[Asterisk-Dev] SIP Connection on

[Pertti Pikkarainen]

Kang.ChenJi at c3smail.monmouth.army.mil wrote:

>I have 2 SIP phones, one registered on ServerA, and the other one
>registered on ServerB. Any one can help me understand how to configure so
>one SIP phone can dial to the other? Configuration sample would be
>appreciated.
>
>Thanks,
>Kang
>
>  
>

Here is what I use a lot. Let's say server B has got extensions 300-399.

This line is in Server A
exten => _3XX,1,dial,sip/${EXTEN}@10.2.1.5

10.2.1.5 is the the address of Server B.

You can use the exact extension numbers intead of _3XX   ( like exten => 
301 ... )

Obviously - In server B you need to do this other way around.
Once the phones get connected the RTP media streams should be direct.

-- Pertti

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