifying cdr_addon_mysql.c I want to have more fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c and replace userfield with ex team (sed -e 's/userfield/team/g' ). When I try to recomplie menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'.

I think were getting closer: I did: - I edited /etc/default/asterisk to include : AST_USER="root" AST_GROUP="root" # systemctl daemon-reload # systemctl start asterisk # ps aux | grep asterisk root 3602 7.1 2.5 60332 26012 ? Ssl 16:00 0:03 /usr/sbin/asterisk -U root -G root -g # rasterisk # pkill -SEGV asterisk Then console showed: Segmentation error (core

2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>: > On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: > > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Where should core file be created when Asterisk is run as a daemon by > > > asterisk user and group ? > > > Is there a setting I

Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing "bs523450017" instead of +6523450017. i tried putting it inside double quotes CALLERID(num)="+6523450017" telco says the same thing. is this possible? thank you Regards, nhadie

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

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

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, I'd like to generate a filled step-function in R as you can see in the attachement (image is generated by grapher, but I need a freely available alternative). The problem is, that I don't know, how to fill it up. My code right now looks like this: pH <- c(0,6.1,6.1,6.3,6.6,7.3,0) Tiefe <- c(0,0,10,40,80,120,10) plot(pH,Tiefe,type="s",main="Tiefenfunktion

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes.... Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio.

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

https://bugs.freedesktop.org/show_bug.cgi?id=50043 Bug #: 50043 Summary: GT240 (nv40) doesn't show anything at boot up, nor X starts Classification: Unclassified Product: xorg Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: blocker

https://bugs.freedesktop.org/show_bug.cgi?id=40138 Summary: nouveau doesn't show screen on DVI on ALiveNF6G-DVI with DVI Graphics-SI Card Product: xorg Version: git Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH