similar to: Feature Request for 1.10 - ISDN power-save mode

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to write out the audio frames

On Oct 22, 2014, at 11:47 AM, BJ Weschke <bweschke at btwtech.com> wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: >> >>> Paul Albrecht wrote: >>>> Really? Shouldn?t something this major affecting the entire Asterisk >>>> community get discussed on the lists?

In putting together the SoundPack code, I am looking at the various language/locale specific code, and wondering how it all really stands... So, share with me, non-English speakers, what is your experience and impression? I heard a few comments during AstriDevCon, that some of the languages are not quite right; some said their language was understandable, but... Would anyone be willing to share

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After

Really? Shouldn?t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial plan model to external control? Here?s a link to the notes posted on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2014

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Source: http://www.nanpa.com/number_resource_info/vsc_assignments.html See also: http://bugs.digium.com/bug_view_page.php?bug_id=0000071 Some (which?) of the codes below are hardcoded into Zap channels only. Is there a European equivalent for this (or ITU / IETF)? Greetings, Philipp VERTICAL SERVICE CODES (US Standard) ASSIGNMENTS *00 - Inward Voice Activated Services (English) *01 -

Rather hasty I think. I think whatever version 1.2.X winds up on should be the most stable release of Asterisk, period. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Olle E Johansson > Sent: Tuesday, May 29, 2007 5:19 AM >

I see the following paragraph in the Asterisk trunk LICENSE file: "In addition, Asterisk implements two management/control protocols: the Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface (AGI). It is our belief that applications using these protocols to manage or control an Asterisk instance do not have to be licensed under the GPL or a compatible license, as we believe

As we plan for our 10th AstriCon, which will be in Atlanta, GA the week commencing October 7th, we want to make sure that our conference sessions are the best they've ever been! That's why we need YOU to submit a speaking proposal - to share you experiences and ideas around Asterisk! The best sessions at AstriCon are always those from the Community that show what you've been up

? Hello-- Boy, it's been a long time since I posted to the user mailing list! Pardon me, I've posted this to dev also, but I thought the general users should also be aware of this. I'd like to announce a proposal to the Asterisk Community, that I introduced at Astridevcon last month. It is a new API for playing sound files (mainly speech). A pdf describing the Proposal in some

On Wed, Oct 21, 2020 at 7:46 AM Michael Maier <m1278468 at mailbox.org> wrote: > Hello! > > On 20.10.20 at 14:00 Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > Asterisk 18.0.0. > > This release is available for immediate download at > > https://downloads.asterisk.org/pub/telephony/asterisk > > I

Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation

Le 11/02/2020 ? 14:11, Jonathan Billings a ?crit?: > I've mentioned on this list countless times about how NetworkManager > is actually pretty good for a general server. Automatic link > detection and activation/deactivation, a dispatch service on link > activation/deactivation, support for bringing up secondary interfaces > after a primary goes up, a dbus interface for

Hello everyone! Asterisk is a large project and is used all over the world. As a result, the Asterisk project is lucky to have a large and thriving user community that communicates in a variety of places: IRC channels, forums, mailing lists, social media, and more. We in the Asterisk community value that diversity - the Asterisk project is stronger due to our many different experiences and

Hi list, I'm configuring apache with https and I've a question about sslv3 deactivation. Running "openssl ciphers -v" I get a list of cypher suite of openssl like: ECDH-RSA-AES128-GCM-SHA256 TLSv1.2 Kx=ECDH/RSA Au=ECDH Enc=AESGCM(128) Mac=AEAD ......... Each lines report relative protocol. Disabling sslv3 with "SSLProtocol all -SSLv3" I can use cypher like:

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

Hi together, i've read a few mails here about the following problem, but the solution comes not clear to me. I'm using SuSE 7.3 (kernel 2.4.10), and samba 2.2.2. Following problem appeared: In the standard settings for pam.d/login,passwd,other and samba SWAT works well. Swat is access able as root and as user. Changing the options in /etc/pam.d to use md5-hashes in shadow, SWAT denies