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BleXXon
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Joined: 13 Aug 2004
Posts: 14
Location: Italia / Alto Adige / Sesto

PostPosted: Thu Feb 16, 2006 11:02 pm    Post subject: X-Lite SIP Softphone for Linux Reply with quote

Hi,

http://www.skypho.net/index.php?option=com_content&task=view&id=17&Itemid=33&

Has somebody an ebuild, I'm not able to make one, sorry...


Greets from Italy
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ectospasm
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Joined: 19 Feb 2003
Posts: 711
Location: Mobile, AL, USA

PostPosted: Fri Feb 17, 2006 1:56 am    Post subject: Reply with quote

Just run the installer found here:

http://www.xten.net/download/X-Lite_Install.tar.gz

No need for an ebuild file, since you'd probably need a manual download anyway (this is not an open source product, I don't think).
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dol-sen
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Joined: 30 Jun 2002
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Location: Richmond, BC, Canada

PostPosted: Fri Feb 17, 2006 2:30 am    Post subject: Reply with quote

It runs from a local user directory without the need for a root type install. I downloaded and started it a few weeks back. I haven't actually tried making a call yet though.
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ectospasm
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Joined: 19 Feb 2003
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PostPosted: Fri Feb 17, 2006 12:20 pm    Post subject: Reply with quote

dol-sen wrote:
It runs from a local user directory without the need for a root type install. I downloaded and started it a few weeks back. I haven't actually tried making a call yet though.


I used it in testing my Asterisk system at my last job. It's really a neat program, it's just a little strange setting it up. Oh, and the SIP protocol sucks ass. If you're behind a NAT, good luck in getting it working right.
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batistuta
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Joined: 29 Jul 2005
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PostPosted: Fri Feb 17, 2006 12:43 pm    Post subject: Reply with quote

Quote:
If you're behind a NAT, good luck in getting it working right.

What type of problems can we expect? I've used it behind my firewall without trouble, both for send and receive. Is this what the STUN is used for?
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ectospasm
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PostPosted: Fri Feb 17, 2006 2:05 pm    Post subject: Reply with quote

batistuta wrote:
Is this what the STUN is used for?


Yes, that's what STUN is used for. The main problem comes when your SIP phone (soft or hard) is behind a NAT, and your VoIP SIP service provider is somewhere else. The SIP protocol doesn't properly account for NAT, but it does work sometimes without using STUN (I think, it's been a while since I've looked at this stuff). I just remember having problems getting my Asterisk server talking to our VoIP provider through a NAT. And if you've got a SIP<==>NAT<--->Internet<--->NAT<==>SIP connection, it's even more difficult. IAX2 does not have this limitation, but I haven't heard of many phones that can speak IAX2, and it's primarily an Asterisk protocol (Inter-Asterisk eXchange v2)...
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