similar to: Stuttered Dialtone for multiple extensions

I have asterisk running on my personal computer and am using Kphone to connect to it. My provider is broadvoice which is Ulaw and I had kphone connected as GSM. The lag was terrible coming from Pots-->--Broadvoice-->Kphone. About 2.5 seconds! Going the other direction seemed fine. I did a: show translation recalc 200 and see that the translation time should be about 2 ms. When I do the

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

I think I've configured everything to have stuttered dialtone on my analog phones (it works fine for my SIP phones). But I still don't have it:-( I'm using asterisk 0.9.0 and zaptel 0.9.1. In voicemail.conf: [local] 21111 => 987654,Robert Withrow,email@here.com In zapata.conf [channels] ... mailbox=21111@local channel => 1

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

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

Does anybody out there run * on an AT&T broadband phone line? I'm not seeing any callerid and I can't tell if its AT&T doing something funky or if its my setup. I do see CID on my normal phones.... Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

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

Hello everyone, I am trying to fit the following model All X. variables are continuous, while the conditions are categoricals. model <- lm(X2

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

I'm working on migrating my home POTS phone system into an Asterisk PBX. Currently I have my FXO and FXS setup and working great. Zapateller is nice! I'm working on my voicemail now and currently have my 3 mailbox answering machine plugged into my FXS along with all my other analog phones. I'd like to slowly migrate away from the FXS answering machine in favor of Asterisk Voice Mail.

Hallo, All, I have a question about changing the height or scale of the y axis. When I use following two R codes, I can get two plots. Please look at the y axes, the number of indices (x1, x2, ?) on the y axis in the first plot is smaller than that in the second plot, and hence the space between any two indices in the first plot is wider than that in the second plot. As the number of indices

Hi everybody... I'm having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

You wrote: : Do your two bogus servers cause stuttering under 1.3.7? There's a big : difference in the errors you can get from trying to make a connection, : no route to host, connection refused, just really slow, blah blah. Good point. I put 1.3.7 back and tested. The initial bogus servers worked pretty much okay. : You're looking for test cases where the connect time would be

On 28/08/13 23:03, Sam Varshavchik wrote: > I'm running xorg-x11-drv-nouveau-1.0.9-1.fc19.x86_64 and > xorg-x11-server-Xorg-1.14.2-9.fc19.x86_64 > > On the following chipset, the display suffers from "stuttering": > > 06:00.0 VGA compatible controller: NVIDIA Corporation GT200b [GeForce > GTX 285] (rev a1) (prog-if 00 [VGA controller]) > Subsystem:

Hi All, I have a query about time based sequences. I know such questions have been asked a lot on forums, but I couldnt find the exact thing that I was looking for. I want to create a time-based sequence which will mimic the trading window AND would span multiple days. Something like below: "2011-01-03 09:15:00 IST" "2011-01-03 09:15:01 IST" .... .... .... "2011-01-03

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key