[asterisk-users] rtp problem with 1.8.0-rdc1

[covici at ccs.covici.com]

Leif Madsen <leif.madsen at asteriskdocs.org> wrote:

> On 10-09-23 05:40 PM, covici at ccs.covici.com wrote:
> > Hi.  I am having a very strange problem --aren't they all -- with the
> > release candidate.  I have softphone which talks to asterisk from behind
> > nat -- the asterisk is on a public ip -- and when I hit mute on the
> > softphone, all rtp traffic ceases!  Now, a version which does work is
> > r281875, this does not happen in that vrsion, but right after that this
> > strange thing starts and is not fixed in the current one.
> >
> > Any assistance here would be appreciated.
> 
> We're probably going to need some sort of debugging information such as a 
> console trace and SIP (I assume chan_sip) debug.
> 
> More information here:
> 
> doc/HOWTO_collect_debug_information.txt
> 
> Leif.
I certainly can do a  sip set debug, is that what you need?  I did do
an rtp set debug and this is how I found out that when I hit the mute
button on the soft phone all rtp traffic ceased between the phone and
the asterisk box.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com