Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] SIP to SIP calls


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
Adrian.Marsh at ubiqui...
Guest





PostPosted: Tue May 27, 2008 10:55 am    Post subject: [asterisk-users] SIP to SIP calls Reply with quote

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
Back to top
rj2807 at gmail.com
Guest





PostPosted: Wed May 28, 2008 2:26 am    Post subject: [asterisk-users] SIP to SIP calls Reply with quote

On Tue, May 27, 2008 at 11:55 AM, Adrian Marsh
<Adrian.Marsh at ubiquisys.com> wrote:
Quote:
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.

Quote:
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.

Quote:
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

You 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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services