>so the asterisk is in middle in all version, right? thank you for your
explanation<br>is the one whom everyone goes and says "hey I'm 101 and live downstairs can I play with you guys?"<br><br>>my goal is asterisk is on internet - WAN IP address and the softphones
are in NAT but the xlite supports the ICE function that is >why i ask the
media should be go directly between softphones and no need go through
the asterisk <br>
<br>I guess you still don't fully get it :)<br><br>The scenario you mention is similar (at least for the "direct call" thingy) ICE doesn't mean you don't need to know where the callee is it just means it will play a little with the SDP part of the SIP. Have a look at <a href="http://www.voiptraversal.com/ice_methodology.htm">http://www.voiptraversal.com/ice_methodology.htm</a> to better understand what's ICE about.<br>
<br>Alyed<br><br><br><div class="gmail_quote">2010/3/26 haloha <span dir="ltr"><<a href="mailto:haloha201@gmail.com">haloha201@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi Alyed<br><br>so the asterisk is in middle in all version, right? thank you for your explanation<br>all devices i mean are asterisk + softphones<br>my goal is asterisk is on internet - WAN IP address and the softphones are in NAT but the xlite supports the ICE function that is why i ask the media should be go directly between softphones and no need go through the asterisk <br>
<br>will check the SJphone feature, thank you for your suggestion<br><br><br>Thank you<br><br><div class="gmail_quote">On Sat, Mar 27, 2010 at 9:04 AM, Alyed <span dir="ltr"><<a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">If your sofphones are registering to the asterisk, then asterisk needs to be in the middle, otherwise there's no way your 101 sofpthone user can actually know where (by where I mean which IP) is the 102 softphone user. <br>
<br>UNLESS (yes, there's a big unless) you dial from 101 DIRECTLY to 102. How? well dialing directly to 102's IP.That's where Xlite doesn't work, but SJphone does.<br><br>SJphone supports the advanced SIP URI syntax which for a user is: sip:username@the.user.ip<br>
<br>Nevertheless...... if you are inside a LAN, why wouldn't you want those calls to go through asterisk??? If you have collision problems I suggest you fix them instead of asking everyone to call using SIP uri.<br><br>
Alyed<br><br><br><div class="gmail_quote"><div>2010/3/26 haloha <span dir="ltr"><<a href="mailto:haloha201@gmail.com" target="_blank">haloha201@gmail.com</a>></span><br></div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi Alyed<div><div></div><div><br><br>xilte softphone work perfectly on other sip server(opensips server)<br><br>Don't remember the exact syntax but guess it's something like
sip:username@the.phones.ip:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">5060</blockquote><br>>>>you mean i config the extension.conf look like exten => 1000,1,Dial(SIP/1000@ip address:5060), is it right?<br>
<br>the problem i got here is the asterisk server to stay middle of media first, then redirect the media later, how to fix it,asterisk no need stay in middle of media because all devices are in the same LAN<br><br>is there another hint<br>
<br>Thank you <br><br><br><div class="gmail_quote">On Fri, Mar 26, 2010 at 11:56 PM, Alyed <span dir="ltr"><<a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
I guess to do what you want you need to dial directly between the phones. Can't do it with xlite but you can with SJphones<br><br>Don't remember the exact syntax but guess it's something like sip:username@the.phones.ip:5060<br>
<br>Alyed<br><br><br></blockquote></div><br>
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