Sorry my question wasn't clear enough, so I ask again.

UA1,UA2,UA3===(NAT) ====(Internet)====(FreeSwitch)===(PSTN)

When UA1 dial UA2 the traffic go over the internet connection to FreeSwitch and then comes back to UA2. Can UA1 dial UA2 without going to internet? In other words, after SIP signaling, RTP traffic flow between UAs without FreeSwitch in between? I know in Asterisk you can configure UA with "caninvite" enable, wondering if FS also able to this and how?

Sent from my iPhone

On Mar 3, 2011, at 5:58 PM, Brian West <brian@freeswitch.org> wrote:

give each phone a different client side sip port.

/b

On Mar 3, 2011, at 3:44 PM, Ali Zaidi wrote:

My both phones are Linksys SPA942, please let me know any good phones that
can work best with free switch over wan connection.

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