See the contents of the Contact header in A's REGISTER messages. They tell FreeSWITCH where to send the call to. NAT can confuse matters when picking the correct value though. It needs to be the external IP & port of the NAT router that the internal IP/port of the SIP messages are mapped to. Sometimes the phone will put the internal details instead which aren't routable externally. Sometimes it can detect it correctly (eg via STUN). If it can't some routers will contain a SIP ALG that will rewrite the header for a phone sending the internal ip/port to the external ip/port, but sometimes this can cause more problems than it solves if it doesn't do this correctly and it can't modify the packet if you're using TLS. On top of that that internal to external port mapping will expire on that NAT router if you don't re-REGISTER frequently enough so that could stop the INVITE getting through even if you're sending to the correct place. If you're having issues like that getting the SIP packets through then it's likely If you can't fix it on the phone/router then you can also look at the NDLB (no device left behind) options. For example there's one that'll use the address the REGISTER is received from instead of the Contact header. This differs from how SIP is supposed to work but works in most cases (usually a phone will ask you to call it directly not via a proxy or route you elsewhere). On 11 February 2016 at 09:46, David Wafula <davidwaf@gmail.com> wrote:
Hi all, I have two users who registered in the same domain: user A and B. A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here:
Am not sure what is causing it.
Regards
-- David W
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