similar to: Pulse dialed digit recognization

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

Hello, I've discovered that in Italy ISDN lines can be programmed to generate a "billing pulse" every n seconds (it dipends from the pricebook). The pulse has these figures: frequency .................................................................... 12 kHz ? 1% level .......................................................................... 200 mVrms on 200

Just a quick note that the priority in the Pulse AO plugin (ao/src/plugins/pulse/ao_pulse.c) should be changed from 41 to 50 (in the ao_pulse_info struct). Since the plugin now ships as part of libao, its priority should be a factor of 5, as that's how I designed the plugin priority system. Plugins that ship with libao should be a factor of 5, so that 3rd party plugins can fit in between

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port I get a motorboating sound or warble - or - just not clear audio. When I switch that to ALSA direct it sounds just fine. What might be happening with pulse audio that it does not sound clear??? asound.conf below. Thanks, Jerry more /etc/asound.conf # # Place your global alsa-lib configuration here... # @hooks [ {

I am looking for some great instructions on using asterisk with pulse. I'm using centos 7 and pulse as a user and not having much luck. I have changed all permissions for the asterisk directories. set asterisk.conf user and group to be my user that is running. No go. Anyone done this? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I know that wine currently doesn't support the pulse audio, but can we expect this feature in upcomming versions? While I used other audio drivers I had a lot of sound problems with other applications, pulse audio works perfectly so it would be nice to add a support for it. Cheers!

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards

Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

Hello. I have not seen that asterisk software have a possibility to dial pulse on outgoing calls. Don't you know is there any plan to do it? Thanks. Good luck. Lev.

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

hi guys, Most of the conversations about the Pulse newsletter have gone away ( and I suspect to private emails ). Can those be brought back here to this list, if not - what needs doing to make them shareable ? - KB -- Karanbir Singh : http://www.karan.org/ : 2522219 at icq

I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo

Thanks for your suggestion. I have compiled according to http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse +dialing dialing at 10 pps works fine with Asterisk with the newly compiled wctdm. but when I dial at 20 pps, the pulses cannot be decoded correctly. I tried changing the make/break ratio but dialing at 20 pps still has the decoding problem. Does anyone have any

I have three time series, x, y, and z, and I want to analyse the relations between them. However, they have vastly different resolutions. I am writing to ask for advice on how to handle this situation in R. x is a stimulus, and y and z are responses. x is a rectangular pulse 4 sec long. Its onset and offset are known with sub-millisecond precision. The onset varies irregularly -- it doesn't

Hi all, I have started working on the next Pulse issue #0902 and therefor copied the template to the Editing section. I would like to contribute the following articles: A short introduction of the Artwork SIG Getting started with Multimedia on CentOS Also I would suggest to create a monthly newsletter instead of bi-weekly, as we are all very busy, I guess ;) It would be great if ppl like Russ

Just read a post that Dag has resigned from the CentOS project... Is that the end of the CentOS Pulse newsletter, too?

Hi, I am pleased to announce the first edition of the bi-weekly CentOS newsletter which we dubbed "CentOS Pulse". This first issue centers around improving communication within the CentOS community and how that relates to the CentOS Promo SIG. We also look at the recent announcements regarding the CentOS LiveCD and the CentOS Directory Server. And dive into interesting community

I have a situation where I am receiving DID calls using Immediate Start Pulse signalling on a Loop Start trunk. The line terminates on a Newbridge Mainstreet 3624 channel bank, which provides battery etc. The channel is converted and routed to Asterisk. The lines are configured as follows: /etc/asterisk/zapata.conf ; Channels 1-24 service MainStreet 3624 channel bank context=infrom-did group=1