Re: [asterisk-users] ControlPlayBack

On 6/30/2021 9:50 AM, Dovid Bender wrote: > Yes that works. It's an "ugly hack". Would this be classified as a bug > or feature? It's an existing bug: https://issues.asterisk.org/jira/browse/ASTERISK-27871 Kind regards, Sean --

Re: [asterisk-users] ControlPlayBack

Sean, Yes that works. It's an "ugly hack". Would this be classified as a bug or feature? On Wed, Jun 30, 2021 at 9:30 AM Sean Bright wrote: > On 6/30/2021 8:55 AM, Dovid Bender wrote: > > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779 > > ast_openstream_full: File

Re: [asterisk-users] ControlPlayBack

On 6/30/2021 8:55 AM, Dovid Bender wrote: > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779 > ast_openstream_full: File http://localhost/test.gsm?foo=bar > does not exist in any format > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:1252 >

[asterisk-users] Combine audio and video from two different sources

I'm trying to combine audio and video from two different sources. The application is a door intercom system with a separate RTSP video camera. I'd like the door intercom to be able to call an extension and that receiving extension will show the video and have two-way audio. Also it would be nice

[asterisk-users] ControlPlayBack

Hi, I am trying to use ControlPlayBack but pass along some values in the GET request. For instance if I try ControlPlayBack(http://localhost/test.gsm?foo=bar) I get an error in Asterisk [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779 ast_openstream_full: File

[asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello I see the following event from the Asterisk Manager : 2021-06-30 11:20:55 Array (     [0] => Event: PeerStatus     [1] => Privilege: system,all     [2] => SystemName: tstv7     [3] => ChannelType: SIP     [4] => Peer: SIP/testacc7700921     [5] => PeerStatus: Unregistered     [6] =>

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens wrote: > Hello > > > I see the following event from the Asterisk Manager : > > 2021-06-30 11:20:55 > Array > ( > [0] => Event: PeerStatus > [1] => Privilege: system,all > [2] => SystemName: tstv7 > [3] => ChannelType: SIP > [4] =>

[asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

Hello! Short question: Is it possible to set a=silenceSupp:off in the SDP for alaw / ulaw for fax calls? Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

On Wed, Jun 30, 2021 at 1:36 PM Michael Maier wrote: > > Hello! > > Short question: Is it possible to set > > a=silenceSupp:off > > in the SDP for alaw / ulaw for fax calls? > No. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and