similar to: Loud noise instead of MOH

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

Hi everybody I'm trying to make Asterisk 1.2.1 run under VMWARE and Suse 9.2. I use ZTDUMMY module for timing and ZTTEST gets an average precision of 98,4 %. Is there any way to improve it? Best regards Mauro Zanin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060117/945d544a/attachment.htm

Hi dear group members, I have finally an Asterisk box working, capable of receiving and making calls. I have this issue while calling mobiles from our SIP softphones: -------------------------- linux*CLI> -- Executing NoOp("SIP/2000-6850", "3487024125") in new stack -- Executing Dial("SIP/2000-6850", "ZAP/g1/3487024125") in new stack -- Called

Hi everybody, I'm strugling between two devices: the both TE but one was set up as a TN. I have no current on that interface. I have tried to find some circuit over the net to power the connection, both commercial and home made. Can anybody give some hint? Ciao Mauro

Hi everybody, I have this issue: I need to automatically park an incoming call, play a welcome prompt and then connect to some extension but under extension user's command. I was thinking to use a small database to comunicate between asterisk and the main application. Has anybody had this kind of experience? Best regards Mauro

Hello I'm working with mixed effects models using lmer() and have some problems to get all variance components of the model's random effects. I can get the variance of the random effect out of the summary and use it for further calculations, but not the variance component of the residual term. Could somebody help me with that problem? Thanks a lot! Below an example. Aline ## EXAMPLE

Hi. I want to do an lmer model but have doubts of what family I should use. My response variable was originally a proportion, however I standarized it for each year of data collection (20 in total). After standarizing it I checked for normality with the Kolmogorov-Smirnov test, and it turns out it is not normal. It ranges from -3 to 4. Since it is no longer a proportion I can't use a

Hallo Everybody Consider the following vector a=1:10 b=20:30 c=40:50 x=c(a,b,c) I need a function that can tell me that there are three set of continuos sequences and that the first is from 1:10, the second from 20:30 and the third from 40:50. In other words: a,b, and c. regards Christiaan [[alternative HTML version deleted]]

Yes, I saw that they are different I was just willing to test something similar. Actually, I'm searching for a Samba feature that allow transparent failover, or continuos availablity in a cluster setup (Samba + ctbd + gluster) Based on the following link my understanding is that such feature is not currently available in Samba:

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

I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro

Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: "Chi lascia la via vecchia per la nuova, sa

Hi, I am using Asterisknow 1.5. And TDM400P card for interfacing with PSTN line. This setup was working without any problem. But now it is showing issues. When I try to call through PSTN, there is a continuous large noise is hearing from the SIP phone. And can't make the call. When I try to call the PSTN number from mobile there is only engaged tone is hearing. And also the Asterisk

On Mon, 2005-01-10 at 08:01 -0600, asterisk-users- request@lists.digium.com wrote: > > I am new to asterisk but learn a lot about it to this mailing list > and > > wiki currently i am facing problem about sip phone i have "PA 1688" > > chipset ip-phone and i have iptel.org sip account i registered > locally > > and through iptel.org comfortably my problem

Hi R users: How can I obtain with bwplot boxplots with bwplot whose box width will vary acording to other variable. bwplot(categ1~continuos|categ2,box.ratio=continuos2,data=data.base) But it doesn't work as I expected. Thank you for your help. Kenneth

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Hi guys I'm using Icecast , Ezstream for creating my Webtv for creating a continuos streaming of some files In the filename filed of Ezstream xml configuration i put a simple m3u playlist. My problems begin when ezstream begin to encode and stream the second file: If a use a system player , like Vlc or totem , the stream it's continous , if I use a web player like Cortado or Itheora

Hi guys I'm using Icecast , Ezstream for creating my Webtv for creating a continuos streaming of some files In the filename filed of Ezstream xml configuration i put a simple m3u playlist. My problems begin when ezstream begin to encode and stream the second file: If a use a system player , like Vlc or totem , the stream it's continous , if I use a web player like Cortado or Itheora