OK my sip.conf looked OK and to save ports, I changed the rtp.conf to
specify the following and altered the linksys to match.
rtpstart=10000
rtpend=10500
I don't think I remember seeing Stun on the sonicwall, but it does seem
to have a lot of VOIP stuff. They do have a help for VOIP so I'll play
with that.
On 12/25/2010 01:35 PM, Gilles wrote:
On Sat, 25 Dec 2010 09:49:29 -0500, John Ervin<[email protected]>
wrote:
So, assuming your Asterisk box is behind one firewall (Linksys/Tomato
Software) and your Wireless SIP phone is behind another firewall
(SonicWall 1260 Enhanced). Is there anything special that I have to do
to the firewalls.
If the SonicWall firewall supports STUN, just configure each SIP
client to use this to connect to the Asterisk server.
In Asterisk, provided the firewall also provides NAT, use the
following settings in sip.conf so that Asterisk knows that SIP packets
should be rewritten and how:
===========
[general]
externip=<public IP of Tomato router>
;the LAN where Asterisk lives
localnet=192.168.0.0/255.255.255.0
nat=yes
;all RTP packets go through Asterisk
canreinvite=no
;template for SIP users
[sets](!)
type=friend
context=my-phones
host=dynamic
qualify=yes
nat=no
[1234](sets)
secret=mysecret
===========
As for the RTP part :
- 10001 is wrong, since RTP always starts on even ports, and AFAIK
uses two ports (one for sound TX/RX, and one for RTCP)
- unless you need to support 500 concurrent conversations, you can
trim it down. Make sure the range mapped on the Tomato matches what it
says in rtp.conf
HTH,
--
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John F. Ervin
Central Florida TeleSource
407-679-6238
http://jervin.com/cft
[email protected]
--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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