A follow up to this issue: I resolved this issue by setting the ext-sip-ip and ext-rtp-ip to 10.0.1.252.
in the vars.xml as this freeswitch is going nowhere to the internet. So basically this one way audio issue is the NAT issue, the typical SIP issue. now the sip default profile looks like this:

freeswitch@callcenter> sofia status profile default
API CALL [sofia(status profile default)] output:
=================================================================================================
Name            default
Domain Name     10.0.1.252
DBName          sofia_reg_default
Dialplan        XML
RTP-IP          10.0.1.252
Ext-RTP-IP      10.0.1.252
SIP-IP          10.0.1.252
Ext-SIP-IP      10.0.1.252
URL             sip:mod_sofia@10.0.1.252:5060
BIND-URL        sip:mod_sofia@10.0.1.252:5060;maddr=10.0.1.252
HOLD-MUSIC      local_stream://moh
CODECS          G722,PCMU,PCMA,GSM,H263-1998
TEL-EVENT       101
CNG             13
SESSION-TO      0
MAX-DIALOG      0

Registrations:
=================================================================================================
Call-ID         Zjg0YTVjMmU0NzRhN2NhZjA1MmNiOTEyZWExN2M4YTY.
User            1000@10.0.1.252
Contact         "1000" <sip:1000@10.0.1.253:36784;rinstance=6f9ee3cff586843a>
Agent           Bria release 2.2 stamp 45414
Status          Registered(unknown) EXP(2008-04-11 00:15:24)

Call-ID         1858798975-5064-1@10.0.1.160
User            1006@10.0.1.252
Contact         "user" <sip:1006@10.0.1.160:5064>
Agent           Grandstream GXW-4004  V1.1A 1.0.0.67
Status          Registered(unknown) EXP(2008-04-11 00:20:37)

Call-ID         1412946610-5066-1@10.0.1.160
User            1007@10.0.1.252
Contact         "user" <sip:1007@10.0.1.160:5066>
Agent           Grandstream GXW-4004  V1.1A 1.0.0.67
Status          Registered(unknown) EXP(2008-04-11 00:20:37)

Call-ID         51e43082-5491ec6d@10.0.1.251
User            1010@10.0.1.252
Contact         1010 <sip:1010@10.0.1.251:5060>
Agent           Linksys/PAP2-3.1.12(LS)
Status          Registered(unknown) EXP(2008-04-11 00:21:11)

Call-ID         f37727a1-518ac7da@10.0.1.251
User            1011@10.0.1.252
Contact         1011 <sip:1011@10.0.1.251:5061>
Agent           Linksys/PAP2-3.1.12(LS)
Status          Registered(unknown) EXP(2008-04-11 00:21:11)

Call-ID         445786933-5060-1@10.0.1.160
User            1004@10.0.1.252
Contact         "user" <sip:1004@10.0.1.160:5060>
Agent           Grandstream GXW-4004  V1.1A 1.0.0.67
Status          Registered(unknown) EXP(2008-04-11 00:21:21)

Call-ID         1697459759-5062-1@10.0.1.160
User            1005@10.0.1.252
Contact         "user" <sip:1005@10.0.1.160:5062>
Agent           Grandstream GXW-4004  V1.1A 1.0.0.67
Status          Registered(unknown) EXP(2008-04-11 00:21:21)

=================================================================================================

I really appreciate your valuable advice to me for the happy freeswitch journey.
Lots of good stuffs to be explored further.

Thanks,

Chris




On Thu, Apr 10, 2008 at 3:16 PM, Nicolas Brenner <nicolas@medularis.com> wrote:
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@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@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@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@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@freeswitch.org
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users@lists.freeswitch.org
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
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> > > http://www.freeswitch.org
> > >
> > >
> >
> > ________________________________
>
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> >
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> >
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> >
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>
>
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>


Regards,

--
Nicolás Brenner

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