[Freeswitch-users] No outgoing media on SIP endpoints

[Nicolas Brenner]

Hi Chris,

Your problem seems a lot like what's happening to me, although I'm
only using xlite softphones on extensions 1000 and 1001. The phones
are able to register and make calls (well two calls at least, before
FS stops responding), but the media is not getting through. I'm also
using CentOS 5, but the server and softphone computers are on public
IPs connected directly to the Internet, and there's no firewall
running.

I made a test of registering with extension 1000 and calling the
number 9998 2 times, the first I can hear 3 secs of the tetris sound,
and hanging up on the softphone ends the call, but on the second call
I get no audio at all, and when I hang up on the softphone, the call
is not ended on FS, so it stays open for a while until it times out
and FS starts deperately trying to send BYE messages to the softphone.

Here are my logs and sip trace (user/pass: freeswitch/mailing):
- http://www.medularis.com/fs/freeswitch.log
- http://www.medularis.com/fs/consolelog.txt
- http://www.medularis.com/fs/siptrace.log

I guess I'm not being very useful to you in solving the problem, but I
thought the issues could be related, so maybe we could be able to get
the same help.

On 4/10/08, Chris Chen <chris.chen2004 at gmail.com> wrote:
> Firewall is disabled on the centos 5 as this is purely LAN environment to
> start with. I believe there is no signaling issue here as every SIP endpoint
> is able to be registered and with presence information. The issue is about
> the media can be heard but not able to be sent out.
>  on the sip endpoints.
>
> Thanks
>
> Chris
>
>
>
> On Thu, Apr 10, 2008 at 2:49 PM, Chris Danielson <chris at maxpowersoft.com>
> wrote:
> >
> > Just tossing this out there.  Are iptables running?
> > as root run:  `iptables -L`
> >
> > Make sure that ports 5060-5080 (tcp and udp) are not being blocked.
> >
> >
> > Chris Danielson
> > chris at maxpowersoft.com
> >
> >
> >
> > Chris Chen wrote:
> >
> >
> >
> > Without the change, the domain name and rtp-ip sip-ip would be 127.0.0.1
> which is the loopback IP address  of the freeswitch server. After those
> changes, at least I had all sip end-points registered and I heard dialtones
> when off the hook.
> > Thanks
> >
> >
> >
> > On Thu, Apr 10, 2008 at 2:31 PM, Brian West <brian at freeswitch.org> wrote:
> >
> > >
> > > Can you elaborate on why you did this change?
> > >
> > >
> > > /b
> > >
> > >
> > >
> > > On Apr 10, 2008, at 1:25 PM, Chris Chen wrote:
> > >
> > > I am using the default profile, with the SIP-IP and RTP-IP changed to
> 10.0.1.252
> > >
> > >
> > >
> > > Brian West
> > > sip:brian at freeswitch.org
> > >
> > >
> > >
> > >
> > > _______________________________________________
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> > >
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> > >
> > >
> >
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Regards,

-- 
Nicolás Brenner