similar to: Using stutter dialtone like the PSTN does

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

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

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

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 /

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

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

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

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

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

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, 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

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

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

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

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

asterisk-users-request@lists.digium.com is believed to have said: >I don't mean to be rude to everyone who responded to this question, but >I think that everyone is answering the wrong question. The point is that >the message waiting indicator doesn't light up, at all, ever. All that >happens when messages are waiting is that the display blinks and the >phone gives a

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,