[asterisk-users] Remote VOIP/SIP Phones through two routers

[John Ervin]

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<jervin at jervin.com>
> 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
jervin at jervin.com

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