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 <pawzl...@gmail.com> wrote:
> 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: 1...@10.0.0.12
> Contact: "124.254.81.250"
> <sip:1...@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 1...@$$
> {domain} as follows:
>
> <extension name="Jake">
> <condition field="destination_number"
> expression="^(0746029001)$">
> <action application="bridge"
> data="USER/1...@$${domain}"/>
> </condition>
> </extension>
>
> When I try and make a call from my mobile (0451282630) to the DID, it
> says it's bridging to USER/1...@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/0451282...@203.161.130.132 bridge(USER/1...@10.0.0.12
> )
>
> But the sofia address for the connection is shown as
> sofia/internal/sip:1...@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:
> 1...@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
>
_______________________________________________
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
