Re: [asterisk-dev] Packet Loss Concealment in confbridge

On Thu, Oct 14, 2021 at 2:37 PM Pascal Cadotte wrote: > > Hello everyone, > > We've been trying to improve the quality of our video conferences using > confbridge. We've been able to figure out how to get the video usable for all > participants even for users using bad internet connections

[asterisk-dev] Open Position on Asterisk Development Team

Hey All, Sorry to interrupt your regularly scheduled Asterisk development related discussions. Just wanted to let any of you that may be interested know that we have an opening on the Asterisk Development team at Sangoma. I figured this might be a good place to start with to find people who

Re: [asterisk-dev] Off next week

Sorry about that guys, sent that out to the wrong mailing list. My desire remains the same though - happy holidays to all of you :-) Matthew Fredrickson On Thu, Dec 26, 2019 at 2:58 PM Matt Fredrickson wrote: > > Hey All, > > Assuming there's no significant fallout from the

[asterisk-dev] Off next week

Hey All, Assuming there's no significant fallout from the key updates of this last week, I'm going to take off all of next week to spend time with the wife and kids. If there are any emergencies, my mobile number is on the wiki. Hope you all are doing well, and have a good holiday. -- Matthew

[asterisk-dev] Message from Sangoma's CEO

Hey All, Bill Wignall, Sangoma's CEO (and the corporate sponsor of the Asterisk project) has asked that I forward information about a blog post that he wrote to the open source community. You can find information about it here:

Re: [asterisk-dev] Video calling

On Fri, Nov 15, 2019 at 7:38 PM Troy Bowman wrote: > > On Fri, Nov 15, 2019 at 3:56 PM John Kiniston wrote: >> >> I do not recommend using chan_sip, chan_sip is no longer receiving >> development. >> chan_pjsip is where the development focus is at. > > > Sure, chan_pjsip is where the feature

Re: [asterisk-dev] Advice of Charge (AOC) in PJSIP

Hey Alistair, On Fri, Nov 15, 2019 at 9:32 AM Alistair Cunningham wrote: > > We're currently using Advice of Charge (AOC) with chan_sip. Since chan_sip is > becoming deprecated, we'd like to get this feature added to PJSIP. What would > be needed to have this added? My suggestion would be to

Re: [asterisk-dev] AstriDevCon ideas

Thanks for posting your thoughts Dan! You'll be missed this year. Matt On Thu, Oct 24, 2019 at 12:20 PM Dan Jenkins wrote: > > Thanks Josh! I'd seen the no dialplan needed but hadn't seen the move > functionality (woot!) > > On Thu, 24 Oct 2019, 18:09 Joshua C. Colp, wrote: &g

[asterisk-dev] AstriDevCon Reminder

Hey All, Just a quick additional reminder - AstriDevCon 2019 signup is open [1]. AstriDevCon will be on Monday, October 28th, at The Omni at The Battery in Atlanta. If you would like to attend, discussion is intended to be focused on developmental and project level topics. Seating space is

Re: [asterisk-dev] Memory leak since Asterisk 16.5.x / pjsip

On Mon, Sep 16, 2019 at 12:39 PM Andreas Wehrmann wrote: > > > >> BTW: I'm not really happy with the fact, that an existing LTS / stable > >> version gets a new pjsip version "on the fly". From my point of view, > >> this should have been done > >> during a normal development cycle and not during

[asterisk-dev] AstriDevCon 2019

Hey All, It's that time again! AstriDevCon 2019 is fast approaching and signup is now open[1]. AstriDevCon will be on Monday, October 28th, at the Omni Hotel at the Battery in Atlanta, Georgia. If you would like to attend, discussion is intended to be focused on developmental and project level

[asterisk-dev] Asterisk 17 Feature Freeze Deadline Today

Hey Everybody, As previously announced, today is the last day that you can submit a code level contribution to gerrit and have it included in the first version of Asterisk 17. Historically, we have been lenient as to the definition of when "today" ends, but generally speaking it will be tonight

Re: [asterisk-dev] Audio to/from Asterisk

On Wed, May 15, 2019 at 7:13 PM Sylvain Boily wrote: > > Hello, > > On 2019-01-25 4:14 p.m., Sylvain Boily wrote: > > Hello, > > > > On 2018-10-16 3:18 p.m., Sylvain Boily wrote: > >> Hello, > >> > >> On 2018-10-15 3:29 p.m., Matt Fredri

[asterisk-dev] Asterisk 17 Feature Freeze Announcement

Hey Everybody, Just a friendly reminder that today is the official two month notification [1] prior to feature freeze on Wednesday July 17th, 2019. For those who are not aware, we try to "feature freeze" the master branch in order to test and stabilize the code base prior to release of a new

Re: [asterisk-dev] No MTU discovery and packet sizing in DTLS implementation by RTP engine (res_rtp_asterisk) which leads to IP fragmentation

On Fri, Apr 19, 2019 at 5:29 AM Mohit Dhiman wrote: > > As per the RFC 4347 section-4.1.1 > >Each DTLS record MUST fit within a single datagram. In order to >avoid IP fragmentation [MOGUL], DTLS implementations SHOULD determine >the MTU and send records smaller than the MTU. DTLS

Re: [asterisk-dev] Adding new ARI for application execute

On Tue, Apr 2, 2019 at 6:05 PM Kevin Harwell wrote: > > On Tue, Apr 2, 2019 at 4:01 PM Sungtae Kim wrote: >> >> So, here's some API draft. >> >> I will introduce 2 new ARI requests and 1 ARI event. > > > ARI is meant to be an alternative to diaplan applications, and not act as a > layer on top

Re: [asterisk-dev] Adding new ARI for application execute

On Mon, Apr 1, 2019 at 2:23 PM naruto360 wrote: > > Hello Are you talking to me about the problem that was asked by me They're having a discussion about Asterisk ARI development. ARI is Asterisk's REST API. Matthew Fredrickson > > في الاثنين، ١ ابريل، ٢٠١٩ ٨:٤٣ م Joshua C. Colp كتب: >> >> On

Re: [asterisk-dev] Looking for help with gerrit for submitting an issue fix for review

On Fri, Mar 29, 2019 at 3:18 PM Dan Cropp wrote: > > I have an issue fix I would like to commit for review. > > > > I was able to retrieve the asterisk master code from gerrit, edit the code, > and put it through tests. > > I believe I have everything completed with the git message. > > > > When

Re: [asterisk-dev] fastAGI vs CURL

nowing that I have explained the problem to your security team > > > > > > > > في الثلاثاء، ٢٦ مارس، ٢٠١٩ ٩:٣١ م Matt Fredrickson كتب: >> >> Hey Naruto, >> >> Sounds like we're having some communications misunderstandings. I >> sent you a reply to

Re: [asterisk-dev] fastAGI vs CURL

Hey Naruto, Sounds like we're having some communications misunderstandings. I sent you a reply to your email that you sent to secur...@asterisk.org about this. Let's handle this there. Matthew Fredrickson On Tue, Mar 26, 2019 at 1:56 PM naruto360 wrote: > > But what you say is not true this

Re: [asterisk-dev] Which steps to add Clearmode (rfc4040) passthrough support ?

It seems like this question has come up once or twice in the past 10 years or so. Replies are inline, below. On Thu, Mar 21, 2019 at 4:31 AM Olivier wrote: > > Hello, > > I'm working on request to support SIP trunking with IAD boxes connected to > legacy PBXs. > Those PBXs are using ISDN for

Re: [asterisk-dev] dahdi_span_assignments fails because of missing "spantypes"

On Wed, Mar 13, 2019 at 4:54 AM Tzafrir Cohen wrote: > > Hi, > > > It seems that a relatively recent change in DAHDI-linux (Was this change made > by me? maybe) broke the userspace interface a bit: as of kernel 4.13 the > dahdi device has the attibute "dahdi_spantypes" instead of "spantypes". >

[asterisk-dev] Asterisk Usage Survey

Hey All, For those of you that do not know me, my name is Matthew Fredrickson and I'm the project lead for the Asterisk project. First off, I wanted to thank all of you that contribute in various ways to the project - whether it be at a developmental level, answering questions on forums and

Re: [asterisk-dev] ConfBridge Bug in Asterisk 16.2.0

On Thu, Feb 28, 2019 at 12:07 PM Daniel McFarlane wrote: > > Hello, > > I've installed and been testing Asterisk 16.2.0 in order to upgrade from > Asterisk 11.15.0 and it seems I found a bug: > > I use AMI to set up Conference Bridge variables dynamically and my code > works perfectly with

Re: [asterisk-dev] Asterisk Video voicemail.

On Tue, Jan 22, 2019 at 8:10 PM John T. Bittner wrote: > Before I post as a bug, I want to make sure it's not something stupid I am > doing wrong. > > > > Have 8845 Cisco video phones setup using chan_sip on asterisk 13 / 16. > Video between the phones works perfect, but If one of the phones

Re: [asterisk-dev] slin Transcode + JB = Console warning spam

On Tue, Jan 22, 2019 at 5:05 AM Steve Davies wrote: > > Hi, > > This is possibly an OLD bug that has existed since at least 1.8.x and > persists into 16.x. > > In asterisk 16, main/translate.c ~line 646 is the following code: > > if (f->samples != current->samples && ast_test_flag(current, >

Re: [asterisk-dev] Idea: make app_agent_pool dynamic

On Sun, Dec 2, 2018 at 11:34 AM Igor Goncharovsky < igor.goncharov...@gmail.com> wrote: > Hello All! > > We are using currently app_agent_pool to make agents login into queues and > get queue calls connected to agents without delay. It is how this module > designed. > > But this module is used

Re: [asterisk-dev] Manipulate re-INVITE response in user written Asterisk module

On Tue, Oct 30, 2018 at 4:50 PM Andreas Wehrmann wrote: > > Hello folks, > > I'm currently writing a module for Asterisk > to make it divert a video stream to an external videoserver. > The scenario is this: > > - some external participant is calling into Asterisk (offering audio+video) > > -

Re: [asterisk-dev] redirecting lost over IAX

On Thu, Nov 1, 2018 at 9:13 AM Kaloyan Kovachev wrote: > > На 2018-10-31 20:44, Matt Fredrickson написа: > > On Mon, Oct 29, 2018 at 10:33 AM Kaloyan Kovachev > > wrote: > >> > >> Hello list, > >> I have followed the example from > >> h

Re: [asterisk-dev] redirecting lost over IAX

On Mon, Oct 29, 2018 at 10:33 AM Kaloyan Kovachev wrote: > > Hello list, > I have followed the example from > https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information > to unconditionally forward a call between two Asterisk instances > connected over IAX, but it seems there

Re: [asterisk-dev] Chief Minister - Government of Telangana : Speed Up The Widening Of Road Under Bridge At Hitech City Railway Station

On Sun, Oct 28, 2018 at 6:01 AM wrote: > > Hey, > > I just signed the petition "Chief Minister - Government of Telangana : Speed > Up The Widening Of Road Under Bridge At Hitech City Railway Station" and > wanted to see if you could help by adding your name. > > Our goal is to reach 84

Re: [asterisk-dev] How about to support multi video tracks of the webrtc sfu?

On Tue, Oct 23, 2018 at 8:21 PM Xiemin Chen wrote: > > Anyone can help? Thanks very much. > > Xiemin Chen 于2018年10月10日周三 下午4:40写道： >> >> Hi there, >> >> I was told that it's not support by asterisk to have multi video tracks of >> one client in the following post. How about if I want to achieve

Re: [asterisk-dev] Voice Messaging Platform

Alright, sounds like the window of constructive answers has closed. Hasan, I don't think any deliberate insult was intended in their responses, but I do think that James and others are technically correct. You're probably better looking for this level of custom development on top of Asterisk as

Re: [asterisk-dev] Audio to/from Asterisk

On Tue, Oct 9, 2018 at 12:09 PM Seán C. McCord wrote: > > Because several people raised the issue at DevCon, I figured it may be worth > mentioning this: app_audiosocket. I haven't submitted it mainly due to the > thought that no one else would fine it interesting. There exist other, >

Re: [asterisk-dev] Not able to see stun logs

On Mon, Oct 8, 2018 at 4:31 PM Balraj Singh wrote: > > Hi all, > Following is my rtp.conf config for Asterisk installed on AWS which also has > a NAT in front of it and using a gateway too. > > [general] > rtpstart=10002 > rtpend=2 > rtcpinterval=9998 > rtpchecksums=no > strictrtp=no >

[asterisk-dev] Asterisk 15 Security Fix Only Mode

Hey All, Just a quick reminder to all of you. As of today, the Asterisk 15 branch transitions into security fix only mode. What this means is that that 15 branch will stop receiving normal bug-fixes/contributions and only receive security patches. For those of you that are on this release,

Re: [asterisk-dev] Problems with reregister schedule

On Tue, Oct 2, 2018 at 4:29 AM Torbjörn Abrahamsson wrote: > > > Hey Torbjorn, > > > > I don't think anybody is intentionally ignoring your post - it's probably > > not being responded to because you are > > working on code in the deep dark belly of chan_sip and any people still > > familiar

Re: [asterisk-dev] Problems with reregister schedule

On Fri, Sep 28, 2018 at 9:06 AM Torbjörn Abrahamsson wrote: > > Hello again! > > > > As no one have provided any insights into this yet, I will combine a bump > with providing some more information… I have unfortunately not resolved this > yet. > > > > I tried to put a custom lock around the

[asterisk-dev] Asterisk 14 EOL Today

Hey Everybody, For those of you that missed my reminder last week, as of today, Asterisk 14 is officially end of life. It now joins the rank of all the great branches that have been left behind - 12, 11, 10, 1.8, etc. :-) Thanks to all of you that put time and effort into the making of the 14

[asterisk-dev] Asterisk 14.x going EOL in a week

Hey All, Just a friendly reminder for those still running 14.x versions of Asterisk. In one week (on Sept 26th, 2018) the 14.x branch will finally transition into the unsupported stage of its life cycle. It's already been in the state where it has been receiving only security fixes for the past

Re: [asterisk-dev] GPG Key signing

On Mon, Sep 17, 2018 at 11:24 PM James Finstrom wrote: > > I personally am offering to sign GPG keys for the #WebOfTrust at Astricon. > Note this is not an official venture and is not related to my employer or > related projects. Submit a request at: https://t.co/0ti9v9rpMr I will setup >

Re: [asterisk-dev] write my self app. Debug

On Mon, Sep 17, 2018 at 9:00 AM i...@magnussolution.com wrote: > > Hi, sorry bother you. > > I get a new backtrace > > Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. > Program terminated with signal 11, Segmentation fault. > #0 0x7f82ad124c80 in pthread_mutex_lock () from

Re: [asterisk-dev] doxygen offline

<http://doxygen.asterisk.org/trunk/index.html> > >> > >> Was migrated to another domain? > > > > It hasn't been migrated and is on Matt Fredrickson's list to look at. Even > > before now it was not updated in quite a few years. If you need it >

Re: [asterisk-dev] PJSIP and RTP address selection

On Tue, Sep 11, 2018 at 1:51 PM, Jaco Kroon wrote: > Hi, > > I've got a scenario where (when using PJSIP, using chan_sip does what I > expect) PJSIP will advertise one address in the SDP during a > conversation but then start transmitting from another. In my case PJSIP > is advertising

Re: [asterisk-dev] write my self app. Debug

On Wed, Sep 12, 2018 at 6:10 PM, i...@magnussolution.com wrote: > thanks for you help. > > > I try use backtrace. But I no a expert. > > I using cents 7 64x > > gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c > core.13414 > /tmp/backtrace.txt > > warning: exec file is newer

[asterisk-dev] AstriDevCon Reminder

Hey All, Just a quick additional reminder - AstriDevCon 2018 signup is open [1]. AstriDevCon will be on Tuesday, October 9th, at the Omni Hotel in Orlando. If you would like to attend, discussion is intended to be focused on developmental and project level topics. Seating space is limited, so

Re: [asterisk-dev] asking

On Sat, Sep 1, 2018 at 5:31 AM, modou lo wrote: > Hello Dear can i have your helping i'm student and i'm in the end of my > studies i must present a memory on taxation voip/toip services This list is for the purpose of developmental discussion of the Asterisk source code. For discussion of

[asterisk-dev] Sangoma Annoucement

Greetings! For those of you who have not noticed, it was just announced that Sangoma is intending to acquire Digium. Since discussion about this will likely come up, I figured that it would be good to post something here first. For questions about the acquisition and how it impacts Asterisk and

Re: [asterisk-dev] chap_sip guest over tcp + insecure=port

Hey, Responses inline. On Tue, Aug 14, 2018 at 6:16 AM, Jaco Kroon wrote: > Hi All, > > The following bugs refers: > > ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS) if > existing peer from same IP > commit b2c4e8660a9c89d07041271371151779b7ec75f6 > > ASTERISK-27881 - PBX calls

[asterisk-dev] Asterisk 16 Feature Freeze Deadline is Today!

Hey Everybody, It’s everybody’s favorite time again, feature freeze! As you may well recall, today is the last day that you can submit a code level contribution to gerrit and have it included in the first version of Asterisk 16. Historically, we have been lenient as to the definition of when

[asterisk-dev] Reminder: AstriDevCon on Oct 9th, 2018 in Orlando, FL, USA - Asterisk Developers invited!

Hello Asterisk Developers, This year's AstriDevCon will be held Tuesday, October 9th, 2018 at the Omni Orlando[1] in Orlando, FL, United States. Your attendance at AstriDevCon is welcome. AstriDevCon is a developers-only event held in conjunction with the annual AstriCon conference. It

[asterisk-dev] Reminder: Feature freeze in less than a week

Hey All, Just a friendly reminder again :-) The feature freeze deadline for the 16.0.0 release will occur in less than a week. Wednesday, July 18 will be the last day to submit code reviews to gerrit that can be included in the 16.0.0 release. For further details on how this process works, see

Re: [asterisk-dev] Question regarding SIP MESSAGE log verbosity in chan_pjsip

On Tue, Jul 3, 2018 at 7:18 AM, Floimair Florian wrote: > I’m not exactly sure if the current implementation (tested with 15.4.1) of > SIP MESSAGE in chan_pjsip is logging with the correct loglevel. > > > > E.g.: If a SIP MESSAGE is sent to an endpoint (in my case via ARI) where > there is

Re: [asterisk-dev] Dead code around HAVE_VIDEO_CONSOLE

On Sun, Jun 3, 2018 at 11:38 AM, Alexander Traud wrote: > In the script configure, while going through all AST_EXT_LIB_SETUP, I find > more and more code which is dead. That means, external projects are not used > anymore too. This time, this is HAVE_FFMPEG, HAVE_SDL, HAVE_SDL_IMAGE, >

Re: [asterisk-dev] ./configure --with-pjproject-bundled: GitHub changed/s to TLS 1.2-only!

On Sun, Jun 3, 2018 at 9:15 AM, Alexander Traud wrote: > Currently, Asterisk downloads the tarball of the PJProject not from the > original source at Teluu but GitHub. GitHub forces the user to use HTTPs > instead of HTTP. This is already an issue as described in ASTERISK-27665 > because many

Re: [asterisk-dev] Headers: Incomplete and Cyclic Referenced

On Sun, Jun 3, 2018 at 9:47 AM, Alexander Traud wrote: > The Google tool include-what-you-use (iwyu) works in Asterisk (for more > details see ASTERISK-25591). You can pick one single file, it goes > through the code and checks that all #include are > there. I love iwyu, because it helps me to

[asterisk-dev] Test from Digium address

Testing... -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To

Re: [asterisk-dev] Doxygen docs update ?

On Fri, May 18, 2018 at 10:55 AM, Rodrigo Ramírez Norambuena wrote: > On Wed, May 10, 2017 at 10:25 AM, Joshua Colp wrote: >> On Wed, May 10, 2017, at 10:14 AM, Ludovic Gasc wrote: >>> Hi, >>> >>> The Doxygen documentation: http://doxygen.asterisk.org/

[asterisk-dev] Asterisk 16 Feature Freeze Reminder

Hey Everybody, Just a friendly reminder that today is the official two month reminder [1] prior to feature freeze on Wednesday, July 18th, 2018. For those who are not aware, we try to "feature freeze" the master branch in order to test and stabilize the code base prior to release of a new major

[asterisk-dev] AstriCon Approaching, Super Earlybird Pricing Expires In 3 days

Hey All, So one of the jobs that I get to do as head of the Asterisk project is to help inform people about the yearly conference we have about Asterisk named Astricon. For those who are not familiar with it, AstriCon is a fantastic event for anyone that is serious about Asterisk. This year,

[asterisk-dev] Asterisk Community Services Outages

Dear Asterisk Community, For the past 24 hours or so, Digium’s upstream provider has had a few outages that have affected Asterisk community services, including Asterisk.org, the mailing lists, and potentially other services. We apologize for any inconvenience that it has caused. Hopefully

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

On Thu, Apr 5, 2018 at 9:30 AM, Jared Smith wrote: > 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

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

On Mon, Apr 2, 2018 at 9:40 PM, Steve Murphy wrote: > Someone complained about errant behavior of the Busy and Hangup apps... > and I see some strangenesses that might make them credible. > > Boy, did I have to chase this around to find what I think is the problem! > > What

Re: [asterisk-dev] starpy

all the things in the same >> place. >> >> On Tue, Mar 27, 2018 at 10:48 AM, Matt Fredrickson <cres...@digium.com> >> wrote: >>> >>> On Mon, Mar 26, 2018 at 4:55 AM, Corey Farrell <g...@cfware.com> wrote: >>> > I'm working on some PR'

[asterisk-dev] Sorry for interruption of service

Hey All, Just as a public service announcement, we had a 12-16 hour window with mailing list service interruption. Last night we scheduled a time to update the mailing list server but today found some problems impacting mailing service after the updates. Due to this discovery, we quickly

Re: [asterisk-dev] starpy

On Mon, Mar 26, 2018 at 4:55 AM, Corey Farrell wrote: > I'm working on some PR's to update the Asterisk testsuite to be compatible > with Python3 (without breaking Python2). An issue I've hit is starpy. A PR > to deal with compatibility was started but never finished. Can we

Re: [asterisk-dev] Authenticated downloads of external stuff?

On Sat, Feb 10, 2018 at 7:29 AM, Alexander Traud wrote: > Asterisk downloads a lot of external stuff while configuring and > installing - via HTTP - for example sound files, Digium modules, and the > PJProject. These downloads are guarded by checksum/hashes which are > -

Re: [asterisk-dev] Webpage Downloads

t;> can be removed, not to confuse novice users. Furthermore, I do not >> understand the state of AsteriskNOW-10.13.66*17* because FreePBX is on >> 10.13.66*22* already: >> - <http://www.freepbx.org/downloads/> >> - <http://wiki.freepbx.org/display/PPS/10.13.66+Release+No

Re: [asterisk-dev] Compiler feature requirements

On Wed, Jan 24, 2018 at 3:50 PM, Corey Farrell wrote: > I've posted ASTERISK-27619 [1] proposing that we drop support for GCC > versions older than 4.1.2. Specifically we'd be requiring that either > __sync or __atomic builtin functions be available (I'm unsure what this will >

[asterisk-dev] FOSDEM

Hey All, For any interested in potentially meeting up to talk about Asterisk and other fun things, Ben Ford from Digium's Asterisk development team and myself will be in Brussels for FOSDEM Feb 3-4. I hope to see many of you there! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445

Re: [asterisk-dev] QueueLog Linkedid

On Thu, Jan 11, 2018 at 6:00 AM, marek cervenka wrote: > hi, > > because of some call analytics we need linkedid in queue_log > > if we create patch for it (app_queue), will you accept it? My preferences are that any changes to the queue log format would be done in master

Re: [asterisk-dev] Problems with the ASTERISK-27206 patch.

On Wed, Jan 3, 2018 at 2:04 PM, Joshua Colp wrote: > On Wed, Jan 3, 2018, at 3:53 PM, Richard Mudgett wrote: >> On Wed, Jan 3, 2018 at 5:08 AM, Joshua Colp wrote: >> >> > On Wed, Jan 3, 2018, at 12:03 AM, Richard Mudgett wrote: >> > > On Tue, Jan 2, 2018 at

Re: [asterisk-dev] Adding a Key/Value Store mechanism to Asterisk

Matthew Fredrickson > > On 22 December 2017 at 16:54, Matt Fredrickson <cres...@digium.com> wrote: > >> >> >> On Fri, Dec 22, 2017 at 6:58 AM, Nir Simionovich < >> nir.simionov...@gmail.com> wrote: >> >>> Abhay, >>> >>>

Re: [asterisk-dev] Adding a Key/Value Store mechanism to Asterisk

On Fri, Dec 22, 2017 at 6:58 AM, Nir Simionovich wrote: > Abhay, > > Migrating astsb from SQLlite to redis isn't the topic here. I'm talking > adding a new type of storage engine. For example, func_redis, that will > implement the relevant key/value operations that are

Re: [asterisk-dev] reproducible builds of asterisk

Hey Tzafrir, On Wed, Dec 20, 2017 at 3:53 AM, Tzafrir Cohen wrote: > Hi, > > There is a patch in the Debian package to build Asterisk in a > reproducible way[1] if so needed. Patch is really simple, but as I did > not write it, I can't push it. I described what it does.

Re: [asterisk-dev] Minor Release Branches

On Tue, Dec 19, 2017 at 10:46 AM, Kevin Harwell wrote: > > > On Tue, Dec 19, 2017 at 10:12 AM, Corey Farrell wrote: >> >> I'd like to propose that we make minor release branches temporary. What I >> mean is 13.15, 15.0, etc. The sole purpose of the '13.15'

Re: [asterisk-dev] 3PCC patch for AMI "SIPnotify"

Answer below. On Thu, Nov 30, 2017 at 6:46 PM, Yasuhiko Kamata wrote: > Hello asterisk-dev list, > > We have created a patch for use in 3PCC applications. > With this patch, asterisk can let the certain SIP phone answer or hold > through AMI action. > > [summary] > A

[asterisk-dev] Recent Video Interview and Upcoming Webinar about Asterisk 15

, there's a fun 15 minute video interview featuring Matt Jordan and myself discussing Asterisk 15 and what's new with it. It can be found at: https://www.youtube.com/watch?v=0XSDOPftNpM=youtu.be For those who enjoyed the video or would like to get a bit of a deeper dive into Asterisk 15

Re: [asterisk-dev] Ad: Passing Variables to Voicemail's Email Notification

Just to be clear - in order to create an issue on the issue tracker, it doesn't require signing a license agreement. So there shouldn't be any problems with opening an issue for it. Hope that's more straightforward. Matthew Fredrickson On Wed, Nov 15, 2017 at 4:08 PM, Matt Fredrickson <c

Re: [asterisk-dev] Ad: Passing Variables to Voicemail's Email Notification

On Fri, Nov 10, 2017 at 2:20 PM, butrus.but...@gmail.com wrote: > Hello! > > While playing with Asterisk's voicemail and it's IMAP storage I found > a bug - and later I realized how old this bug is. The problem is that > variable substitions in voicemal.conf works only

Re: [asterisk-dev] Open Invitation - Open Source VoIP Documentation Project

On Sat, Nov 4, 2017 at 2:53 PM, Nir Simionovich wrote: > Hi All, > > Following Astricon 2017, and my own personal interest in providing the > community with more detailed documentation sources and specifically - best > of practice and how-to documents, I've opened a

Re: [asterisk-dev] autoconf version requirement increase

Hey Corey, On Tue, Oct 31, 2017 at 3:52 PM, Corey Farrell wrote: > Hello all, > > autoconf is required to run ./bootstrap.sh after changing configure.ac or > *.m4 in any folder of the Asterisk source tree (including menuselect and > third-party folders). Currently we require

[asterisk-dev] Asterisk 11 EOL Announcement

Dearly Beloved, We have gathered here today to mourn the passing of a deeply regarded branch of Asterisk - Asterisk 11. As of today, it has officially reached its end of life. It was a good branch, having served 5 years faithfully in the service of its users. As far as history goes, 11.0.0 was

Re: [asterisk-dev] chan_sip maintenance mode

On Wed, Oct 11, 2017 at 7:24 AM, Corey Farrell wrote: > I propose that we restrict the kind of bugs/patches that are accepted > against chan_sip to only major/critical bugs, regressions and security > fixes. This means rejecting all new features, improvements and most minor >

Re: [asterisk-dev] One sip stack to rule them all....

such as marking chan_sip as deprecated) my opinion is that they are still premature - as mentioned in my other reply. Best wishes, Matthew Fredrickson > > On Tue, Oct 10, 2017 at 2:40 PM, Matt Fredrickson <cres...@digium.com> > wrote: > >> >> >> On Sun, Oc

Re: [asterisk-dev] One sip stack to rule them all....

On Sun, Oct 8, 2017 at 2:00 PM, Seán C. McCord wrote: > As James mentioned at the top, chan_sip is already de facto deprecated. > The discussion (at devcon) was centered around making it _officially_ > deprecated. > > For clarity, deprecation is NOT the same thing as removal.

Re: [asterisk-dev] RTCP feedback for codec modules

On Mon, Oct 9, 2017 at 9:01 AM, marek cervenka wrote: > hi, > > i'm writing article about new features in Asterisk 15 > > can you explain if > > https://issues.asterisk.org/jira/browse/ASTERISK-26584 > > is only part of the building block for function "change codec or codec >

[asterisk-dev] Asterisk 14 Security Fix Only Mode

Hey all, For those who may not be aware Asterisk 14 transitioned from bug fix mode to security-fix-only mode a few weeks ago (Sept 26th). For those of you that are still on this release, it's a good time to consider building an upgrade plan for moving to 15.x.x. I sincerely apologize for the

Re: [asterisk-dev] Plans for legacy APIs in Asterisk ?

Welcome! Reply is inline. On Wed, Oct 4, 2017 at 9:42 AM, Hans Petter Selasky wrote: > Hi, > > I maintain an external channel driver called chan_capi for ISDN4BSD (not > the same like chan_capi for Linux). Every time there is a new major release > of Asterisk I need to update

Re: [asterisk-dev] AstDB mySQL implementation

On Fri, Sep 22, 2017 at 12:12 PM, Ryan Wagoner wrote: > I've been scaling out FreePBX horizontally with Kamailio and custom > FreePBX modules mainly to handle call center outbound dialing (around 20k > calls per day). One of the issues I ran into was FreePBX uses the AstDB >

Re: [asterisk-dev] Want to see what's new and fun in Asterisk 15?

On Fri, Sep 22, 2017 at 5:09 PM, Ludovic Gasc <gml...@gmail.com> wrote: > Hi Matt, > > Indeed, it's a very exciting piece of news with SFU :-) > BTW, do you think it would be possible to share also the screen instead of > video ? > sip.js seems to support that: https:/

[asterisk-dev] Want to see what's new and fun in Asterisk 15?

Hey Everybody, For those of you who have not been following gerrit or the blog posts on blog.asterisk.org, there has been a lot of work put into 15 to turn it into a first class citizen in the video SFU (Selective Forwarding Unit) world. We're very excited to have people start playing with it

[asterisk-dev] Help with recently discovered RTP/RTCP issues

Hey All, Many of you may have noticed the most recent security release for correcting a potential RTP hijacking vulnerability when strictrtp is enabled in conjunction with certain nat settings. In reality, it’s very challenging to get gain and plunder from the bug due to several mitigation

Re: [asterisk-dev] Asterisk amqp (ARI, AMI...)

On Wed, Aug 30, 2017 at 1:54 PM, Sylvain Boily wrote: > Hello! > > We start a proof of concept based on thie review board > (https://reviewboard.asterisk.org/r/4365/) to get all ARI messages and all > AMI messages in rabbitmq instead of using a proxy like we did in

Re: [asterisk-dev] Asterisk 15 Beta Released

On Fri, Aug 25, 2017 at 8:15 AM, Ryan Wagoner <rswago...@gmail.com> wrote: > > On Wed, Aug 23, 2017 at 9:48 AM, Matt Fredrickson <cres...@digium.com> > wrote: >> >> On Fri, Aug 18, 2017 at 8:14 AM, Sidney VanNess >> <sid...@oncallcentral.com> w

Re: [asterisk-dev] Asterisk 15 Beta Released

On Fri, Aug 18, 2017 at 8:03 AM, Seán C. McCord <ule...@gmail.com> wrote: > Matt, > > would you care to expound on your reasons against 14.6 -> LTS? I don't > have all the data, so I certainly don't discount the possibility that there > are compelling "reas

Re: [asterisk-dev] Asterisk 15 Beta Released

Hopefully that answers your question appropriately. Matthew Fredrickson > > On Fri, Aug 18, 2017 at 9:03 AM, Seán C. McCord <ule...@gmail.com> wrote: >> >> Matt, >> >> would you care to expound on your reasons against 14.6 -> LTS? I don't &

Re: [asterisk-dev] Asterisk 15 Beta Released

On Tue, Aug 15, 2017 at 1:15 PM, Dan Jenkins <dan.jenkin...@gmail.com> wrote: > > > On Wed, Aug 2, 2017 at 10:57 PM, Matt Fredrickson <cres...@digium.com> > wrote: >> >> It is with great pleasure I wish to inform you of the first beta >> release of the ne

Re: [asterisk-dev] Asterisk 15 Beta Released

On Tue, Aug 15, 2017 at 1:15 PM, Dan Jenkins <dan.jenkin...@gmail.com> wrote: > > > On Wed, Aug 2, 2017 at 10:57 PM, Matt Fredrickson <cres...@digium.com> > wrote: >> >> It is with great pleasure I wish to inform you of the first beta >> release of the ne

[asterisk-dev] Asterisk 15 Beta Released

It is with great pleasure I wish to inform you of the first beta release of the new Asterisk 15 branch. It's a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or so years. There has been a lot of work done in the

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