Re: [asterisk-dev] Mailing List Future

> On 4 Dec 2023, at 13:38, Joshua C. Colp wrote: > > The mailing lists have remained unchanged since deployed. …and the archives for everything is still around. Mail is boring but very very long-term stable. Forums are cool, sexy and keeps changing so we loose history because the cost of

Re: [asterisk-dev] Deprecating users.conf

> On 30 Jun 2023, at 13:45, aster...@phreaknet.org wrote: > > Hi folks, > > I've put up a PR to deprecate users.conf[1], following a discussion > earlier this year about this, but I think that was on IRC so wanted to > discuss here as well. > > Mark introduced users.conf at some point

Re: [asterisk-dev] chan_sip deprecation

> On 22 Nov 2022, at 09:13, Jon Bonilla (Manwe) wrote: > > El Tue, 22 Nov 2022 08:00:48 + > Henning Westerholt escribió: > >> Hello, >> >> I am really wondering why people are trying to keep chan_sip alive. No >> offence to the past developers, but pjsip is a much better SIP stack >>

Re: [asterisk-dev] SIP subscription with expires=0

A subscription with expire:0 should get at least ONE notify, right? I’ve just that to check the status without setting up a dialog. It is not invalid. Report this as a bug. /O > 25 okt. 2021 kl. 09:22 skrev Nikša Baldun : > > Hello, > > I see the following in res_pjsip_pubsub.c: > >

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

> On 19 Oct 2021, at 21:56, Matt Fredrickson wrote: > > 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 >>

Re: [asterisk-dev] Gerrit ssh host key changed again :(

> On 26 Aug 2021, at 16:32, George Joseph wrote: > > During the upgrade of Gerrit today, it decided to regenerate its ssh host key > again, The good news is that Jira issues will now list the active reviews > associated with an issue again. The bad news is that 'gerrit review' and > other

Re: [asterisk-dev] Feature request: Allow the use of pjsip client only transports in Asterisk pjsip

> 13 juni 2021 kl. 16:32 skrev Michael Maier : > > > Hello! > > pjsip provides the ability to create (TCP / TLS) transports without opening > any listener. This is handy if you don't need any listening transport at all > for a sip device. > > One of the typical use cases is for dial up

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

wait a > further 4 years. > > > > On Fri, Oct 2, 2020 at 8:48 AM Olle E. Johansson <mailto:o...@edvina.net>> wrote: > Hi! > I like adding product management and actually removing stuff that the company > can’t keep maintaining - and don’t wan’t to. > >

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

Hi! I like adding product management and actually removing stuff that the company can’t keep maintaining - and don’t wan’t to. Compared with years ago a lot of users never bother building asterisk any more and don’t interface with the project, they just run “apt-get install asterisk” and they

Re: [asterisk-dev] Support for amr codec?

> On 18 Aug 2020, at 10:49, Carsten Bock wrote: > > Hi, > > one of the big issues with AMR/AMR-WB and Asterisk is that usually devices > speaking AMR/AMR-WB do use quite a lot of comfort noise, which is not > supported by Asterisk as far as I know. Not in the regular Asterisk - I made a

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

list and you’ll get the proper answers! Cheers, /O > > > On Sun, Nov 19, 2017 at 12:56 PM Olle E. Johansson <o...@edvina.net > <mailto:o...@edvina.net>> wrote: > >> On 16 Nov 2017, at 22:18, Nir Simionovich <nir.simionov...@gmail.com >> <mailto:ni

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

> On 16 Nov 2017, at 22:18, Nir Simionovich wrote: > > and that the RPM repo for Kamailio is - how shall we put it, not providing > all the possible modules. You are very welcome to contribute updates directly to the Kamailio project. We’ve been looking for

Re: [asterisk-dev] AstDB mySQL implementation

ard > enough, making it configurable, seems > like a pain in the butt. Look for the “appleraisin” branch if you want to see code :-) /O > > > > On Thu, Oct 26, 2017 at 4:23 PM Olle E. Johansson <o...@edvina.net > <mailto:o...@edvina.net>> wrote: >> On 26 Oc

Re: [asterisk-dev] AstDB mySQL implementation

'm > wrong - turning this into a configurable thing > would be more or less an open-heart surgery. My patch wasn’t that bad, but it was before sqlite. /O > > > > On Thu, Oct 26, 2017 at 4:16 PM Olle E. Johansson <o...@edvina.net > <mailto:o...@edvina.net>> wrot

Re: [asterisk-dev] AstDB mySQL implementation

Somewhere in Asterisk space, there’s an old patch where I added ASTDB over realtime, meaning you can use any realtime storage. If I remember correctly there was a bit of chicken-and-egg problem with some astdb calls happening before realtime got launched, but otherwise it worked just fine in

Re: [asterisk-dev] Team branches in the asterisk repository

> On 25 Feb 2017, at 19:35, George Joseph wrote: > > No hurry but since the git migration and the availability of GitHub, do we > really need to keep the team branches at all? An important aspect for me with all my subversion branches was that all the code was contributed

Re: [asterisk-dev] Registration state for SIP over TCP or TLS

> On 09 Jan 2017, at 19:52, Joshua Colp wrote: > > On Mon, Jan 9, 2017, at 02:10 PM, Steve Davies wrote: >> Hi, >> >> I believe that the current state of affairs with Asterisk's SIP over TCP >> or >> TLS registration is that if a connection is dropped or closed, then the >>

Re: [asterisk-dev] Subscription behavior when an incoming registration goes away?

> On 22 Dec 2016, at 18:13, George Joseph <gjos...@digium.com> wrote: > > > > On Thu, Dec 22, 2016 at 9:22 AM, Olle E. Johansson <o...@edvina.net > <mailto:o...@edvina.net>> wrote: > >> On 22 Dec 2016, at 17:14, Matthew Jordan <mjor...@d

Re: [asterisk-dev] Subscription behavior when an incoming registration goes away?

> On 22 Dec 2016, at 17:14, Matthew Jordan wrote: > > > > On Thu, Dec 22, 2016 at 9:32 AM, George Joseph > wrote: > When an incoming registration goes away, either because it expires or it was > explicitly expired, what

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

Let’s extend this “how to communicate with a codec” discussion and add the need for feedback on silence. I think there’s an old mail from Matt Jordan touching this earlier. My code for silence suppression and comfort noise adds un-needed transcoding and needed a way for a codec to communicate

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

Hi! From my perspective I know that maintaining a SIP stack requires *A LOT* of effort, so I understand that a project can’t maintain two of them. I suggest that a working group is created for the transition and that the first task is to compare the functionality. Last time I checked the

Re: [asterisk-dev] strictrtp seems to be not so strict

> On 26 Aug 2016, at 14:29, Joshua Colp wrote: > > Torrey Searle wrote: >> I wouldn't dare change the default :-) >> >> But the way I understand the code is that it would end up being a >> switching, as getting a packet from the current source doesn't seem to >> re-set the

Re: [asterisk-dev] Development of asterisk 1.4.23 Can we please get some development?

> On 15 Jul 2016, at 16:51, Matthew Jordan wrote: > > The modules you are referring to were removed quite awhile ago due to not > having an active maintainer contributing to the project [1]. Since no one > active in the project used the modules, or could verify their

Re: [asterisk-dev] Menuselect missing from 1.8 branches

> On 31 May 2016, at 18:23, George Joseph <gjos...@digium.com> wrote: > > > > On Tue, May 31, 2016 at 9:11 AM, Olle E. Johansson <o...@edvina.net > <mailto:o...@edvina.net>> wrote: > Hi! > > I think Menuselect used to be an include from another

[asterisk-dev] Menuselect missing from 1.8 branches

Hi! I think Menuselect used to be an include from another svn repo and now it’s missing from my git repos. I need to resurrect some of them for some certification tests. Can anyone help this old man on how to get menuselect back? Is there a specfic commit I can cherrypick or should I just copy

Re: [asterisk-dev] Configuring Request URI with outbound proxyu

> On 13 Apr 2016, at 22:05, Nitesh Bansal wrote: > > Hello, > > I want to use Asterisk to use Kamailio as an outbound proxy for routing calls > to remote SIP end points, one option could be to use a default peer, but in > my case, my outbound proxy can change >

[asterisk-dev] Certified asterisk patches not in trunk?

From the web page: "Certified Asterisk releases have undergone additional testing and are made less frequently - typically two to four times a year. Certified Asterisk releases are generally identical to the Long Term Support release they are based on, save for additional bug fixes that were

Re: [asterisk-dev] Certified asterisk patches not in trunk?

> On 11 Feb 2016, at 13:06, Joshua Colp <jc...@digium.com> wrote: > > Olle E. Johansson wrote: >> From the web page: >> >> "Certified Asterisk releases have undergone additional testing and >> are made less frequently - typically two to four

Re: [asterisk-dev] Certified asterisk patches not in trunk?

> On 11 Feb 2016, at 16:29, Rusty Newton <rnew...@digium.com> wrote: > > On Thu, Feb 11, 2016 at 6:43 AM, Olle E. Johansson <o...@edvina.net> wrote: > > > >> THat last sentence does not agree with the quote. Please update the web page >> to assure

[asterisk-dev] Bug marshals back !

Yay! I notice in a bug report that the response talks about a Bug Marshal. I am very happy that we have bug marshals back in the process. Is there a document somewhere outlining the process of becoming one and what they do nowadays? Cheers, /O --

Re: [asterisk-dev] Team branches and gerrit

On 24 Apr 2015, at 15:42, Russell Bryant russ...@russellbryant.net wrote: On Fri, Apr 24, 2015 at 8:31 AM, Joshua Colp jc...@digium.com wrote: Olle E. Johansson wrote: Playing around following Matt's wiki page on gerrit usage, I created a team branch and did two commits. When pushing

Re: [asterisk-dev] Team branches and gerrit

On 27 Apr 2015, at 14:55, Russell Bryant russ...@russellbryant.net wrote: On Mon, Apr 27, 2015 at 8:20 AM, Olle E. Johansson o...@edvina.net wrote: On 24 Apr 2015, at 15:42, Russell Bryant russ...@russellbryant.net wrote: On Fri, Apr 24, 2015 at 8:31 AM, Joshua Colp jc...@digium.com wrote

[asterisk-dev] Team branches and gerrit

Playing around following Matt's wiki page on gerrit usage, I created a team branch and did two commits. When pushing it with git review {branch} only the last commit shows up. Is that the way it's supposed to be? I thought the whole branch was the review subject, not just a single commit. /O

Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

On 21 Apr 2015, at 17:55, James Cloos cl...@jhcloos.com wrote: OEJ == Olle E Johansson o...@edvina.net writes: OEJ It's a bug in chan_sip that I fixed a while ago in one of my branches. OEJ SNOM sends an SDES key but RTP/AVP in the offer and chan_sip OEJ chokes. It's a one or two line fix

Re: [asterisk-dev] s?rtp via SIP/TLS/TCP

On 20 Apr 2015, at 17:41, James Cloos cl...@jhcloos.com wrote: I'm not sure whether this is a bug, so I'm starting here. My remote asterisk (debian's compile of 13, currently 13.1.0) and my snom had been unable to rtp for some time. I still use chan_sip. It took a few hours of testing,

[asterisk-dev] Branches

How do I push changes to my branches? If I try a git push my gerrit username/password doesn't work. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or

Re: [asterisk-dev] Review Request: DNS NAPTR/SRV Test plan for PJSIP

First quick observation: please don't use .internal. There are domains for example use, like example.com or internal use - and that's .local There are a lot of tests in the SIPit site you can play with to get more inspiration. /O On 13 Apr 2015, at 20:11, Mark Michelson mmichel...@digium.com

[asterisk-dev] Subjects for e-mails

Can we possibly have different Subject: lines in e-mails for a commit and for a review. Seems like everything is a change in asterisk now. I would like to be able to prioritize reading the commits. THanks, /O Begin forwarded message: From: Matt Jordan (Code Review) asteriskt...@digium.com

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

...@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 a commit and for a review. Seems like everything

Re: [asterisk-dev] [Code Review] 4597: res_pjsip: add CLI commands for global and system configuration

On April 7, 2015, 9:55 p.m., Olle E Johansson wrote: global is used for global variabels. Other CLI commands have show settings - can't we reuse the same with some extra options for system? rmudgett wrote: Where are you getting global variables from the command? pjsip

Re: [asterisk-dev] [Code Review] 4597: res_pjsip: add CLI commands for global and system configuration

show settings - can't we reuse the same with some extra options for system? - Olle E Johansson On April 7, 2015, 6:05 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4563: chan_sip: handle IPv4 mapped clients when NAT and IPv6 socket is enabled

versions of Linux? We had a discussion about it when the IPv6 support was added, but I don't remember all the details - but something with the mapping was O/S and sysctl dependent. - Olle E Johansson On March 30, 2015, 4:23 p.m., Valentin Vidić wrote

Re: [asterisk-dev] [Code Review] 4597: res_pjsip: add CLI commands for global and system configuration

On April 7, 2015, 9:55 p.m., Olle E Johansson wrote: global is used for global variabels. Other CLI commands have show settings - can't we reuse the same with some extra options for system? rmudgett wrote: Where are you getting global variables from the command? pjsip

[asterisk-dev] DNS work :: IDNA domäner ;-)

An additional decision to be made: Do we need to support IDNA in asterisk or not? If I want to set up a sip trunk do blåbärsmjölk.ästerisk.org - do I add that in the config file? For those that create web interfaces to asterisk - should they do the conversion or will Asterisk? There are now

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

to Unicode-art. ◼️⚫️◼️  Please add this to the agenda of the Asterisk-internationalization meeting in Stockholm next month. We also need to discuss the babel-fish module. - Olle E Johansson On April 1, 2015, 9:59 p.m., Matt Jordan wrote

Re: [asterisk-dev] [Code Review] 4441: Enable TLS Dual-Certificates (ECC+RSA)

attention to that. I am not sure about adding DSA, but adding ECC is a good thing. I would suggest going for more config parameters instead of guessing file names. We are not doing that anywhere else (that I know of) and I don't think it's a good thing. - Olle E Johansson On March 30, 2015, 10

Re: [asterisk-dev] RFC: Refactor qualify and res_pjsip/endpt_send_request

On 30 Mar 2015, at 16:54, Mark Michelson mmichel...@digium.com wrote: On 03/28/2015 08:06 PM, Joshua Colp wrote: George Joseph wrote: The fact that it goes to unavailable would be a bug. Why does it do so? Mark should probably chime in here but I think it's because the earliest you could

Re: [asterisk-dev] [Code Review] 4532: PJSIP: Create transactions for out-of-dialog responses

/asterisk/res_pjsip.h https://reviewboard.asterisk.org/r/4532/#comment25514 ...and send the SAME answer to every retransmission... /branches/13/res/res_pjsip.c https://reviewboard.asterisk.org/r/4532/#comment25515 - Olle E Johansson On March 26, 2015, 11:54 p.m., Mark Michelson wrote

[asterisk-dev] Asterisk 11 - where's the code documentation?

Friends, Going through some Asterisk 11 code for my RTCPFB work. There are a lot of new code in the RTP module - almost zero comments. Those that are there are generally not doxygen formatted. Can we please try to add more comments as you add new code? Please. Names doesn't explain your logic

[asterisk-dev] Apologies for messy commit of the uacreg counters

Apologies for the messy merge of my branch. One learns all the time. I tried merging the branch locally instead of using the pull request web interface. One has to be careful with every commit - even to a local branch - or just copy files between the branches and commit without a proper

Re: [asterisk-dev] AstriDevCon Follow Up - Asterisk and Kamailio - smoother integration

So far most of authorization between Kamailio and Asterisk relies on IP addresses, but those need to be provisioned one by one in both sides. The new module is practically adding a custom header with a hash over parts of the message or other environment attributes (eg., IP address) and a

Re: [asterisk-dev] ARI - Add Support for custom SIP Headers with Originate

. /O On Mar 8, 2015 6:18 PM, Matthew Jordan mjor...@digium.com wrote: On Sun, Mar 8, 2015 at 10:51 AM, Nir Simionovich nir.simionov...@gmail.com wrote: Ok, I'll have a look into that one. On Sun, Mar 8, 2015 at 1:03 PM, Olle E. Johansson o...@edvina.net wrote: On 08 Mar 2015, at 09:52

Re: [asterisk-dev] ARI - Add Support for custom SIP Headers with Originate

On 08 Mar 2015, at 17:17, Matthew Jordan mjor...@digium.com wrote: You should be able to do it with just the channel variable SIPADDHEADER, that is: SIPADDHEADER=X-CustomHeader-1: foo SIPADDHEADER=X-CustomHeader-2: bar I think there's a serial number in the name of the variable. /O--

Re: [asterisk-dev] ARI - Add Support for custom SIP Headers with Originate

On 08 Mar 2015, at 09:52, Nir Simionovich nir.simionov...@gmail.com wrote: Hi All, So, I've been banging my head against an issue with ARI. While Channel Originate enables you to originate channels, you can't really do a SIPAddHeader type functionality in there. Originally, I

Re: [asterisk-dev] [Code Review] 4437: dns: Define a core DNS API with examples.

On 03 Mar 2015, at 22:30, Joshua Colp jc...@digium.com wrote: Olle E. Johansson wrote: On 03 Mar 2015, at 21:10, Mark Michelson reviewbo...@asterisk.org mailto:reviewbo...@asterisk.org wrote: for (i = 0; i ast_query_set_num_queries(query_set); ++i) { struct ast_dns_query *query

Re: [asterisk-dev] [Code Review] 4437: dns: Define a core DNS API with examples.

On 03 Mar 2015, at 21:10, Mark Michelson reviewbo...@asterisk.org wrote: for (i = 0; i ast_query_set_num_queries(query_set); ++i) { struct ast_dns_query *query = ast_dns_query_set_get(query_set, i); ...do stuff... } Why not use DNS terminology? /O--

Re: [asterisk-dev] [Code Review] 4453: Asterisk 14: RTP improvements

On 02 Mar 2015, at 16:54, Matt Jordan reviewbo...@asterisk.org wrote: {quote} Buffering/reordering RTP may be received in bursts, out of order, or in other less-than-ideal ways. Asterisk will implement reception buffers to place incoming RTP traffic into, potentially reordering packets

Re: [asterisk-dev] [Code Review] 4453: Asterisk 14: RTP improvements -session

Mark, One thing I think we have to look into is grouping of media sessions. You do not mention that, but you point to the bundle RFC. There are a few cases, like video conferences with left/right video and different options for audio - from high bandwidth to low bandwidth, lossy compression.

Re: [asterisk-dev] AstriDevCon Follow Up - Asterisk and Kamailio - smoother integration

On 23 Feb 2015, at 16:19, Matthew Jordan mjor...@digium.com wrote: On Wed, Feb 11, 2015 at 11:12 AM, Olle E. Johansson o...@edvina.net wrote: On 11 Feb 2015, at 17:50, Matthew Jordan mjor...@digium.com wrote: On Tue, Feb 3, 2015 at 6:11 AM, Daniel-Constantin Mierla mico

Re: [asterisk-dev] AstriDevCon Follow Up - Asterisk and Kamailio - smoother integration

On 23 Feb 2015, at 16:27, Matthew Jordan mjor...@digium.com wrote: On Fri, Feb 13, 2015 at 5:42 AM, Ben Langfeld b...@langfeld.me wrote: On 11 February 2015 at 15:12, Olle E. Johansson o...@edvina.net wrote: On 11 Feb 2015, at 17:50, Matthew Jordan mjor...@digium.com wrote

Re: [asterisk-dev] AstriDevCon Follow Up - Asterisk and Kamailio - smoother integration

On 23 Feb 2015, at 20:10, Matthew Jordan mjor...@digium.com wrote: On Mon, Feb 23, 2015 at 11:16 AM, James Cloos cl...@jhcloos.com wrote: MJ == Matthew Jordan mjor...@digium.com writes: MJ What I'm trying to reason out is: given a set of routing constraints - MJ which includes not

Re: [asterisk-dev] [Code Review] 4419: SDES-SRTP: Handle SRTP keys negotiated with key lifetime/MKI (oej branch lingon-srtp-key-lifetime-1.8) - Asterisk 11

examples in the source helps anyone wanting to modify bugs we miss in the future. Hopefully I can backport part of your forward port to my branch :-) - Olle E Johansson On Feb. 14, 2015, 4:26 a.m., Matt Jordan wrote

Re: [asterisk-dev] Advanced feature query

On 12 Feb 2015, at 09:55, David Radcliffe david.radcli...@clockworkit.co.uk wrote: Hi, Does anyone know if Asterix has the ability to send a network message (TCP/IP packet) indicating that a call is in progress through the PBX? This is not really a question about asterisk development -

Re: [asterisk-dev] DNS Support in Asterisk

There is one version of c-ares in resiprocate as well. C-ares has been in use for a long time and is in use every single day for you as part of most curl installs. I am not sure there is much to do there. Libunbound adds a lot if that is what we want. Why is a cache a good thing? You surely

Re: [asterisk-dev] AstriDevCon Follow Up - Asterisk and Kamailio - smoother integration

On 11 Feb 2015, at 17:50, Matthew Jordan mjor...@digium.com wrote: On Tue, Feb 3, 2015 at 6:11 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Thanks Matt for all the valuable details -- even quite some time since your answer, I still have to digest parts of it, given that I had

Re: [asterisk-dev] OT: Opus Asterisk 13

On 26 Jan 2015, at 14:03, Sean Bright sean.bri...@gmail.com wrote: Hi, I've just finished updating codec_opus for Asterisk 13. Unfortunately it still requires a small patch to the core of Asterisk, but the size of that patch is getting smaller with each new major version of Asterisk.

Re: [asterisk-dev] [Code Review] 4345: Use SIPS Contact headers as prescribed by RFC 3261 (res_pjsip)

On 15 Jan 2015, at 21:07, Mark Michelson reviewbo...@asterisk.org wrote: it feels like a bug that I can send a request to a SIPS URI over UDP and that Asterisk will accept the request. +1 I can't remember any SIPS-using clients, can't say I've looked hard though. Anyone that can help

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

policy, especially in cases where NAPTR doesn't exist. - Olle E Johansson On Jan. 12, 2015, 2:33 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

that SRV is not done if there's any port in the URI. If there's a port in the URI, the hostname is to be used for A/ lookups. If there's no port, go for NAPTR, then SRV. As a side note, SRV may change your port address. - Olle E Johansson On Jan. 12, 2015, 2:33 p.m., Joshua Colp wrote

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

On Jan. 15, 2015, 9:43 a.m., Olle E Johansson wrote: You write that SRV is done if the URI is not IP. Check that SRV is not done if there's any port in the URI. If there's a port in the URI, the hostname is to be used for A/ lookups. If there's no port, go for NAPTR, then SRV

Re: [asterisk-dev] [Code Review] 4318: res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown

On 07 Jan 2015, at 16:24, George Joseph reviewbo...@asterisk.org wrote: This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for persistent connections. To do this, I've had to add a new AST_OPT_FLAG_SHUTTING DOWN to options so we know that a shut down is in progress.

Re: [asterisk-dev] [Code Review] 4318: res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown

On 07 Jan 2015, at 17:05, George Joseph reviewbo...@asterisk.org wrote: I'm open to other suggestions but you absolutely can't send a NOTIFY/terminated to a subscriber and expect to pick up where you left off. They'll respond with a 481. Also, testing for restart vs shutdown isn't

Re: [asterisk-dev] git migration update

On 23 Dec 2014, at 21:53, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Dec 23, 2014 at 7:20 AM, Leif Madsen lmad...@thinkingphones.com wrote: On 22 December 2014 at 18:34, Russell Bryant russ...@russellbryant.net wrote: On Mon, Dec 22, 2014 at 3:08 PM, George Joseph

Re: [asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127

NOT be used. Specifically, dynamic RTP payload types SHOULD be chosen in the range 96-127 where possible. Values below 64 MAY be used if that is insufficient, in which case it is RECOMMENDED that payload type numbers that are not statically assigned by [7] be used first. - Olle E Johansson

Re: [asterisk-dev] Adding the support for NACK in asterisk

On 18 Dec 2014, at 10:42, Nitesh Bansal nitesh.ban...@gmail.com wrote: Hello folks, I am contemplating adding the support for NACK in asterisk hoping for better video quality with Asterisk WebRTC calls. Can somebody please give me an estimate of the challenges involved, how complicated

Re: [asterisk-dev] pjsip vs ca path

On 01 Dec 2014, at 16:21, Mark Michelson mmichel...@digium.com wrote: On 11/25/2014 02:46 PM, James Cloos wrote: Now that 13 has hit sid, I've started converting to pjsip. Chan_sip supports one's preference of a ca path or ca file, but res_pjsip does not. At least not on the 13 branch.

Re: [asterisk-dev] The state of Asterisk (new topic)

On 28 Oct 2014, at 17:44, Russell Bryant russ...@russellbryant.net wrote: I said this in my talk last week and will reiterate it here. I think the Asterisk dev community has been doing an amazing job over the last few years. Internal refactorings have been achieved that we used to dream

Re: [asterisk-dev] [Code Review] 2478: Support multiple Require: and Supported: headers in the same request

--- Tested with the server that caused the issue. Problem solved and we did reach interoperability! Thanks, Olle E Johansson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] [Code Review] 3437: chan_sip: Add support for a few more 4xx error responses

/channels/chan_sip.c 414046 Diff: https://reviewboard.asterisk.org/r/3437/diff/ Testing --- A lot during interoperability tests. Thanks, Olle E Johansson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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

It is critical that a group of developers ask themself questions along these lines - what if??? - What if we removed AGi and AMI? - What if we made a pluggable PBX? - What if we restarted working on a SIP channel? - What if we made a whole new bridge architecture? - What if we skip the idea of

Re: [asterisk-dev] open appliance platform

Glenn, You are misusing this list - this is not for commercial information, it's for discussions about asterisk development. By doing this you are hurting your company as well as your own reputation. IN the future, please be a bit more careful before sending out to mailing lists and posting

Re: [asterisk-dev] [svn-commits] gtjoseph: branch 12 r425964 - /branches/12/

On 19 Oct 2014, at 19:04, SVN commits to the Digium repositories svn-comm...@lists.digium.com wrote: If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh and ./configure to regenerate the build files. You shouldn't have to do this for Intel or SPARC. Good

Re: [asterisk-dev] [Code Review] 4050: Add ability for Channel Drivers to provide Presence State information

On 14 Oct 2014, at 15:51, Matt Jordan reviewbo...@asterisk.org wrote: exten = hint,SIP/aliceCustomPresence:alice Will *always* use the presence provided by SIP/alice, instead of the custom presence provider. That feels like a loss of functionality. Prior to this patch, we would only use

Re: [asterisk-dev] [Code Review] 2783: Fix SIP/TLS reading - random connection drop

E Johansson On Aug. 21, 2013, 10:24 p.m., Tzafrir Cohen wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2783

[asterisk-dev] License issue

Hi! A customer showed interest in supporting the SIP GIN standard - registering for multiple contacts according to RFC 6140, part of the SIPconnect work. I remembered an old patch on reviewboard that has an issue in the bug tracker - #18705 The patch was never uploaded to the bug tracker,

Re: [asterisk-dev] License issue

On 13 Oct 2014, at 15:46, Russell Bryant russ...@russellbryant.net wrote: On Mon, Oct 13, 2014 at 9:28 AM, Olle E. Johansson o...@edvina.net wrote: Hi! A customer showed interest in supporting the SIP GIN standard - registering for multiple contacts according to RFC 6140, part

Re: [asterisk-dev] [Code Review] 4073: res_pjsip: Add 'user_eq_phone' option for placing 'user=phone' parameter in request URI if user is number.

On Oct. 13, 2014, 7:33 p.m., Matt Jordan wrote: /trunk/res/res_pjsip/pjsip_configuration.c, line 1735 https://reviewboard.asterisk.org/r/4073/diff/1/?file=68003#file68003line1735 Alembic! Thinking about it, we also need to update Alembic with your optimistic

Re: [asterisk-dev] [svn-commits] mjordan: branch 12 r424624 - /branches/12/res/res_pjsip/pjsip_options.c

On 06 Oct 2014, at 02:59, SVN commits to the Digium repositories svn-comm...@lists.digium.com wrote: An OPTIONS request that is sent to Asterisk but not to a specific endpoint is currently sent a 404 in response. This is because, not surprisingly, an empty extension is never going to be

Re: [asterisk-dev] [svn-commits] mjordan: branch 12 r424624 - /branches/12/res/res_pjsip/pjsip_options.c

On 06 Oct 2014, at 16:21, Matthew Jordan mjor...@digium.com wrote: On Mon, Oct 6, 2014 at 1:25 AM, Olle E. Johansson o...@edvina.net wrote: On 06 Oct 2014, at 02:59, SVN commits to the Digium repositories svn-comm...@lists.digium.com wrote: An OPTIONS request that is sent to Asterisk

Re: [asterisk-dev] Timeline ZRTP Implementation

On 11 Sep 2014, at 19:51, Matthew Jordan mjor...@digium.com wrote: On Thu, Sep 11, 2014 at 1:26 PM, Jonathan Brown jbr...@fmcllc.com wrote: Are there any plans to add ZRTP functionality anytime soon to this product? ZRTP is the only way for end users to ensure their conversations are

Re: [asterisk-dev] [svn-commits] mjordan: branch 11 r422294 - in /branches/11: ./ LICENSE

On 28 Aug 2014, at 23:53, SVN commits to the Digium repositories svn-comm...@lists.digium.com wrote: This patch updates the LICENSE text to allow users to link Asterisk with UniMRCP and distribute the resulting binaries. Thank you! Good decision. /O --

Re: [asterisk-dev] [Code Review] 3952: Add 'rtpbindaddr' setting for chan_sip

for the archives: I always wanted to do this with an array of IP addresses to get some sort of load sharing. In my John Todd Dragster tests a few years ago the limiting factor was the gigabit ethernet interface. What if we could split the load between multiple interfaces? - Olle E Johansson On Aug

Re: [asterisk-dev] [Code Review] 3780: res_pjsip_outbound_publish / res_pjsip_publish_asterisk: Add outbound PUBLISH support with 'asterisk' event type.

On 31 Jul 2014, at 17:28, Joshua Colp reviewbo...@asterisk.org wrote: This adds two PJSIP modules which add outbound PUBLISH support and an 'asterisk' event type. I don't think it's a good idea to mix different events in one event tag. Will make it hard to handle in proxys and stuff. We

Re: [asterisk-dev] [Code Review] 3709: configure.ac: Check OpenSSL for support of Elliptic Curve cryptography

On 04 Jul 2014, at 14:41, wdoekes reviewbo...@asterisk.org wrote: And generally there is a period of rest between submitting the review and committing the code, so you would normally have been in time for review. I would kindly like to remind everyone about this. Especially if it affects

Re: [asterisk-dev] [svn-commits] mjordan: trunk r418019 - in /trunk: ./ addons/ apps/ channels/ channels/h323/...

Matt! I thank you oh our leader great For this message following the farewell of a lot of old crap, old mate, which no one longer could sell. But please don't change the commit rules! /O ;-) On 04 Jul 2014, at 15:26, SVN commits to the Digium repositories svn-comm...@lists.digium.com wrote:

Re: [asterisk-dev] Timestamps in RTP bridged calls

On 02 Jul 2014, at 13:00, Olle E. Johansson o...@edvina.net wrote: On 02 Jul 2014, at 11:58, Olle E. Johansson o...@edvina.net wrote: Related issue: https://issues.asterisk.org/jira/browse/ASTERISK-23142 In the big jitterbuffer patch in 2006 ther was code that sets a flag

Re: [asterisk-dev] Timestamps in RTP bridged calls

On 03 Jul 2014, at 19:45, Matthew Jordan mjor...@digium.com wrote: On Wed, Jul 2, 2014 at 4:58 AM, Olle E. Johansson o...@edvina.net wrote: Related issue: https://issues.asterisk.org/jira/browse/ASTERISK-23142 In the big jitterbuffer patch in 2006 ther was code that sets a flag

Re: [asterisk-dev] Timestamps in RTP bridged calls

On 03 Jul 2014, at 19:45, Matthew Jordan mjor...@digium.com wrote: On Wed, Jul 2, 2014 at 4:58 AM, Olle E. Johansson o...@edvina.net wrote: Related issue: https://issues.asterisk.org/jira/browse/ASTERISK-23142 In the big jitterbuffer patch in 2006 ther was code that sets a flag

[asterisk-dev] Timestamps in RTP bridged calls

Related issue: https://issues.asterisk.org/jira/browse/ASTERISK-23142 In the big jitterbuffer patch in 2006 ther was code that sets a flag on a AST_FRAME that it contains time stamp information. This is set on all incoming RTP audio frames. When sending RTP we reset the timestamp to the one

Re: [asterisk-dev] Timestamps in RTP bridged calls

On 02 Jul 2014, at 11:58, Olle E. Johansson o...@edvina.net wrote: Related issue: https://issues.asterisk.org/jira/browse/ASTERISK-23142 In the big jitterbuffer patch in 2006 ther was code that sets a flag on a AST_FRAME that it contains time stamp information. This is set on all

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