similar to: Loud pop at the end of messages causing level problems

Hello all, This is my first attempt at posting a patch. So if I screw this all up, my apologies and please someone let me know without beating me up too bad. To use this patch. You need to have an extra line in /etc/asterisk/voicemail.conf that looks like volgain=10.0 The 10.0 gets passed to sox which you will need installed on your system. 10.0 is what works for me, anything over 1.0 will

Hello List. I am having problems retreiving voicemails on my system. I noticed when someone leaves a message through the pstn line I can't hear anything. I tested leaving a message from one of the extensions and that can be heard. I don't know if is the type of card I'm using for analog ( cheap X100p modem ) calls but, can't hear any message coming in from that line. Any

Group In voicemail.conf I would like to having the following setup per context not per-mailbox settings serveremail userscontext fromstring usedirectory emailbody pagerfromstring dialout sendvoicemail callback review operator volgain nextaftercmd forcename forcegreetings tempgreetwarn Can this be done? Thanks! -------------- next part -------------- An HTML

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I have a contingency table with a 0/1 variable (fwr) that flags a particular condition, a weight variable (cnt) and other variables (e.g. zz) characterizing that observation of the table. I am trying to use GLM. By converting the variables to factors and running the following regression: fwr1<-factor(fwr) zz1<-factor(zz1) res1<-glm(fwr1 ~ zz1,weights=cnt) Is that appropriate? Paul M.

I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is having problems with Voicemail. They can listen to their voicemail but on the weekend it stopped delivering messages via email. The only thing I can notice is that the permissions for the files on teh voicemail directories are created with no permissions at all. Here is the listing on one of the mailboxes: 4 -------rw- 1

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

I have an issue getting any incomming calls When the phone rings something picks up and gives it a fast busy. There is no one using Zap/2 it does the same thing with voicemail and voicemail 2 you can see the console output below, I would love any help anyone could shead on this issue, Michael NOTICE[1192484144]: File chan_zap.c, Line 4270 (ss_thread): Got event 2 (Ring/Answered)... --

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

In the rockchalk package, I want to provide functions for regression objects that are "well behaved." If an object responds to the methods that lm or glm objects can handle, like coef(), nobs(), and summary(), I want to be able to handle the same thing. It is more difficult than expected to ask a given fitted model object "do you respond to these functions: coef(), nobs(),

> On Mar 31, 2016, at 1:00 PM, Paul Johnson <pauljohn32 at gmail.com> wrote: > > In the rockchalk package, I want to provide functions for regression > objects that are "well behaved." If an object responds to the methods > that lm or glm objects can handle, like coef(), nobs(), and summary(), > I want to be able to handle the same thing. > > It is more

What causes the error report: logical(0) to arise in the rms function lrm? Here's my data: But both the dependent and the independent variable seem fine... > str(AABB) 'data.frame': 1176425 obs. of 9 variables: $ sex : int 1 1 0 1 1 0 0 0 0 0 ... $ faint : int 0 0 0 0 0 0 0 0 0 0 ... Here's the simplified model and error AABB$model1 < lrm (faint ~ sex)

Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what

Dear all, I have invested substantial amount of work in a complicated, yet on screen perfect looking graph that uses image(). Unfortunately saving it as pdf (or quartz.save at high resolution) all generate very disturbing border lines around each small rectangle that image() has drawn. While using Preview it helps to turn off antialiasing to make those faint lines disappear on screen

Hi Everyone, I cannot seem to get the voicemail gain option g(#) work in Asterisk 1.4.11. I am using it like so... Voicemail(4444 at mycontext,bg(10)) ; for busy announce and 10dB record gain This has absolutely NO affect on the resulting voicemail wav file. I have also tried using "format=wav" instead of wav49 in voicemail.conf to increase the volume as well. This also has no affect

Hi, Would anyone have a clue on this issue? I'm running asterisk 1.4.13. Trying to get a WAV voicemail file attachment sent to my email address. Voicemail is working fine. Email notification of a new message works fine. However, when I set up voicemail.conf to have an attachment of the file sent to email I get an error message like this on the console: [Feb 17 13:33:37]

Hey, I've come across two interesting problems today. First, when recording long calls using Monitor(), it appears the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help. Now it seems that

Ok, Here's another bizarre one (no strange curve balls to throw this time :P). I have several mp3 files of some easy listening music that I pulled off some CDs we have. They sounds fine and are at a nice volume level. When I run this script I wrote: (I run it by doing ./script < filename) mpg123 -s --rate 44100 --mono $1.mp3 > $1.raw sox -r 44100 -w -s -c 1 $1.raw -r 8000 -c 1 $1.wav

Hello, I am struggling with what seems a common unresolved problem, changing the password from voicemailman when using a realtime engine (adaptive_odbc in my case, connected to mysql). I have seen messages dating back to 2007 with this problem and the last one was bug 5168, reported as closed, but without explaining the fix