Hi. I am having a problem with a conference call on my server which a
vps in the cloud. I am using chan_sip and meetme. What I get is a
bit of a staticy or robotic sound, but it goes away if the user lowers
the volume a bit which we can do with *4 in meetme.
So, is the problem with the
On Fri, Feb 11, 2022 at 9:44 AM Jonas Kellens
wrote:
>
> So if "DeviceStateChange" is not reporting the real state of a SIP
> user/device (like 180-ringing), which event does ?!
>
I don't know of anything else. Someone else may have an idea.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma
On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
Hello
I notice a major difference in what Asterisk console is telling me
(which seems correct) and what Asterisk Manager is telling.
A SIP user is called, and the phone does not ring.
On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens
wrote:
> Hello
>
>
> I notice a major difference in what Asterisk console is telling me (which
> seems correct) and what Asterisk Manager is telling.
>
>
> A SIP user is called, and the phone does not ring. This is the situation.
>
>
> On Asterisk
Hello
I notice a major difference in what Asterisk console is telling me
(which seems correct) and what Asterisk Manager is telling.
A SIP user is called, and the phone does not ring. This is the situation.
On Asterisk console I see (which seems to be in line with an unreachable
phone) :
