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asterisk and BT100

Hope someone can point me in the right direction with this problem. I have 2 BT100 phones connecting to asterisk (on a Debian box) and when those 2 phones (including the asterisk box) are all behind a linksys befsx41 they work fine, but as soon as one phone is at a different location (off-site) and on high-speed internet and behind a linksys (befsr41) also the calls can never connect. That 2nd phone gets a dialtone and can ring the 1st one but the call just can't connect.

In sip.conf the 2nd phone has nat=yes and canreinvite=no (was yes but made no difference) but does anyone have any hints on why it would not work?

Reply to
radar
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Have you got all the relevent ports open? and forwarded ? I have not yet tried double Nat!!!

On the BT100 the * server should be the visible ip address that is forwarded. you will also need to forward all the RTP ports as well.

Feel free to email me direct or call me 9 - 7pm (UK time) on FWD 43143

Ian voipATcyber-cottage.co.uk

Reply to
Ian

The dialtone on the Grandstream is from the phone so you'll always have it.

Using double NAT you will have to be sure that on both NAT routers, the appropriate ports are forwarded.

On the asterisk side (assume its ip were 192.168.1.10) you would forward

5060 and 10000-10100 to 192.168.1.10

On the phone side, you'd forward the same ports to the phone's ip.

You can set up multiple SIP clients by using a range of ports like

5060-5063 and forwarding them appropriately.

People on mailing lists love to argue about this, but it has always worked for me using X-Lite, Grandstreams and Linksys NAT routers.

Reply to
BlueRinse

I know this is a silly question but for the ports 10000-10100 should I open just UDP or UDP&TCP?

Reply to
radar

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