Ugh I used the wrong keyboard shortcuts and the message sent before I was
done. Below is the rest :-)
On Wed, Jan 29, 2020 at 3:42 PM Kevin Harwell wrote:
> On Wed, Jan 29, 2020 at 3:12 PM Michael Maier
> wrote:
>
>>
>>
>
>>
>>
>> From my point of view, it should always be possible to
On Wed, Jan 29, 2020 at 5:12 PM Michael Maier wrote:
> Hello Kevin,
>
> On 29.01.20 at 20:22 Kevin Harwell wrote:
> > Greetings!
> >
> > Over the years there have been numerous requests to improve the codec
> > negotiation process in Asterisk. Specifically, regarding what codecs are
> > offered,
On Wed, Jan 29, 2020 at 3:12 PM Michael Maier wrote:
>
>
> Will Asterisk 18 be a LTS version?
>
I'll defer the answer to that question to others :-)
>
>
> From my point of view, it should always be possible to prevent
> transcoding as long as there is one codec which can be used on
Hello Kevin,
On 29.01.20 at 20:22 Kevin Harwell wrote:
> Greetings!
>
> Over the years there have been numerous requests to improve the codec
> negotiation process in Asterisk. Specifically, regarding what codecs are
> offered, in what order, how Asterisk chooses which codec(s) to use, and of
>
Hello
I’m to try execute AMD on the 183 signalisation, to detect audio on early media.
I’m work in app_dial.c and it work ok when the audio start on the beginner,
but when keep ringing I need to continue analyse until receive the 200 OK or
some error. I already do it
but, always there a
For those of you who actually process SIP MESSAGE requests... Do you use
any of the AMI events generated by the "Message/ast_msg_queue" channel?
We want to change that channel to an "internal" channel that doesn't
generate AMI events (for performance reasons) but we need to know if
anyone's
Greetings!
Over the years there have been numerous requests to improve the codec
negotiation process in Asterisk. Specifically, regarding what codecs are
offered, in what order, how Asterisk chooses which codec(s) to use, and of
course how transcoding is affected by that.
Well hopefully that
