similar to: detect telco voicemail stutter-tone

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

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'd like to be able to configure Asterisk to detect and respond to message waiting indications from my telco, as I'm using the telco voicemail as a fallback in case the Asterisk line is busy, Asterisk or it's server is down, etc. Detecting either the FSK (preferred) or the stutter-tone, would allow Asterisk to notify me that there it missed a message. I see periodic queries about

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

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

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

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

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-- 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

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

Hi. I noticed that if new messages are recorded with voicemail2 , they're not detected by the message waiting indicator, so the mailbox=XXXX param has no effect, and no message waiting is sent to the phone (sip & adsi, or stutter dialtone) Any hint? -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl

Hello, We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume environment. At least twice a day there are complaints of 'dropped calls'. Examining the debug logs, I see that in each case, an "on hook" event is detected, followed by the zap channel being hung-up and * saying "BYE" to the sip phone: Jun 23 14:17:22 DEBUG[2441232]: Exception on

Hi I am fairly new to Asterisk. I will have few questions on one Asterisk system: Description: I looked at Asterisk Mall says that "The TrueLine SMB PBX is perfect for the small office providing service for up to 28 Telco lines". This is a 1U Rack Mount Single T1 Asterisk Appliance. Question: Does it mean it has 7 PCI bus and takes 7x4=28 channels (if they used 7 TDM400P card

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

Has anyone seen the following problem? Until recently, I couldn't understand why some extensions on my * system would have a "congestion tone" as soon as I picked up the handset. A little sleuthing through the logs and the source code led me to understand that * thought it had seen the extension go off-hook, send some DTMF tones, and then wait. * treated this situation as a

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

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

The subject says it all... I have an X100P and I have (for now anyway) Bell Canada's Call Answer which will notify you through one of those nifty CLASS signals that you either do or do not have voicemail. This is not only a "stuttering dialtone" but some actual signal passed so CLASS-aware phones can detect it and flash their message waiting indicator. I'd like to detect