Audio problems with remote extensions on asterisk

I have a problem with the external extensions of my PBX Asterisk.

Inside my office’s intranet everything works perfectly with the SIP Numbers and extensions, tested on the Grandstream and YeaLink telephones. It also works perfectly on Softphones such as Zoiper and X-Lite.

The problem is when you try to connect to an external or remote extension, the necessary ports are already open (5060 and UDP 10000-20000), and the Softphone programs are registered correctly on the PBX. While dialing to another extension or a conventional number the link is established and the phone rangs, but when the call is answered neither of us is able to listen anything.

Eventually the call ends, 5 to 20 seconds later after.

I’ve been told that I might have the RTP protocol blocked in my D-link router DIR-857.

Does anyone know what could be happening?
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grbladesConnect With a Mentor Commented:
On your asterisk server make sure you have localnet= and sipip= (or siphost) defined correctly in your sip.conf file. That will enable it to detect that the remote user is on a different network.
Also for the sip accounts the remote client logs in as make sure you have nat=yes defined.

You probably dont have RTP blocked as such. What is most likely happening is that your sip client is telling the server the ip address to send the packet back to but its telling it your internal ip address behind the dlink. The 'nat=yes' will make asterisk work around this issue.
WebserviceMXAuthor Commented:
Your answer solves the problem, thank you very much :D
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