similar to: Help with dial command and h, H and g parameters

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Hi, I am trying to dial # or *411, in order to understand what the * box should answer me. In both cases, I only ear "Good-Bye" (italian , "arrivederci") dialing # -- Executing Wait("SIP/555-a2e5", "1") in new stack -- Executing AGI("SIP/555-a2e5", "directory||ext-local|lo") in new stack -- Launched AGI quitScript

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

Hi All I am working on voicemail , mailbox , after reading documents, I had setup of three users for mailbox to make things simpler , I had kept the user name and passord same for all the sip users, Now I am able to record the message and I do get voicemail in my email , But as defined in extensions.conf The Asterisk console messages, part of the sip.conf ,

I have this code which was taken from the phpagi project page along with the following in extensions_conf and the output from the asterisk CLI. When I call the 311 extension, I does nothing then hangs up. What am I doing wrong?? ----php code------------ #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); $agi = new AGI(); $agi->answer(); $cid =

I am having problems getting incoming calls from the PSTN to route to extensions, digital receptionist and even voicemail. When I call a DID number for one of the lines, it rings twice then says: "Goodbye" and hangs up. (logs to follow below configuration info). When I dial 7777 it goes to the digital receptionist without any problems. The system setup is simple; I have 8 PSTN

Hello, >From: "Hermann Wecke" <hermann@wecke.com> >Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern => >extensions.conf >Date: 8 May 2004 22:03:57 +0000 > >Is it possible to strip some numbers from the *end* of a number? > >I know that ${EXTEN:1} will remove 1 position from the beggining... but >how to remove N numbers from

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi, I want to count the number of open Zap channels on my server. [outgoingzap] exten => _0NXXXXXX,1,NoOp(Outgoing Local - 7 digs - ${EXTEN:1}) exten => _0NXXXXXX,2,Set(ZAP01=$[${ZAP01} + 1]|g) exten => _0NXXXXXX,3,Set(UPDATED=true) exten => _0NXXXXXX,4,Dial(${TRUNK}/${EXTEN},60) exten => _0NXXXXXX,6,Busy exten => _0NXXXXXX,7,Playback(thank-you) include => hangupcontext

Dear Sir, I got a problem with my program. I would like to classify my data using bagging support vector machine ensemble. I split my data into training data and test data. For a given data sets TR(X), K replicated training data sets are first randomly generated by bootstrapping technique with replacement. Next, Support Vector Mechine (SVM) is applied for each bootstrap data sets. Finally, the

Hi, I'm using the AbsoluteTimeout and Dial (with L() option) commands to set a timeout for my calls, but when the calling user transfers a call the timeout doesn't work and the call last until hanging-up. If the call is not transfered the limit works just fine. ?How can I make this work? Thanks in advance. My asterisk version is 1-0-9-07 and here's an example of one of the macros

Hi, it's me again with a cdr-issue. I have the following example extensions.conf: # dial in exten => 100,1,Playback(hello) exten => 100,n,Dial(local/200,20,rtg) exten => 100,n,Playback(pleasewait) exten => 100,n,wait(10) exten => 100,n,Playback(goodbye) exten => 100,n,Hangup # for local dial exten => 200,1,Playback(hello) exten => 200,n,wait(10) exten =>

Hi Karthik, Thank you very much, you made me much more relaxed. Below is getfattr output for a file from all the bricks: root at gv2 ~ # getfattr -d -e hex -m . /data/glusterfs/testset/306/30677af808ad578916f54783904e6342.pack getfattr: Removing leading '/' from absolute path names # file: data/glusterfs/testset/306/30677af808ad578916f54783904e6342.pack

Hi, I am trying to construct a svmpoly model using the "caret" package (please see code below). Using the same data, without changing any setting, I am just changing the seed value. Sometimes it constructs the model successfully, and sometimes I get an ?Error in indexes[[j]] : subscript out of bounds?. For example when I set seed to 357 following code produced result only for 8

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

hello, i am so glad to write you. i am dealing now with writing my M.Sc in Applied Statistics thesis, titled " Data Mining Classifiers and Predictive Models Validation and Evaluation". I am planning to compare several DM classifiers like "NN, kNN, SVM, Dtree, and Naïve Bayes" according to their Predicting accuracy, interpretability, scalability, and time consuming etc. I have

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

Hi everyone I am trying to do cross validation (10 fold CV) by using e1071:svm method. I know that there is an option (?cross?) for cross validation but still I wanted to make a function to Generate cross-validation indices using pls: cvsegments method. ##################################################################### Code (at the end) Is working fine but sometime caret:confusionMatrix