similar to: Stutter Tone

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

Stutter tone when voicemail in boxHi, I'm having a problem getting Asterisk to make a stutter tone when a voicemail is received. It used to work, but since I rebuilt my server I can't get it going. I've set put a line in the sip.conf file: mailbox=100 to check it, but it's not doing the stutter tone ! I am using the stable version of asterisk from the CVS tree. Any help /

I have the following entry in zapata.conf, but I don't get stutter dialtone when there is a message waiting. Suggestions? Please? callgroup=1 pickupgroup=1 callerid="Paul mahler" <100> context=inside mailbox=100 channel => 1 Thanks, Paul

I'm having problems enabling stutter dialtone for users connected to channel banks. Half of our users are on iaxy's and the other half are connecting to channel banks. The users on ixay's are getting the stutter dialtone on new voicemails, but the ones on the channel banks are not. Currently, all users are in the default context in the voicemail.conf file. I've tried the

Any way to get the Asterisk to do the "broken dial tone" on phones that have voice mail in their mailbox? Similar to the phone company's built-in voice mail - awh

On 2016-11-22 07:49 PM, Pete Mundy wrote: > > One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. I have to be careful here as I auto-provison these devices and changes would propogate to every user. Echo cancellation is off. Do you think it should be on?

Hi-- If I have a voicemail box with a number of 1, and say (among other things) in zapata.conf: mailbox = 1 group = 3 context = workext callerid = "Steves Extension"<(999)999-9999> channel => 6 (assuming I have a TDM 4 port card, and 2 FXO T100P's ) Yes, I really have a mailbox, number 1. From voicemail.conf: [voicemails] 1 => 9999,Steve,user@wherever.com And, the

Im checking the wiki for call files info and seems somebody has a wake up script that runs call files at certain times. Do you know if its possible to run a call file by using some other methods different from cron jobs or at? The wiki mentions that it might be possible to do this is you modify the files creation date and time by using touch -f.. Is it true? The wakeup script creates call files

May be I have fiddled too much with my sipura settings but I can't get it to give the stutter tone when there is a new voice mail waiting on the asterisk box. I can either get a stutter tone all the time or not at all. Anyone got this working. Thanks Chris

Besides email notification, is there another way to get asterisk notify the user that they have a message? Example: Some analog phones have a blinking light that lets the user know that they have a voicemail message. Is asterisk capable of doing this?

AFAIK, this is not possible - but I'll throw it out there anyhow... I subscribe to telco voicemail, for the event that all my pstn lines are in use. Telco gives me a stutter-tone dialtone when I have a message waiting. Can a Zap card detect this stutter-tone and perform some action? I'm using TDM400P+FXOs and SIP devices. Thanks

I am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like most of my clients who use an ATA. He can make call and they are crystal clear. However, when he tries to use phone menus ("dial 234 for John Doe" for example) it

A question: is there any way to get * to answer certain DTMF sequences entered on an extension with a stutter tone? Long version: I would like to add features to my dialplan like "Caller ID Unblock" which work in the same way that the PSTN works: I pick up the phone, get a regular dialtone, press *82, and get a short stutter dialtone which confirms acceptance of the request, and then

Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy

Has anybody had any problems with their soundcards like this: Feb 20 01:05:22 WARNING[3420]: chan_oss.c:271 sound_thread: Read error on sound device: Resource temporarily unavailable This shows on the console and I have no clue what it is.. voice prompts sound good.... Any clues? __________________________________________________________________ Anton Krall

Thanks for the help. I was able to get my application to load with Asterisk, now I just need to get it to work. After reading your comment, I don't know that I fully understand what's going on as far as the channels and extensions. Are you saying that the MWI is tied to the channel? If that is the case, then for my SIP phones, would the channel be (SIP/XXXX)? If it should be, then

After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have "phone company" compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no

Hi to all!! I'm successful to connect Asterisk to MySQL database... Can anyone learn me how to store sip user in MySQL database and how to configure voicemail?? Thanks for all!!!

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex