Re: [asterisk-dev] Configuring multistream in chan_pjsip

On 06/05/2017 03:17 PM, Matt Fredrickson wrote: On Mon, Jun 5, 2017 at 2:31 PM, Joshua Colp <jc...@digium.com> wrote: On Mon, Jun 5, 2017, at 04:21 PM, Mark Michelson wrote: Hi folks, For those of you following along at home, I recently published review https://gerrit.asterisk.org/#/

[asterisk-dev] Configuring multistream in chan_pjsip

that would pertain to a single call. We can get to that discussion after this part gets settled. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUB

Re: [asterisk-dev] Asterisk 13 Queues - eventwhencalled=vars

sent in these events, channel variables can already be included. In your manager.conf file, in the general section, set channelvars = foo,bar,baz Every time a channel is sent in a manager event, the channel variables foo, bar, and baz will be s

Re: [asterisk-dev] Symmetric RTP

On 02/23/2017 08:28 AM, Andreas Krüger wrote: Hi, I was wondering with the following setup, if you would use Symmetric NAT. +--+ +--+| 89.192.76.11 | +--+ |Client 1 | LAN| Router | Internet|

Re: [asterisk-dev] Deadlock GIT 13

.6 Should an issue be created on Jira? Regards, Ross Hi Ross, This may be the same issue that George discovered earlier today while doing some testing. If you apply the code change on this review https://gerrit.asterisk.org/#/c/4902/ , do you still see the iss

[asterisk-dev] SDP API Wiki Page

API design is more important at this stage. If people could take a look at the SDP API page and offer feedback, that would be great. There are some specific questions at the bottom of the wiki page that would be good to get answered. Thanks, Mark Michelson [1] https://wiki.asterisk.org/wiki/displ

Re: [asterisk-dev] Problems with custom C-file: undefined reference

On 12/05/2016 04:53 PM, Julian Fleischhauer wrote: Hey all, I have a problem using a custom C-file. The error ouput received when compiling is given below. error output .../system.c:1330: undefined reference to `XML_ParserCreate' ... .../system.c:1465: undefined reference to `sip_get_header'

[asterisk-dev] ARI versioning in 13 and 14

sk and ARI will be out of sync (e.g. a user might assume that Asterisk 14.3.0 would require ARI version 14.3.0, when that's not actually the case). Feel free to voice your support for one of these options or to suggest something of your own. Thanks, Mark

Re: [asterisk-dev] Sending control frame outside a call

On 11/09/2016 01:11 PM, wferre...@nc.rr.com wrote: Hi, I have a customer who is running 12.8.2 and since their software needs to be certified at considerable expense and time will not be upgrading Asterisk anytime soon. In addition due to security requirements the only connection between

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

I have opened https://issues.asterisk.org/jira/browse/ASTERISK-26492 and have attached the patch there. Feel free to try it out and let me know on the issue how it works for you. On Fri, Oct 21, 2016 at 8:37 AM, Sébastien Duthil <sdut...@proformatique.com > wrote: > On 19/10/16 11:59

Re: [asterisk-dev] app_queue retry limit

On 10/20/2016 09:51 AM, marek cervenka wrote: hi, we have questions from busy call centers if the retry time parameter can be 0 (wait time before next agent call) in app_queue its prohibited } else if (!strcasecmp(param, "retry")) { q->retry = atoi(val);

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

On 10/19/2016 06:45 AM, Matthew Jordan wrote: On Tue, Oct 18, 2016 at 4:02 PM, Joshua Colp 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 emitted out of the

Re: [asterisk-dev] I want to make my first patch but have run into a problem and don't know how to progress.

in_features.c. This approach would be better than copying/pasting because it keeps the code in one spot and makes use of already tested code. Hope this was helpful, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http:/

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

n requested. One side-effect is that VarSet > events would never be produced for this variable, not sure how much this > would matter given better events to monitor ConfBridge joins/parts? > > On Tue, Aug 9, 2016 at 7:01 PM, Mark Michelson <mmichel...@digium.com> > wrote: > >

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

the variable. Let me know what your thoughts are on the matter. Thanks, Mark Michelson [1] https://gerrit.asterisk.org/#/c/3445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing

Re: [asterisk-dev] Icelandic numbers, dates and voicemail

I think you should feel free to submit this patch if you want it included in mainline Asterisk. If you are interested in submitting you patch, see https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage . That page details how to submit a patch to the Asterisk project. Just as a note, I don't

Re: [asterisk-dev] Issues with ARI python example for recording

you look at the HTTP traffic, you may be able to tell more clearly why the 404 is being sent. It's certainly possible there's a bug in Asterisk, but I would be more willing to bet that there's some sort of error in the python example. Although, li

[asterisk-dev] LibPRI and LibSS7 Migrated to git

that had been used in subversion. Mark Michelson [1] https://gerrit.asterisk.org/#/admin/projects/libpri [2] https://gerrit.asterisk.org/#/admin/projects/libss7 [3] https://github.com/asterisk/libpri [4] https://github.com/asterisk/libss7 [5] http://git.asterisk.org/gitweb/?p=libpri.git;a=summary [6

Re: [asterisk-dev] reloading XML docs for 'core show application'

On 03/04/2016 09:14 AM, Jeremy Kister wrote: after installing a module in asterisk, I want to also install some docs for it for "core show application" I can easily insert the appropriate xml in documentation/core-en_US.xml but the only way I've found Asterisk will use the new xml is to

Re: [asterisk-dev] Mixmonitor omits some frames

In most cases, flushing an audiohook is not a bad thing, nor is it something to be worried about. That's why it's a debug message. If you want it in your logs, then set the core debug level to 1 so it shows up. You will get more messages than you are used to seeing, but by only setting the core

Re: [asterisk-dev] Asterisk 13.7.0 app PAGE plays participant count

Either with Page or Confbridge itself. Either 1) The Page application is not telling Confbridge not to announce the number of users in the conference properly. 2) The Confbridge application is announcing the number of users despite the Page application's request for quiet. As a workaround, you

Re: [asterisk-dev] Proposing change to Queue and missed calls behavior

On 12/04/2015 01:00 PM, Stian Hvatum wrote: Hi, I have a problem with an accompanying solution that I wish to share, but I am not sure if it is valuable enough or correct enough to be suggested as a patch to Asterisk. When SIP-phones are members of a queue, they tend to accumulate "missed

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

On 12/06/2015 07:57 PM, Matthew Jordan wrote: Hello all - One of the efforts that a number of developers in the community here at Digium have been at work at are cleaning up test failures exposed by Jenkins [1]. One of these, in particular, has been rather difficult to resolve - namely,

[asterisk-dev] ODBC improvements: Community opinions requested

we modify the existing connection pooling code in res_odbc to be more user-friendly? Thanks in advance, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To U

Re: [asterisk-dev] Transfer cmd (PJSIP not sending Referred-By but chan_sip does)

The answer to this is actually pretty simple: adding Referred-By in outgoing SIP REFERs is simply not implemented in chan_pjsip's chan_pjsip_transfer() function. As far as the syntax required for the Transfer() application, that's probably a case where that needs to be clarified in

Re: [asterisk-dev] Transfer cmd (PJSIP not sending Referred-By but chan_sip does)

) Thank you Mark *From:*asterisk-dev-boun...@lists.digium.com mailto:asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] *On Behalf Of *Mark Michelson *Sent:* Tuesday, August 25, 2015 10:30 AM *To:* Asterisk Developers Mailing List *Subject:* Re: [asterisk-dev

Re: [asterisk-dev] Transfer cmd (PJSIP not sending Referred-By but chan_sip does)

); / End of changes / pjsip_xfer_send_request(sub, packet); ast_queue_control_data(session-channel, AST_CONTROL_TRANSFER, message, sizeof(message)); } *From:*asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] *On Behalf Of *Mark Michelson *Sent

Re: [asterisk-dev] new module

ast_tcptls_close_session_file() in order to tear down the TCP connection. Hopefully that points you in the correct direction. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list

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

for ld(1), it sounds like load-time binding would, at most, cause module loading to take longer. Are there any other potential issues to making this change? Mark Michelson -- _ -- Bandwidth and Colocation Provided by http

[asterisk-dev] Sorcery Caching Wiki Page

Hey fellow developers, I have created a wiki page that details the finer points of sorcery caching [1]. The intended audience for the page is Asterisk administrators. If you have any critiques of the page, please feel free to mention them. Mark Michelson [1] https://wiki.asterisk.org/wiki

Re: [asterisk-dev] Review Request: Sorcery Caching

I've had a look at the page now, and here are my thoughts: 1) One thing that isn't really made clear is the interaction between multiple sorcery wizards. In a real-world example, you'd want to hit the memory_cache first and then hit the database afterwards if you couldn't retrieve the object

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

On 04/14/2015 12:11 PM, Matthew Jordan wrote: snip Yup. SO! The question is: is this change worth having, or should it be scrapped in favour of some alternate approach that makes use of other technology? My feelings won't be hurt if the answer is ditch it and do something else - this was a fun

[asterisk-dev] Change in asterisk[master]: res_pjsip: Add external PJSIP resolver implementation using ...

Mark Michelson has posted comments on this change. Change subject: res_pjsip: Add external PJSIP resolver implementation using core DNS API. .. Patch Set 1: (3 comments) https://gerrit.asterisk.org/#/c/75/1/main

[asterisk-dev] Change in asterisk[1.8]: .gitignore: Ignore tarballs (*.gz)

Mark Michelson has posted comments on this change. Change subject: .gitignore: Ignore tarballs (*.gz) .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/86 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[12]: .gitignore: Ignore tarballs (*.gz)

Mark Michelson has posted comments on this change. Change subject: .gitignore: Ignore tarballs (*.gz) .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/85 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[1.8]: .gitignore: Ignore tarballs (*.gz)

Mark Michelson has submitted this change and it was merged. Change subject: .gitignore: Ignore tarballs (*.gz) .. .gitignore: Ignore tarballs (*.gz) This patch updates the root .gitignore file to ignore files with a .gz

[asterisk-dev] Change in asterisk[1.8]: Add .gitignore and .gitreview files

Mark Michelson has posted comments on this change. Change subject: Add .gitignore and .gitreview files .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/82 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[1.8]: Add .gitignore and .gitreview files

Mark Michelson has submitted this change and it was merged. Change subject: Add .gitignore and .gitreview files .. Add .gitignore and .gitreview files Add the .gitignore and .gitreview files to the asterisk repo. NB: You can

[asterisk-dev] Change in asterisk[1.8]: main/editline: Add .gitignore.

Mark Michelson has submitted this change and it was merged. Change subject: main/editline: Add .gitignore. .. main/editline: Add .gitignore. This patch adds a .gitignore for main/editline to ignore all build results. Change

[asterisk-dev] Change in asterisk[1.8]: main/editline: Add .gitignore.

Mark Michelson has posted comments on this change. Change subject: main/editline: Add .gitignore. .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/84 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[12]: main/editline: Add .gitignore.

Mark Michelson has posted comments on this change. Change subject: main/editline: Add .gitignore. .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/83 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[12]: main/editline: Add .gitignore.

Mark Michelson has submitted this change and it was merged. Change subject: main/editline: Add .gitignore. .. main/editline: Add .gitignore. This patch adds a .gitignore for main/editline to ignore all build results. Change

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

I have created a wiki page [1] with a test plan for DNS NAPTR/SRV for PJSIP. Please take a moment to have a look at the page and provide any feedback (on this list, not on the page please) about the test plan. Are there any test cases that I left out that should be covered? Are there any test

[asterisk-dev] Change in asterisk[12]: Add .gitignore and .gitreview files

Mark Michelson has submitted this change and it was merged. Change subject: Add .gitignore and .gitreview files .. Add .gitignore and .gitreview files Add the .gitignore and .gitreview files to the asterisk repo. NB: You can

[asterisk-dev] Change in asterisk[12]: Add .gitignore and .gitreview files

Mark Michelson has posted comments on this change. Change subject: Add .gitignore and .gitreview files .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/81 To unsubscribe, visit https

[asterisk-dev] Change in repotools[master]: mapmantis: Remove dependency on digium_jira

Mark Michelson has posted comments on this change. Change subject: mapmantis: Remove dependency on digium_jira .. Patch Set 2: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/68 To unsubscribe, visit https

[asterisk-dev] Change in repotools[master]: digium_jira: Refactor module to wrap the Atlassian JIRA REST...

Hello Jared K. Smith, I'd like you to reexamine a change. Please visit https://gerrit.asterisk.org/69 to look at the new patch set (#2). Change subject: digium_jira: Refactor module to wrap the Atlassian JIRA REST client

[asterisk-dev] Change in asterisk[master]: Fixing extconf compile

Mark Michelson has submitted this change and it was merged. Change subject: Fixing extconf compile .. Fixing extconf compile During the mass code deletion for clang support, a stray backslash was left behind that was causing

[asterisk-dev] Change in asterisk[13]: Fixing extconf compile

Mark Michelson has submitted this change and it was merged. Change subject: Fixing extconf compile .. Fixing extconf compile During the mass code deletion for clang support, a stray backslash was left behind that was causing

[asterisk-dev] Change in asterisk[11]: Fixing extconf compile

Mark Michelson has submitted this change and it was merged. Change subject: Fixing extconf compile .. Fixing extconf compile During the mass code deletion for clang support, a stray backslash was left behind that was causing

[asterisk-dev] Change in asterisk[master]: Fixing extconf compile

Mark Michelson has posted comments on this change. Change subject: Fixing extconf compile .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/90 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[13]: Fixing extconf compile

Mark Michelson has posted comments on this change. Change subject: Fixing extconf compile .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/89 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[11]: Fixing extconf compile

Mark Michelson has posted comments on this change. Change subject: Fixing extconf compile .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/88 To unsubscribe, visit https

[asterisk-dev] Change in asterisk[12]: .gitignore: Ignore tarballs (*.gz)

Mark Michelson has submitted this change and it was merged. Change subject: .gitignore: Ignore tarballs (*.gz) .. .gitignore: Ignore tarballs (*.gz) This patch updates the root .gitignore file to ignore files with a .gz

[asterisk-dev] Change in repotools[master]: digium_jira: Refactor module to wrap the Atlassian JIRA REST...

Mark Michelson has posted comments on this change. Change subject: digium_jira: Refactor module to wrap the Atlassian JIRA REST client .. Patch Set 2: (2 comments) https://gerrit.asterisk.org/#/c/69/2/digium_jira.py File

Re: [asterisk-dev] [Code Review] 4609: chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices.

channels, frames that come into the core don't have to go through any extra translations in order to be used in audiohooks. - Mark Michelson On April 9, 2015, 8:50 p.m., rmudgett wrote: --- This is an automatically generated e-mail

Re: [asterisk-dev] [Code Review] 4604: loader/main: Don't set ast_fully_booted until deferred reloads are processed

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4604/#review15153 --- Ship it! Ship It! - Mark Michelson On April 9, 2015, 2:57

[asterisk-dev] Change in testsuite[master]: rest_api/applications/stasisstatus: Make run-test executable

Mark Michelson has posted comments on this change. Change subject: rest_api/applications/stasisstatus: Make run-test executable .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/36

Re: [asterisk-dev] [Code Review] 4603: pjsip: Add external PJSIP resolver implementation using core DNS API

do another NAPTR lookup on the returned record? After all, the next NAPTR lookup might give me back the SIP records I want. What if I get a mix of empty flags and empty service records AND SIP+D2X records with s flag? What then? - Mark Michelson On April 8, 2015, 7:37 p.m., Joshua Colp wrote

[asterisk-dev] Change in testsuite[master]: rest_api/applications/stasisstatus: Make run-test executable

Mark Michelson has submitted this change and it was merged. Change subject: rest_api/applications/stasisstatus: Make run-test executable .. rest_api/applications/stasisstatus: Make run-test executable If it isn't executable

Re: [asterisk-dev] [Code Review] 4605: translate.c: Only select audio codecs to determine the best translation choice.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4605/#review15161 --- Ship it! Ship It! - Mark Michelson On April 9, 2015, 4:50

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

/trunk/include/asterisk/dns_test.h PRE-CREATION /trunk/include/asterisk/dns_internal.h 434218 Diff: https://reviewboard.asterisk.org/r/4598/diff/ Testing --- All DNS unit tests continue to pass. Thanks, Mark Michelson

Re: [asterisk-dev] [Code Review] 4594: Asterisk manager output escape message control characters

is in the same location where manager events are sent, you may also be able to get away with fewer heap allocations, too, which is another bonus. - Mark Michelson On April 7, 2015, 6:25 p.m., warren smith wrote: --- This is an automatically

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: Verified+1 -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: Code-Review+2 -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: -Verified Removing the Verified since there are still outstanding findings

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: Code-Review+1 (2 comments) It's looking good by me! https://gerrit.asterisk.org

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

be replaced with a key-value store that has the concept of replication built into it. If that were done, then presumably, there would be no need to involve Stasis or SIP PUBLISH in the process of sharing values between Asterisk boxes. - Mark Michelson On April 6, 2015, 2:22 a.m., Matt Jordan wrote

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 10: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/18

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has submitted this change and it was merged. Change subject: stasis: set a channel variable on websocket disconnect error .. stasis: set a channel variable on websocket disconnect error This test is to ensure

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

be relegated to a separate option. - Mark Michelson On April 6, 2015, 6:32 p.m., Kevin Harwell wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4592

[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.

Mark Michelson has uploaded a new patch set (#3). Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions. The sip_attended transfer test

[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.

Mark Michelson has uploaded a new patch set (#3). Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions. The sip_attended transfer test

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

was present in pjsip.conf. If default system configuration setup on startup fails, then res_pjsip.so should fail to load. I think an assertion here that cfg is non-NULL would do the trick instead of trying again to create default configuration settings - Mark Michelson On April 7, 2015, 4:05

Re: [asterisk-dev] [Code Review] 4589: res_pjsip_t38: Add missing initialization of udptl-far_max_datagram in t38_initialize_session()

is not filled in at all. What has been done to test this change? - Mark Michelson On April 6, 2015, 3:53 p.m., Juergen Spies wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4589

Re: [asterisk-dev] [Code Review] 4541: clang compiler warning: -Wpointer-bool-conversion

On April 7, 2015, 1:05 a.m., rmudgett wrote: /branches/13/apps/app_minivm.c, line 1842 https://reviewboard.asterisk.org/r/4541/diff/3/?file=73407#file73407line1842 Missing the ! if (!ast_strlen_zero()) Diederik de Groot wrote: Thanks again for checking my stuff,

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

/trunk/include/asterisk/dns_internal.h 434218 Diff: https://reviewboard.asterisk.org/r/4598/diff/ Testing --- All DNS unit tests continue to pass. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-dev] [Code Review] 4595: Voicemail API: fix handling of mailbox full condition

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4595/#review15101 --- Ship it! Ship It! - Mark Michelson On April 6, 2015, 8:20

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4598/#review15092 --- On April 6, 2015, 10:46 p.m., Mark Michelson wrote

[asterisk-dev] Change in testsuite[master]: Enable support for directory containing custom tests.

Mark Michelson has posted comments on this change. Change subject: Enable support for directory containing custom tests. .. Patch Set 1: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/27 To unsubscribe, visit

[asterisk-dev] Change in testsuite[master]: Testsuite: New test for FAX via PJSIP T38 with authentication

Mark Michelson has posted comments on this change. Change subject: Testsuite: New test for FAX via PJSIP T38 with authentication .. Patch Set 4: Code-Review+2 -- To view, visit https://gerrit.asterisk.org/28 To unsubscribe

Re: [asterisk-dev] [Code Review] 4522: segfault-queue-ASTERISK-23319

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4522/#review15105 --- Ship it! Ship It! - Mark Michelson On March 25, 2015, 4:10

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

/group/dns/main/dns_core.c 433885 /team/group/dns/include/asterisk/dns_internal.h 433885 Diff: https://reviewboard.asterisk.org/r/4542/diff/ Testing --- All previous DNS tests continue to pass, and all new tests added in this review pass as well. Thanks, Mark Michelson

[asterisk-dev] [Code Review] 4579: res_pjsip_messaging: Serialize outgoing MESSAGEs

://reviewboard.asterisk.org/r/4579/diff/ Testing --- The bug itself is incredibly difficult to have happen under normal circumstances, but I have confirmed that this patch has not hindered operations any. Thanks, Mark Michelson

Re: [asterisk-dev] [Code Review] 4571: Add usegmtime option to cel_pgsql

the Posting Code to Review Board section on this wiki page: https://wiki.asterisk.org/wiki/display/AST/Review+Board+Usage - Mark Michelson On April 3, 2015, 4:56 p.m., Rodrigo Ramirez Norambuena wrote: --- This is an automatically

[asterisk-dev] Change in testsuite[master]: Add SIP attended transfer for Asterisk 11.

Mark Michelson has abandoned this change. Change subject: Add SIP attended transfer for Asterisk 11. .. Abandoned -- To view, visit https://gerrit.asterisk.org/20 To unsubscribe, visit https://gerrit.asterisk.org/settings

[asterisk-dev] Change in testsuite[master]: Add SIP attended transfer for Asterisk 11.

Mark Michelson has posted comments on this change. Change subject: Add SIP attended transfer for Asterisk 11. .. Patch Set 1: I am abandoning this change in favor of change /c/29/ -- To view, visit https

[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.

Mark Michelson has posted comments on this change. Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. Patch Set 1: (1 comment) https://gerrit.asterisk.org/#/c/29/1/tests/channels/SIP

[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.

Mark Michelson has uploaded a new change for review. https://gerrit.asterisk.org/29 Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions

[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.

Mark Michelson has uploaded a new patch set (#2). Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions. The sip_attended transfer test

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

--- On March 27, 2015, 2:45 p.m., Mark Michelson wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4542

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

added in this review pass as well. Thanks, Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-dev] [Code Review] 4536: iax2_poke_noanswer expiration timer too short

On April 1, 2015, 1:07 a.m., Matt Jordan wrote: trunk/channels/chan_iax2.c, line 12370 https://reviewboard.asterisk.org/r/4536/diff/1/?file=72980#file72980line12370 The usage of max_retries here feels arbitrary. I'm not against this being controlled more dynamically based on the

Re: [asterisk-dev] [Code Review] 4572: RFC: Refactor Qualify and res_pjsip/endpt_send_request

transaction, then there would only ever be a single timer running, and the possibility of races is eliminated. - Mark Michelson On March 31, 2015, 5:57 a.m., George Joseph wrote: --- This is an automatically generated e-mail. To reply, visit

[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error

Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 1: Code-Review+1 (4 comments) I noticed that I gave the Code-Review a 0 last time. I

Re: [asterisk-dev] [Code Review] 4556: dns: Add res_resolver_system module.

even lower than that is probably fine). - Mark Michelson On March 29, 2015, 6:05 p.m., Joshua Colp wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4556

[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.

/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson

[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.

Mark Michelson has uploaded a new patch set (#3). Change subject: Rewrite sip_attended_transfer test to stop failing. .. Rewrite sip_attended_transfer test to stop failing. The sip_attended_transfer test has been bouncing

[asterisk-dev] Change in testsuite[master]: rest_api/channels/snoop_spy: Stop test on bridge destruction

Mark Michelson has posted comments on this change. Change subject: rest_api/channels/snoop_spy: Stop test on bridge destruction .. Patch Set 1: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/21 To unsubscribe

Re: [asterisk-dev] [Code Review] 4574: pjsip: resolve compatibility problem with ast_sip_session struct

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4574/#review15021 --- Ship it! Ship It! - Mark Michelson On April 1, 2015, 9:20

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