Hi everyone-
I’m just getting started with Asterisk on Fedora 4
All I want to do right now is get four or five H.323 clients able to register from out on the Internet
(trying to set up a family-calling system where our family across the Caribbean and the States can call each other without any involvement with any companies that might change their pricing/free-ness…)
I have a PIX firewall, so I’m not too worried about opening up the system for h.323 calls or SIP calls from just anywhere- I have ways to control that I won’t go into
What I don’t know is what and where the minimum info I’d need to configure to set up 5 SIP or H.323 extensions that can call each other is.
No outbound trunks, no VM (yet), no hold music…
what is the minimum I have to know about their SJphone clients?
(or NetMeeting) to be either H.323 or SIP?
not sure which to try to use- SIP or H.323 either…
[quote=“triniphone”]Hi everyone-
I’m just getting started with Asterisk on Fedora 4
All I want to do right now is get four or five H.323 clients able to register from out on the Internet
(trying to set up a family-calling system where our family across the Caribbean and the States can call each other without any involvement with any companies that might change their pricing/free-ness…)[/quote]
Well done, a hero for the family.
[quote=“triniphone”]I have a PIX firewall, so I’m not too worried about opening up the system for h.323 calls or SIP calls from just anywhere- I have ways to control that I won’t go into
What I don’t know is what and where the minimum info I’d need to configure to set up 5 SIP or H.323 extensions that can call each other is.[/quote]
I would strongly recommend SIP over H323. For info on opening up the firewalls have a look here:
voip-info.org/wiki-Asterisk+firewall+rules
[quote=“triniphone”]No outbound trunks, no VM (yet), no hold music…
what is the minimum I have to know about their SJphone clients?
(or NetMeeting) to be either H.323 or SIP?[/quote]
For for SJPhone or X-Lite or something similar, avoid NetMeeting!
SIP, maybe even IAX2.
voip-info.org/wiki-IAX
So, then…
SJPhone it is then – or some other client you’d recommend for WinXP machines… (which is what they have)
I’m guessing it’s all done in sip.conf or iax.conf and extensions.conf?
I guess the real question is-- can I do what I’m proposing here without much difficulty on the other familys’ end of things???
How do clients ‘find’ each other-- I’m assuming that the remote phones will have dynamic IP addresses, they might be able to use something like no-ip, so that they have a resolvable host record for me to ‘know’ them by or something…
Other than installing and configuring a SIP/IAX capable client, do they need to open a given port in their Linksys/dlink/netgear firewalls to their PC?? so that the calls that get set-up can stream to each other?