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Hi anyone can help me to resol my big problems ? thanks bye

Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold: -- Executing [youngandson-test at

Hi, I am new with Puppet and I am very glad using it, but I have a basic doubt, when I use puppet to install packages to a debian machine with: debian:/etc/puppet# cat manifests/packages/lm-sensors package { lm-sensors: ensure => installed } Does it run apt-get update before installing the package? If not how is the best way to do it? How Puppet deals with packages installations that

Dear R-users I have some doubts about with converting dates... e.g. dates data 23/12/1975 0.8678 03/01/1976 0.8736 28/01/1976 0.765 13/03/1976 0,9654 ... I think that converting these dates can easily my plots... is there a R function to convert these dates to a julian dates and so plot that variables?? The help files as.POSIXlt, POSIXt and others don't aswer my doubts...

Hello, i have been looking over inet for an answer to a problem iam unable to resolve, and after not being able to find a solution, i hope to get an aswer in the list. I have been trying to reach my /home (Dom0) from one of my virtual machines (i''m runnning them in debian etch) and i''m unable to do it, what i''m trying to do is to catch a file from the /home of my Dom0,

I made the changes you proposed but still my system must be messed up. Please have a look at the screenshots below... may be you or some other guru would know how to come out of this mess. http://pasteboard.co/Ba9Ex0NQD.png http://pasteboard.co/3Iq39El98.png By the way, is there a clean way to reset ADDC and DM so that I could start fresh? those two virt machines are just lab machines. I can do

...the next ones or a web where I could find this information? That will be very helpful. - alsa.conf - enum.conf - modem.conf - modules.conf - oss.conf: what is oss? - parking.conf: what is parking? - rpt.conf: what is radio repeter? - queues.conf - skinny.conf - voip.conf Thanks a lot for all the aswers. Mireia

Hi, I have been searching the mailing list, but i couldn't found the aswer for this one : How do i give root privilege for other users in swat, so that other users can create shares, and change other configs using swat ? Thanks in advance. Alexandre.

Hi, alhtough i googled for details concerning ASTCC i did not found an aswer to the following: Should i put in my extensions.conf the configuration of the astcc? I ask this because as i see it, in the end of the extensions.conf there is an include statement : #include /var/lib/astcc/astcc-exten.conf Should the config been done in the astcc-exten.conf file or the initial extensions.conf

Hi, I have the following dialplan. Problem is, if the user authenticates, * starts counting as billable seconds even if i hangup the phone before the called party answers..And also as disposition.. it accepts all calls authenticated as 'answered' If i commentout the authentication line everything works as it should be. How should i use authentication that, it should accept it as aswered by

Hello! I'm new in Asterisk and I hope that my trouble is very simple. We're implementing a Education Project of a e-Learning system (LMS) that uses conferencing (video and audio) over internet. The e-Learning system will be on GPL license, and for that, we're using only free software to implement. Asterisk is our first choice for video and audio conferencing, and making tests,

Hello friends of R list, Im a physician and Im not that good in statistics. I have posted similar email in the epi-sig list but no one aswered so far. Im cunducting a systematic review on two diagnostic test for a particular tropical disease. I found a free software that make most of the analysis called MetaDiSc. However there is a paticular analysis that I wuould like to try that it is not

Wlcome in the club :-) I wrote a topic "Limit of group membership for a user?" with the same problem, and I don't have any answer by the samba team. I have the most recent Samba 3.0.21b, OpenLDAP 2.2.28 and Glibc 2.3.6 version on a Gentoo (kernel 2.6.15) and I have the same problem. I check the NGROUP_MAX (65535) and the NSS_BUFLEN_GROUP (2048) value of the glibc. For the AMD64

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

Hai, Sorry if you feel like that but i always have to ask this. Your aswer is not help also for me, because without smb.conf OS and samba version we know nothing.. We dont like guessing, because everybody's time is precious. So please tell more about your system, i suggest start with. OS Samba version Smb.conf Upgraded from samba 3.??? To samba 4.?? The more you tell us te better we

hi, we have successfully implemented at tang/clevis environment for automatically entering luks keys and booting hosts without operator intervention. Now we would like to use this as well on ipv6 networks, but I do not seem to get it to work. I have already posted this issue to the dracut devs github issue tracker ( https://github.com/dracutdevs/dracut/issues/554) but no response so far. Maybe

On Tue, 21 Feb 2017 16:25:24 -0600 Lin Pro <linforpros at gmail.com> wrote: > I made the changes you proposed but still my system must be messed up. > Please have a look at the screenshots below... may be you or some > other guru would know how to come out of this mess. > > > http://pasteboard.co/Ba9Ex0NQD.png > http://pasteboard.co/3Iq39El98.png It looks fairly

Hello, I'm having some problems recording calls. Here is the situation. Call come in Zap/1-1, rings through to SIP/101. Call is answered, talk talk talk, hangup Zap/1-1-in.wav is full of loud static Zap/1-1-out.wav sounds fine Call comes from SIP/101, rings to SIP/102 Call is aswered, talk talk talk, hangup SIP..in.wav sounds fine SIP..out.wav sounds fine. So, it's something about

Hi, When I make a call from my asterisk and it is passed thru another astrisk eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting as soon I get connected to the other asterisk, and not if the party on the other asterisk server pick up the phone. So IF the other party are not aswering the call at all, I still get Answerd and billsec in cdr. Whats wrong ? Can I do

Yeah sorry about that. But I didnt see my message in the list, so I thought it didn't ame through. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Dave Cotton Sendt: 14. september 2005 19:45 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] RxFax problems On Wed,