similar to: Sound playback problems

I have the raw data with 9 column and 197977 row: > dummy State Months Bedroom 1 xxx Jan 1 2 xxx Jan 2 3 xxx Jan 1 4 yyy Jan 1 5 yyy Jan 2 6 yyy Jan 1 7 zzz Jan 3 8 zzz Jan 1 9 zzz Jan 2 10 xxx Feb 3 11 xxx Feb 4 12 xxx Feb 2 13 yyy Feb 1 14

Hi, I would like to change the sequence of messages that was playbacked when I call to the voicemail extension, VoicemailMain. Where can I changed it, can't find the definition for VoicemailMain in extension.conf and voicemail.conf __________________________________________________ Do You Yahoo!? Download the latest ringtones, games, and more! http://sg.mobile.yahoo.com

List, I purchased an SPA-1001 from voipsupply.com (which was great.. no hassle), but now anytime there is silence and then Playback() sound files the first half second gets cut off. For example, when I login to the VoiceMailMain() function I don't hear "Password", I hear "-assword". Then after I login I hear "- have no messages..." instead of "YOU have no

Ladies and gentleman, I'm running Ubuntu Dapper Drake (amd64 version). I managed to set a 32bit Ubuntu dapper chroot on /chroot. It works excellent! on that chroot environment I have installed firefox, in order to run macromedia flash player (i know this is irrelevant to my question). Then I downloaded the latest wine sources from winehq.com website. I compilled and installed it on that

Is there a way to make the asterisk voicemail app play back messages in NEWEST FIRST order, instead of OLDEST FIRST? I see the situation repeatedly where someone needs to dip into their voicemail archive to get something from a recently saved voicemail message, and they have to slog through lots of irrelevant stuff to get there. I have seen this question come up previously on this list without

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

HI i'm trying to detect breaks points in various flow time series, they all contains seasonality and trend my question is : i have to remove this seasonality and trend before apply the function breakpoints du package strucchange?? another question, the function breakpoints is similar to de Pettit tests ? or how does it realy works? THANKS!!!! DENISSE -- View this message in context:

Hi, This is a message I already posted on the chan_sccp mailing list, but since this list has a lot of active members, I'm hoping someone might be able to help (And my problem is * related, so I guess it's ok if I post it here also ;) ). I'm trying to get SCCP ATA188s to run with Asterisk. The Asterisk box uses the latest Asterisk@Home image (Version 2.6). I have compiled and

I have some shares on a media server that are considdered "Local, offline content", namely they should be accessible if the rest of the network is down, and each system has it's own group of users who are allowed to maintain it. the media servers in the livingroom are only for my wife and I, but each person can modify the one in their own bedroom and noone elses bedroom.

Hello, I am new to Ubuntu (Linux in general) and I am trying to install R: so far, I am stuck. I have followed the instructions at the CRAN website with little luck. I read an exchange about the same issue posted before, but did not give me enough information to solve things by myself, hence, this email. Based on the former exchange I read, I am trying to include enough information to see if

Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an

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tirosien dfstype over-intense'jepp ~cscs m'e.ds from e,u.ropean & 0v^er.n_ight s.hippi^ng http://pharm-martworld.com Suddenly the minister cried, Ha! and, pointing to the figure of a man disembarking from an English boat at Calais, he said, excitedly: That, your Excellency, is the Duke of Orleans, in disguise! I must leave you for a time, that I may issue some necessary orders to

Greetings folks, Since the sourceforge apt repository is woefully limited to just one distribution version, and because it is exceedingly slow and times out a lot, budgetdedicated.com has graciously donated us a blazing fast server to host the fresh Ubuntu packages. To use the APT repository, simply edit 'sudo gedit /etc/apt/sources.list' and then add the following lines, depending on

I use debian/sid with the Dapper wine package. winecfg crash when i click on Audio with WINEDEBUG=wave,dscapure,mixer winecfg i get: trace:wave:OSS_DriverProc (00000000, 0x7fd30270, 00000001, 00000000, 00000000) trace:wave:OSS_drvLoad () trace:wave:OSS_WaveInit () trace:wave:OSS_WaveOutInit (0x7dc0e140) /dev/dsp trace:wave:OSS_OpenDevice (0x7dc0e140,1,(nil),0,-1,-1,ffffffff)

Hi all, Daniele Favara was kind enough to package Xapian (and Pinot :-) for Dapper and Xubuntu. I understand Daniele used cdbs to build these packages. http://packages.debian.org/unstable/devel/cdbs I am not sure if the differences with the official packages built for Debian and Breezy are significant... Anyway, if you would like to give them a go, add the following to your sources.list : deb

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: -- Executing [301 at from-test:1] Dial("SIP/300-00000045", "SIP/301,60,rjtTg") in new stack -- Called SIP/301 -- SIP/301-00000046 is ringing

Hi everyone Last night I set up ubuntu 6.06 - on a virtual machine. I installed wine and wine doors on it to test run a software or too - before making a switch to ubuntu ultimate edition next week. However wine and wine doors are not working. I used the links below to register on software properties and I included the authentication keys wine deb http://wine.budgetdedicated.com/apt dapper

Dear r-sig Debian list, I subscribe to the 'r-help' forum but as requested on the "Download and Install" section of CRAN I am reporting a suspected bug on the " bin/linux/ubuntu dapper/" installation binaries, I show an extract from my Ubuntu(dapper) terminal console below---- # Attempted unsuccessful installation copy from terminal-- root at ubup:/etc/apt# sudo