Martin, Not my post, but yes that is correct.
Please note that when you do port forwarding it may affect other devices on the same network using the same ports through the same router. In that case you can use totally different ports, and Asterisk will still respond to it. Thanks, Bjorn PS: STUN is not that reliable, but sometimes the only alternative. -----Original Message----- From: Martin Glazer [mailto:[EMAIL PROTECTED] Sent: Thursday, June 08, 2006 1:19 PM To: [email protected] Subject: Re: [on-asterisk] Remote extensions - SIP, NAT Mark, Thanks for the response and tips - Didn't know about the IAXy and DNS names. So if I understand correctly, I will only have to setup port forwarding on the Asterisk PBX side and nothing on the client side (as long as I enable NAT, keep alive, etc)? Martin On Thursday 08 June 2006 10:25, Mark Palser wrote: > As long as you're not afraid to "tinker" it's really not that hard. As > far as I know IAXy's don't support DNS names you have to have an IP > address, so wouldn't work for you unless you have a static external > address. Configure your sip.conf as below, exterhost can be a FQDN, if > your external IP is dynamic go to http://www.dyndns.com/ and sign up. > I use Aastra, Linksys and Polycom and have been able to get them all > to work with a NAT'ed Box with them being behind a firewall as well. > > [general] > nat=1 > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > disallow=all > allow=ulaw > allow=alaw > localnet = 10.144.1.0/255.255.255.0 ; local network your Asterisk > server is in, plus networkmask. > externhost=your external IP address > externrefresh=1 > context = from-sip-external ; Send unknown SIP callers to this context > callerid = Unknown > > Check that you have the correct ports forwarded to your Asterisk box > SIP:5060, RTP:10000-20000 depends on the SIP client you're using. In > your client software check any boxes that refer to NAT keep alive or > anything like it. > > > ----- Original Message ----- > From: "Martin Glazer" <[EMAIL PROTECTED]> > To: <[email protected]> > Sent: Thursday, June 08, 2006 11:45 AM > Subject: **SPAM: [>>> SPAM <<<] - [on-asterisk] Remote extensions - > SIP, NAT - Email found in subject > > > Hi, > > Here's the scenario - we have an Asterisk server setup on our internal > network (firewalled using NAT). We would like to setup a few remote > SIP phones at various peoples home/offices (some behind firewalls as > well). > > I am aware that there are problems with SIP and NAT. > > Has anyone got any real world experience doing something similar? What > is the best method of doing this (STUN, port forwarding, SIP proxy, > etc)? > > The phones we are looking to use are the GXP-2000's and the Snom 360's > or potentially softphones as well. > > Would it be easier to just purchase an IAXy and have people use their > regular analog phones as IAX does not have the same issues as SIP with > firewalls? > > Thanks > > Martin > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] For additional > commands, e-mail: [EMAIL PROTECTED] > > --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
