similar to: unable to enable stutter dialtone

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 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 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

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

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

We have 1 queue called helpdesk and are setting up a second one called isp. The helpdesk queue is for internal support calls and isp for our ISP customer calls. Both of these queues will be directed to the same agents (helpdesk phone extensions). We want to have the separate queues for tracking purposes but the queued calls need to be ordered and answered as if there was only one queue. For

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

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

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 /

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 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

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

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

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option

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

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

On Sun, May 26, 2013 at 5:38 PM, Gard Spreemann <gspreemann at gmail.com> wrote: > Hi, > > If I should contact a list instead of you directly, I sincerely apologize! You should have cc'ed relevant lists at least ;-) Fixed up. > After a recent kernel upgrade, I found that my system (using Nouveau) would > stutter whenever eDP-1 was turned off. That display being turned

I am not sure if this is the place for Digium user-to-user discussion, but... We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no dialtone and no registration with the server. Has anyone else experienced high failure rate

As far as I can tell, in order to have caller ID show up for calls from other internal phones, I have to set the caller ID on each channel in zapata.conf. This is tedious, and redundant (since Asterisk knows which extension is making the call, and it could look up the name from the voicemail configuration--if the extension matched the mailbox). Is there a way to set the caller ID for these calls