VoIP Mailing List Archives

[pawzlion at gmail.com]

Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ
machine on my modem so it can receive incoming connections without any
NAT related problems.

I'm trying to get a user outside on the internet to connect to my FS
box and register as an internal user. He is using X-Lite on his laptop
behind his own NAT. His external IP is 203.206.171.118.

His registration looks like this:

Call-ID: NGU1NjE4ZmIxODMwMWM1OTZlMDYzMzk0NjMyMjM1YWY.
User: 1001@10.0.0.12
Contact: "124.254.81.250" <sip:1001@203.206.171.118:40168;rinstance=c5779e159bbe8bc7
Agent: X-Lite release 1014k stamp 47051
Status: Registered(UDP)(unknown) EXP(2009-05-18 19:32:03)
Host: kira
IP: 203.206.171.118
Port: 40168
Auth-User: 1001
Auth-Realm: 124.254.81.250

I note that it's registered as plain UDP, not UDP-NAT like my own
internal extensions are.

The dialplan is set to route this DID (0746029001) to user 1001@$$
{domain} as follows:

<extension name="Jake">
<condition field="destination_number" expression="^(0746029001)$">
<action application="bridge" data="USER/1001@$${domain}"/>
</condition>
</extension>

When I try and make a call from my mobile (0451282630) to the DID, it
says it's bridging to USER/1001@10.0.0.12, but when the person
answers, we get no audio in either direction. It rings and answers
fine, it just doesn't send any audio in either direction so I'm
suspecting a bridging problem.

The log file of the connection is on the web at http://pastebin.freeswitch.org/8990

The bridge line is:
EXECUTE sofia/external/0451282630@203.161.130.132 bridge(USER/1001@10.0.0.12
)

But the sofia address for the connection is shown as sofia/internal/sip:1001@203.206.171.118:40168;rinstance=c5779e159bbe8bc7

Is this correct ? Am I missing something fundamental ? His user
address is @10.0.0.12, but his sofia address is sip:
1001@203.206.171.118. Is this OK or should his user ID be at his
actual ip address ? This seems normal to me as I believe the 10.0.0.12
address is the "domain" of the FS box. Is it OK that he's in the same
domain as my own users on my LAN or am I supposed to configure a
different domain for him because he's "outside".

I thought maybe it was a double-NAT problem, but the log doesn't show
any fs_nat=yes entries so I assume it's not trying to NAT him (as it
shouldn't). The situation is an external mobile rings my DID, so the
call comes in from my provider's address, hits my FS box, which
successfully sends at least the ringing information out to his
softphone at his external IP, but then when it bridges, it seems not
to send the audio to the right place.

I'm terrible with FS log files so I have no idea whether any of the
entries are wrong. What's likely to be my issue here ? Is it NAT-
related, or routing related ? Any suggestions appreciated.

David

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[jim at evolutiontel.net]

Can you post the INVITE and 200 OK messages from your mates end of the
call. Even if you forward the ports on the router, the RTP will not
traverse correctly if the advertised IP address is an internal one for
both ends.

On Mon, May 18, 2009 at 6:20 PM, David Robinson <pawzlion@gmail.com> wrote:

[pawzlion at gmail.com]

[anthony.minessale at g...]

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)


On Mon, May 18, 2009 at 8:02 PM, David Robinson <pawzlion@gmail.com (pawzlion@gmail.com)> wrote:

[brian at freeswitch.org]

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:

[jim at evolutiontel.net]

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@freeswitch.org> wrote: