Howdy,
I have the following issue and would like to know if anyone has got around this
before.
IP Phones - Linksys 942
Sip server - Asterisk 1.4.13
Stun server - Vovida
Ok heres the issue. We have multiple client phones on their own network behind
a natted connection. We have setup the phones to be natted and also pointing to
our stun server. Now when the phones make an outside call to the PSTN stun
kicks in and their rtp streams are carried from the phones to the sip provider
without any issues.
Now when the phones dial each other internally the rtp stream is still carried
via stun and therefore fails as its pointing to the same ip on the same router.
Now by adding t to the asterisk dial commands for each internal phone the
inbound calls work fine but the rtp streams are carried through asterisk rather
than between themselves on their network.
Also in this scenario when you try conference an outside phone with an inside
phone it fails due to stun and outside address problems.
So my question is can we set up or change something on the phones or asterisk
to allow the phones rtp to go across the local network on internal calls and
via stun for outbound pstn calls?
Thanks....
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