[Asterisk-Users] * behind NAT, client behind NAT(handytone 286), very strange behavior

[Ohad.Levy at infineon.com]

Hi All,

 

I've an Asterisk Server behind a NAT.

Using DNAT, I've opened port 5060 and all 10000:20000 udp.

Sip configured with externalip and subnet.

 

I've another site, also with NAT, where I map the rtp port (as defined
in the client) to map to the local client (DNAT).

Using Xlite, this configuration works, it requires using the quality=yes
and NAT=yes/always in the sip ext configuration but works quite well.

However, lately I've purchased a Grandstream ATA Handytone 286 and tried
to apply the same settings but...

 

When doing an echo test, I can't hear myself, but I can hear the
asterisk server (meaning asterisk can reach the client behind the NAT).

When doing some tcpdump, it looks like some packets are coming from the
client to asterisk, so the network setting looks ok.

When calling to another sip device, with or without canreinvite (yes/no)
the rtp stream is unable to establish it self, no matter where the
second client is (inside/outside NAT).

 

But! When calling using a zap channel (which is on the asterisk server)
everything works! I can hear the person I'm talking to and he can hear
me.

 

I'm a bit confused..... How could it be that this works and echo test
doesn't?

Any help would be appreciated!

 

Thanks,

Ohad

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/0d950839/attachment.htm