On Fri, Oct 4, 2019, at 1:45 AM, Andreas Wehrmann wrote:
>
> On 03/10/2019 16:24, Joshua C. Colp wrote:
> > In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately
> > codec negotiation is not written or implemented in the way you need. There
> > are some hints provided
On 03/10/2019 16:24, Joshua C. Colp wrote:
In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately
codec negotiation is not written or implemented in the way you need. There are
some hints provided internally for outgoing legs but the result is still
ultimately
On Thu, Oct 3, 2019, at 11:10 AM, Andreas Wehrmann wrote:
>
> On 03.10.19 15:08, Administrator TOOTAI wrote:
>
> > Before calling the gatreway add
> >
> > same = n,set(SIP_CODEC=alaw)
> >
> > [...]
> >
>
> Hey there,
>
> that doesn't work as it seems to be implemented for chan_sip only;
> I'm
On 03.10.19 15:08, Administrator TOOTAI wrote:
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[...]
Hey there,
that doesn't work as it seems to be implemented for chan_sip only;
I'm using chan_pjsip; sorry if I didn't explain myself properly.
Anyway, in my case that would
Hi
Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :
[...]
- Even if direct_media is disabled: Is there a way to make Asterisk
always use a common codec between SIP endpoints,
so it doesn't need to transcode?
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[...]
--
Hello people,
I've ran into two problem that I can't seem to be able to solve on my own.
Here's my scenario (running Asterisk 13.28.1):
In short: - Asterisk behaves unexpectedly (at least to me) when
negotiating between endpoints
that have a different but intersecting set of codecs
