Re: [asterisk-dev] ARI events order

to those aggregation topics, but that's a lot of overhead.) You could argue that it *should* be ordered with respect to both the channel and bridge's lifetime, but generally enforcing that ordering imposes a penalty on the system, as it requires more synchronization. If someone went down that pat

Re: [asterisk-dev] Asterisk 14 doesn't cache media, really

alue of max-age or s-maxage, or an expires header value). If it expired, we get a new version of the file. So - you'll need to check the HTTP responses from your server to determine why we would get a new version each time. Having an expiration value, no ETag, or cache-control values that indicate

Re: [asterisk-dev] Help with AOC (Advice Of Charge) - price info on outbound calls

ted correctly. For what it's worth, the data models wiki page shows the various events that can be received: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Hu

[asterisk-dev] Line length restrictions in code changes

than: (1) Should there even be a line length rule? (2) If there is a line length, what is a reasonable length given some of our function names? (Looking at things like ast_sip_get_mwi_disable_initial_unsolicited) (3) Should we simply advocate for readability, with examples of what is readable and

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

6852a60310df96). > > Enjoy! > --- > Dennis Guse Thank you Dennis for all your hard work on this feature! Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- __

Re: [asterisk-dev] asterisk-dev Digest, Vol 151, Issue 14

On Wed, Feb 22, 2017 at 1:20 AM, Dương Nguyễn Văn <vanduong...@gmail.com> wrote: > > Re: Contents of asterisk-dev digest... > Thanks for pointing that individual out. They've been unsubscribed. -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - U

Re: [asterisk-dev] Subscription Persistence Issues

han the other to implement. It's > just a matter of backwards compatibility vs extra traffic and the 'SHOULD' > uncertainty. > > Thoughts? > > If you go with the first approach, do the changes to pjproject to fix the expiration timer issues cause any API changes? Or is it merely fi

Re: [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip

On Mon, Jan 30, 2017 at 4:44 PM, George Joseph <gjos...@digium.com> wrote: > > > On Mon, Jan 30, 2017 at 3:32 PM, Matthew Jordan <mjor...@digium.com> > wrote: > >> >> >> On Mon, Jan 30, 2017 at 3:22 PM, Matt Fredrickson <cres...@digium.com> >

Re: [asterisk-dev] Wish: adding intelligent codec negotiation to asterisk / pjsip

do transcoding - which is not always easy to figure out. Things get challenging as well when you have a multi-party bridge involved at all - either when a dialed party has to be placed into a multi-party bridge or when you have a chain of Local channels dialing someone and the far end of the Local channe

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

ription, we could have a situation where the phone is re-registered but no longer receives state updates. I'll grant the above scenario is pretty unlikely, but it is plausible. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 3

Re: [asterisk-dev] app_queue missed calls per agent - caller hangup before timeout

r > *membername) > { > if (option_verbose > 2) > ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", rnatime); > ast_queue_log(qe->parent->name, qe->chan->uniqueid, membername, > "RINGNOANSWER", "%d", rnatime); > --- > &

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

as ever useful to anybody so I'm > >> leaning towards option 1 but we need feedback. > >> > >> This would be a change to 13, 14 and master. > > > > Since it was not really useful I'm okay with 1. > > Just as something to consider: > > If

Re: [asterisk-dev] ast_sip_session

etc.) to modify the outbound channel(s) prior to dialing. Doing that gives you more control over exactly _when_ you pass information along, and helps you avoid the many, many edge cases that can occur. -- Matthew Jordan Digium, Inc. | CTO 445

Re: [asterisk-dev] group video calling

not happening, I would suspect that's something that could be fixed relatively easily in app_dial. -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- __

Re: [asterisk-dev] Proposed Working Group Guidelines

nment on github, with github issues and labels etc but its been made > clear that isn't really an option unfortunately. > > There was talk of needing at least one core contributor (doesn't need to > be a digium employee in my mind) to be one of the 3 initial memb

[asterisk-dev] Remote media playback mea culpa

+Documentation [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Project+-+URI+Media+Playback -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk

Re: [asterisk-dev] ARI StasisEnd event vs. channel variables

On Tue, Oct 18, 2016 at 4:02 PM, Joshua Colp <jc...@digium.com> wrote: > Matthew Jordan wrote: > > > > >> There are a few wrinkles with emitting channel variables with >> arbitrary events (of which StasisEnd would qualify). >> >> When an event is emitt

Re: [asterisk-dev] ARI StasisEnd event vs. channel variables

sure how you would convey variables only on that event and not on all the rest. -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- __

Re: [asterisk-dev] Retransmission timeout reached on transmission

> The asterisk-dev list is used for discussion regarding the development of Asterisk, not for assistance in its usage. Please ask your question on the asterisk-users mailing list [1] or on the community forums [2]. [1] http://lists.digium.com/mailman/listinfo/asterisk-users [2] https://community.asteri

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

you have to force it. There will be blood, but >> nothing that can't be cleaned up with a little bleach and some elbow grease >> >> -- >> James >> >> -- >> _ >> -- Bandwidth and Colocation Pr

Re: [asterisk-dev] Working Groups

one would benefit from this process - but the more ambitious the recommendation, the more difficult it is to build consensus and resources to accomplish the goal. Having expectations be set would be good to avoid disappointment. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - H

Re: [asterisk-dev] [asterisk-commits] format ogg opus: New format (asterisk[master])

ts.digium.com/mailman/listinfo/asterisk-dev -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation P

Re: [asterisk-dev] BRIDGEPEER on multi-party conferences: Thoughts?

behavior, but only calculating it when someone asks for BRIDGEPEER - seems like a reasonable middle ground. The downside is just that there won't be any events fed to people, which means people who have built systems looking for those events are going to be broken. I suppose we could do the 'unfortu

Re: [asterisk-dev] BRIDGEPEER on multi-party conferences: Thoughts?

a fair number of testsuite tests, although that shouldn't be a reason not to do the change. It is evidence, however, that we ourselves relied on the setting of BRIDGEPEER to get an indication of who the channel was in a bridge with. -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Hu

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

the experience is like if a lower sampling rate is used. What happens if the sampling rate used in the conference is 8kHz or 16kHz? > 4. Is the dependency to libfftw3 an issue? libfftw3 if GPLv2, so no, it should not be an issue. https://github.com/FFTW/fftw3/blob/master/COPYING That being said

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

ary. That means that all the rest can be put up for code review, and anyone who wants to take advantage of the new feature(s) can simply load an externally available codec_opus. So - if you're interested - please do contribute the patches back upstream. -- Matthew Jordan Digium, Inc. | CTO 445 Ja

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

rg/wiki/display/AST/Code+Review [5] https://wiki.asterisk.org/wiki/display/AST/Code+Review+Checklist [6] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation [7] http://git.asterisk.org/gitweb/?p=asterisk/infrastructure.git;a=summary -- Matthew Jordan Digium, Inc. |

Re: [asterisk-dev] Asterisk Load Performance

ng trying to get 10,000 callers on > one Asterisk server. As Asterisk is not capable on one server for what I am > trying to do, I am going to design a scalable, multi-server architecture > instead. > While that's definitely a more sustainable approach, it has been awfully entertaining/in

Re: [asterisk-dev] ASTERISK-26145 - Task Process Issues possibly caused by HEP

e talking about, a patch should be provided and submitted to Gerrit. This is a patch that anyone - including the issue reporter - could write. To echo what Josh said, this doesn't really merit a lengthy discussion on the mailing list, nor does it require 'poking' developers. Matt -- Matthew J

Re: [asterisk-dev] Asterisk Load Performance

it decreases performance. >> >> With further testing and having implemented your suggestions, I am >> realizing the subm:devService-test-0038 task processor is a major >> bottleneck. I have always read that Asterisk can handle as

Re: [asterisk-dev] Asterisk Load Performance

essages are removed. For example, disabling 'ast_channel_snapshot_type' would break ... most things. You may however be able to streamline your application by looking at what ARI messages it cares about, what messages it doesn't, inspecting the code, and disabling those that you don't care about. Lots o

Re: [asterisk-dev] Help for developing a channel driver module

alog info bodies I'd encourage anyone writing a new channel driver to think of splitting the functionality up into - at a minimum - separate files. Keeping the Asterisk channel core integration separate from the protocol handling itself generally keeps things a lot cleaner. Hope this helps - Mat

[asterisk-dev] Asterisk 14 Feature Freeze Reminder

ki.asterisk.org/wiki/display/AST/Software+Configuration+Management+Policies#SoftwareConfigurationManagementPolicies-NewFeatures -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com &

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

templates, you could conceivably update the JSON and Python interpreters and not have to know a single line of C code. [1] https://github.com/apigee-127/swagger-tools/issues/335 -- Matthew Jordan Digium, Inc. | Director of Technol

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

to calculate. [1] https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- ___

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

On Wed, Jan 20, 2016 at 11:04 AM, Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote: > On Wed, Jan 20, 2016 at 09:49:32AM -0600, Matthew Jordan wrote: > > On Wed, Jan 20, 2016 at 8:22 AM, Jared Smith <jaredsm...@jaredsmith.net> > > > So why is pjproject such a pain? >

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

port was added to pjproject. If you go back in time > almost everything needed to make it work in a bundled configuration is > there already. > Would we automatically download and link pjproject when someone runs 'make', or would it require some different make target? -- Matthew Jorda

Re: [asterisk-dev] Asterisk - 13.6.0 - While bring up Asterisk seeing error "undefined symbol":

ration of this function pointer. The fact that your Errors.txt file is littered with many other symbol resolution issues makes me think that you have 'tweaked' something, attempted to build on a very non-standard platform, or are otherwise operating outside the bounds of the usual build system.

Re: [asterisk-dev] Asterisk - 13.6.0 - Registering OWN Module(Application):

E_INFO macros. Failure to add those will cause your module to not be registered, and hence not be loaded. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- ___

Re: [asterisk-dev] Implementing Native Bridging in a DAHDI kernel driver

module (outside of the channel driver) and selected based on criteria it presents to the core. The DAHDI native bridging is now implemented in the bridge_native_dahdi module. Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

Re: [asterisk-dev] Lock inversion deadlock in asterisk-11.21.0-rc1 and probably 12.x and 13.x

best. > My bad - I must be slowing down for Xmas. > > > Better to make sure everyone knows before a release gets made than to let it slip through. Thanks Steve! -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

Re: [asterisk-dev] Implementing Native Bridging in a DAHDI kernel driver

> > -Fabio Urquiza > Hi Fabio: The answer to this depends on which version of Asterisk you're targeting. Which version did you have in mind? Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Dav

Re: [asterisk-dev] app_queue: wrapuptime is intermittently disregarded

/* Let wrapuptimes override device state availability */ if (mem->lastcall && q->wrapuptime && (time(NULL) - q->wrapuptime < mem->lastcall)) { available = 0; } return available; } As it would be prone to the race condition you're descr

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

. Most of these tests simply use SIPp to send/receive an offer from Asterisk and receive/send a corresponding answer: * tests/channels/SIP/SDP_offer_answer * tests/channels/SIP/SDP_attribute_passthrough * tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/audio/basic

[asterisk-dev] Bridges, T.38, and other good times

jsip_directmedia_reinvite_t38/ [3] https://gerrit.asterisk.org/#/c/1761/ [4] https://jenkins.asterisk.org/jenkins/job/periodic-asterisk-master/80/testReport/junit/%28root%29/AsteriskTestSuite/tests_masquerade/ -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsvil

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

uire a CLA to be assigned to log into Gerrit. It makes it a lot simpler for everyone participating in the project to know what rules they are playing by. That being said, everything you've proposed sounds like it would be admissible. Codec modules are the only thing to be c

Re: [asterisk-dev] SIP channel X Local Channel

Asterisk 11 is an option, you may want to look into that functionality. The control frame that handles cause code propagation (which could reasonably be extended for more information) is AST_CONTROL_PVT_CAUSE_CODE. -- Matthew Jor

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

er is another way to deploy Asterisk, that may or may not use packages. (And as Chad pointed out, we're using Docker here at Digium for that purpose.) As Leif mentioned in his blog post, there's definitely benefits to using packages wit

Re: [asterisk-dev] Bug marshals back !

Let me know if there's anything you think might be missing on those wiki pages - we're always trying to improve the documentation of the project! Thanks - Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - U

Re: [asterisk-dev] Asterisk 1.8 deadlock with Kernel 4.1

either attach the backtraces, or send them along in the text of the e-mail. (Plus, that way we'll have them for posterity.) -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-dev] Asterisk 1.8 deadlock with Kernel 4.1

On Wed, Nov 18, 2015 at 2:01 PM, Stefan Priebe <s.pri...@profihost.ag> wrote: > Am 18.11.2015 um 19:46 schrieb Matthew Jordan: > >> >> >> On Wed, Nov 18, 2015 at 12:37 PM, Stefan Priebe <s.pri...@profihost.ag >> <mailto:s.pri...@profihost.ag>> wrote

Re: [asterisk-dev] Lockups in Asterisk 11

g occurring here that is outside the control of Asterisk. Either way, nothing above makes me think there is a bug in Asterisk. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-dev] Asterisk-11 Trying to get Channel Name related to RTP/SIP Private

exceptions, but FAR fewer than what you'll find in Asterisk 11.) Any other approach is liable to not work. You may want to look at the original patch Jaco Kroon had for this feature on the issue tracker: https://issues.asterisk.org/jira/browse/ASTERISK-20754 If you read the comments however, you'll note

Re: [asterisk-dev] PJSIP realtime scalability problem

; >> lot better in this case. > >> > >> > > >> > Why would asterisk need to load the whole list of endpoints more than >> > 300 > >> > times is just completely beyond me. > >> > > >> > >> Hyperbole aside, it's becau

Re: [asterisk-dev] PJSIP realtime scalability problem

On Sat, Oct 17, 2015 at 10:03 AM, Michael Ulitskiy wrote: > Matthew, > > > > Thanks for the reply. > > Yes I do have caching enabled. While caching does somewhat help (there are > different problems there) Which problems? > with ongoing load it has nothing to do with

Re: [asterisk-dev] contrib/realtime sources

it.asterisk.org/gitweb/?p=asterisk/repotools.git;a=tree [2] http://git.asterisk.org/gitweb/?p=asterisk/repotools.git;a=blob;f=alembic_creator.py;hb=HEAD [3] http://git.asterisk.org/gitweb/?p=asterisk/repotools.git;a=blob;f=mkrelease.py;hb=HEAD -- Matthew Jordan Digium, Inc. | Director of Technology 445 J

Re: [asterisk-dev] PJSIP realtime scalability problem

d sorcery caching yet? https://wiki.asterisk.org/wiki/display/AST/Sorcery+Caching It is new in 13.5.0, and was designed explicitly for this kind of scenario. It probably didn't exist when Josh answered you on the -users list, but I would take a look at putting a cache in sorcery.conf for your

Re: [asterisk-dev] SIP/SDP: ptime in translation module?

- and packetization is a media session attribute - why not just store it on the ast_translator on the channel? Frankly, it is the closest to something keeping track of the state of the media session that I can think of, and that's really where packetization should pr

Re: [asterisk-dev] Discovering Asterisk with consul

On Fri, Oct 9, 2015 at 6:57 AM, Sylvain Boily <sbo...@avencall.com> wrote: > Hello, > > Le 2015-10-04 22:09, Matthew Jordan a écrit : >> >> On Thu, Oct 1, 2015 at 4:11 PM, Sylvain Boily <sbo...@avencall.com> wrote: >>> >>> Hello, >>>

Re: [asterisk-dev] Asterisk 13.6.0-rc2 Possible CDR Bug

ime on the CDR should not be set. The CDR should allow manipulation of its properties via the CDR function. Does that sound right? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://aste

Re: [asterisk-dev] SIP/SDP: ptime in translation module?

store it on the frames generated by that media session. What's nice about this is that codec modules are fed frames already; that means there's no additional lookups you'll have to do in your codec module. Of the two approaches I like #2 best, but some others may want to chime in on this as well. Matt

Re: [asterisk-dev] Discovering Asterisk with consul

reat - there's a few minor coding guideline [1] things you might want to take a look at first, but nothing too major. If you'd like to get it into Asterisk 13 as well as master (which will eventually be 14), we can talk about writing automated tests :-) [1] https://wiki.asterisk.org/wiki/display

Re: [asterisk-dev] Asterisk + WebRTC: No audio on any direction

gt; allow=ulaw > dtlscertfile=/etc/asterisk/keys/asterisk.pem > dtlsprivatekey=/etc/asterisk/keys/asterisk.pem > dtlssetup=actpass > context=myContext > > ... Thanks in advance The asterisk-dev mailing list is for discussions regarding the actual source code of Asterisk. Please us

[asterisk-dev] New Features Coming in 13.5.0

that as with all new features in Asterisk, these features have automated tests in the Asterisk Test Suite. However, if something did get missed, please make sure you file an issue and let the developer community know. Thanks! Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis

Re: [asterisk-dev] R: Using the Call Manager events queue to identify incoming calls (from DAHDI)

. For versions 12 and forward: yes: https://wiki.asterisk.org/wiki/display/AST/AMI+v2+Specification -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-dev] ARI Snoop

#Asterisk13ChannelsRESTAPI-startSilence That *should* start silence frames going to the target of the Snoop channel, at which point, there's at least something flowing through to allow the audiohook to start sending Whispered audio. (If I'm wrong, Josh will tell me why :-) -- Matthew Jordan Digium, Inc. | Director

Re: [asterisk-dev] ARI Snoop

we can figure out what is going on with your Whisper scenario. If the channel is answered and the Whisper isn't working, please do open up an issue and attach the logs illustrating it so we can't get to the bottom of the problem. Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan

Re: [asterisk-dev] Deadlock in pthread_exit due to lazy binding with libgcc

/listinfo/asterisk-biz -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

-maxcapturerate=48000 it should not even have to load up the opus patch because it is just a passthrough have you changed anything to chan_sip.c to make this work? Do you have res_format_attr_opus loaded? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW

Re: [asterisk-dev] To set realm in asterisk

of Asterisk who could also answer your question. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth

Re: [asterisk-dev] text2wav in ARI

feature, folks on the mailing list could certainly help point you in the right direction. Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-dev] pjsip vs cel

the context/extension later on in chan_pjsip_new after the channel has been created; it should be trivial to refactor that to pass that information into ast_channel_alloc_with_endpoint. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-dev] Asterisk Beacon Module Proposal

-- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-dev] systemd sd_notify() [was: Re: Journald support for Asterisk]

of Shutdown in asterisk.c::really_quit -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth

Re: [asterisk-dev] (unreported) uninitialized: struct ast_sockaddr

the existing usage semantics, you get the best of both worlds: your new feature gets what it needs, and the mainline Asterisk project - as well as other third party modules - aren't broken. So yes, I'd just use a char array with a fixed length. -- Matthew Jordan Digium, Inc. | Director of Technology 445

Re: [asterisk-dev] Journald support for Asterisk

without impacting the core - or impacting users who don't care for it. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-dev] Asterisk Beacon Module Proposal

On Fri, May 8, 2015 at 4:54 PM, Rodrigo Ramírez Norambuena decipher...@gmail.com wrote: On Fri, May 08, 2015 at 11:49:39AM -0400, Moises Silva wrote: On Thu, May 7, 2015 at 10:35 PM, Matthew Jordan mjor...@digium.com wrote: Hey everyone - At the past several AstriDevCon events, we've had

Re: [asterisk-dev] Asterisk Beacon Module Proposal

On Thu, May 7, 2015 at 11:40 PM, George Joseph george.jos...@fairview5.com wrote: On Thu, May 7, 2015 at 8:35 PM, Matthew Jordan mjor...@digium.com wrote: Hey everyone - At the past several AstriDevCon events, we've had an open discussion about adding a module to Asterisk that would gather

[asterisk-dev] Asterisk Beacon Module Proposal

suggestions for improvements. Thanks! Matt [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Beacon+Module -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org

[asterisk-dev] Asterisk 14 Status and Project Timeline

/AST/Asterisk+14+Projects [19] http://lists.digium.com/pipermail/asterisk-dev/2015-February/072748.html [20] http://lists.digium.com/pipermail/asterisk-dev/2015-March/073217.html -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-dev] send DTMF not Detected on SIP/XXXX using AMI +AGI+EXEC+SendDTMF

. The asterisk-dev list for discussing the actual source code of Asterisk; this appears to be a configuration issue. Please ask this question on the asterisk-users mailing list. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-dev] A Week with GIT/Gerrit

On Sat, Apr 18, 2015 at 8:26 PM, Matthew Jordan mjor...@digium.com wrote: On Thu, Apr 16, 2015 at 5:00 PM, George Joseph george.jos...@fairview5.com wrote: The Emails: Overall I think they're too verbose. Change in asterisk[master]: bridge.c: NULL app causes crash during attended transfer

Re: [asterisk-dev] A Week with GIT/Gerrit

instructions for squashing commits before review somewhere on the Git or Gerrit usage page. Ack. I'll try to get to that today. Maybe on the Git usage or Commit message page, since it isn't explicitly tied to using Gerrit? -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive

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

by chan_pjsip, currently, an offer with RTP/AVPF with crypto attributes is currently rejected by chan_sip. See https://issues.asterisk.org/jira/browse/ASTERISK-23989. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-dev] Branches

paste the full command + error so I can help track it down. Thanks - Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-dev] A Week with GIT/Gerrit

as well. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-dev] Git Migration update

On Tue, Apr 14, 2015 at 9:24 AM, Russell Bryant russ...@russellbryant.net wrote: On Mon, Apr 13, 2015 at 6:52 PM, Matthew Jordan mjor...@digium.com wrote: For *right now*, we are going to try cherry-picking the changes to the affected branches when the change is first up for review

[asterisk-dev] Asterisk 1.8/12 build system changes

and contains both the menuselect/Git changes as well as the security release changes. Note that in any of the cases mentioned above, the UPGRADE notes will clearly state what has changed in the version, including the menuselect alterations. Thoughts? Suggestions? Flames? -- Matthew Jordan Digium, Inc

Re: [asterisk-dev] [Code Review] 4490: astdb: Allow clustering of the Asterisk Database between multiple Asterisk servers

in writing an alternative approach, so if the expectation is that an AstDB wrapper around RabbitMQ or Redis will magically appear if I hit the delete key, that expectation is likely to be wrong. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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

On Tue, Apr 14, 2015 at 9:43 AM, Dan Jenkins dan.jenkin...@gmail.com wrote: 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

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

already be set up for the asterisk-commits list (although the last tweaks on it just went in yesterday). -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-dev] Git Migration update

On Mon, Apr 13, 2015 at 5:52 PM, Matthew Jordan mjor...@digium.com wrote: On Sun, Apr 12, 2015 at 1:57 AM, George Joseph george.jos...@fairview5.com wrote: On Sat, Apr 11, 2015 at 10:15 PM, Matthew Jordan mjor...@digium.com wrote: snip Further updates after Day 2 (3?): 1. Due to popular

Re: [asterisk-dev] Gerrit plugin suggestions

to be marked as not ready for review, and not show up in reviewers list of reviews. Updating the patch set or setting it back to Ready for Review causes it to show back up. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-dev] Git Migration update

On Sun, Apr 12, 2015 at 1:57 AM, George Joseph george.jos...@fairview5.com wrote: On Sat, Apr 11, 2015 at 10:15 PM, Matthew Jordan mjor...@digium.com wrote: I'm wondering if we can do this... You submit a review on the lowest target branch, using 13 as an example. The review gets reviewed

Re: [asterisk-dev] How to get peer review for patch to deprecated module

On Sun, Apr 12, 2015 at 11:26 AM, Alex Villací­s Lasso a_villa...@palosanto.com wrote: El 11/04/15 a las 22:59, Matthew Jordan escribió: On Sat, Apr 11, 2015 at 4:31 PM, Alex Villacís Lasso a_villa...@palosanto.com wrote: snip I'd recommend doing the following: * Re-open ASTERISK-20347

Re: [asterisk-dev] Gerrit plugin suggestions

at this one next. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-dev] How to get peer review for patch to deprecated module (was: Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition)

that support that. [1] https://issues.asterisk.org/jira/secure/DigiumLicense.jspa [2] https://gerrit.asterisk.org/#/ [3] https://gerrit.asterisk.org/#/admin/projects/asterisk [4] https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan

[asterisk-dev] Git Migration update

-- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-dev] Git Migration update

On Sat, Apr 11, 2015 at 11:15 PM, Matthew Jordan mjor...@digium.com wrote: 1. We need to determine the best way to handle maintaining the long running branches. While rebasing is appropriate for topic branches (team branches) that closely track a mainline branch, the mainline branches

[asterisk-dev] Asterisk is moving to Git (next week)

/pipermail/asterisk-dev/2015-April/074129.html [2] https://gerrit.asterisk.org/#/admin/projects/TEST-Asterisk [3] http://lists.digium.com/pipermail/asterisk-dev/2015-March/074026.html -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

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