Re: [asterisk-dev] Call unhold/topology change indication order

On Wed, May 11, 2022 at 9:54 AM Joshua C. Colp wrote: > On Wed, May 11, 2022 at 11:50 AM Fridrich Maximilian < > m.fridr...@commend.com> wrote: > >> > You're in off-nominal untested un thought of territory, so the code >> behavior >> > probably reflects that. Specifically having both audio and

Re: [asterisk-dev] Doxygen \since

Thanks for the feedback everyone. It seems like generally speaking using /since Doxygen command for the C API calls is not that useful and can be left off. A couple you brought up version documentation for public facing user API's and protocols (AMI, ARI, dialplan functions etc...). AMI [1] and

[asterisk-dev] Asterisk External Application Protocol

[1] http://lists.digium.com/pipermail/asterisk-dev/2021-March/078244.html [2] http://lists.digium.com/pipermail/asterisk-dev/2021-March/078258.html -- Kevin Harwell Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk.org --

[asterisk-dev] Doxygen \since

/commands.html#cmdsince [2] https://github.com/asterisk/asterisk/blob/master/include/asterisk/stasis.h#L305 -- Kevin Harwell Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk

Re: [asterisk-dev] Deadlock in SIP processing in Asterisk 16.

Response inline. On Tue, Nov 23, 2021 at 2:06 PM Steve Sether wrote: > We've had a couple instances of a deadlocks recently. This happened while > we were trying to move phones from one Asterisk server to another. It's > too much to describe here what goes in in this process, but from an >

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

I can't speak much directly about the confbridge packet loss scenario, but can talk a bit about OPUS. No guarantees it'll be helpful to this situation, or not information you don't already know :-) Most folks will want to leave codec OPUS alone, and let it do its thing. Meaning, OPUS is pretty

Re: [asterisk-dev] Asterisk 16 deadlock (maybe in AMI) stops responding to SIP

On Mon, Oct 4, 2021 at 4:36 PM Steve Sether wrote: > > We're running Asterisk certified/16.8-cert5 > > I looked through the changelog for the cert versions 6 - 11, but didn't > see this fix ported to this branch. Did I miss something, or has this fix > just not been back-ported yet? > You did

Re: [asterisk-dev] Detecting B-leg channels

What's the overall scenario you are trying to solve? Perhaps there is another way to do what you want to do without even modifying Asterisk code? For example, maybe this is something an ARI application could handle, or even straight dialplan using a combination of app_dial, pre-dial handlers, and

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

On Mon, Aug 23, 2021 at 12:24 PM Dan Jenkins wrote: > Based on > https://wiki.asterisk.org/wiki/display/AST/Module+Deprecation I think > you’re right when you say it shouldn’t be built by default > > On Mon, 23 Aug 2021 at 15:15, Alexander Traud > wrote: > >> While creating a minimal

Re: [asterisk-dev] =?UTF-8?B?44CQcmVzX3Bqc2lwX3JlZ2lzdHJhcg==?=.c: Error】

ea957658165021a6ad28.patch > > > [4] <https://wiki.asterisk.org/wiki/display/AST/PJSIP-pjproject> > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list >

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

_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Kevin Harwell Software Devel

[asterisk-dev] PJSIP_MEDIA_OFFER

. So thoughts, ideas, and/or expectations on how it should work? [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_MEDIA_OFFER -- Kevin Harwell Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk

Re: [asterisk-dev] Pjsip aor with multiple contacts

sed. So depending on how a particular contact is hashed, and stored in the container affects its location in said container. If all you have are static contacts then the same one should be chosen each time. However when including dynamic contacts it's possible (once registered, and reachable) i

Re: [asterisk-dev] ARI Text messaging : inconsistencies in the API ?

On Wed, Feb 26, 2020 at 1:12 AM Jean Aunis wrote: > Le 25/02/2020 à 19:09, Kevin Harwell a écrit : > > > > > I could never get (2). When trying to send variables in the > TextMessageReceived event I would get a validation error unless they are > formatted like (3).

Re: [asterisk-dev] ARI Text messaging : inconsistencies in the API ?

. Option C: Leave sendMessage as is (1), update the TextMessageVariable API definition to be similar to (1), e.g { "var name": "var value" }, and not { "key": "var name", "value": "var value" }. This of course breaks the current API definition, a

Re: [asterisk-dev] Asterisk 18 Planning: Codec Negotiation

On Fri, Jan 31, 2020 at 1:06 AM Michael Maier wrote: > On 30.01.20 at 23:20 Kevin Harwell wrote: > ... > > No worries it's my bad. I can see how what I wrote was ambiguous. What I > > meant was "if *either* remote *or* local is chosen". As in it doesn't > > ma

Re: [asterisk-dev] AMD

(reading of frames, etc.). -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-dev] Asterisk 18 Planning: Codec Negotiation

you don't provide any option to outgoing_sdp_send_prefs. > Maybe I missunderstood some more ... . > No worries it's my bad. I can see how what I wrote was ambiguous. What I meant was "if *either* remote *or* local is chosen". As in it doesn't matter which option value you choose.

Re: [asterisk-dev] AMD

you can please describe the overall scenario you are trying to implement, or the problem you are trying to solve. With a bigger picture of what you are trying to do someone might be able to point you in a direction. -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: htt

Re: [asterisk-dev] AMD

u need. A third way is to check if the channel hung up using the "ast_check_hangup" or "ast_check_hangup_locked" function (maybe the best option?). -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://aster

Re: [asterisk-dev] Asterisk 18 Planning: Codec Negotiation

based on list 3 and 1 >(remote ...) > >Or > >the list is based on 1 and 3 (local ...). > >The Codec order may differ between those >variants. > >This list only contains codecs which can be >found in list 3 and 1 at the same time >(

Re: [asterisk-dev] Asterisk 18 Planning: Codec Negotiation

Ugh I used the wrong keyboard shortcuts and the message sent before I was done. Below is the rest :-) On Wed, Jan 29, 2020 at 3:42 PM Kevin Harwell wrote: > On Wed, Jan 29, 2020 at 3:12 PM Michael Maier > wrote: > >> >> > >> >> >> From my p

Re: [asterisk-dev] Asterisk 18 Planning: Codec Negotiation

uld always > prefer the codecs desired by the caller. > > Did I got this correctly? > We're still working through the idea of the "transcode" option, and how it might work in practice. But what you have is the general idea. To better avoid it in the setup you have above I'd

[asterisk-dev] Asterisk 18 Planning: Codec Negotiation

+Negotiation Thanks! -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-dev] acl.c - option to turn off logging

gt; now. Each consumer should not need to be touched, unless they are to be > switched to silent. We have an obligation to maintain ABI and behavior of > API functions as best we can in case there are any outside consumers as > well. > Unfortunately due to how consumer possibly use of the cur

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

__ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Kevin Harwell

Re: [asterisk-dev] res_config_sqlite3 segfault?

debug Asterisk log [3] at startup could be helpful as well. [1] https://issues.asterisk.org/ [2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Thanks! -- Kevin Harwell Digium - A Sangoma Company | Senio

Re: [asterisk-dev] bad AMI events order

"112879.7","destlinkedid": > "112879.0" } > { "event": "Hangup", "uniqueid": "112879.8" ,"linkedid": > "112879.0" } > { "event": "AgentComplete&q

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

done without changes to the ARI definitions and in a way that does not undermine the design of ARI. > -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & h

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

> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Kevin Harwell Digium

Re: [asterisk-dev] Dynamic server side event type filtering in ARI

for the feedback! [1] https://gerrit.asterisk.org/#/c/asterisk/+/10978/ [2] https://gerrit.asterisk.org/#/c/asterisk/+/10977/ -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https://asterisk

Re: [asterisk-dev] AMI events in Asterisk 16.1 happening 26+ seconds after parking a call.

On Fri, Dec 28, 2018 at 3:56 PM Steve Sether wrote: > On Thur, Dec 20, 2018 at 10:57 AM Kevin Harwell > > > > Have you tested, or currently running with the exact same setup but with > > another version of Asterisk and you are not seeing the delay? > > I tested As

Re: [asterisk-dev] Dynamic server side event type filtering in ARI

rs? Perhaps a DELETE? But that would be twisting the > semantics of the DELETE, since the eventFilter resource would still be > there. A POST on /applications/.../eventFilter/reset ? Or a POST with > {"allow": null, "disallow": null} ? It's probably worthwhile

Re: [asterisk-dev] AMI events in Asterisk 16.1 happening 26+ seconds after parking a call.

Asterisk: https://wiki.asterisk.org/wiki/display/AST/Call+Parking https://blogs.asterisk.org/2016/03/30/setup-call-parking/ -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https://as

Re: [asterisk-dev] Dynamic server side event type filtering in ARI

On Thu, Dec 13, 2018 at 3:42 PM George Joseph wrote: > > > On Thu, Dec 13, 2018 at 2:27 PM Kevin Harwell wrote: > >> Greetings, >> >> I'm looking into adding the ability for an ARI application to dynamically >> control which event types it would like sent

[asterisk-dev] Dynamic server side event type filtering in ARI

. Thanks! [1] https://issues.asterisk.org/jira/browse/ASTERISK-28106 -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https://asterisk

Re: [asterisk-dev] Web Downloads

sterisk 15. > > The update might take a few minutes to sync, but the button should now show the latest version: 15.4.0. -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

[asterisk-dev] Asterisk release branch deletions

can be recreated from a given tag. Current plans are to initiate this process on Monday March 12, 2018. If anyone has any valid reasons, or concerns about the release branch deletions please speak up. Thanks! -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville

Re: [asterisk-dev] Branching in the Testsuite

On Mon, Jan 22, 2018 at 12:41 PM, Kevin Harwell <kharw...@digium.com> wrote: > I'm getting ready to start the process of branching the testsuite. I'm > going to give a basic overview here of what I'm going to do, so as things > get posted on gerrit people will have some idea as to

Re: [asterisk-dev] Branching in the Testsuite

this way it will minimize the size of the each review and help separate out the different types of changes. Thanks! -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- __

Re: [asterisk-dev] Minor Release Branches

shows "13.15" first (and selects it by default). > > Anyone have thoughts on this? > > Sounds good to me. You might even be able to automate this by making it part of the mkrelease script. Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-dev] Branching in the Testsuite

I went ahead and created an issue to track this: https://issues.asterisk.org/jira/browse/ASTERISK-27492 Feel free to add comments or suggestions on the issue as well. On Fri, Dec 15, 2017 at 10:59 AM, Kevin Harwell <kharw...@digium.com> wrote: > Greetings, > > We're thinki

[asterisk-dev] Branching in the Testsuite

. Especially any potential pitfalls or problems you might see with it. Thanks! -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk

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

> > > > > > Thanks for any suggestion. > > Marin > Have you tried the subscribe to all events[1] option in ARI? If not give that a try and see if the event(s) show up. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Events+REST+API -- Kevin Harwell Digium, Inc.

Re: [asterisk-dev] AstDB mySQL implementation

ng really basic, but much > has changed since then. > Are you referring to the coding guidelines? If so those can be found on the wiki[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsvil

Re: [asterisk-dev] New application for MQTT send

o block while it is in the "loop"). So in your application exec handler you'd initiate a thread that runs the "loop". > > Thanks, > Gabriel > > > -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at

Re: [asterisk-dev] Opus Codec not in the translation array?

On Tue, Aug 8, 2017 at 5:45 PM, Steve Murphy <m...@parsetree.com> wrote: > > On Tue, Aug 8, 2017 at 4:21 PM, Kevin Harwell <kharw...@digium.com> wrote: > >> On Tue, Aug 8, 2017 at 11:29 AM, Steve Murphy <m...@parsetree.com> wrote: >> >>> When I lo

Re: [asterisk-dev] Opus Codec not in the translation array?

it manually and then copy the *.so files into the Asterisk lib directory? -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org --

Re: [asterisk-dev] MOH still playing after attended transfer

shed up for code review others will have a better idea of your proposed changes and will comment appropriately. [1] https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage > > > Thanks for any advice! > > > > Jason > > > > > Hope that helps and thanks fo

Re: [asterisk-dev] AMI/ARI versioning

[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+%28AMI%29+Changes Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http:

Re: [asterisk-dev] OPUS horrible quality with packet loss

On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev wrote: > after fixing attr->fec in open source edition of OPUS and applying > Alexanders patch: ASTERISK-25629 (Native Packet-Loss Concealment) > I have following results: > > I setup 2 hosts with asterisk 13.14.0 and made

Re: [asterisk-dev] OPUS horrible quality with packet loss

On Mon, Apr 3, 2017 at 3:38 PM, Yury Tsaregorodtsev wrote: > Even forced enabled jitter doesn't make asterisk to ignore late arrived > packets. > During my tests jb was always enabled (forced). > Hrm I'd think the jitter buffer should ignore or drop the late packets. That

Re: [asterisk-dev] OPUS horrible quality with packet loss

On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev wrote: > > MOS on calls using open source opus higher almost twice. > Subjective opinion regarding audio quality: using open source codec > quality almost same as in example on http://opus-codec.org/examples/ with > 30%

Re: [asterisk-dev] OPUS horrible quality with packet loss

;fec=yes". When a packet has been lost and the decoder receives a frame with FEC data (and fec is enabled) it will attempt to rebuild the lost packet (current packet minus one) from the given FEC information. Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsv

Re: [asterisk-dev] AMI/ARI versioning

On Thu, Mar 30, 2017 at 6:54 PM, Corey Farrell <g...@cfware.com> wrote: > On 03/30/2017 07:14 PM, Kevin Harwell wrote: > I think it's worth referencing a previous discussion on this [1]. > Yes, thank you! I looked for this and for some reason my searches turned up nothi

Re: [asterisk-dev] AMI/ARI versioning

> > > [asterisk-branch-number].[minor].[patch] > > Actually, the proposal might be better represented as the following: [asterisk-branch-number].[major].[minor/patch] Or another way to state it: [asterisk-branch-number].[api breaking].[api non breaking] --

[asterisk-dev] AMI/ARI versioning

then start with 13.x.x for Asterisk 13 and 14.x.x for Asterisk 14. Thoughts and opinions? [1] http://semver.org/ -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk

Re: [asterisk-dev] Dynamic Payloads

to move forward with just using the current payload values assigned by Asterisk as defaults. This should cover the majority of user cases with less complexity. -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

Re: [asterisk-dev] Dynamic Payloads

eld > [2] <https://tools.ietf.org/html/rfc4566#section-6> attribute =rtpmap: > [3] <https://tools.ietf.org/html/rfc4566#section-8.2.3> > > > -- Kevin Harwell Digium, Inc. | Software Developer 445 J

Re: [asterisk-dev] Line length restrictions in code changes

t is readable and what is not? Readability should be first and foremost, and some examples would probably help with that. -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- __

[asterisk-dev] Dynamic Payloads

format? Would it lessen the chance of this being a potential breaking change for people (with the option of course people can always fall back to the way things used to work if need be)? [1] https://issues.asterisk.org/jira/browse/ASTERISK-26515 -- Kevin Harwell Digium, Inc. | Software Developer

Re: [asterisk-dev] app_queue: RINGNOANSWER event

d option (mentioned on the code review [1] and by Troy) would be to include the agent name/info in the ABANDON event. [1] https://gerrit.asterisk.org/#/c/4649/ Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com &am

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

versions 1.X.Y, Asterisk 14 will have ARI > versions 2.X.Y, and Asterisk 15 will end up with Asterisk 3.X.Y > > +1 for option 2. Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http:/

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

On Fri, Oct 21, 2016 at 10:49 AM, Corey Farrell wrote: > > I'm in favor per app config. I do not yet use ARI, but when I do I > will have '#tryinclude /etc/asterisk/ari.d/*.conf' in ari.conf. My > hope is that each ARI app would install it's own config to >

Re: [asterisk-dev] [Code Review] 4606: chan_sip: make progressinband default to no

the setting to the other values and reloaded/checked between each to make sure those got set correctly as well. Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev

Re: [asterisk-dev] [Code Review] 4606: chan_sip: make progressinband default to no

between each to make sure those got set correctly as well. Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-dev] [Code Review] 4606: chan_sip: make progressinband default to no

/ Testing --- Started Asterisk and loaded chan_sip with the new default value for progressinband. Check to make sure that is what it was set to. Changed the setting to the other values and reloaded/checked between each to make sure those got set correctly as well. Thanks, Kevin Harwell

Re: [asterisk-dev] [Code Review] 4592: chan_pjsip: 183 sent when inband_progress=no

it will only send it when 'inband_progress' is set to 'yes' Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-dev] progressinband in chan_sip default value

on this? Leave it defaulting to never? Changing it to no would be fine? [1] https://issues.asterisk.org/jira/browse/ASTERISK-23972 [2] https://reviewboard.asterisk.org/r/3700 -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-dev] [Code Review] 4597: res_pjsip: add CLI command to show global and system configuration

not. Changed some settings again and restarted and made sure both global and system changes took effect. Also removed the sections completely from the pjsip.conf file and made sure the defaults were shown. Thanks, Kevin Harwell

Re: [asterisk-dev] [Code Review] 4575: bridge.c: Hangup attended transfer target after it has been swapped out

manual testing with different attended transfer scenarios. Thanks, Kevin Harwell -- _ -- 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] 4597: res_pjsip: add CLI commands for global and system configuration

and made sure global settings changed, but system ones did not. Changed some settings again and restarted and made sure both global and system changes too effect. Also removed the sections completely from the pjsip.conf file and made sure the defaults were shown. Thanks, Kevin Harwell

Re: [asterisk-dev] [Code Review] 4598: Refactor duplicated DNS routines into common sections

/4598/#comment25770 Extraneous merge nomenclature. - Kevin Harwell On April 6, 2015, 5:46 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4598

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4597/#review15095 --- On April 7, 2015, 11:05 a.m., Kevin Harwell wrote

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4597/#review15109 --- On April 7, 2015, 11:05 a.m., Kevin Harwell wrote

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

effect. Also removed the sections completely from the pjsip.conf file and made sure the defaults were shown. Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing

Re: [asterisk-dev] [Code Review] 4598: Refactor duplicated DNS routines into common sections

://reviewboard.asterisk.org/r/4598/#comment25786 moar nitpicking :-) this could be changed to actual_len as well if you were so inclined. - Kevin Harwell On April 7, 2015, 12:49 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail

[asterisk-dev] [Code Review] 4592: chan_pjsip: 183 sent when inband_progress=no

to 'yes'. Diffs - branches/13/channels/chan_pjsip.c 434021 Diff: https://reviewboard.asterisk.org/r/4592/diff/ Testing --- Duplicated the issue with chan_pjsip always sending the 183. After the patch it will only send it when 'inband_progress' is set to 'yes' Thanks, Kevin Harwell

[asterisk-dev] Change in testsuite[master]: non_stasis_bridge_to_stasis_bridge: Update regex for ami events

Kevin Harwell has uploaded a new patch set (#3). Change subject: non_stasis_bridge_to_stasis_bridge: Update regex for ami events .. non_stasis_bridge_to_stasis_bridge: Update regex for ami events Due to a bug in Asterisk

Re: [asterisk-dev] [Code Review] 4582: res_pjsip: config option 'timers' can't be set to 'no'

branches/13/res/res_pjsip.c 433966 Diff: https://reviewboard.asterisk.org/r/4582/diff/ Testing --- Made sure the option can now be set to 'no' and that it clears the bit. Also set it to the other values and reloaded to make sure the field was updated correctly. Thanks, Kevin Harwell

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

both global and system changes too effect. Also removed the sections completely from the pjsip.conf file and made sure the defaults were shown. Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-dev] Change in testsuite[master]: non_stasis_bridge_to_stasis_bridge: Update regex for ami events

Kevin Harwell has posted comments on this change. Change subject: non_stasis_bridge_to_stasis_bridge: Update regex for ami events .. Patch Set 3: Updated the per review based on feedback and other findings. A couple of things

[asterisk-dev] Change in testsuite[master]: non_stasis_bridge_to_stasis_bridge: Update regex for ami events

: I0a3bcb1a0df7e7bdca02be827288f5f08b5140ce Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Kevin Harwell kharw...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: John Bigelow jbige...@digium.com

Re: [asterisk-dev] [Code Review] 4582: res_pjsip: config option 'timers' can't be set to 'no'

: https://reviewboard.asterisk.org/r/4582/#review15048 --- On April 3, 2015, 2:58 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https

Re: [asterisk-dev] [Code Review] 4582: res_pjsip: config option 'timers' can't be set to 'no'

://reviewboard.asterisk.org/r/4582/diff/ Testing --- Made sure the option can now be set to 'no' and that it clears the bit. Also set it to the other values and reloaded to make sure the field was updated correctly. Thanks, Kevin Harwell

Re: [asterisk-dev] [Code Review] 4582: res_pjsip: config option 'timers' can't be set to 'no'

, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

On April 2, 2015, 6:13 p.m., Kevin Harwell wrote: It is probably always the case that framehooks should not be attached twice. If this is true then it might be better to add a check in 'ast_framehook_attach' that first makes sure the hook is not already in the list. If so don't add

[asterisk-dev] [Code Review] 4575: bridge.c: Hangup attended transfer target after it has been swapped out

that initially caught the problem. Ran the modified test (had been set to always pass) after applying the patch and it worked as expected. Also did some manual testing with different attended transfer scenarios. Thanks, Kevin Harwell

[asterisk-dev] Change in testsuite[master]: non_stasis_bridge_to_stasis_bridge: Update regex for ami events

Kevin Harwell has uploaded a new patch set (#2). Change subject: non_stasis_bridge_to_stasis_bridge: Update regex for ami events .. non_stasis_bridge_to_stasis_bridge: Update regex for ami events Due to a bug in Asterisk

[asterisk-dev] Change in testsuite[master]: non_stasis_bridge_to_stasis_bridge: Update regex for ami events

Kevin Harwell has uploaded a new change for review. https://gerrit.asterisk.org/25 Change subject: non_stasis_bridge_to_stasis_bridge: Update regex for ami events .. non_stasis_bridge_to_stasis_bridge: Update regex for ami

Re: [asterisk-dev] [Code Review] 4577: res_pjsip_t38: T38 fax fails when using authentication with PJSIP sender

not be attached twice. If this is true then it might be better to add a check in 'ast_framehook_attach' that first makes sure the hook is not already in the list. If so don't add it again. - Kevin Harwell On April 2, 2015, 2:08 p.m., Jonathan Rose wrote

Re: [asterisk-dev] [Code Review] 4542: DNS: Add NAPTR support and tests

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4542/#review15031 --- Ship it! Ship It! - Kevin Harwell On April 1, 2015, 9:51

Re: [asterisk-dev] [Code Review] 4542: DNS: Add NAPTR support and tests

/group/dns/tests/test_dns_naptr.c https://reviewboard.asterisk.org/r/4542/#comment25606 Should these failures break the loop and just goto cleanup as well? - Kevin Harwell On March 27, 2015, 9:45 a.m., Mark Michelson wrote

Re: [asterisk-dev] [Code Review] 4520: Testsuite: stasis: set a channel variable on websocket disconnect error

On March 24, 2015, 1:15 p.m., Kevin Harwell wrote: ./asterisk/trunk/tests/rest_api/applications/stasis_status/test_case.py, lines 22-23 https://reviewboard.asterisk.org/r/4520/diff/1/?file=72750#file72750line22 A lot of the code in this object as well as others (AriClient

Re: [asterisk-dev] [Code Review] 4523: res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4523/#review14854 --- Ship it! Ship It! - Kevin Harwell On March 26, 2015, 11:46

Re: [asterisk-dev] [Code Review] 4477: app_confbridge (11): file playback blocks dtmf

file playback would stop (no longer have to wait) and a new option was executed when appropriate. Also ran the app_confbridge testsuite tests to make sure they still passed. Thanks, Kevin Harwell -- _ -- Bandwidth

Re: [asterisk-dev] [Code Review] 4510: app_confbridge (13): file playback blocks dtmf

users to enter. Ran through various menu options to make sure the sound file playback would stop (no longer have to wait) and a new option was executed when appropriate. Also ran the app_confbridge testsuite tests to make sure they still passed. Thanks, Kevin Harwell

Re: [asterisk-dev] [Code Review] 4510: app_confbridge (13): file playback blocks dtmf

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4510/#review14798 --- On March 19, 2015, 4:59 p.m., Kevin Harwell wrote

Re: [asterisk-dev] [Code Review] 4510: app_confbridge (13): file playback blocks dtmf

to make sure the sound file playback would stop (no longer have to wait) and a new option was executed when appropriate. Also ran the app_confbridge testsuite tests to make sure they still passed. Thanks, Kevin Harwell

Re: [asterisk-dev] [Code Review] 4498: res_pjsip: Enable unload of all modules at shutdown

be fixed at a later time. - Kevin Harwell On March 20, 2015, 11:17 p.m., Corey Farrell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4498

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