Re: [asterisk-dev] git migration update

+1 from me as well. We use the methodology of using personalized repos for projects and it works really well. We use either GitHub or BitBucket, depending on the project - but both work equally well. I'm confident that Atlassian will be happy to show their support by contributing a Stash license

Re: [asterisk-dev] [Code Review] 4292: app_macro: Don't restore the calling location on a channel redirect.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4292/#review14027 --- You'll need to write a test that verifies this code change. No

Re: [asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4293/ --- (Updated Dec. 22, 2014, 10:19 p.m.) Review request for Asterisk Developers

[asterisk-dev] [Code Review] 4293: testsuite: Add a test for PJSIP keep alive packets for connection oriented transports

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4293/ --- Review request for Asterisk Developers. Repository: testsuite Descriptio

Re: [asterisk-dev] git migration update

Hi, Thanks for the update. On Mon, Dec 22, 2014 at 01:03:17PM -0600, Samuel Galarneau wrote: > Just wanted to update everyone on the git migration status and illicit some > feedback on a few items. > > We setup an instance of Gerrit and Jenkins internally, imported the > Asterisk testsuite into

Re: [asterisk-dev] [Code Review] 4292: app_macro: Don't restore the calling location on a channel redirect.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4292/ --- (Updated Dec. 22, 2014, 7:19 p.m.) Review request for Asterisk Developers.

[asterisk-dev] [Code Review] 4292: app_macro: Don't restore the calling location on a channel redirect.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4292/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23850 https://i

Re: [asterisk-dev] [asterisk-commits] rmudgett: trunk r430028 - /trunk/channels/chan_vpb.cc

Curse you chan_vpb! On Mon, Dec 22, 2014 at 2:25 PM, SVN commits to the Asterisk project wrote: > Author: rmudgett > Date: Mon Dec 22 14:25:40 2014 > New Revision: 430028 > > URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=430028 > Log: > Fix compilation since the patch for ASTERISK-2436

Re: [asterisk-dev] git migration update

On Mon, Dec 22, 2014 at 3:08 PM, George Joseph wrote: > On Mon, Dec 22, 2014 at 12:03 PM, Samuel Galarneau > wrote: > >> 2 - we have a few options as far as team branches go. We could configure >> user branches using refs/heads/team/${username}/* permissions in Gerrit to >> allow users to create

Re: [asterisk-dev] confbridge feature request

Has anyone thought of using some sort of crowd funding method for new features? I'm sure I'm not the only one who could use a good feature of this type. Regards -Original Message- From: Matthew Jordan Sent: Tuesday, December 16, 2014 8:53 PM To: Asterisk Developers Mailing List Subje

[asterisk-dev] [Code Review] 4291: Testsuite: Add external bridging tests for Stasis application interaction - Stasis bridge to non-stasis application.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4291/ --- Review request for Asterisk Developers and Mark Michelson. Bugs: ASTERISK-

Re: [asterisk-dev] [Code Review] 4259: DTMF atxfer: Setup recall channels as if the original transferrer originated the call.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4259/ --- (Updated Dec. 22, 2014, 3:18 p.m.) Status -- This change has been mar

Re: [asterisk-dev] git migration update

On Mon, Dec 22, 2014 at 12:03 PM, Samuel Galarneau wrote: > Just wanted to update everyone on the git migration status and illicit > some feedback on a few items. > > We setup an instance of Gerrit and Jenkins internally, imported the > Asterisk testsuite into a git repo, and configured Gerrit an

Re: [asterisk-dev] [Code Review] 4282: queue_log: Post QUEUESTART entry when Asterisk fully boots.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4282/ --- (Updated Dec. 22, 2014, 1:39 p.m.) Status -- This change has been mar

Re: [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

On Mon, Dec 22, 2014 at 12:01 PM, Matthew Jordan wrote: > On Mon, Dec 22, 2014 at 12:53 PM, Joshua Colp wrote: > > Currently within PJSIP there is no mechanism for the dialplan to examine > > things at an AOR or contact level. The depth to which you can venture is > > simply the endpoint. > > >

Re: [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

Matthew Jordan wrote: I think this gets 90% of the way towards what people (such as X-Rob, who filed ASTERISK-24341) were looking for. I don't think this quite addresses the use case of "I want to create an outbound INVITE request to all contacts on an AoR that do not currently have an activ

[asterisk-dev] git migration update

Just wanted to update everyone on the git migration status and illicit some feedback on a few items. We setup an instance of Gerrit and Jenkins internally, imported the Asterisk testsuite into a git repo, and configured Gerrit and Jenkins together. A few issues and questions have come up in the c

Re: [asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

On Mon, Dec 22, 2014 at 12:53 PM, Joshua Colp wrote: > Currently within PJSIP there is no mechanism for the dialplan to examine > things at an AOR or contact level. The depth to which you can venture is > simply the endpoint. > > In practice this can be annoying if you would like to examine things

[asterisk-dev] PJSIP_AOR and PJSIP_CONTACT Dialplan Functions

Currently within PJSIP there is no mechanism for the dialplan to examine things at an AOR or contact level. The depth to which you can venture is simply the endpoint. In practice this can be annoying if you would like to examine things and change an outgoing INVITE (to add a header for auto an

[asterisk-dev] [Code Review] 4290: pjsip_options: Fix continued qualifies after endpoint/aor deletion.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4290/ --- Review request for Asterisk Developers. Repository: Asterisk Description

[asterisk-dev] [Code Review] 4289: pjsip_options: Fix duplicate qualify schedules on reload.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4289/ --- Review request for Asterisk Developers. Repository: Asterisk Description

Re: [asterisk-dev] [Code Review] 4230: add capath support to res_pjsip

> On Dec. 11, 2014, 2:13 p.m., Joshua Colp wrote: > > trunk/res/res_pjsip/config_transport.c, line 220 > > > > > > As this requires a pjproject change this needs a configure check so it > > can be optional instead

Re: [asterisk-dev] [Code Review] 4279: chan_sip: Send CANCEL via proxy if CANCEL is to be sent after an UPDATE

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4279/ --- (Updated Dec. 22, 2014, 9:39 a.m.) Status -- This change has been mar

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4050/ --- (Updated Dec. 22, 2014, 8:33 a.m.) Status -- This change has been mar

Re: [asterisk-dev] ARI Stasis Application Examples and Tutorials

The examples on github[1] use client libraries for both the Python and JavaScript examples. Client libraries abstract out some things for ease of use, such as namespacing events for a given instance (channel, playback, et cetera) but more importantly perform a lot of boilerplate that you would othe

[asterisk-dev] ARI Stasis Application Examples and Tutorials

Hi All, I'm not sure if the dev list is the proper list of this, however, due to the fact that the issue at hand revolves around documentation and proper usage, I think bringing it up here is a good place. So, during the past few days, I've been trying to implement the Dial application using