Conference Ports

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Hi Nanang,

Yes you are right it is not a new behavior of the conference bridge
but i still have a problem even when i'm not talking to Asterisk. I
setup two PJSIP user agents on two separate PCs and make a call
between them. Everything works fine but as soon as i disconnect my
sound card by doing "cd 0 1" the incoming audio becomes hugely
distorted, on either user agents. Is this normal? Because port 1, the
call i made, is still transmitting to my sound card.

I'm a bit confused, do i always have to have my sound card
transmitting to port 1 or should I do what Norman did, that is
creating a new port that just transmit low-level noise?

Best Regards,

Hubert


On Tue, Oct 7, 2008 at 5:56 AM, Nanang Izzuddin <nanang at pjsip.org> wrote:
> Hi,
>
> AFAIK, there is no such 'new' behavior of the conference bridge. And I
> just recalled I read this somewhere that (by default) Asterisk's clock
> for sending RTP is triggered by receiving RTP. So if it doesn't
> receive any RTP packets, it won't send any RTP packets too. This might
> be the case.
>
> Regards,
> nanang
>
>
> On Mon, Oct 6, 2008 at 5:17 AM, Hubert Langevin
> <hubertlangevin at gmail.com> wrote:
>> Hi everyone,
>>
>> I just realised a "new" behavior from pjsip that i haven't seen
>> before. I make a call from my user agent to a soft PBX, asterisk in
>> this case, in order to get music on hold. When i make the call, the
>> conference port for the new call gets created and i hear music on my
>> headset. Eveything looks fine. My sound card sits on port 0 and the
>> new call i made sits on port 1 and they are both transmitting to each
>> other. However if i do "cd 0 1", that is stop my sound card to
>> transmit to port 1, i can't hear the music anymore, even though i
>> still have Port 1, the new conference port playing music on hold,
>> transmitting to port 0.
>>
>> Any ideas?
>>
>> Hubert
>>
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