similar to: Pulse dial: outbound?

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.

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

Good day, does anyone have pulse dialing working ? http://voip-info.org/tiki-index.php?page=Asterisk+zaptel+pulse+dialing At the link above there is a statement: configuration for European telephone lines will look like: make_time=63 break_time=37 pause_time=800 So where these pamameters should go to ? zapata.conf ignore them. I have tried on both version of asterisk (cvs and stable) with

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

Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a "pulse dial" telephony. Could * detect digits dialed on pulse telephones? Daniel

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

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

I really should be doing something better on this beautiful weekend, but I'm trying to save myself some time for later projects by documenting some things that have been particularly troublesome in the past. That being said... I've written up a configuration guide for the Cisco ATA-186, which describes some of the features that are possible to set in the ATA and specifically what

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!

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:

I prefer to use Pulseaudio, but since apperantly I'm not allowed to do that anymore if I want to hear things out of Wine, I removed it for now. And guess what, it still doesn't work. I have two sound cards, the HDMI one in my graphics card and my regular one hooked up to the speakers, which is why I like Pulse because I can easily send different apps to different cards. Here's my

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

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

Hello all - Second-to-last notice to collect RSVP's for interested parties for the Asterisk Birds of a Feather dinner/beer meeting. There are five people signed up for this at the moment. Looks like it's going to be the 23rd of September, since that fits well with the VON conference schedule. Please email me directly for details. JT >Date: Thu, 21 Aug 2003 01:52:52 -0700

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

John, I spoke with Level(3) last week regarding SIP termination. They quoted $0.01/minute, with an 11 Million Minute / Month minimum. Ugh! -dg -------------------------------------------------------------- Darnell Gadberry President binaryMedia darnell AT binmedia DOT com ------------ Date: Fri, 19 Dec 2003 21:12:22 -0500 To: asterisk-users@lists.digium.com From: John Todd

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

OK, that "VOIP dialtone?" thread was getting really out of hand, so I'll condense my answers into one big ugly message: 1) 911 service. Yes, that is one of three reasons to keep your PSTN line. The other two reasons are: Inbound calls from local callers still should work on a POTS line, for now. You can't find VOIP providers in most area codes, so you'll most