Re: [asterisk-dev] git migration update

On Wed, Dec 24, 2014 at 5:06 PM, Olle E. Johansson wrote: > > You are missing one thing. When committing to the current team branches, > the code is contributed under the license agreement. > > The code in my branches is available for Digium to use at any point in > time. If I had to have it in m

Re: [asterisk-dev] git migration update

On 23 Dec 2014, at 21:53, Paul Belanger wrote: > On Tue, Dec 23, 2014 at 7:20 AM, Leif Madsen > wrote: >> On 22 December 2014 at 18:34, Russell Bryant >> wrote: >>> >>> On Mon, Dec 22, 2014 at 3:08 PM, George Joseph >>> wrote: On Mon, Dec 22, 2014 at 12:03 PM, Samuel Galarneau >>

Re: [asterisk-dev] [Code Review] 4296: PJSIP: Fix bugs and improve documentation of remote attended transfers

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Re: [asterisk-dev] Volume Control

On Wed, Dec 24, 2014 at 1:14 PM, Murthy Gandikota wrote: > To correct myself, it has nothing to do with hardware. I am using > ast_openstream acquire the stream object. I then use ast_write to send a > frame on the stream to all the conference members. Meanwhile another > thread is waiting for

Re: [asterisk-dev] Volume Control

To correct myself, it has nothing to do with hardware. I am using ast_openstream acquire the stream object. I then use ast_write to send a frame on the stream to all the conference members. Meanwhile another thread is waiting for events on the channel (ast_waitfor) and if a spoken frame is s

Re: [asterisk-dev] Volume Control

On Wed, Dec 24, 2014 at 1:02 PM, Murthy Gandikota wrote: > Using > > > > set(Volumn(TX)=1) > That wouldn't be the correct invocation of the function (VOLUME, not Volumn). > set(Volume(RX)=1) > > > > in the dialplan had no effect. > Did it have no effect on a channel not in App Konference? H

Re: [asterisk-dev] Volume Control

Using set(Volumn(TX)=1) set(Volume(RX)=1) in the dialplan had no effect. I understand that App Konference is not part of the Asterisk code base. I am using it because the source code to set up a dynamic conference--as opposed to confbridge or meetme--is given away at sourceforge.net. In gener

Re: [asterisk-dev] [Code Review] 4283: Testsuite: Dual channel redirect tests

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4283/ --- (Updated Dec. 24, 2014, 1:01 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] Volume Control

Per channel, there is a dialplan function called volume. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_VOLUME AFAIK App Konference is not part of the Asterisk code base. On Wed, Dec 24, 2014 at 3:08 PM, Murthy Gandikota wrote: > Hello All > > > > What is the standard practic

[asterisk-dev] Volume Control

Hello All What is the standard practice to adjust the volume on a channel? I am using App Konference where they have a talk volume and listen volume. No matter what I try, it's not making a difference. By the way, I know that the phone comes with a volume control. I am interested in the softw

Re: [asterisk-dev] [Code Review] 4281: Testsuite: Single channel redirect tests

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Re: [asterisk-dev] [Code Review] 4294: testsuite: Add a test for user_eq_phone setting in PJSIP

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

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