similar to: Local sip client stuttered audio

Hey all..I'm looking to start with a single FXS card but with 3 extensions for VM purposes only. I'd like to know if there is a way that you can have different stuttering dialtones depending on which extension has a VM. For example If x103 and x104 have VM can there be a distinctly different stutter for each mail box and have them play back at once or back to back so that when you

I've noticed on a few installations that the very first audio played after a call in answered (eg: Greeting), the first part of the audio is cutoff/stuttered. Is this because Asterisk needs some RTP to create a sync for audio - and the first 1 second is lost? Should one play 1 sec of silence first? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

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

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

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

The configuration for kphone in sip.conf: [177204] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid="Jane Smith" <5678> host=dynamic ;nat=yes ;

Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

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

My only problem now is this: I have a broadvoice number that I can dial out on, and i ncomming calls word fine. Once the ip phone handset picks up, the audio bi-directionally is perfect. The problem is the audio BEFORE the handset picks up is silent on the broadvoice side. The user calling in over the sip number doesnt hear the menu, or even the ringing. Once the handset picks up then

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 using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with

Hi, I'm using the SPA-2000 with firmware 1.06 on the Asterisk PBX, which works great for taking and placing calls, but for for some reason I can't seem to clear the stutter dialtone by either calling the extension I'm on, or the voicemail system on the Asterisk PBX. If I call my voicemail access extension directly, It tells me I have no messages waiting, yet when I hang up, then

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

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

Hi, I have set up the asterisk in Redhat 9 with kphone, kphone is successfully registered under asterisk console.....so i had some tests on the extension......but there were messages saying that something(probably .gsm files) were being played, but i heard nothing.....there were also some warning messages from the asterisk console(when i launch it asterisk -vvvc) saying that ERROR:sound device

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 /