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On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. > That's probably not something you can hack away at with simple > Asterisk

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

...h participant individually, giving a higher quality), I came up with the idea to use Asterisk. Now I've installed it and got it all set up and did a test call, and it mostly works, but there's an issue. First, here's my dialplan section: exten => 15,1,Answer exten => 15,n,Monitor(wav16,,o) exten => 15,n,ConfBridge(10) As you can see, it's pretty simple. It begins monitoring the caller's audio only, and dumps them into the conference. I tested this with myself and just one other person. I line up the starts of the WAV files, and it's fine at first. But after about...

...of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations. This release candidate contains fixes since the last beta release as reported by the community. A sampling of the changes in this release candidate include: * Add slin16 support for format_wav (new wav16 file extension) (Closes issue #15029. Reported, patched by andrew. Tested by Qwell) * Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded. (Closes issue #17917. Reported by lmadsen. Patched by bbryant) * Var...

...of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations. This release candidate contains fixes since the last beta release as reported by the community. A sampling of the changes in this release candidate include: * Add slin16 support for format_wav (new wav16 file extension) (Closes issue #15029. Reported, patched by andrew. Tested by Qwell) * Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded. (Closes issue #17917. Reported by lmadsen. Patched by bbryant) * Var...

...chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported...

...chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported...

I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.

...(Reported by Sebastian Gutierrez) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported...

...(Reported by Sebastian Gutierrez) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported...

...chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported...

...eported by Niklas Larsson) - [ASTERISK-26853 <https://issues.asterisk.org/jira/browse/ASTERISK-26853>] - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) - [ASTERISK-26613 <https://issues.asterisk.org/jira/browse/ASTERISK-26613>] - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) - [ASTERISK-26169 <https://issues.asterisk.org/jira/browse/ASTERISK-26169>] - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) - [ASTERISK-21856 <https://issues.asterisk.org/...