Re: [asterisk-dev] AstriCon 2024: February 15th, 2024 - Fort Lauderdale, Florida

On Tue, Sep 5, 2023 at 10:06 AM Joshua C. Colp wrote: > The 25th anniversary of Asterisk is upon us! We’ll be celebrating it at > AstriCon 2024 held on February 15th, 2024 in Fort Lauderdale, Florida as > part of IT Expo. We’d love it if you would join us. You can register > here[1], and if you

Re: [asterisk-dev] Stir Shaken

On Sun, Jun 27, 2021 at 1:43 PM Dovid Bender wrote: > I am as equally confused. They sent me a list of questions including what > IP's we would be connecting from. Has anyone gotten anywhere with them? > This is typically just a security thing on the part of the PA or CA, as they'd like to

Re: [asterisk-dev] Stir Shaken

know that the number presented belongs to your customer, then you can give them an "A" level attestation. If you know the customer but not that they're authorized to use that particular number, you can give the calls a "B" level a

Re: [asterisk-dev] Video IVR

On Mon, Dec 21, 2020 at 4:36 AM Abhay Gupta wrote: > Now that video over LTE is supported by majority of mobile providers do we > have a plan to enable video in Playback , Record and ARI applications so > that we have a video IVR (Interactive video response ) ? > > Or do we have a way where this

Re: [asterisk-dev] Module Deprecation, Default Not Building, and Removal

On Mon, Nov 9, 2020 at 8:24 AM Joshua C. Colp wrote: > Since this is the first real time formalizing this once all the things are > in place (process documented on wiki, deprecation list created from > existing state of things) I'll likely send out an email to -users and also > post on the

Re: [asterisk-dev] Module Deprecation, Default Not Building, and Removal

On Tue, Oct 6, 2020 at 2:23 PM Joshua C. Colp wrote: > As a packager and someone who has been in the community and user world, > what's your opinion and thoughts on the 2 year strategy? > I'm fine with it... for faster-moving distributions (such as Fedora), users are used to following new

Re: [asterisk-dev] Module Deprecation, Default Not Building, and Removal

On Fri, Oct 2, 2020 at 11:50 AM Dan Jenkins wrote: > sorry, I thought I was agreeing with you :) we need to engage package > maintainers to potentially help ease the shift - if packages are a > thing but as far as I'm concerned most package managers have out of > date versions of Asterisk,

Re: [asterisk-dev] Module Deprecation, Default Not Building, and Removal

On Thu, Oct 1, 2020 at 10:04 AM Joshua C. Colp wrote: > Not really, and I think part of the problem is that this entire thing > hasn't really been documented, communicated, or been a strict part of the > release or development process. It's been more organic. Going forward it > would be

Re: [asterisk-dev] Module Deprecation, Default Not Building, and Removal

On Thu, Oct 1, 2020 at 9:20 AM Joshua C. Colp wrote: > 1. All the changes listed below initially occur in standard releases - in > my opinion beginning the process to remove a module is a big thing and we > should gradually introduce it, gaining feedback from those who run standard > releases

Re: [asterisk-dev] Proposal for New Major Version Process Change

ix weeks is more than adequate, given the maturity of the project. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists

Re: [asterisk-dev] PJSIP_MEDIA_OFFER

; endpoint's configuration? > That has always been my expectation -- that PJSIP_MEDIA_OFFER completely overrides the configured codecs -- but that may be a misunderstanding on my part. That being said, I can't think of a reason why it wasn't coded that way.

Re: [asterisk-dev] Asterisk and CentOS 8

ive them a shot and give me feedback. Over the next few weeks, I'll try to get these packages pushed into EPEL proper. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-d

Re: [asterisk-dev] [Asterisk 16.x / pjsip TLS] Memory leak with core reload

On Thu, Oct 3, 2019 at 10:01 AM Sean Bright wrote: > In the future, please feel free to skip the mailing list and submit > issues directly to https://issues.asterisk.org/jira for any Asterisk > problems. > > FreePBX issues like this one can go directly to their issue tracking > system (I don't

Re: [asterisk-dev] Unable to compile asterisk12 with MALLOC_DEBUG

You do realize that Asterisk 12.x is past it's End of Life, and is no longer receiving security updates or bug fixes, right? You should try with the latest 13.x or 16.x release. -Jared On Sat, May 25, 2019 at 3:34 PM bala murugan wrote: > Hi , > > I am trying to compile asterisk12 with

Re: [asterisk-dev] Dynamic server side event type filtering in ARI

On Fri, Dec 14, 2018 at 12:34 PM Kevin Harwell wrote: > I'm not a REST expert by any means, but I thought POST aligned more with > create and PUT create/update. But I guess since we are working on the whole > list then we can get away with just a PUT, and as you say it will add or >

Re: [asterisk-dev] The "Busy" App.... isn't.

On Wed, Apr 4, 2018 at 7:17 PM, Richard Mudgett wrote: > The argument is used when the channel is already answered. The channel > will then send > the busy tone inband for the specified number of seconds and hangup. The > behavior also > depends upon the channel driver. >

Re: [asterisk-dev] RPM Repository for Asterisk/Kamailio

ules. > As one of the maintainers of the Asterisk and dahdi-tools packages in Fedora and EPEL and a long time RPM packaging fanatic, I'd really like to collaborate together on the spec files you're using. -- Jared Smith -- _ -- Bandwi

Re: [asterisk-dev] Attn: pjproject packagers

ora Rawhide now. Once it's got a little more testing, I'll push updates back to earlier releases and to EPEL. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] Proposal to bring pjproject back into the fold

h dynamic linking? Thoughts? Ideas? Improved testing? More robust code? -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-dev] Proposal to bring pjproject back into the fold

On Tue, Jan 19, 2016 at 9:30 AM, George Joseph <george.jos...@fairview5.com> wrote: > ​I understand the packaging issue and I'd like to hear from packagers like > Jared Smith. Not sure what to say here -- bundling pjproject with Asterisk causes me a world of hurt from a packagin

Re: [asterisk-dev] Asterisk Docker Containers: Phase 1

hose releases. If it's easier, I'd be happy to setup some repositories for newer versions of Asterisk on RHEL/CentOS 6 and RHEL/CentOS 7. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-dev] Asterisk Docker Containers: Phase 1

get a tighter feedback loop between the Asterisk development community and the packagers in Fedora/RHEL/CentOS/etc. -- Jared Smith -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ast

Re: [asterisk-dev] [Code Review] 4573: trunk: Can't touch this

pants have an shorter inseam that begins around the knees. Also, the command contains no XML documentation. - Jared Smith On April 1, 2015, 3:59 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit

Re: [asterisk-dev] [Code Review] 4544: clang compiler warning: -Wself-assign

comments are about the coding standards, the meat of this patch is very straightforward, and shouldn't cause any problems. - Jared Smith On March 27, 2015, 12:02 p.m., Diederik de Groot wrote: --- This is an automatically generated e-mail

Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

and ideas. If people feel threatened or belittled, they're much more likely to be hostile to your opinions or ideas. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] Queue discussion at Astricon

On Fri, Oct 17, 2014 at 9:44 AM, Matthew Jordan mjor...@digium.com wrote: I'd love to discuss how things have been going. Maybe after AstriDevCon? I'd love to discuss things as well, but will only be around on Tuesday, before I have to fly off for more conferences. I like the idea of getting

Re: [asterisk-dev] [Code Review] 3540: chan_local+app_dial: Propagagate call answered elsewhere over local channels.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3540/#review11888 --- Ship it! Looks simple and straightforward to me. - Jared

Re: [asterisk-dev] Dundi library

library in Perl. More details at https://github.com/heytensai/dundi and on his blog post at https://www.zmonkey.org/blog/content/perl-dundi-library. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-dev] [Code Review] 1164: [patch] Improve debug of ast_hangup

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/1164/#review11582 --- Looks fine to me at first glance. - Jared Smith On April

Re: [asterisk-dev] [Code Review] 3343: res_pjsip: Enable DNS support.

not be perfect, but I think there are more serious sins than allowing an end user to shoot themselves in the foot. (We already give them plenty of firearms and ammunition with the wide range of configuration options and files and modules available today.) -- Jared Smith

Re: [asterisk-dev] CentOS packaging

could just get a patch to have pjproject use the system version of ilbc instead of the bundled copy, I'd be a happy man. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing

Re: [asterisk-dev] CentOS packaging

/show_bug.cgi?id=728302, but I've got pjproject 2.2 packaged up and in fairly good shape. I still need a patch to use the system copy of ilbc instead of the bundled version, but other than that, I think we have something that could be approved for Fedora and EPEL. -- Jared Smith

Re: [asterisk-dev] CentOS packaging

the Asterisk fork of pjproject packaged up in Fedora/EPEL shortly, so that we can begin getting Asterisk 12 packaged. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] CentOS packaging

-portaudio asterisk-postgresql asterisk-radius asterisk-skinny asterisk-snmp asterisk-sqlite asterisk-tds asterisk-unistim asterisk-voicemail asterisk-voicemail-imap asterisk-voicemail-odbc asterisk-voicemail-plain -- Jared Smith

Re: [asterisk-dev] CentOS packaging

. -- Jared Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] CentOS packaging

with getting better Asterisk packages available for RPM-based systems, and am willing to put some of my limited spare time into improving the situation. (I'd love to help for dpkg-based systems as well, but I don't know enough about dpkg to code myself out of a wet paper bag.) -- Jared Smith

Re: [asterisk-dev] measure call quality for performance test

look up RTCP and start looking to see if your endpoints will send RTCP reports. I've found this is a first good step in monitoring call quality. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http

Re: [asterisk-dev] Unstable releases lately

in the first place. If you have questions or concerns, drop me an email and I'll do what I can to help you. [1]http://lists.digium.com/pipermail/asterisk-biz/2008-January/024819.html -- Jared Smith Community Relations Manager Digium, Inc

Re: [asterisk-dev] Implementation of Broadsoft Sip Access in Asterisk to enable SLA for Sipura/Linksys

, the item that was sent to list says it's confidential and can only be distributed under license. I hate to imagine the problems it might cause the Asterisk developers to use it without proper permissions. In short, the Boy Scout in me is saying Let's be careful and play by the rules! -- Jared Smith

Re: [asterisk-dev] MeetMe DTMF menus

(or perhaps *) for adjusting volume as presently. Personally, I think it's a great idea -- you've got my vote. But since I'm not really one of the Asterisk developers, I'm not sure my opinion counts for much. --- Jared Smith Community Relations Manager Digium, Inc

Re: [asterisk-dev] Tab-completion borked in trunk

channel name, so I can't cut/paste to sip show channel blah-320932 for debugging.) It's not at all obvious, but the output of core show channels concise may give you the information you're looking for. At least, it's worked for me in the past. -- Jared Smith Community Relations Manager

Re: [asterisk-dev] Thoughts on Asterisk release management

On Thu, 2007-09-20 at 09:30 -0500, Power, Paul C. wrote: I would like to see some test harnesses exist to really wring out various aspects of atserisk code. I would do what I could to donate a machine to run regular testing. I've offered to help coordinate such an effort (run by the community,

Re: [asterisk-dev] Xorcom claims

On Mon, 2007-08-06 at 14:36 -0400, Andrew Latham wrote: Just for everyone else out there, where does a person buy/get these cables or should we just make our own? You can buy them from Digium. -- Jared Smith Community Relations Manager Digium, Inc

[asterisk-dev] AstriCon -- Last chance to speak!

on the right-hand side of the page. If you know of someone who we should invite to speak, we're open to suggestions as well. Feel free to email me (off the list, please!) with your recommendations. I'm looking forward to a great conference, and hope to see you all there! -- Jared Smith Community

Re: [asterisk-dev] SIP Missed Calls when calling SIP phones parallel

On 6/21/07, Olle E Johansson [EMAIL PROTECTED] wrote: Yes, this is my code. But please check out the branch and try it out instead of sending out random patches. I've worked with Frank Sautter who has been helping me test this and I have implemented other changes since the code you send out.

Re: [asterisk-dev] Docs converted to TeX?

On 5/22/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Have you thought about using DocBook? http://www.docbook.org/whatis I've advocated using DocBook for quite a while, but it's always turned into a my favorite is better than your favorite religious war. Items I can add to the discussion are:

Re: [asterisk-dev] Enhanced AGI

On 4/16/07, Loic Didelot [EMAIL PROTECTED] wrote: So my question is do I have access to the sound channel and how can I debug? I try to open file descriptor 3 but writing to it does not give me anything. EAGI only allows you to *read* from file descriptor 3 for inbound audio, but does not

Re: [asterisk-dev] Lost Query with MySQL

On 3/22/07, C. Savinovich [EMAIL PROTECTED] wrote: Hello everyone, I am writing a C module that uses a SELECT query to a mysql server upon starting a phone call... somehow the query works the first time or two, afterwards, it doen't, although connection is still on... any hints or guesses

Re: [asterisk-dev] Re: [svn-commits] kpfleming: trunk r50538 - /trunk/main/channel.c

On 1/12/07, John Todd [EMAIL PROTECTED] wrote: While I understand the sentiment here, I'm not sure this is a good idea. This builds in a 500ms post-dial delay issue into every call. I've been building systems for three years now, and everywhere there is an Answer (which, I believe, should be

Re: [asterisk-dev] Re: [svn-commits] kpfleming: trunk r50538 - /trunk/main/channel.c

On 1/12/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: Note that the Answer() argument was also mis-documented; it was already a post-answer delay, and I've updated all three branches to reflect that. I thought it was, but then I went back and looked at the documentation and it said before, so

Re: [asterisk-dev] daemonising after possible errors

On 12/7/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Something that I find bothering when I try to debug Asterisk is that it deamonizes before most errors can occour. It will fork into background before many potential fatal errors occour. Such fatal errors are in the module loading time.

Re: [asterisk-dev] IAX2 very CPU hungry

On 11/22/06, Loic DIDELOT [EMAIL PROTECTED] wrote: it really is a lot better in 1.4. One of the things that Asterisk 1.4 added was a multi-threaded IAX implementation (thanks Mark!). From the little bit of testing on it that I've done, it's made a huge improvement on how well IAX works under

Re: [asterisk-dev] OriginateSuccess and OriginateError incomplete (in 1.4 beta3)

On 11/24/06, dmb [EMAIL PROTECTED] wrote: Is that correction included in the next beta or in the release candidate? I suggest you file a bug in the bug tracker, so that your request doesn't get lost. -Jared ___ --Bandwidth and Colocation provided by

Re: [asterisk-dev] Bug marshals - won't you take a look at 8188

On Thu, 2 Nov 2006, Stephen Davies wrote: I posted up a change that gets chan_iax2 to log jitter buffer stats into the logs regularly during active calls.I've also responded to the bug (#8188), but I'll repost my comments below as they're likely to get a greater audience of developers:

Re: [asterisk-dev] Re: SIP to IAX (Jeremy McNamara)

with the wiki... -JaredOn 10/6/06, John Lange [EMAIL PROTECTED] wrote: On Thu, 2006-10-05 at 16:30 -0400, Jared Smith wrote: On 10/5/06, John Lange [EMAIL PROTECTED] wrote: FYI, there's still work being done as part of the Asterisk Documentation Project, although a lot of it is going on behind

Re: [asterisk-dev] Configuration for recording quality?

On Fri, 2006-08-04 at 13:42 +0200, Jan du Toit wrote: I have recently posted a mail on the users mailing list, asking around how to change the quality setting of files that asterisk record for you. For instance change the 8kHz for meetme recordings to 32kHz. Unfortunately, this list isn't the

Re: [asterisk-dev] Patch to app_queue to add annoucements to the caller

On Tue, 2006-07-04 at 15:15 +0200, Tristan wrote: Hi, here is a patch against 1.2.9.1 to add annoucements to the caller... Thanks for submitting a patch! It's always good to have more people contributing to the code base. Instead of posting your patches to the mailing list, please post them to

Re: [asterisk-dev] extra parameter for DB read function

On Wed, 2006-05-31 at 14:30 +0100, Julian Lyndon-Smith wrote: There are often times that I want to read a DB value from the dialplan, and if this family/key pair does not exist, set it to some default value. for example: 1234,1 = Set(EMAILADDR=${DB(x/y)} 1234,2 = GotoIf($[${EMAILADDR} =

Re: [Asterisk-Dev] Voicemail to email volume change patch

On Fri, 2006-01-13 at 11:15 -0600, Aaron Daniel wrote: My boss and I have been working on a patch to the voicemail code, and I'd like to see what everyone thinks of it. I'd like suggestions and stuff on anything that needs to be changed, as this is the first time we've patched the code,

Re: [Asterisk-Dev] The Zaptel init scripts must die!

On Tue, 2005-12-13 at 12:50 -0500, Greg Boehnlein wrote: I may also modify the zaptel Makefile at a later date to allow you to to do make config (same as Asterisk). I'm shooting from the hip here, so don't quote me on this -- but if I remember correctly, this is already supported in the

Re: [Asterisk-Dev] Patch to allow Dial() to ignore call forward

On Tue, 2005-11-08 at 11:57 -0600, John Lange wrote: The following patch applies against 1.0.9 and adds another option to the Dial command. Would you please add this patch to the bug tracker, and if possible, create one against CVS HEAD? Asterisk 1.2 is getting quite close to being released,

Re: [Asterisk-Dev] Send tone to caller on answer

On Mon, 2005-10-31 at 08:39 -0800, Ed Greenberg wrote: I've been asked to configure Asterisk to send a tone back to the caller when the call is answered - to indicate answer supervision. Has anybody done this before? Is there interest in it? Am I missing something? My gut reaction would

RE: [Asterisk-Dev] getting started

On Sun, 2005-09-11 at 18:41 -0400, Sherwood McGowan wrote: Hey all, I wanted to start getting in there too, but dev.asteriskdocs.org does not resolve under my 6 different DNS servers. Is there a mistake here? Yes, I changed DNS on the asteriskdocs.org domain to point to a new server until we

Re: [Asterisk-Dev] help needed

On Mon, 2005-07-25 at 16:34 +0200, Hoai-Anh Ngo-Vi wrote: But Asterisk didn't really launch that a.out programm (I didn't get any message via CLI console, that programm would have put some messages into CLI console via stderr if it ran correctly). Don't be too sure. In most cases,

Re: [Asterisk-Dev] chan_sip crash w/ Refer [patch]

On Tue, 2005-06-14 at 00:09 -0400, Jared Mauch wrote: Index: chan_sip.c === RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.759 diff -u -r1.759 chan_sip.c --- chan_sip.c9 Jun 2005 22:41:18

Re: [Asterisk-Dev] Loss of functionality with depreciation ofDBGet/Put?

On Tue, 2005-05-17 at 10:04 -0500, Matthew Boehm wrote: /me raises hands and directs the chorus of angels to sing: Hallelujah! (geez..its only tuesday?) While I agree that this is a great idea, I'm a little concerned about communicating this to the Asterisk community at large -- Is there any

Re: [Asterisk-Dev] rfc2833 DTMFs sent with bad timestamps (patch)

your patch in this mailing list so that it can be discussed. On the technical side, it looks pretty straightforward to me. Does anybody with more Asterisk coding experience have anything to say about this patch? -Jared Smith ___ Asterisk-Dev mailing list

Re: [Asterisk-Dev] variable sample period?

On Wed, 2005-02-23 at 16:14 +0100, Peter Svensson wrote: For ulaw/alaw on a local connection between machines 10ms or even less may be perfectly acceptable, especially combined with trunking. If you're using IAX2 trunking with a codec frequency other than 20ms, you'll also want to set the

Re: [Asterisk-Dev] Reload crash backtrace

For what it's worth, I've seen a lot of crashes on restart nows as well. Jared On Mon, 2003-08-18 at 13:41, Tilghman Lesher wrote: This was an odd crash. Crashed on a reload. Current CVS: #0 __ast_context_destroy (con=0x0, registrar=0x404ad5ad pbx_config, lock=1) at pbx.c:3983 #1