[asterisk-users] Multiple SIP phones behind a Linksys firewall

[shadowym]

Do you have a range of registration ports configured and forwarded through
the firewall on the server end?  Ie. 5060-5065 for example.  

On the Phone side you should forward 5060 to phone1 and 5061 to phone 2 etc.
and configure the phones to use that port for registration.  You may need to
forward ports for the actual voice as well. 2 ports per phone so 10000-10001
for phone1 and 10002-10003 for phone2.  It's either that or mess around with
STUN or Proxy servers or whatever.

SIP+NAT=headache



 

-----Original Message-----
From: john at quonix.net [mailto:john at quonix.net] 
Sent: Saturday, February 02, 2008 8:23 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys firewall

The server is at a remote datacenter - no nat, no firewall, pure public 
IP. 

The phones are at home offices (i.e. DSL or Cable with Linksys-type 
firewall/routers). 

My initial testing was with a single SIP phone at the home office - and 
everything worked fine. But when I have two SIP phones at the home office, 
things start behaving badly. 

I understand the issue of phone-to-phone, where both phones are behind a 
nat at the home office - but that is not the issue I am having. 

My main problem is when I have two phones at the home office, the second 
phone cant register, and/or, you cant here the voicemail greeting when you 
try to check messages.