[asterisk-users] crossed channels

[Johan Wilfer]

2013-02-19 17:10, Juan Carlos Agudelo skrev:
> I don't have analog channels, this happens with SIP Trunk...
> 
> Juan..
> 

I've seen this with one of our sip-trunks. Our provider used opensips
for that platform I think. If had the same account registered in two
asterisk-servers and they answered the call at the same time, audio was
from both asterisk-servers and the phone from pstn, like a 3-way conference.

I also watched a presentation about rtp ports, mentioned on this list
some time ago, explaining quite a bit on rtp ports and security:
http://lists.digium.com/pipermail/asterisk-users/2013-January/277342.html
http://media.ccc.de/browse/congress/2010/27c3-4193-en-having_fun_with_rtp.html

So my guess is that if you get two devices using the same port, or one
device that don't stop sending, you will hear that injected in your call.


-- 
Johan Wilfer