similar to: Asterisk 10.0.0-rc1 Now Available

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

The Asterisk Development Team is pleased to announce the first beta release of Asterisk 10.0.0-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

The Asterisk Development Team is pleased to announce the first beta release of Asterisk 10.0.0-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk

The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show profile bridge default_bridge" I see: Video Mode: no video I can change it

On 11/14/17 3:38 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: >> I am trying to get the "Mega Phone" demo working on my office PBX >> but there seems to be a problem when trying to set the default bridge to >> sfu mode. I have the following configuration in confbridge.conf in the >> default_bridge section: video_mode

On 11/14/17 3:55 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >> I followed the blog post and I can get video from the conference if >> I configure the bridge as follow_talker so I know everything is working >> on the pjsip side. The only problem is that video_mode = sfu is >> apparently not valid in either confbridge.conf or

On 11/14/17 4:27 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >> On 11/14/17 3:55 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>>> I configure the bridge as follow_talker so I know everything

I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Where does video support for confbridge stand? I need to be able to take in multiple video callers and have the active speaking caller displayed to all participants. Are we there yet in Asterisk 10 or 11? Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference

confBridge in Asterisk 1.6.2.0-rc1 doesn't stable. It causes segment fault very often and results in asterisk crash. Ian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090908/cdcc6674/attachment.htm

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ?

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single video source of the conf. The video mode stays follow_talker. I

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.14 and 1.6.1.5. Asterisk 1.6.0.14 and 1.6.1.5 are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.6.0.14 is the first full, non-security release since 1.6.0.10. The release candidate 1.6.0.11-rc1 was redone as 1.6.0.14-rc1 (which this release has been created