Re: [asterisk-dev] AMI Disconnect/Sudden Asterisk Call Manager/1.3 received

On Wed, May 7, 2014 at 1:26 PM, Olle E. Johansson o...@edvina.net wrote: On 06 May 2014, at 19:13, Daniel McFarlane dan...@szeto.ca wrote: On 05/06/2014 10:56 AM, Olle E. Johansson wrote: On 06 May 2014, at 16:11, Daniel McFarlane dan...@szeto.ca wrote: Hi All, I've been working

Re: [asterisk-dev] Asterisk with socket.io

On Tue, Feb 3, 2015 at 7:40 AM, Vipul Rastogi vipul.rast...@temasys.com.sg wrote: Today, I got Asterisk working with respoke channel code as well but on startup Asterisk connects (register=yes) with node.js server with HTTP GET which does not have following headers... Connection: Upgrade

Re: [asterisk-dev] Asterisk with socket.io

On Wed, Jan 21, 2015 at 2:36 AM, Vipul Rastogi vipul.rast...@temasys.com.sg wrote: Anybody tried asterisk connecting to Socket.io server as websocket client ? I am not getting websocket established. See below error, after successful 101 Switching Protocols res_http_websocket.c:576

Re: [asterisk-dev] Subjects for e-mails

On Tue, Apr 14, 2015 at 3:18 PM, Russell Bryant russ...@russellbryant.net wrote: On Tue, Apr 14, 2015 at 8:47 AM, Matthew Jordan mjor...@digium.com wrote: On Tue, Apr 14, 2015 at 2:15 AM, Olle E. Johansson o...@edvina.net wrote: Can we possibly have different Subject: lines in e-mails for

Re: [asterisk-dev] Plan for updating the ARI Swagger Version

On Tue, Feb 9, 2016 at 4:26 PM, Matthew Jordan <mjor...@digium.com> wrote: > > > On Tue, Feb 9, 2016 at 3:56 AM, Dan Jenkins <dan.jenkin...@gmail.com> > wrote: > >> Hi Everyone, >> >> I've been looking at how we can add proxy support (1) to the Node

[asterisk-dev] Plan for updating the ARI Swagger Version

Hi Everyone, I've been looking at how we can add proxy support (1) to the Node.js ARI Client for the past couple of days and have hit a few issues which I'm sure we'll be able to work out. But this has led me down the path of looking into the current status of swagger. Swagger recently donated

Re: [asterisk-dev] node-ari-client users: what version of Node do you use?

On Sun, Aug 7, 2016 at 4:42 PM, Chad McElligott wrote: > Hi everyone, > > We recently merged a critical bug fix to node-ari-client master branch > that only affected users of Node 6.3+ [1][2]. This fix was in the 'ws' > library. By upgrading to the necessary version to

Re: [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required

jssip? > > thanks > > best regards > > > On Jan 18, 2017, 10:52 -0600, Dan Jenkins <dan.jenkin...@gmail.com>, > wrote: > > Hi All, > > I've been working with a company who utilise WebRTC using Asterisk behind > Kamailio to connect browser users and their SIP

Re: [asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required

On Thu, Jan 19, 2017 at 9:09 AM, Dan Jenkins <dan.jenkin...@gmail.com> wrote: > Hi Sebastian, > > I haven't opened an issue in Asterisk yet - ran out of time yesterday - > first thing on my list to do today. > > I have now tested the flag in jssip (jssip makes i

[asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required

Hi All, I've been working with a company who utilise WebRTC using Asterisk behind Kamailio to connect browser users and their SIP infrastructure and just came across an issue making/receiving calls in Chrome Canary and Chrome Dev. Long story short; the issue is that rtcp-mux has now been set as

Re: [asterisk-dev] Working Groups

On Tue, Oct 4, 2016 at 11:09 PM, Matt Fredrickson wrote: > Hey all, > > Welcome back to all of you who attended AstriDevCon. Thanks so much > for all of you that attended and gave so much of your time to be able > to contribute. > > One of the ideas proposed in AstriDevCon

Re: [asterisk-dev] Viva Chan_Sip, may it rest in peace

On Thu, Oct 6, 2016 at 10:01 AM, marek cervenka wrote: > > Michael, >> >> What would be amazing is for you to tell us which features you are >> missing (or were missing when you tried) >> >> If we start a working group around PJSIP migration then these points will >> help

Re: [asterisk-dev] Working Groups

On Wed, Oct 5, 2016 at 11:11 PM, Rodrigo Ramírez Norambuena < decipher...@gmail.com> wrote: > On Tue, 2016-10-04 at 17:09 -0500, Matt Fredrickson wrote: > > Hey all, > > > Hello!, > > > Welcome back to all of you who attended AstriDevCon. Thanks so much > > for all of you that attended and gave

Re: [asterisk-dev] Viva Chan_Sip, may it rest in peace

On Wed, Oct 5, 2016 at 8:21 AM, Leandro Dardini wrote: > Your analysis of the chan_sip/PJSIP is really great and I agree with you. > Being a grey haired tech, I can check what drives similar changes in the > latest 20 years. We moved from Netware networks to TCP/IP, we moved

Re: [asterisk-dev] Viva Chan_Sip, may it rest in peace

On Wed, Oct 5, 2016 at 4:04 PM, Michael Ulitskiy wrote: > I am in the same situation. All my systems are business-critical and I'm > > yet to see a convincing argument to spend a lot of man power to migrate > the systems. > > Yes, pjsip supposed to be more stable, but

Re: [asterisk-dev] Working Groups

On Fri, Oct 7, 2016 at 3:10 AM, Rodrigo Ramírez Norambuena < decipher...@gmail.com> wrote: > On Thu, Oct 6, 2016 at 7:44 AM, Dan Jenkins <dan.jenkin...@gmail.com> > wrote: > > > > Hi Rodrigo, > > > > Hi Dan!, > > > > [ .. ] > > >

Re: [asterisk-dev] Proposed Working Group Guidelines

On Mon, Nov 7, 2016 at 2:51 PM, Matthew Jordan <mjor...@digium.com> wrote: > > > On Mon, Nov 7, 2016 at 5:00 AM, Dan Jenkins <dan.jenkin...@gmail.com> > wrote: > >> >> On Fri, Nov 4, 2016 at 2:07 PM, Matt Fredrickson <cres...@digium.com> >> wrot

Re: [asterisk-dev] Proposed Working Group Guidelines

On Fri, Nov 4, 2016 at 2:07 PM, Matt Fredrickson wrote: > Hey All, > > I've been thinking a lot about how working groups might work within > the context of the Asterisk project. Here are a few guidelines that I > have come up with governing working groups. Some of these

Re: [asterisk-dev] ARI versioning in 13 and 14

+1 to option 2 On Thu, Nov 17, 2016, 19:44 Bryant Zimmerman wrote: > > > +1 to option 2. > > Thanks > > Bryant Zimmerman (ZK Tech Inc.) > 616-855-1030 Ext. 2003 > > > -- > _ > -- Bandwidth and Colocation

Re: [asterisk-dev] Viva Chan_Sip, may it rest in peace

On Wed, Oct 12, 2016 at 2:43 PM, Matt Fredrickson wrote: > I've been deliberately waiting to weigh in on this discussion - > however, my thoughts are as follows: > > 1.) Marking a module as deprecated is *not* the same thing as moving > it out of the code base. It simply

Re: [asterisk-dev] Possible change to the AMI PJSIPShowRegistrationsInbound command

On Tue, Dec 6, 2016 at 2:43 PM, George Joseph wrote: > We just discovered that the PJSIPShowRegistrationsInbound command really > only dumps all aors regardless of whether there's an inbound registration > associated with it or not. Fixing this would mean a change to what

Re: [asterisk-dev] git issues pulling asterisk 13, 14, 15, master branches

We've all been there and done that :D On Wed, Aug 2, 2017 at 1:39 AM, George Joseph wrote: > I messed up. The next time you pull one of the asterisk branches from git > you may get an error like this... > > Switched to branch 'master' > Your branch and 'origin/master' have

Re: [asterisk-dev] Asterisk 15 Beta Released

On Wed, Aug 2, 2017 at 10:57 PM, Matt Fredrickson wrote: > 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

Re: [asterisk-dev] Making pjproject_bundled the default in Asterisk 15

On Tue, Aug 8, 2017 at 10:44 PM, George Joseph wrote: > > > On Tue, Aug 8, 2017 at 1:15 PM, George Joseph wrote: > >> The option to use the bundled version of pjproject has been available >> since January 2016 and is the only "supported" method of using

Re: [asterisk-dev] Making pjproject_bundled the default in Asterisk 15

On Tue, Aug 15, 2017 at 2:07 AM, George Joseph <gjos...@digium.com> wrote: > > > On Mon, Aug 14, 2017 at 1:04 PM, Dan Jenkins <dan.jenkin...@gmail.com> > wrote: > >> >> >> On Tue, Aug 8, 2017 at 10:44 PM, George Joseph <gjos...@digium.com> >

Re: [asterisk-dev] Asterisk 15 Beta Released

On Thu, Aug 17, 2017 at 11:11 PM, Matt Fredrickson <cres...@digium.com> wrote: > 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> > > w

Re: [asterisk-dev] Configuring multistream in chan_pjsip

On Tue, Jun 6, 2017 at 6:28 PM, Mark Michelson wrote: > On 06/05/2017 03:17 PM, Matt Fredrickson wrote: > >> On Mon, Jun 5, 2017 at 2:31 PM, Joshua Colp wrote: >> >>> On Mon, Jun 5, 2017, at 04:21 PM, Mark Michelson wrote: >>> Hi folks,

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

On Thu, Sep 7, 2017 at 10:03 AM, Sylvain Boily wrote: > > > Le 2017-09-05 à 23:54, Matt Fredrickson a écrit : > >> On Wed, Aug 30, 2017 at 1:54 PM, Sylvain Boily >> wrote: >> >>> Hello! >>> >>> We start a proof of concept based on thie

Re: [asterisk-dev] Asterisk 15 RC1 and RTP/RTCP leak ?

He means the rtpbleed issue I presume. And I would presume that its already been patched. But I can't tell you for sure https://rtpbleed.com/ On Tue, Sep 26, 2017 at 7:04 PM, Sean Bright wrote: > On 9/26/2017 12:37 PM, sean darcy wrote: > >> Are there any RTP/RTCP leak

Re: [asterisk-dev] NET::ERR_CERT_SYMANTEC_LEGACY: Re-issue your RapidSSL certificate!

Ha! Already informed them on Friday via other means. I'm told there is now an IT ticket open On Sun, 5 Aug 2018, 11:18 Alexander Traud, wrote: > All asterisk.org (sub-) domains are secured by a SSL/TLS certificate from > RapidSSL which chains up to the trust anchor "GeoTrust Global CA". That >

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

Thanks for the reminder Matt. The minutes from devcon can be found here if anyone is interested - https://wiki.asterisk.org/wiki/display/AST/astridevcon+2018+minutes At devcon we talked about the growing need to be able to read audio from a channel and be able to pump audio back into a channel

Re: [asterisk-dev] ARI, Stasis, and Dialplan

On Wed, Dec 12, 2018 at 4:24 PM Ben Ford wrote: > Hey all, > > We’re looking into AstriCon feedback and one of the things we want to > touch on is ARI -- specifically, the ability to not have to create an > extensions.conf in order to dial into Stasis. Before we start working on > this, we’d

Re: [asterisk-dev] ARI, Stasis, and Dialplan

Oh I do remember the context idea - which seemed like a good idea at the time because of not having to change much internally On Wed, Dec 12, 2018 at 7:07 PM Seán C. McCord wrote: > Correction: when I said the "latter," I should have said the "third" > option. The last option does not really

Re: [asterisk-dev] ARI, Stasis, and Dialplan

t;> >>> On Wed, Dec 12, 2018 at 11:42 PM Anil Gupta >>> wrote: >>> >>>> +1 to the context idea >>>> >>>> Something like *context = stasis:app_name* would be nice. I presume >>>> this could be achieved by adding the functionality to t

Re: [asterisk-dev] ARI, Stasis, and Dialplan

alls will dial into that Stasis > application. The context parameter shouldn't matter in this case, and with > the addition of the new "move" REST API call, you can send your channel to > any other active application. Overall, less work! > > Hopefully this answers your questi

Re: [asterisk-dev] ARI, Stasis, and Dialplan

:thumbsup: On Tue, Feb 5, 2019 at 11:03 AM Joshua C. Colp wrote: > On Tue, Feb 5, 2019, at 5:27 AM, Dan Jenkins wrote: > > so in this scenario a context will be created with "stasis-master" > > and we can do all our magic if we want to; but its quite possible

Re: [asterisk-dev] ARI, Stasis, and Dialplan

Ah so just to confirm - above in the thread theres a variable in the url passed in called context so that you could have an app name of foo but a context of bar and also an ari application without a context called alice and therefore alice could still be addressable from dialplan. So do we want

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

Just caught up on this - makes sense to me. DNS is important transport - WSS to make it easy to just send binary down a WSS and have it secure. Dan On Wed, Aug 7, 2019 at 4:41 PM Seán C. McCord wrote: > Sounds like good reasoning to me. > > On Wed, Aug 7, 2019, 11:23 Joshua C. Colp wrote: >

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

Just going to chime in and say I don't see a one way audio solution as enough and I'd worry that doing that would lead to "oh but only so many people need to chuck audio in". This has been discussed at at least 3 devcons now. On Thu, Jul 18, 2019 at 2:09 PM Seán C. McCord wrote: > I certainly

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

ould be able to address an asterisk independently maybe this has just proved that it needs to work both ways? On Wed, Jul 24, 2019 at 4:19 PM George Joseph wrote: > > > > On Mon, Jul 22, 2019 at 2:01 AM Dan Jenkins wrote: > >> Also coming back to this with some

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

>>> *From:* asterisk-dev *On Behalf >>> Of *Luca Pradovera >>> *Sent:* Monday, July 22, 2019 3:12 AM >>> *To:* Asterisk Developers Mailing List >>> *Subject:* Re: [asterisk-dev] Audio to/from Asterisk >>> >>> >>> >>> He

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

I multiplex the ARI amongst a wide range of processing nodes, but if > you want to minimize additional routing and orchestration, you can't beat > just simply using the same websocket that your control channel is on. > That's about as coupled as it gets. > > On Wed, Jul 24, 2019 at

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

;> - time-wise >>> - after connection (what if nother ever connects?) >>> - by command only >>> - security >>> - DoS vulnerability >>> >>> Technically, you could say that interface binding is a problem with >>> outbound, t

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

this mechanism at devcon? Dan On Sat, Jul 20, 2019 at 2:39 PM Dan Jenkins wrote: > Just going to chime in and say I don't see a one way audio solution as > enough and I'd worry that doing that would lead to "oh but only so many > people need to chuck audio in". This has been disc

Re: [asterisk-dev] [asterisk-users] [asterisk-app-dev] Proposed change to External Media API

seems fair to me! Looking forward to this George! On Fri, Oct 18, 2019 at 1:57 PM George Joseph wrote: > When we created the External Media addition to ARI we created an > ExternalMedia object to be returned from the channels/externalMedia REST > endpoint. This object contained the channel

[asterisk-dev] AstriDevCon ideas

Unfortunately I won't be at DevCon this year (I'll be on a plane flying from West to East US) - I will of course try to listen in etc but I'd like to raise one new idea and another maintenance issue to be discussed at DevCon. Idea: I find it tough to get people (even myself) to build actual

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

Ultimately whats stopping package maintainers from releasing "asterisk-full" which still has all the deprecated modules enabled and "asterisk" which follows the defaults? Nothing Other packages get released in such a way so why not asterisk? I'm 100% not qualified to talk about it because

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

don’t think anyone has > missed it - it’s been far too long with two channels, which is confusing. > > There are propably a list of modules I would remove quickly if I was under > Josh’s hat. > /O > > On 2 Oct 2020, at 12:18, Dan Jenkins wrote: > > Ultimately whats stop

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

be enabled by default and ultimately thats what the changelog/upgrade.txt is for isn't it? 4 years seems like a long time to get rid of something thats already been decided isnt being looked after any more. On Thu, Oct 1, 2020 at 5:27 PM Joshua C. Colp wrote: > On Thu, Oct 1, 2020 at 1:15 PM

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

; On 10/1/20 12:25 PM, Joshua C. Colp wrote: > > On Thu, Oct 1, 2020 at 1:15 PM Dan Jenkins wrote: > >> Firstly, thank you Josh for trying to outline the process so that it's >> something that can be followed and while I agree mostly with the steps... >> the fact th

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

Firstly, thank you Josh for trying to outline the process so that it's something that can be followed and while I agree mostly with the steps... the fact that its going to take 4 years for a module to be moved from "deprecated" to being removed just feels too long but understandable if we're

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

Completely agree with Sean on the "what's going to be deprecated" question months before a cut. And I like the laid out plan that would be involved for 2 year process of deprecation,default enabled no and then remove. On Thu, 1 Oct 2020, 22:42 BJ Weschke, wrote: > I don’t think anyone would

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

Thu, Oct 1, 2020 at 3:56 PM Dan Jenkins wrote: > >> If there was an additional message attached to minor releases, does that >> mean we can accelerate the steps? >> >> On the question of why I'm opposed to 4 years? 4 years is an eternity to >> be in limbo - we've al

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

If there was an additional message attached to minor releases, does that mean we can accelerate the steps? On the question of why I'm opposed to 4 years? 4 years is an eternity to be in limbo - we've already seen this with chan_sip - even though its deprecated in 17, people still start using

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

I was hoping you'd pipe in again Jared! Thank you! On Fri, 2 Oct 2020, 20:20 Jared Smith, wrote: > 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 s

Re: [asterisk-dev] Advanced Codec Negotiation: Invite without Offer

Hi George, Funnily enough I saw this the other day and the client's Kamailio/RTPProxy setup wasn't set up to handle the reinvite scenario afterwards. From what I understand this is becoming more and more common. So in this case it was an initial request to get the dialog going, once everything

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

YES YES YES. I never really got why it was done like it was... I agree, 4-6 weeks is good. If someone's going to take the time to upgrade to an 18 RC then they'll be looking to test it and give feedback etc etc so you don't need a huge amount of time... just enough to actually action bug fixes

Re: [asterisk-dev] Asterisk console hangs up

Without any logs we can’t really help you… Also a reminder that 13 is in security fix only mode and goes end of life in October - it’s time to upgrade! On Tue, 27 Apr 2021 at 06:13, Jaco Kroon wrote: > Hi Rajesh, > > On 2021/04/27 05:46, Rajesh wrote: > > > We are using Asterisk 13.18.2.

Re: [asterisk-dev] Asterisk Module Deprecation

Hi Josh, Thanks for this! I have a question regarding this line in your email. These changes allow the information regarding when a module was deprecated and when it will be removed to be communicated to users in earlier branches. Does this mean that app_foo being marked as deprecated in

Re: [asterisk-dev] Asterisk 19: res_adsi built although deprecated?

Based on https://wiki.asterisk.org/wiki/display/AST/Module+Deprecation I think you’re right when you say it shouldn’t be built by default On Mon, 23 Aug 2021 at 15:15, Alexander Traud wrote: > While creating a minimal installation of the upcoming Asterisk 19 with > > ./configure > make full >