Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <65404> Message-ID: <[email protected]>
A while back, a change was made to sipx 4.4 to allow sipx to listen for inbound ITSP trunks who expect to send to udp port 5060 (instead of the sipx default of 5080) MANY, MANY TIER1 sip trunk providers insist on this. YES, we can trick many firewalls into doing PAT (port address translation), and in fact, that is what I am currently doing on sipx 4.2. But, in sipx 4.4, when this fix was put in and tested, it only worked if you were NOT natted. (someone want to prove me wrong: get a voip.ms account. set it for ip based authentication, no user/password registration. Tey to make it work (as in inbound calls, be able to transfer back out, moh working, correct called id in forwarded calls) you will see that voip.ms sends the invite to port 5060. Which will sorta work, but sipx sees this as an 'anonymous' call.. something like: sip://[email protected] Which limits what can be done with the call. So, has sipx 4.5 fixed this? or would I still need an sbc/and or use port natting on my firewall? -- -- Michael Scheidell, CTO SECNAP Network Security Corp _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
