RTP problem?

Hi,

I am running Asterisk 1.4.21 behind a NAT

I have setup my asterisk server and my phones located intothe building are working fine.
I have a VoIP provider that I am using -> fine !
I can make phone called and receive without any problems.

Now, I have ONE phone located outside the building.
I can register
If I dial the extension -> the phone is ringing
This phone can make ring my local phone,
But :

  • I can’t hear anything
  • The person over there can hear me

I thought about an NAT and RTP problems, but I really don’t know anymore what to do ?! ?

I place a another phone, registering to my asterisk server via Public IP address, thinking that I will force the COM to get out from the router and come back. Not sure about this is really happen like this or not
I forced it into the SIP file with
CANIVITE=no
CANREINVITE=no
NAT=yes

And this phone INSIDE the building works fine
the other one OUTSIDE the building doesn’t work ( And both have the some SIP configuration [ CANINVITE=no, NAT=yes…etc,)

I am lost …

Any Idea ?

Thanks and regards,

Found the solution…
all I setup was :
NAT=yes
caninvite
Canreinvite=…

=> This is only for client behind a NAT


My Asterisk Server is also behind a NAT
=> Added into SIP.CONF

externip=
Localnet=…
srvlookups=yes