Hi all, I'm testing some call scenarios with my asterisk server. I've two questions to make sure my understanding of SIP is correct: 1) What does the localnet setting in sip.conf actually effect? Does it mean that inbound SIP calls, from outside the LAN get treated differently from those on the LAN? Is it just a NATing thing?? 2) The really basic one: If I wanted to make a really simple SIP to SIP call, eg Xlite to Zoiper client to client, would the URI be : sip:2001 at 192.168.0.1 on Zoiper? I'm trying to make calls to sip: ext at serverip, but I'm not sure if the format should be: ext at serverip or sip:ext at serverip or sip://ext_to at serverip/ext_from etc, At the moment inbound SIP calls work, and I'm trying to understand *how* its working.. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080527/fdf30d69/attachment.htm
On Tue, May 27, 2008 at 11:55 AM, Adrian Marsh <Adrian.Marsh at ubiquisys.com> wrote:> 1) What does the localnet setting in sip.conf actually effect? Does > it mean that inbound SIP calls, from outside the LAN get treated differently > from those on the LAN? Is it just a NATing thing??The localnet flag tells Asterisk about the private subnet that it is sitting in. This allows Asterisk to use its own private IP address (as opposed to NAT's external IP) when communicating with entities that are also behind the same NAT.> 2) The really basic one: If I wanted to make a really simple SIP to > SIP call, eg Xlite to Zoiper client to client, would the URI be : > sip:2001 at 192.168.0.1 on Zoiper?Normally when both your devices are registered with Asterisk, you would simply dial the extension number (e.g. 2001) with no SIP-URI syntax.> I'm trying to make calls to sip: ext at serverip, but I'm not sure if the > format should be: > > ext at serverip or > > sip:ext at serverip or > > sip://ext_to at serverip/ext_fromYou should need to dial the extension number only. You typically enter a SIP-URI (sip:user at host) in your SIP phone when you want to make a direct call to a SIP entity that is not registered with your local SIP Registrar. -- Raj Jain