[SR-Users] sips to sip with TLS proxy

Hi all, We have a working Kamailio setup, lets call it a transparent proxy for Asterisk boxes. Its based on domain and dispatcher modules and everything is working as expected with the test clients (more or less microsip, softphone for ios, etc). We are tried to register with a Grandstream deskphone today, and we see that the phone sending sips:xxx in the Reg Contact field for example. Because the sips schema, the register is working, but we cannot initiate calls from this phone. If we are turning SIP scheme to sip from sips in the phone, then everything is working as expected. I think we can transform those requests from sips to sip with Kamailio, but currently we dont know where can we start. Has anybody a suggestion about this issue? I know that we can transform ruri, contact, etc with textops, nathelper and a lot of other modules, but what is the best for this sips->sip translation? Thanks for your help. With kind regards, Zoltan