[Freeswitch-users] Dialing extensions behind NAT

[Steven Ayre]

<action application="set" data="bypass_media=true"/> before bridge.

You'll need to be sure they aren't behind different nat routers though.

Steve on iPhone

On 4 Mar 2011, at 02:11, "Ali G. Zaidi" <aligzaidi at gmail.com> wrote:

> 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 at 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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