similar to: A@H 2.6 : problem connecting call from PSTN

Dear friends from the R-community, I am djipie, bokaha and live in Germany. I am a student and user of R. During the installation of the package "Metrics" I had a pronlem as shown below: > install.packages("Metrics") Warnung in install.packages("Metrics") : Argument 'lib' fehlt: nutze 'C:\Users\Guyso\Documents/R/win-library/2.10' --- Bitte einen

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Now that I have a little more time, I was debugging my dialplan and it was of the following way: - ------------------------------------------------------------------------- ; DGB - 20090615 [macro-dial] exten => s,1,Dial(${ARG1},15) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)

Hi All, I'm having a strange pronlem with winbind. For users it seems to be working fine but for groups its not enumerating most of the groups! A getent group produces only 325 lines of domain groups whereas wbinfo -g produces 2839 lines of groups. I'm not seeing any errors logged and all commands are exiting with status 0. Winbind related sections of smb.conf are shown below,

Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

Hi, I just found this solution for identifying a job in the job table: http://stackoverflow.com/questions/3638250/how-to-cancel-scheduled-job-with-delayed-job-in-rails They are using this code: class MyJob < Struct.new(:user_id); def perform # ... end def display_name return "MyJob-User-#{user_id}" end end # store reference to a User my_job =

> Olle E Johansson wrote: >> >> 23 apr 2007 kl. 19.55 skrev Russell Bryant: >> >>> John Todd wrote: >>>> To morph this into a -dev thread: if this patch were to become (again) >>>> useful and error-free, is there any objection or usefulness in adding it >>>> to TRUNK? Personally, I think there is, if there is a method by which

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/104a07f3/attachment.htm -------------- next part -------------- Hello In my IVR menu whenever user select the option number 1 then it should jump to echo context, I think call did jump to "echo" context but I always get the following warning and I hear couple of beeps and then

Hi again I'd like to configured my Asterisk to use german sounds for the "Say"-commands... I installed the sounds-files and I tried them with "Playback(de/demo-echodone)" and it works. Now I tried to add an extension to say the current time: exten => 24,1,Verbose(2,Time asked by ${CALLERID(num)}) Exten => 24,n,Set(CHANNEL(language)=de) Exten =>

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a

Hi All, This one has me stumped, but I've done quite a bit of debugging to hopefully isolate it down enough. Basically, I hear an echo *delay* when I do a 'echo' test (key- there's an annoying delay). Here's what I have: Asterisk 1.0.3 current as of Dec 7. x100p (although, irrelevant in this case) (2) Cisco 7940's with 7.3 firmware All communications between asterisk and

Hi! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

Hi list users I?m trying to do an incluye statement from the Database In my dialplan I have different contexts that defines common services for example ---------------------------------------------------------------------------- ----------------------- [basic_services] exten => 100,1,VoicemailMain() exten => 600,1,Playback(demo-echotest) exten => 600,2,Echo() exten

Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), -----> [ GSM Gateway ] -------> [ BN8S0 ] ==== asterisk Port connected to GSM gatway is in TE mode , gateway is in NT mode , When I dialin to cellphone numer , call goes to 'from-eragsm' context, to Echo application. [from-eragsm] exten => 700,1,Goto(600,1) exten

Hi, For some reason, I seem to have issues with dailing toll free numbers and can't seem to find out why, sometimes, I get a busy signal. Some other times I get weird errors from the phone. The error below was a simple busy signal. Here's couple of my info relevant to the problem: -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling

Hello! I am trying to come up with a way to read a digit *before* the call is answered. My Asterisk version is 1.6.2.0-rc6 SIP early media works fine (I can receive and transmit audio before the call is answered), but as soon as I start the read application, Asterisk answers the call which is not what I want. Here is how to reproduce the problem: send incoming calls from a SIP provider that