[Freeswitch-users] Unable to successfully bridge calls to an "external" user

[Jim Burke]

Hey David,

IMHO first you need to decide if you want to proxy the media traffic
or not (look at bypass_media), as you are behind a NAT it suggests
that you are perhaps using a cable or adsl connection to the internet
and may not want to give up some of your bandwidth for VOIP calls to
external connections.  If you choose to bypass the media, you will
then need to make sure the IP address reported in the 200 OK by the
terminating user on answer is reported correctly to Faktortel your
ITSP.  You might find this mode will work as Faktortel will probably
be able to determine the path to the terminating phone based on the IP
and PORT it received the voice packets from.

Alternatively if you want to proxy the media traffic, you will need to
make sure that FS reports the correct External IP address in the
INVITE message to the terminating user.  These settings are mentioned
by Anthony below.

I use both NGREP and TCPDUMP heavily when trying new things on FS,
because when you determine what comes out gets easier to findout what
parms to change.

Regards,
Jim

On Tue, May 19, 2009 at 11:24 PM, Brian West <brian at freeswitch.org> wrote:
> You will also need to modify the dial-string in conf/directory/default.xml
> because it only looks on internal for registered users.
> /b
> On May 19, 2009, at 7:56 AM, Anthony Minessale wrote:
>
> edit your sip profile and comment out every line that contains the string
> nat to disable all the nat auto-detection.
> for dmz, you need to set the rtp-ext-ip and sip-ext-ip to be the live ip and
> sip-ip and rtp-ip to be the lan ip (the real one)
>
> Brian West
> brian at freeswitch.org
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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