>so the asterisk is in middle in all version, right? thank you for your explanation is the one whom everyone goes and says "hey I'm 101 and live downstairs can I play with you guys?"
>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 I guess you still don't fully get it :) 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 http://www.voiptraversal.com/ice_methodology.htm to better understand what's ICE about. Alyed 2010/3/26 haloha <[email protected]> > Hi Alyed > > so the asterisk is in middle in all version, right? thank you for your > explanation > all devices i mean are asterisk + softphones > 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 > > > will check the SJphone feature, thank you for your suggestion > > > Thank you > > On Sat, Mar 27, 2010 at 9:04 AM, Alyed <[email protected]> wrote: > >> 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. >> >> 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. >> >> SJphone supports the advanced SIP URI syntax which for a user is: >> sip:[email protected] >> >> 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. >> >> Alyed >> >> >> 2010/3/26 haloha <[email protected]> >> >>> Hi Alyed >>> >>> >>> xilte softphone work perfectly on other sip server(opensips server) >>> >>> Don't remember the exact syntax but guess it's something like >>> sip:[email protected]: >>>> >>>> 5060 >>> >>> >>> >>>you mean i config the extension.conf look like exten => >>> 1000,1,Dial(SIP/1...@ip address:5060), is it right? >>> >>> 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 >>> >>> is there another hint >>> >>> Thank you >>> >>> >>> On Fri, Mar 26, 2010 at 11:56 PM, Alyed <[email protected]> wrote: >>> >>>> 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 >>>> >>>> Don't remember the exact syntax but guess it's something like >>>> sip:[email protected]:5060 >>>> >>>> Alyed >>>> >>>> >>>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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