Routing problem

Here is my scenario

i have a sip account with
everything is working fine - no firewall issues

however when i try to use a sip client on the smart phone and dial in - i get no audio

after much investigation on the firewall side this is what happens

Working scenario

outside ip talks to the outside if on the firewall - and response is outside to outside ip

Non working scenario

sip call comes in from outside ip to outside interface - is then natted to the pbx
pbx calls internal extension and the internal device replies to the packet
this packet is then dropped as it doesn’t match the originating packet with is to the pbx

so the question:
how do i force my “internal” call to go trough pbx and reply for asterisk and not from the device itself…

i hope i am explaining my self correctly without having to post the firewall cap log

Configure correctly the asterisk NAT settings, in your firewall don’t masquerade IPs in order to send the reply to the peer’s public address.

Have you considered join an Asterisk training or give a full read to the book at

After further investigation here is an update on this problem as diagnosed with help of CISCO support

I can use the soft phone on the “outside” to register with my asterisk and when i make the call out via PSTN i get 2 way audio all is good
which means my soft phone to asterisk and then out via pstn works fine

the issue is when i try to dial internal extension
at that point i get no audio
the bottom line is that if it is an “internal” call the ip is treated as internal and no natting is performed - hence the firewall is dropping the ip mismatch - the sip client receiveing the call is replying to my pbx ip - instead of the external ip of the sip soft phone

how do you deal with this situation - sip user has nat enabled but it’s not making any difference ?

and this jsut got solved

i added the canreinvite=no the the softphone user in sip.conf