SIP Client cannot receive call

We have a few phones connecting to our Asterisk over the internet. These phones are able to call extensions in the office with success - we can hear the ring, answer and talk both ways. However, we are unable to dial their extensions from internal - they do not get a ring and cannot answer.

We have other phones at other internet locations connecting just fine. It must be a firewall issue at the Peer’s end, right? If I look at SIP peers, I see odd ports in use.

Bad phone over internet is this:
442/442 XX.XX.XX.XXX D N A 1025 OK (187 ms)

Good phone over internet is this:
450/450 XX.XX.XX.XXX D N A 55056 OK (189 ms)

Internal phone is this:
433/433 XX.XX.XX.XXX D N A 5060 OK (36 ms)

A good phone internally shows port 5060 and not 1025. However, the good phone at another internet location has port 55056 and it works fine.

Any suggestions? I have the RTP port range in the 10000 range too. These all seem out of that range.

Probable firewall problem. The varying port numbers will be the result of NAT.

It is most likely a firewall issue, but also make sure this line exist in your sip.conf file for every peer (phones).

nat = force_rport,comedia