similar to: FW: smbmount problem

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

Jeremy There is NetApp simulator that may help you ! Check now.netapp.com -- Yair

Hi all, i'm trying to integrate cepstral and asterisk, and i have a problem i'd appreciate any help with (i know it's a bit tangential, but i figure this is the place with the most knowledge of app_swift and asterisk). I've installed swift from cepstral.com with alison's voice, and it works fine, from the command line i can do swift "hello there" -o test.wav and then

Hello, I have a problem, I can't install the package 'mgu74av2cdf'. I downloaded the zip file, yet when asked the R console to install it from a zip file, I got the answer: "Error in file(file, "r") : unable to open connection In addition: Warning messages: 1: error -1 in extracting from zip file 2: cannot open file `mgu74av2cdf/DESCRIPTION'

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

I started to learn the R language, but I didn't suceed to use an external file. Let say that I have an excel file called "test1.xls" in the directory "C:/program files/R/rw2000/external_files" that looks like that: name mark yair 80 yosi 70 ... In the appropriate directory I wrote this: x<-read.delim("test1.xls") or this:

Hi, I'm using rsync to keep several file servers in sync. Uploads to the data can be placed at any time on any server. I'm using the extended N-way backup, following the example from the two way backup in the FAQ: rsync --archive --update rsync::first/files files (on second, third) rsync --archive --update rsync::second/files files (on first, third) rsync --archive --update

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833.

Hello. I hope that this is the proper place to ask this. Here is my problem. I have several smbmounts under Red Hat Linux 5.1. They need to be accessed by my apache web server. One of the mounts keeps getting disconnected, but this is not noticed by Linux. So the Linux box thinks that the files under the smbmount are still accessable when they aren't actually mounted. What would be the best

https://bugzilla.samba.org/show_bug.cgi?id=3718 Summary: RSync should verify permission/time before commiting a change Product: rsync Version: 2.6.5 Platform: Sparc OS/Version: Solaris Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org

First, many thanks to Frank Harrell for once again helping me out. This actually relates to the next point, which is my contribution to the 'why don't social scientists use R' discussion. I am a hybrid social scientist(child psychiatrist) who trained on SPSS. Many of my difficulties in coming to terms with R have been to do with trying to apply the logic underlying SPSS, with dire

Hey, Here's a new swfdec-gnome release to go with the just released Swfdec 0.6 release. Changes: - Depend on Swfdec 0.6 - Support automatic install of missing audio/video plugins - Really translate all relevant strings (Wouter Bolsterle, Bastien Nocera) - Updated ca: Gil Forcada - Updated ar: Ignacio Casal Quinteiro, Djihed Afifi - Updated cs: Petr Kovar - Updated et: Priit Laes - Updated gl:

hello, if anyone is using asterisk as a voicemail system for SER I would be grateful if i could see a working ser.cfg and extensions.conf of such a setup. I am having some issues with rollover to voicemail when busy, and in setting up a VM extension for users to retrieve their mail without having to enter their own extension. When i get this working i'll write it up clearly for the wiki

(sorry, posted without subject) hello, if anyone is using asterisk as a voicemail system for SER I would be grateful if i could see a working ser.cfg and extensions.conf of such a setup. I am having some issues with rollover to voicemail when busy, and in setting up a VM extension for users to retrieve their mail without having to enter their own extension. When i get this working i'll write

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of

hi list, is anyone succesfully using asterisk with libretel port-of-call (www.libretel.com)? If so, i would be grateful for configs..i set up libretel to forward to mynumber@myserver.com:5070 (asterisk is running on 5070 and SER on 5060) and when i call the number i see SIP messages with ngrep but the asterisk CLI doesn't seem to catch them. I assume i need to register...is this even possible

hi all, does anyone know what time variables are fed to to the "calldate" field in cdr_mysql? I have my system time set to israel time zone, have restarted mysql and a show variables shows timzone as "IST" which means now() should return israel time, but the calldate field keeps getting the system clock. I don't have the source for asterisk-addons handy so i can't