[asterisk-dev] Problem with configure and FreeTDS

Howdy, I was building 1.4 earlier today and ran into a problem with ./configure --with-tds=/usr/local. I resolved it by changing line 787 of configure.ac from: case `grep TDS_VERSION_NO ${FREETDS_DIR:-/usr/include}/tdsver.h` in to: case `grep TDS_VERSION_NO

[asterisk-dev] Re: [asterisk-commits] russell: trunk r53047 - in /trunk: ./ apps/ channels/ main/ pbx/

FYI, You're adding redundant calls to pthread_attr_destroy in apps/app_rpt.c in that patch (see below). Sean Modified: trunk/apps/app_rpt.c URL: http://svn.digium.com/view/asterisk/trunk/apps/app_rpt.c?view=diffrev=53047r1=53046r2=53047

Re: [asterisk-dev] please tell me

Hello Mehul, This is not the correct mailing list for this type of question. If you post this message to the asterisk-users list, you are much more likely to receive a helpful response. Good luck, Sean On 3/21/07, mehul shah [EMAIL PROTECTED] wrote: Hello , i installed

Re: [asterisk-dev] LDAPget in Asterisk

Sravana, You'll want to direct this question to the correct people, the asterisk-users mailing list. The asterisk-dev list has to do with the development _of_ Asterisk, not developing _with_ Asterisk. Good luck. Sean On 4/2/07, sravana [EMAIL PROTECTED] wrote: Hi All, Anybody have

Re: [asterisk-dev] Error while setup asterisk

This is a known bug in GNU make. You need to upgrade. On 4/23/07, Mahmoud Shouman [EMAIL PROTECTED] wrote: Dear All, Kindly note that I have the below error while setup asterisk make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory

Re: [asterisk-dev] compile asterisk in arm-linux

This type of question should be directed to the asterisk-users list from now on... That being said, a search for asterisk termcap support not found on google yields plenty of results. All suggesting that installing ncurses and ncurses-devel will resolve this problem. On 5/10/07, lizhong zhu

Re: [asterisk-dev] RTP bridiging optimization

I wonder if you considered sending the same message to the same mailing list 3 times in a row? (Yes, I know I am not helpful.) On 5/10/07, Vadim Lebedev [EMAIL PROTECTED] wrote: Hello I wonder if somebody considered to optimize rtp bridging using Linux splice and tee syscalls? Thanks Vadim

Re: [asterisk-dev] Transfer a call

We got it the first time you sent it, thanks :) You're more likely to get a timely response if you direct this to the correct mailing list. Specifically the asterisk-users mailing list. Good luck, Sean On 5/22/07, Ram Narayan Mishra [EMAIL PROTECTED] wrote: Dear All, I have installed

[asterisk-dev] Re: [asterisk-commits] oej: branch group/astridevcon2007 r65399 - /team/group/astridevcon2007/

+Meetings in small groups to cover one topic, in the afternoon - propably in the autrium outside of the +lab room Sure you didn't mean ...probably in the atrium...? :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing

[asterisk-dev] Re: [asterisk-commits] oej: branch group/astridevcon2007 r65399 - /team/group/astridevcon2007/

(And I have a disclaimer on file ;-) On 5/22/07, Sean Bright [EMAIL PROTECTED] wrote: +Meetings in small groups to cover one topic, in the afternoon - propably in the autrium outside of the +lab room Sure you didn't mean ...probably in the atrium

Re: [asterisk-dev] Queue retry value

Awesome. Thanks a bunch. On 7/17/07, Sean Bright [EMAIL PROTECTED] wrote: Hey guys, I know this is the wrong list, but I'm more interested in the rationale behind this decision... Why is the 'retry' value in queues.conf limited to values 0? I am using rrmemory with a queue, and I am

Re: [asterisk-dev] Difference between trunk and branch/1.4

The trunk version of chan_mobile is designed to work with the trunk version of asterisk, not the 1.4 branch of asterisk. You can try an older revision of chan_mobile and see if that works for you: http://svn.digium.com/view/asterisk-addons/trunk/chan_mobile.c?revision=425 Try that one. On

Re: [asterisk-dev] Need help regarding G.729 Codec purchase

Um... http://store.digium.com/productview.php?category_id=5product_code=8G729CODECmain_category_id=5 On Nov 15, 2007 10:59 AM, Lokesh Agrawal [EMAIL PROTECTED] wrote: Hi All, I want to purchase commercial license of G.729 Codec through Digium. Can anyone help me for the same and tell me the

Re: [asterisk-dev] Bug #9650 - no ringing on transfer

Not sure if you saw or not, but file updated the issue in mantis: Fixed in 1.4 as of revision 90548 and trunk as of revision 90550. On Dec 5, 2007 4:08 PM, Nic Bellamy [EMAIL PROTECTED] wrote: This bug has been closed by file with the comment This has been fixed in 1.4 and trunk. There's no

Re: [asterisk-dev] [Code Review] 2723: Add pass through support for both VP8 and Opus

/2723/#comment20096 Shouldn't this be a_audio instead of a_text, or does it matter? - Sean Bright On Aug. 23, 2013, 3:42 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 2723: Add pass through support for both VP8 and Opus

On Jan. 16, 2014, 5:16 p.m., Sean Bright wrote: /trunk/channels/chan_sip.c, lines 13419-13421 https://reviewboard.asterisk.org/r/2723/diff/5/?file=44381#file44381line13419 Shouldn't this be a_audio instead of a_text, or does it matter? Matt Jordan wrote: Good catch - feel

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

it automatically update with /etc/resolv.conf is changed... /branches/12/res/res_pjsip/config_global.c https://reviewboard.asterisk.org/r/3343/#comment20797 The comment and function name are stale, but this was already resolved in your development branch. Just pointing it out. - Sean Bright

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

On 3/13/2014 4:42 PM, Paul Belanger wrote: +1 with Dan. Comments aside on DNS functionality (I have opinions but sitting this one out). Any functionality should be channel agnostic. I too am a little concern'd that statement seems to have changed. In order to make this channel agnostic you

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

On 3/14/2014 2:41 AM, Olle E. Johansson wrote: On 13 Mar 2014, at 22:13, Sean Bright sean.bri...@gmail.com mailto:sean.bri...@gmail.com wrote: On 3/13/2014 4:42 PM, Paul Belanger wrote: +1 with Dan. Comments aside on DNS functionality (I have opinions but sitting this one out). Any

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

On 3/14/2014 10:02 AM, Olle E. Johansson wrote: It would mean continuing to maintain Asterisk's pjproject fork until those changes were (hopefully) accepted upstream, released, and then waiting for the rpm/deb packages to catch up. Not to mention that someone would actually have to _do_ all

Re: [asterisk-dev] DNS PJSIP

On 3/17/2014 5:47 AM, Olle E. Johansson wrote: - We are still in control of our own product and make our own decisions about Asterisk architecture. Any arguments like PJSIP has it so we have to enable it falls to the ground as not valid and disappear in a cloud of smoke. Those that are in

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3343/#review11257 --- Ship it! Ship It! - Sean Bright On March 17, 2014, 1:15

Re: [asterisk-dev] [asterisk-commits] mjordan: branch 11 r409990 - /branches/11/res/res_fax_spandsp.c

On 3/17/2014 9:48 AM, Matthew Jordan wrote: Ugh. That stinks - there was no g711_free in 0.0.5. Looks like we're going to need a patch for this that examines the libspandsp version and uses one or the other. This has been fixed:

[asterisk-dev] [Code Review] 3372: Allow local ICE candidates to be overridden with specified address

/ Testing --- Ran in a static NAT environment with externaddr set and media setup succeeded as expected. Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev

Re: [asterisk-dev] [Code Review] 3372: Allow local ICE candidates to be overridden with specified address

this patch, so that might throw a wrench in this. - Sean Bright On March 18, 2014, 5:04 p.m., Sean Bright wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3372

Re: [asterisk-dev] [Code Review] 3372: Allow local ICE candidates to be overridden with specified address

On March 18, 2014, 5:33 p.m., Sean Bright wrote: I hadn't considered multihomed machines when I wrote this patch, so that might throw a wrench in this. There are a log of things wrong with this. Going to re-think it some. - Sean

Re: [asterisk-dev] [Code Review] 3372: Allow local ICE candidates to be overridden with specified address

/res_rtp_asterisk.c 410875 /trunk/CHANGES 410875 Diff: https://reviewboard.asterisk.org/r/3372/diff/ Testing --- Ran in a static NAT environment with externaddr set and media setup succeeded as expected. Thanks, Sean Bright

Re: [asterisk-dev] [Code Review] 3372: Allow local ICE candidates to be overridden with specified address

On 3/19/2014 6:30 AM, Olle E. Johansson wrote: The major one being that ICE was create so that we don't have to mess around like this. The external IP is just a server-reflexive address. You may want to make a decision on what you prefer in the C-line depending on the old logic - what's the

Re: [asterisk-dev] [Code Review] 3372: Allow local ICE candidates to be overridden with specified address

On 3/19/2014 6:41 AM, Olle E. Johansson wrote: On 19 Mar 2014, at 11:34, Sean Bright sean.bri...@gmail.com wrote: On 3/19/2014 6:30 AM, Olle E. Johansson wrote: The major one being that ICE was create so that we don't have to mess around like this. The external IP is just a server-reflexive

[asterisk-dev] [Code Review] 3391: ARI: Don't complain about missing ARI users when we aren't enabled

://reviewboard.asterisk.org/r/3391/diff/ Testing --- Basic runtime tests with ARI enabled and disabled. Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] [Code Review] 3391: ARI: Don't complain about missing ARI users when we aren't enabled

logging and error or iterating the container to validate the users. Diffs - /branches/12/res/ari/config.c 411084 Diff: https://reviewboard.asterisk.org/r/3391/diff/ Testing --- Basic runtime tests with ARI enabled and disabled. Thanks, Sean Bright

Re: [asterisk-dev] [Code Review] 3491: res_pjsip: Allow cipher to be specified by name

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3491/#review11787 --- Ship it! Ship It! - Sean Bright On April 29, 2014, 3:21

Re: [asterisk-dev] [Code Review] 3439: chan_sip: Support a=rtcp attribute in SDP

://reviewboard.asterisk.org/r/3439/#comment21785 Should this be an 'else if?' If not it should be on its own line. - Sean Bright On May 16, 2014, 12:36 p.m., Olle E Johansson wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] libpri 1.4.15 Now Available

On 6/16/2014 4:07 PM, Yves A. wrote: Maybe the development team of libpri could remove the 35 char maxlength constraint when using USERUSERINFO in the next release so I don´t have to patch it always... Or is there a specific reason for it? According to the Q.931 recommendation, user-user has

Re: [asterisk-dev] [Code Review] 3773: Add menuselect to Asterisk, remove mxml

into contrib/scripts/install_prereq while you're at it. This also might be a good opportunity to dump mxml altogether as we already use libxml2 in asterisk itself. One less dependency to install. I'd be happy to take a crack at it if you don't have the bandwidth. - Sean Bright On July 14

Re: [asterisk-dev] [Code Review] 3773: Add menuselect to Asterisk, remove mxml

On July 14, 2014, 8:37 p.m., Sean Bright wrote: Looks fine to me. Should probably throw libmxml-dev into contrib/scripts/install_prereq while you're at it. This also might be a good opportunity to dump mxml altogether as we already use libxml2 in asterisk itself. One less

[asterisk-dev] [Code Review] 3830: Fix build when pjproject is installed in non-standard location

/pjsip/lib/pkgconfig ./configure Diffs - /trunk/configure.ac 418979 /trunk/configure UNKNOWN Diff: https://reviewboard.asterisk.org/r/3830/diff/ Testing --- Configured, compiled. Thanks, Sean Bright

Re: [asterisk-dev] [Code Review] 3830: Fix build when pjproject is installed in non-standard location

/ Testing --- Configured, compiled. Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-dev] [Code Review] 3988: res_pjsip: Don't require a password when doing userpass authentication

Description --- An empty password is valid for username/password authentication so we shouldn't barf on it. Diffs - /branches/12/res/res_pjsip/config_auth.c 422963 Diff: https://reviewboard.asterisk.org/r/3988/diff/ Testing --- Compiles. Thanks, Sean Bright

Re: [asterisk-dev] [Code Review] 3988: res_pjsip: Don't require a password when doing userpass authentication

. Registered a device with auth_type=userpass and no password set. Tested registration with a password which failed, and again without a password (an empty password) and it succeeds. Thanks, Sean Bright -- _ -- Bandwidth

Re: [asterisk-dev] [Code Review] 3988: res_pjsip: Don't require a password when doing userpass authentication

that no freaking occurs. - Sean --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3988/#review13278 --- On Sept. 15, 2014, 7:27 p.m., Sean Bright

Re: [asterisk-dev] [Code Review] 3988: res_pjsip: Don't require a password when doing userpass authentication

/config_auth.c 422963 Diff: https://reviewboard.asterisk.org/r/3988/diff/ Testing --- Compiles. Registered a device with auth_type=userpass and no password set. Tested registration with a password which failed, and again without a password (an empty password) and it succeeds. Thanks, Sean

[asterisk-dev] [Code Review] 4106: configure: Add autoconf check for libopus

://reviewboard.asterisk.org/r/4106/diff/ Testing --- Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-dev] [Code Review] 4106: configure: Add autoconf check for libopus

/autoconfig.h.in 426095 /trunk/configure.ac 426095 /trunk/configure UNKNOWN /trunk/build_tools/menuselect-deps.in 426095 Diff: https://reviewboard.asterisk.org/r/4106/diff/ Testing --- Thanks, Sean Bright

[asterisk-dev] OT: Opus Asterisk 13

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. You can download the codec implementation and patch file here:

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

On 1/26/2015 8:16 AM, Olle E. Johansson 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. You can download the codec

Re: [asterisk-dev] [Code Review] 4371: Update res_format_attr_opus res_format_attr_silk to new media formats architecture

to codec_silk for Asterisk 13 so I cannot test. Diffs - /branches/13/res/res_format_attr_silk.c 431089 /branches/13/res/res_format_attr_opus.c 431089 Diff: https://reviewboard.asterisk.org/r/4371/diff/ Testing --- Ran a test call with codec_opus. Thanks, Sean Bright

Re: [asterisk-dev] [Code Review] 4389: Memory leak cleanups

/r/4389/#comment24872 You're missing a leading tab here. - Sean Bright On Jan. 29, 2015, 6:04 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4389

[asterisk-dev] [Code Review] 4371: Update res_format_attr_opus res_format_attr_silk to new media formats architecture

431089 /branches/13/res/res_format_attr_opus.c 431089 Diff: https://reviewboard.asterisk.org/r/4371/diff/ Testing --- Ran a test call with codec_opus. Thanks, Sean Bright -- _ -- Bandwidth and Colocation Provided

[asterisk-dev] Re: [BOUNTY] FATAL: unhandled exception PJLIB/No memory!

On 3/19/2015 6:38 AM, Ross Beer wrote: We are getting the following error with asterisk 12 and 13: [Mar 19 10:14:08] ERROR[40185]: pjsip:0 ?: except.c .!!!FATAL: unhandled exception PJLIB/No memory! Ross, Is there an associated issue in Jira? https://issues.asterisk.org/ Kind

Re: [asterisk-dev] [RTP] Detecting that a packet is SRTP

On 4/29/2015 11:20 AM, Yousf Ateya wrote: In res_rtp_asterisk.c, to detect that a packet is SRTP (not RTP), this check is done (link to gitweb

Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project

On 7/19/2016 10:35 AM, Matt Fredrickson wrote: Response below. On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse wrote: Technical Details (at the moment the modifications are based upon 13.6.0): * Enabled OPUS (with incoming stereo and outgoing stereo

Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project

On 7/19/2016 4:00 PM, Bruce Ferrell wrote: On 07/19/2016 10:59 AM, Sean Bright wrote: On 7/19/2016 10:35 AM, Matt Fredrickson wrote: Response below. On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse <dennis.g...@alumni.tu-berlin.de> wrote: Technical Details (at the moment the modific

Re: [asterisk-dev] Asterisk 13.14.0-rc2 Now Available

On 2/8/2017 5:23 PM, Steve Edwards wrote: I was hoping to see something like 'And now that the patents on g.729 have expired, Digium has contributed their codec to the open source project -- license and fee free' :) Same here. Luckily the licenses are still available for purchase on the

[asterisk-dev] Have you seen me? "Alertpipe on channel X lost O_NONBLOCK?!!"

Hi, Could those of you on the list grep your error logs looking for the message: lost O_NONBLOCK (The full line would be "Alertpipe on channel X lost O_NONBLOCK?!!" where "X" is a channel name). There is a code comment indicating that this is "occasionally" a problem and I would like to

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

On 10/8/2017 10:55 AM, James Finstrom wrote: So one of the things that is needed to finally put Chan sip to bed is feature parody.  Someone brought up CCSS. What features do you feel you would lose going from chan_sip to pjsip. Are there any bugs in pjsip that keep you from migrating? FWIW,

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

On 9/26/2017 12:37 PM, sean darcy wrote: Are there any RTP/RTCP leak issues with 15 RC1 ? What kind of leak? Memory leak? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

Re: [asterisk-dev] Weirdness-- intermittent loss of recordings w/ 13.5.0

On 11/14/2017 3:10 PM, Steve Murphy wrote: This is a preliminary cry for help... We are seeing a 51.2% 'loss' of recordings on one of our 13.5.0 systems. All calls are are in u-law format. The incoming calls are offered gsm and 729, among others,  but we restrict to only u-law. We only offer

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

Hi, On 12/22/2017 7:22 AM, Nir Simionovich wrote:   Every, and I do mean every, Asterisk application requires a key/value store of some form. Most developers will basically butcher (would have used stronger words, but refraining from doing so) AstDB in the process, which will then result in a

Re: [asterisk-dev] Asterisk 16 Parking Lot Full behavior.

On 12/26/2018 2:58 PM, Steve Sether wrote: Jira seems to be closed to outsiders opening bug reports. I'm happy to open one, but I don't have access. Is there some other way I can open a bug report? IIRC Jira charges per user, so if that's the case I can certainly see why they don't want to

Re: [asterisk-dev] Fwd: Bug#925760: libss7: ftbfs with GCC-9

A fix has been merged: https://gerrit.asterisk.org/c/libss7/+/11206 I don’t know if/when it will be released. Kind regards, Sean On Fri, Mar 29, 2019 at 3:39 AM Tzafrir Cohen wrote: > Hi, > > > I wanted to report this as a bug on the libss7 component, but I failed > to do so: there are no

Re: [asterisk-dev] res_config_sqlite3 segfault?

On 7/11/2019 6:11 AM, Dennis Buteyn wrote: Asterisk seems to crash consistently on startup if you don't explicitly define "dbfile => ... " in res_config_sqlite3.conf. (Not having the config file at all is fine). At a glance, I don't see how this is possible with res_config_sqlite3 from

Re: [asterisk-dev] res_config_sqlite3 segfault?

On 7/11/2019 6:11 AM, Dennis Buteyn wrote: Asterisk seems to crash consistently on startup if you don't explicitly define "dbfile => ... " in res_config_sqlite3.conf. (Not having the config file at all is fine). I've been unable to reproduce this with Asterisk 13. If you aren't able to open

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

On 10/2/2019 4:02 PM, Michael Maier wrote: I found one more problem regarding the configuration options, provided by FreePBX, which should be supported by asterisk. I'm referring to the possibility, to add additional options not supported by FreePBX using special config files like

[asterisk-dev] res_calendar_exchange: anyone using it?

Hi, The res_calendar_exchange module currently relies on a library (libiksemel) that is abandoned and is being dropped from recent distros. I've converted it over to use libxml2 (which is already a required dependency of Asterisk), but I don't have an environment in which to test. Would

Re: [asterisk-dev] Asterisk 16.5.x / SIPS / SRTP / pjsip 4.9 - one more memory leak

On 10/3/2019 9:42 AM, Michael Maier wrote: Sorry, but there is one more memory leak even in asterisk 16.6.0-rc2, which can't be seen with pjsip 4.8 instead of 4.9. It can be seen on inbound calls (not sure if it's on outbound calls, too) using SIPS and SRTP. Examples: 1 Call, duration about

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

On 10/3/2019 9:52 AM, Michael Maier wrote: I think this should be enough - just install FreePBX and you will see it. You don't need any special configuration - it's the default configuration provided by FreePBX. Great. Hopefully someone on the FreePBX team also follows the Asterisk issue

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

On 10/3/2019 10:17 AM, Michael Maier wrote: This is not a FreePBX issue OK. Thank you for clarifying. provided by asterisk (take a look at the source code) That's a good idea. I'll take a look, thanks. Kind regards, Sean --

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

On 10/1/2020 7:56 AM, Joshua C. Colp wrote: Around this time each year a discussion always spurs (be it on IRC or at AstriDevCon) about deprecating modules, and removing them. I always find myself asking "what is our real process for doing this?" in my head and end up trying to piece it

Re: [asterisk-dev] External scripts for parsing the security logs

On 11/23/2020 4:09 AM, Mohit Dhiman wrote: can anyone please recommend any existing external scripts that can parse the Asterisk security logs and possibly take appropriate actions like IP blocking. Fail2ban --

Re: [asterisk-dev] Pain Points For Large Scale Instance Provisioning

On 10/20/2020 5:32 PM, Michael Cargile wrote: * Reliable module reloading without core restarts Example: Client lets their SSL certificate lapse on an Asterisk server and they only figure this out when their agents attempting to log in using WebRTC clients. They have dozens or even

Re: [asterisk-dev] PJSIP doesn't seem to process tokens with percent characters correctly

On 9/21/2021 1:36 PM, Dan Cropp wrote: > There seem to be two different issues with PJSIP processing of headers > with the % character in tokenized fields. If you could collect some debug information[1] and create a new issue on the issue tracker[2] that would go a long way towards getting this

Re: [asterisk-dev] Gerrit offline?

On 8/17/2023 9:04 PM, aster...@phreaknet.org wrote: > Would it be at all possible to extend that possibly at least a couple days, > perhaps through Wednesday at least? Shouldn't be necessary, I opened two PRs in your repo that remove the references to gerrit so you should be good to go. Kind

Re: [asterisk-dev] OpenAPI 3.1 API description for ARI

On 6/24/2022 4:36 PM, James Finstrom wrote: > I did convert and manually clean up the spec once. This obviously > isn't maintained Ugh. Same. My intentions were good though: https://github.com/seanbright/asterisk-ari-openapi-spec PRs welcome? Maybe?--

Re: [asterisk-dev] chan_sip realtime port with host dynamic and defaultip

On 3/15/2023 9:43 AM, Marcin Groszek wrote: > Recently I discovered  that when realtime is used the port  is ignored > when used with host=dynamic and defaultip=x.x.x.x. > >     if (port && !realtime && peer->host_dynamic) { >     ast_sockaddr_set_port(>defaddr, port); >    

Re: [asterisk-dev] Infrastructure move to GitHub

On 4/3/2023 11:11 AM, George Joseph wrote: > Last year we announced we were planning to move away from the Atlassian > products (Jira, Confluence, etc), Gerrit and Jenkins to GitHub.  That time is > upon us. This is great news. Thanks to you and everyone else who's been working on this! Kind

Re: [asterisk-dev] Deprecating users.conf

Hi, On 6/30/2023 7:45 AM, aster...@phreaknet.org wrote: > 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. Apologies - I realized after initially commenting on the PR that I could

Re: [asterisk-dev] Deprecating users.conf

Hi, On 6/30/2023 8:19 AM, Sean Bright wrote: > I and my users are using users.conf for nearly every install and removing > support for it would be disruptive to 100s of installs. After some internal discussion I think we can be migrated away from users.conf by the time that Aster

[asterisk-dev] C API Deprecation proposal: ast_gethostbyname()

Hi, Per the C API Deprecation policy¹ I am proposing the deprecation of ast_gethostbyame() in favor of the ast_sockaddr family of functions. No in-tree code currently uses this function. Assuming the function is deprecated in Asterisk 21 it will be removed in Asterisk 23. There is already an